Administering Avaya IP Office™ Platform

with Manager

Release 11.0
February 2019
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 Contents

Chapter 1: Introduction.......................................................................................................... 25
Purpose................................................................................................................................ 25
New in Release 11.0 FP4....................................................................................................... 25
Chapter 2: Overview............................................................................................................... 28
Manager Modes.................................................................................................................... 28
Security Configuration Mode............................................................................................ 29
Standard Mode Configuration Mode.................................................................................. 30
Server Edition Configuration Mode.................................................................................... 32
Shell Server Mode........................................................................................................... 33
Backward Compatibility.................................................................................................... 34
Chapter 3: Getting Started..................................................................................................... 35
PC requirements................................................................................................................... 35
Installing Manager................................................................................................................. 36
Starting Manager................................................................................................................... 37
Opening a Configuration........................................................................................................ 38
Login messages.............................................................................................................. 39
Changing the Manager Language........................................................................................... 41
Chapter 4: Menu Bar Commands.......................................................................................... 42
File Menu............................................................................................................................. 42
File > Open Configuration................................................................................................ 43
File > Close Configuration................................................................................................ 43
File > Save Configuration................................................................................................. 43
File > Save Configuration As............................................................................................ 45
File > Change Working Directory...................................................................................... 45
File > Preferences........................................................................................................... 46
File > Offline................................................................................................................... 55
File > Advanced.............................................................................................................. 57
File > Backup/Restore..................................................................................................... 73
File > Import/Export......................................................................................................... 73
File > Exit....................................................................................................................... 74
View Menu............................................................................................................................ 74
Tools Menu........................................................................................................................... 75
Tools > Extension Renumber........................................................................................... 75
Tools > Line Renumber................................................................................................... 75
Tools > Connect To......................................................................................................... 76
Tools > Export > User..................................................................................................... 76
Tools > SCN Service User Management........................................................................... 77
Tools > Busy on Held Validation....................................................................................... 77
Tools > MSN Configuration.............................................................................................. 78

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Tools > Print Button Labels.............................................................................................. 78

Tools > Import Templates................................................................................................ 79
File > Advanced > Generate WebLM ID............................................................................ 79
Tools > License Migration................................................................................................. 80
Security Mode Menus............................................................................................................ 81
Embedded File Management Menus....................................................................................... 82
Chapter 5: Manager User Interface....................................................................................... 85
Title Bar................................................................................................................................ 85
Toolbars............................................................................................................................... 85
The Main Toolbar............................................................................................................ 86
The Navigation Toolbar.................................................................................................... 87
The Details Toolbar.......................................................................................................... 87
The Navigation Pane............................................................................................................. 87
Expanding and Collapsing the Navigation Tree.................................................................. 88
The Group Pane.................................................................................................................... 88
Sorting the List................................................................................................................ 88
Customizing the Columns Displayed................................................................................. 89
Changing the Column Widths........................................................................................... 89
Adding a New Record...................................................................................................... 89
Deleting an Record.......................................................................................................... 90
Validating an Record........................................................................................................ 90
Show in Groups.............................................................................................................. 90
The Details Pane................................................................................................................... 90
Managing Records.......................................................................................................... 91
The Error Pane..................................................................................................................... 92
Altering the Automatic Validation Settings.......................................................................... 93
Revalidating Configuration Settings.................................................................................. 93
Viewing an Error or Warning............................................................................................. 93
The Status Bar...................................................................................................................... 94
Configuring the Interface........................................................................................................ 94
Resizing the Manager Window......................................................................................... 95
Moving the Border Between the Panes............................................................................. 95
Showing or Hiding Toolbars.............................................................................................. 95
Moving Toolbars.............................................................................................................. 96
Showing or Hiding Panes................................................................................................. 96
Changing the Position of the Details Pane......................................................................... 96
Changing the Size of Configuration Icons.......................................................................... 97
Changing Tab Display...................................................................................................... 97
Chapter 6: Working with the Server Edition Manager User Interface................................ 98
Server Edition Solution View.................................................................................................. 98
System Inventories.............................................................................................................. 101
Default Settings................................................................................................................... 101
Record Consolidation.......................................................................................................... 102

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Telephone Features Supported Across Server Edition and SCN Networks............................... 103
Chapter 7: Security Administration..................................................................................... 105
Service Users, Application Roles, and Rights Groups............................................................. 105
Default Service Users and Rights Groups........................................................................ 107
Default Service Users and Rights Groups for IP Office R 11.0 and earlier........................... 114
Access Control.................................................................................................................... 119
Encryption.......................................................................................................................... 120
Message Authentication....................................................................................................... 121
Certificates......................................................................................................................... 122
Implementing Security......................................................................................................... 123
SRTP................................................................................................................................. 125
Chapter 8: Editing IP Office Security Settings in Manager............................................... 127
Loading Security Settings..................................................................................................... 127
Saving Security Settings...................................................................................................... 128
Resetting a System's Security Settings................................................................................. 128
Chapter 9: Security Mode Field Descriptions.................................................................... 130
General Security Settings..................................................................................................... 131
General........................................................................................................................ 131
System............................................................................................................................... 136
System Details.............................................................................................................. 136
Unsecured Interfaces..................................................................................................... 138
Certificates................................................................................................................... 139
Security Services Settings.................................................................................................... 144
Rights Groups..................................................................................................................... 146
Group Details................................................................................................................ 146
Configuration................................................................................................................ 146
Security Administration.................................................................................................. 148
System Status............................................................................................................... 148
Telephony APIs............................................................................................................. 149
HTTP........................................................................................................................... 149
Web Services................................................................................................................ 149
External........................................................................................................................ 151
Service Users..................................................................................................................... 152
Chapter 10: Editing Configuration Settings....................................................................... 154
Mergeable Settings.............................................................................................................. 156
Configuration Size............................................................................................................... 166
Setting the Discovery Addresses.......................................................................................... 167
Known System Discovery..................................................................................................... 169
Configuring Manager for Known System Discovery.......................................................... 169
Using Known System Discovery..................................................................................... 169
Opening a Configuration from a System................................................................................ 171
Opening a Configuration Stored on PC................................................................................. 173
Creating New Records......................................................................................................... 174

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Adding a New Record Using the Details Pane................................................................. 174

Adding a New Record Using the Group Pane.................................................................. 174
Adding a New Record Using the Navigation Pane............................................................ 174
Other record creation methods....................................................................................... 175
Creating an Offline Configuration.......................................................................................... 175
Importing and Exporting Settings.......................................................................................... 177
Exporting Settings......................................................................................................... 179
Importing Settings......................................................................................................... 179
Copying and Pasting........................................................................................................... 180
Saving a Configuration onto PC............................................................................................ 180
Sending a Configuration....................................................................................................... 181
Erasing the Configuration..................................................................................................... 183
Default Settings................................................................................................................... 183
Chapter 11: Configuration Mode Field Descriptions......................................................... 186
Configuration field display.................................................................................................... 187
Configuration field display in Standard mode................................................................... 187
Configuration field display in Server Edition mode............................................................ 188
BOOTP Record................................................................................................................... 190
Operator............................................................................................................................. 192
System .............................................................................................................................. 192
System......................................................................................................................... 193
LAN1............................................................................................................................ 199
LAN2............................................................................................................................ 214
DNS............................................................................................................................. 215
Voicemail...................................................................................................................... 215
Telephony..................................................................................................................... 222
Directory Services......................................................................................................... 240
System Events.............................................................................................................. 244
SMTP........................................................................................................................... 252
System | SMDR............................................................................................................. 253
VCM............................................................................................................................. 254
Integrated Contact Reporter........................................................................................... 256
VoIP............................................................................................................................. 257
VoIP Security................................................................................................................ 259
Access Control Lists...................................................................................................... 261
Dialer........................................................................................................................... 261
Contact Center.............................................................................................................. 263
Avaya Cloud Services.................................................................................................... 265
Line.................................................................................................................................... 266
Analog Line................................................................................................................... 267
BRI Line....................................................................................................................... 276
PRI Trunks.................................................................................................................... 281
S0 Line......................................................................................................................... 312

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 Contents

H.323 Line.................................................................................................................... 315

IP DECT Line................................................................................................................ 321
SIP Line....................................................................................................................... 326
SIP DECT Line.............................................................................................................. 360
SM Line........................................................................................................................ 362
IP Office Line................................................................................................................ 371
Control Unit......................................................................................................................... 382
Extension........................................................................................................................... 383
Extn............................................................................................................................. 384
Analog.......................................................................................................................... 387
Extension VoIP.............................................................................................................. 390
Extension T38 Fax........................................................................................................ 398
IP DECT Extension........................................................................................................ 400
SIP DECT Base Extension............................................................................................. 401
User................................................................................................................................... 401
User............................................................................................................................. 402
Voicemail...................................................................................................................... 408
User | DND................................................................................................................... 413
Short Codes.................................................................................................................. 414
Source Numbers........................................................................................................... 415
Telephony..................................................................................................................... 421
User | Forwarding.......................................................................................................... 433
Dial In........................................................................................................................... 436
Voice Recording............................................................................................................ 437
Button Programming...................................................................................................... 439
User | Menu Programming............................................................................................. 440
Mobility......................................................................................................................... 441
Group Memberships...................................................................................................... 445
Announcements............................................................................................................ 445
SIP............................................................................................................................... 446
Personal Directory......................................................................................................... 447
Web Self Administration................................................................................................. 449
Group................................................................................................................................. 451
Group........................................................................................................................... 451
Queuing....................................................................................................................... 456
Overflow....................................................................................................................... 459
Fallback........................................................................................................................ 461
Group | Voicemail.......................................................................................................... 463
Voice Recording............................................................................................................ 469
Announcements............................................................................................................ 470
SIP............................................................................................................................... 473
Short Code......................................................................................................................... 474
Service............................................................................................................................... 475

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Contents

Service......................................................................................................................... 476
Bandwidth..................................................................................................................... 477
IP................................................................................................................................. 479
Autoconnect.................................................................................................................. 481
Quota........................................................................................................................... 481
PPP............................................................................................................................. 482
Fallback........................................................................................................................ 484
Dial In........................................................................................................................... 485
SSL VPN Service.......................................................................................................... 485
RAS................................................................................................................................... 488
PPP............................................................................................................................. 489
Incoming Call Route............................................................................................................ 490
Standard....................................................................................................................... 493
Voice Recording............................................................................................................ 497
Destinations.................................................................................................................. 498
WAN Port........................................................................................................................... 500
WAN Port..................................................................................................................... 500
Frame Relay................................................................................................................. 501
DLCIs........................................................................................................................... 501
Advanced..................................................................................................................... 503
Directory Entry.................................................................................................................... 504
Time Profile........................................................................................................................ 505
Firewall Profile.................................................................................................................... 507
Firewall | Standard......................................................................................................... 507
Firewall | Custom........................................................................................................... 509
Static NAT..................................................................................................................... 511
IP Route............................................................................................................................. 511
IP Route | IP Route........................................................................................................ 512
RIP Dynamic Routing.................................................................................................... 513
Account Code..................................................................................................................... 514
Account Code............................................................................................................... 514
Voice Recording............................................................................................................ 514
License............................................................................................................................... 516
License......................................................................................................................... 516
Remote Server.............................................................................................................. 518
Tunnel................................................................................................................................ 522
L2TP Tunnel................................................................................................................. 523
IP Security Tunnel......................................................................................................... 525
Auto Attendant.................................................................................................................... 528
Auto Attendant.............................................................................................................. 529
Actions......................................................................................................................... 531
Authorization Codes............................................................................................................ 533
User Rights......................................................................................................................... 534

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User............................................................................................................................. 534
Short Codes.................................................................................................................. 535
Button Programming...................................................................................................... 535
Telephony..................................................................................................................... 536
User Rights Membership................................................................................................ 540
Voicemail...................................................................................................................... 541
Forwarding................................................................................................................... 542
ARS................................................................................................................................... 543
ARS............................................................................................................................. 543
Location............................................................................................................................. 547
Address........................................................................................................................ 549
Chapter 12: Configure General System Settings............................................................... 552
Applying Licenses............................................................................................................... 552
PLDS licensing.............................................................................................................. 552
Web License Manager (WebLM)..................................................................................... 553
Server Edition Centralized Licensing............................................................................... 554
Distributing Server Edition Licenses................................................................................ 554
Procedures for Applying Licensing.................................................................................. 559
Converting from Nodal to Centralized Licensing..................................................................... 564
Migrating Licenses to PLDS................................................................................................. 565
Certificate Management....................................................................................................... 566
Certificate Overview...................................................................................................... 567
Certificate Support......................................................................................................... 571
On-boarding........................................................................................................................ 580
Configuring an SSL VPN using an on-boarding file........................................................... 580
System Date and Time......................................................................................................... 581
Configuring Time Profiles..................................................................................................... 582
Overriding a Time Profile................................................................................................ 584
Working with Templates....................................................................................................... 585
Importing Trunk Templates............................................................................................. 586
Creating a Template in Manager..................................................................................... 587
Creating a New Record from a Template in Manager........................................................ 587
Creating an Analog Trunk Template in Manager............................................................... 588
Creating a New Analog Trunk from a Template in Manager............................................... 588
Applying a Template to an Analog Trunk.......................................................................... 589
Centralized System Directory............................................................................................... 589
Advice of Charge................................................................................................................. 593
Emergency Call................................................................................................................... 594
Fax Support........................................................................................................................ 595
Server Edition T38 Fax Support...................................................................................... 597
Caller Display...................................................................................................................... 598
Parking Calls....................................................................................................................... 599
Configuring Call Admission Control....................................................................................... 600

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Manager location tab..................................................................................................... 600

Assigning a network entity to a location...................................................................... 601
System actions at maximum call threshold...................................................................... 601
Example....................................................................................................................... 602
Ring Tones......................................................................................................................... 604
Music On Hold.................................................................................................................... 605
System Source.............................................................................................................. 607
Alternate Source........................................................................................................... 607
Conferencing....................................................................................................................... 611
Conference Phones....................................................................................................... 612
Ad-Hoc Conferencing.................................................................................................... 613
Meet Me Conferencing................................................................................................... 615
Routing External Callers................................................................................................ 616
Context Sensitive Conferencing...................................................................................... 617
Paging................................................................................................................................ 619
Paging Via Voicemail Pro............................................................................................... 621
Automatic Intercom Calls..................................................................................................... 623
Wide Band Audio Support.................................................................................................... 624
Configuring Remote H.323 Extensions.................................................................................. 625
System Configuration.................................................................................................... 626
Phone Configuration...................................................................................................... 628
Media Connection Preservation............................................................................................ 628
Configuring ARS................................................................................................................. 630
Example ARS Operation................................................................................................ 631
ARS Operation.............................................................................................................. 632
Configuring IP Routes.......................................................................................................... 642
Creating a Virtual WAN Port................................................................................................. 643
System Events.................................................................................................................... 644
Configuring Alarm Destinations...................................................................................... 645
Configuring authorization codes............................................................................................ 645
Entering an Authorization Code...................................................................................... 647
Preventing Toll Bypass......................................................................................................... 647
Configuring unknown locations....................................................................................... 648
Call Barring......................................................................................................................... 648
Applying Call Barring..................................................................................................... 648
Overriding call barring.................................................................................................... 649
Chapter 13: Configure User Settings.................................................................................. 651
User Management Overview................................................................................................ 651
Configuring User Rights....................................................................................................... 653
Adding User Rights........................................................................................................ 655
Creating a User Right Based on an Existing User............................................................ 655
Associating User Rights to a User................................................................................... 656
Copy User Rights Settings over a User's Settings............................................................ 656

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Configuring Gmail Integration............................................................................................... 657

Call Intrusion....................................................................................................................... 658
Call Tagging........................................................................................................................ 664
Call Waiting........................................................................................................................ 664
Call Restriction.................................................................................................................... 665
Centralized Call Log............................................................................................................ 666
Centralized Personal Directory............................................................................................. 671
Account Code Configuration................................................................................................. 672
Setting a User to Forced Account Code........................................................................... 673
Coverage Groups................................................................................................................ 673
DND, Follow Me and Forwarding.......................................................................................... 674
Do Not Disturb (DND).................................................................................................... 676
Follow Me..................................................................................................................... 678
Forward Unconditional................................................................................................... 680
Forward on Busy........................................................................................................... 682
Forward on No Answer.................................................................................................. 684
Determining a User's Busy Status................................................................................... 686
Chaining....................................................................................................................... 687
Hot Desking........................................................................................................................ 688
Remote Hot Desking...................................................................................................... 690
Call Center Agents........................................................................................................ 691
Hot Desking Examples................................................................................................... 691
Automatic Log Out......................................................................................................... 693
Group Operation................................................................................................................. 694
Group Types................................................................................................................. 697
Call Presentation........................................................................................................... 698
Group Member Availability............................................................................................. 699
Example Hunt Group..................................................................................................... 701
CBC/CCC Agents and Hunt Groups................................................................................ 703
Malicious Call Tracing (MCID).............................................................................................. 704
Message Waiting Indication.................................................................................................. 704
Message Waiting Indication for Analog Phones................................................................ 705
Message Waiting Indication for Analog Trunks................................................................. 706
Mobile Call Control.............................................................................................................. 707
Mobile Direct Access (MDA)........................................................................................... 710
Mobile Callback............................................................................................................. 711
Twinning............................................................................................................................. 712
Private Calls....................................................................................................................... 715
System Phone Features....................................................................................................... 715
The 'No User' User.............................................................................................................. 717
Suppressing the NoCallerId alarm.................................................................................. 718
Transferring Calls................................................................................................................ 718
Off-Switch Transfer Restrictions...................................................................................... 719

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Context Sensitive Transfer............................................................................................. 720

Dial Tone Transfer......................................................................................................... 721
Handsfree Announced Transfers.................................................................................... 723
One Touch Transferring................................................................................................. 725
Centrex Transfer........................................................................................................... 725
Chapter 14: Configure Server Edition system settings..................................................... 727
Synchronizing Server Edition passwords in Web Manager...................................................... 727
Shared Administration User Account..................................................................................... 728
Voicemail Administration...................................................................................................... 728
Server Edition Resiliency..................................................................................................... 729
Resilience..................................................................................................................... 729
Voicemail Pro Resiliency................................................................................................ 731
®
Avaya one-X Portal resiliency....................................................................................... 733
Phone Resiliency.......................................................................................................... 735
Configuring Resiliency................................................................................................... 738
Synchronizing the Configurations.......................................................................................... 742
Starting Web Control........................................................................................................... 743
Chapter 15: Configuring SIP Trunks................................................................................... 744
Overview............................................................................................................................ 744
Configuring a SIP Trunk....................................................................................................... 745
SIP Line Requirements........................................................................................................ 746
SIP Incoming Call Routing................................................................................................... 748
SIP Prefix Operation............................................................................................................ 749
SIP messaging.................................................................................................................... 750
Outgoing call message details........................................................................................ 751
Incoming call message details........................................................................................ 755
Codec selection............................................................................................................ 760
DTMF transmission....................................................................................................... 761
Fax over SIP................................................................................................................. 761
Hold scenarios.............................................................................................................. 761
SIP REFER................................................................................................................... 763
IP Office SIP trunk specifications.......................................................................................... 764
RFCs............................................................................................................................ 766
Transport protocols........................................................................................................ 767
Request methods.......................................................................................................... 767
Response methods........................................................................................................ 767
Headers....................................................................................................................... 768
Chapter 16: Configuring Small Community Networking................................................... 769
Supported Small Community Network Network Layouts......................................................... 770
Telephone Features Supported Across Server Edition and SCN Networks............................... 771
Voicemail Support............................................................................................................... 772
Enabling Small Community Networking................................................................................. 773
Setup the VoIP Line from System A to System B.............................................................. 773

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Setup the VoIP Line from System B to System A.............................................................. 775

Small Community Network Management............................................................................... 775
Enabling SCN Discovery................................................................................................ 776
Creating a Common Admin Account............................................................................... 776
Loading a Small Community Network Configuration......................................................... 777
Editing a Small Community Network Configuration........................................................... 778
Using the Network Viewer.............................................................................................. 779
System Inventory.......................................................................................................... 784
Small Community Network Remote Hotdesking..................................................................... 784
Small Community Network Fallback...................................................................................... 785
SCN Short Code Programming............................................................................................. 786
Chapter 17: Short Code Overview....................................................................................... 788
Short Code Characters........................................................................................................ 790
User Dialing........................................................................................................................ 793
Application Dialing............................................................................................................... 796
Secondary Dial Tone............................................................................................................ 796
? Short Codes..................................................................................................................... 798
Short Code Matching Examples............................................................................................ 798
Default System Short Code List............................................................................................ 803
Chapter 18: Short Code Features........................................................................................ 809
Auto Attendant.................................................................................................................... 812
Auto Intercom Deny Off........................................................................................................ 813
Auto Intercom Deny On....................................................................................................... 813
Break Out........................................................................................................................... 814
Barred................................................................................................................................ 814
Busy On Held...................................................................................................................... 815
Call Intrude......................................................................................................................... 816
Call Listen........................................................................................................................... 816
Call Park............................................................................................................................. 818
Call Park and Page.............................................................................................................. 819
Call Pickup Any................................................................................................................... 819
Call Pickup Extn.................................................................................................................. 820
Call Pickup Group............................................................................................................... 820
Call Pickup Line.................................................................................................................. 821
Call Pickup Members........................................................................................................... 822
Call Pickup User.................................................................................................................. 822
Call Queue......................................................................................................................... 823
Call Record......................................................................................................................... 824
Call Steal............................................................................................................................ 824
Call Waiting On................................................................................................................... 825
Call Waiting Off................................................................................................................... 826
Call Waiting Suspend........................................................................................................... 826
Cancel All Forwarding.......................................................................................................... 827

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Cancel Ring Back When Free.............................................................................................. 827

Change Login Code............................................................................................................. 828
Clear After Call Work........................................................................................................... 829
Clear Call........................................................................................................................... 829
Clear CW............................................................................................................................ 830
Clear Hunt Group Night Service............................................................................................ 830
Clear Hunt Group Out Of Service.......................................................................................... 831
Clear Quota........................................................................................................................ 832
Coaching Intrusion.............................................................................................................. 832
Conference Add.................................................................................................................. 833
Conference Meet Me........................................................................................................... 834
CW.................................................................................................................................... 835
Dial.................................................................................................................................... 836
Dial 3K1............................................................................................................................. 837
Dial 56K............................................................................................................................. 837
Dial 64K............................................................................................................................. 838
Dial CW.............................................................................................................................. 838
Dial Direct........................................................................................................................... 839
Dial Direct Hot Line.............................................................................................................. 839
Dial Emergency................................................................................................................... 840
Dial Extn............................................................................................................................. 840
Dial Fax.............................................................................................................................. 841
Dial Inclusion...................................................................................................................... 842
Dial Paging......................................................................................................................... 842
Dial Physical Extension by Number....................................................................................... 843
Dial Physical Extension By ID............................................................................................... 844
Dial Speech........................................................................................................................ 844
Dial V110............................................................................................................................ 845
Dial V120............................................................................................................................ 845
Dial Video........................................................................................................................... 845
Disable ARS Form............................................................................................................... 846
Disable Internal Forwards.................................................................................................... 846
Disable Internal Forward Unconditional................................................................................. 847
Disable Internal Forward Busy or No Answer......................................................................... 847
Display Msg........................................................................................................................ 847
Do Not Disturb Exception Add.............................................................................................. 849
Do Not Disturb Exception Delete.......................................................................................... 849
Do Not Disturb On............................................................................................................... 850
Do Not Disturb Off............................................................................................................... 851
Enable ARS Form............................................................................................................... 851
Enable Internal Forwards..................................................................................................... 852
Enable Internal Forward Unconditional.................................................................................. 852
Enable Internal Forward Busy or No Answer.......................................................................... 852

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 Contents

Extn Login.......................................................................................................................... 853

Extn Logout........................................................................................................................ 854
Flash Hook......................................................................................................................... 855
FNE Service....................................................................................................................... 855
Follow Me Here................................................................................................................... 856
Follow Me Here Cancel........................................................................................................ 856
Follow Me To....................................................................................................................... 857
Forward Hunt Group Calls On.............................................................................................. 858
Forward Hunt Group Calls Off.............................................................................................. 858
Forward Number................................................................................................................. 859
Forward On Busy Number.................................................................................................... 860
Forward On Busy On........................................................................................................... 860
Forward On Busy Off........................................................................................................... 861
Forward On No Answer On.................................................................................................. 861
Forward On No Answer Off.................................................................................................. 862
Forward Unconditional On.................................................................................................... 863
Forward Unconditional Off.................................................................................................... 863
Group Listen Off.................................................................................................................. 864
Group Listen On.................................................................................................................. 864
Headset Toggle................................................................................................................... 865
Hold Call............................................................................................................................. 866
Hold CW............................................................................................................................. 866
Hold Music.......................................................................................................................... 867
Hunt Group Disable............................................................................................................. 867
Hunt Group Enable.............................................................................................................. 868
Last Number Redial............................................................................................................. 869
MCID Activate..................................................................................................................... 869
Mobile Twinned Call Pickup.................................................................................................. 870
Off Hook Station.................................................................................................................. 870
Outgoing Call Bar Off........................................................................................................... 871
Outgoing Call Bar On........................................................................................................... 872
Private Call Off.................................................................................................................... 872
Private Call On.................................................................................................................... 873
Priority Call......................................................................................................................... 873
Record Message................................................................................................................. 874
Relay On............................................................................................................................ 875
Relay Off............................................................................................................................ 876
Relay Pulse........................................................................................................................ 876
Resume Call....................................................................................................................... 877
Retrieve Call....................................................................................................................... 877
Ring Back When Free.......................................................................................................... 878
Secondary Dial Tone............................................................................................................ 879
Set Absent Text................................................................................................................... 879

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Contents

Set Account Code............................................................................................................... 881

Set Authorization Code........................................................................................................ 881
Set Fallback Twinning Off..................................................................................................... 882
Set Fallback Twinning On..................................................................................................... 882
Set Hunt Group Night Service.............................................................................................. 883
Set Hunt Group Out Of Service............................................................................................ 883
Set Inside Call Seq.............................................................................................................. 884
Set Mobile Twinning Number................................................................................................ 885
Set Mobile Twinning On....................................................................................................... 885
Set Mobile Twinning Off....................................................................................................... 886
Set No Answer Time............................................................................................................ 886
Set Outside Call Seq........................................................................................................... 887
Set Ringback Seq................................................................................................................ 887
Set Time Profile................................................................................................................... 888
Set Wrap Up Time............................................................................................................... 889
Speed Dial.......................................................................................................................... 890
Shutdown Embedded Voicemail........................................................................................... 891
Stamp Log.......................................................................................................................... 892
Startup Embedded Voicemail............................................................................................... 892
Suspend Call...................................................................................................................... 893
Suspend CW....................................................................................................................... 893
Start After Call Work............................................................................................................ 894
Toggle Calls........................................................................................................................ 894
Unpark Call......................................................................................................................... 895
Voicemail Collect................................................................................................................. 895
Voicemail Node................................................................................................................... 898
Voicemail On....................................................................................................................... 898
Voicemail Off....................................................................................................................... 899
Voicemail Ringback On........................................................................................................ 899
Voicemail Ringback Off........................................................................................................ 900
Whisper Page..................................................................................................................... 900
Chapter 19: Button Programming Overview...................................................................... 902
Programming Buttons with Manager..................................................................................... 902
Programming Button via the Menu Key................................................................................. 904
Setting a Button to Dial a Number................................................................................... 904
Setting a Button to a Switch Function.............................................................................. 904
Setting Buttons to Admin Function.................................................................................. 905
Programming Button via an Admin Button............................................................................. 906
Using an Admin Button.................................................................................................. 907
BST Button Programming.................................................................................................... 908
T3 Self-Administration......................................................................................................... 910
Interactive Button Menus..................................................................................................... 912
Label Templates.................................................................................................................. 912

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Chapter 20: Button Programming Actions......................................................................... 914

Abbreviated Dial.................................................................................................................. 922
Abbreviated Dial Pause....................................................................................................... 923
Abbreviated Dial Program.................................................................................................... 923
Abbreviated Dial Stop.......................................................................................................... 924
Account Code Entry............................................................................................................. 925
ACD Agent Statistics........................................................................................................... 925
ACD Stroke Count............................................................................................................... 926
Acquire Call........................................................................................................................ 927
AD Special Functions.......................................................................................................... 927
AD Special Function Mark.................................................................................................... 927
AD Special Function Wait..................................................................................................... 928
AD Suppress....................................................................................................................... 929
After Call Work.................................................................................................................... 930
Appearance........................................................................................................................ 931
Automatic Callback.............................................................................................................. 933
Auto-Intercom Deny............................................................................................................. 934
Automatic Intercom.............................................................................................................. 935
Break Out........................................................................................................................... 936
Bridged Appearance............................................................................................................ 937
Busy................................................................................................................................... 937
Busy On Held...................................................................................................................... 938
Call Forwarding All.............................................................................................................. 938
Call Intrude......................................................................................................................... 939
Call Listen........................................................................................................................... 940
Call Log.............................................................................................................................. 942
Call Park............................................................................................................................. 942
Call Park and Page.............................................................................................................. 943
Call Park To Other Extension................................................................................................ 944
Call Pickup......................................................................................................................... 945
Call Pickup Any................................................................................................................... 946
Call Pickup Group............................................................................................................... 947
Call Pickup Members........................................................................................................... 948
Call Queue......................................................................................................................... 948
Call Record......................................................................................................................... 949
Call Screening.................................................................................................................... 950
Call Steal............................................................................................................................ 952
Call Waiting Off................................................................................................................... 953
Call Waiting On................................................................................................................... 954
Call Waiting Suspend........................................................................................................... 955
Cancel All Forwarding.......................................................................................................... 956
Cancel Leave Word Calling.................................................................................................. 956
Cancel Ring Back When Free.............................................................................................. 957

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Contents

Clear Call........................................................................................................................... 958

Clear CW............................................................................................................................ 959
Clear Hunt Group Night Service............................................................................................ 959
Clear Hunt Group Out Of Service.......................................................................................... 960
Clear Quota........................................................................................................................ 961
Coaching Intrusion.............................................................................................................. 962
Conference......................................................................................................................... 963
Conference Add.................................................................................................................. 963
Conference Meet Me........................................................................................................... 964
Consult............................................................................................................................... 966
Coverage Appearance......................................................................................................... 967
Dial.................................................................................................................................... 968
Dial 3K1............................................................................................................................. 969
Dial 56K............................................................................................................................. 969
Dial 64K............................................................................................................................. 970
Dial CW.............................................................................................................................. 971
Dial Direct........................................................................................................................... 971
Dial Emergency................................................................................................................... 972
Dial Inclusion...................................................................................................................... 973
Dial Intercom....................................................................................................................... 974
Dial Paging......................................................................................................................... 975
Dial Physical Extn by Number.............................................................................................. 976
Dial Physical Number by ID.................................................................................................. 977
Dial Speech........................................................................................................................ 977
Dial V110............................................................................................................................ 978
Dial V120............................................................................................................................ 979
Display Msg........................................................................................................................ 979
Dial Video........................................................................................................................... 980
Directed Call Pickup............................................................................................................ 981
Directory............................................................................................................................. 981
Do Not Disturb Exception Add.............................................................................................. 983
Do Not Disturb Exception Delete.......................................................................................... 984
Do Not Disturb Off............................................................................................................... 984
Do Not Disturb On............................................................................................................... 985
Drop................................................................................................................................... 986
Extn Login.......................................................................................................................... 987
Extn Logout........................................................................................................................ 988
Flash Hook......................................................................................................................... 989
Follow Me Here................................................................................................................... 989
Follow Me Here Cancel........................................................................................................ 990
Follow Me To....................................................................................................................... 991
Forward Hunt Group Calls Off.............................................................................................. 992
Forward Hunt Group Calls On.............................................................................................. 993

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Forward Number................................................................................................................. 994

Forward On Busy Number.................................................................................................... 995
Forward On Busy Off........................................................................................................... 996
Forward On Busy On........................................................................................................... 996
Forward On No Answer Off.................................................................................................. 997
Forward On No Answer On.................................................................................................. 998
Forward Unconditional Off.................................................................................................... 999
Forward Unconditional On.................................................................................................... 999
Group............................................................................................................................... 1001
Group Listen On................................................................................................................ 1002
Group Paging.................................................................................................................... 1003
Headset Toggle................................................................................................................. 1004
Hold Call........................................................................................................................... 1005
Hold CW........................................................................................................................... 1005
Hold Music........................................................................................................................ 1006
Hunt Group Enable............................................................................................................ 1007
Hunt Group Disable........................................................................................................... 1008
Inspect............................................................................................................................. 1008
Internal Auto-Answer......................................................................................................... 1009
Last Number Redial........................................................................................................... 1010
Leave Word Calling........................................................................................................... 1010
Line Appearance............................................................................................................... 1011
MADN Call Appearance..................................................................................................... 1012
Manual Exclude................................................................................................................ 1013
MCID Activate................................................................................................................... 1014
Monitor Analogue Trunk MWI............................................................................................. 1015
Off Hook Station................................................................................................................ 1015
Pause Recording............................................................................................................... 1016
Priority Call....................................................................................................................... 1017
Priority Calling................................................................................................................... 1017
Private Call....................................................................................................................... 1018
Relay Off.......................................................................................................................... 1019
Relay On.......................................................................................................................... 1019
Relay Pulse...................................................................................................................... 1020
Resume Call..................................................................................................................... 1021
Request Coaching Intrusion............................................................................................... 1022
Retrieve Call..................................................................................................................... 1023
Ring Back When Free........................................................................................................ 1023
Ringer Off......................................................................................................................... 1024
Self-Administer.................................................................................................................. 1025
Send All Calls................................................................................................................... 1028
Set Absent Text................................................................................................................. 1029
Set Account Code............................................................................................................. 1030

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Set Hunt Group Night Service............................................................................................ 1031

Set Hunt Group Out Of Service.......................................................................................... 1032
Set Inside Call Seq............................................................................................................ 1033
Set Night Service Destination............................................................................................. 1033
Set No Answer Time.......................................................................................................... 1034
Set Out of Service Destination............................................................................................ 1035
Set Outside Call Seq......................................................................................................... 1036
Set Ringback Seq.............................................................................................................. 1036
Set Wrap Up Time............................................................................................................. 1036
Speed Dial........................................................................................................................ 1037
Stamp Log........................................................................................................................ 1038
Stored Number View.......................................................................................................... 1038
Suspend Call.................................................................................................................... 1039
Suspend CW..................................................................................................................... 1040
Swap CLID Name/Number................................................................................................. 1040
Time of Day...................................................................................................................... 1041
Time Profile...................................................................................................................... 1042
Timer................................................................................................................................ 1044
Transfer............................................................................................................................ 1045
Toggle Calls...................................................................................................................... 1045
Twinning........................................................................................................................... 1046
Unpark Call....................................................................................................................... 1047
User................................................................................................................................. 1048
Visual Voice...................................................................................................................... 1050
Voicemail Collect............................................................................................................... 1052
Voicemail Off..................................................................................................................... 1053
Voicemail On..................................................................................................................... 1054
Voicemail Ringback Off...................................................................................................... 1055
Voicemail Ringback On...................................................................................................... 1055
Whisper Page................................................................................................................... 1056
Chapter 21: Appearance Button Operation...................................................................... 1058
Appearance Button Features.............................................................................................. 1059
Call Appearance Buttons................................................................................................... 1060
Call Appearance Example 1......................................................................................... 1060
Call Appearance Example 2......................................................................................... 1061
How are Call Appearance Buttons Treated?.................................................................. 1062
Call Appearance Button Indication................................................................................ 1063
Bridged Appearance Buttons.............................................................................................. 1065
Bridged Appearance Example 1................................................................................... 1065
Bridged Appearance Example 2................................................................................... 1066
Bridged Appearance Example 3................................................................................... 1067
How are Bridged Appearances Treated?....................................................................... 1068
Bridged Appearance Button Indication.......................................................................... 1068

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 Contents

Call Coverage Buttons....................................................................................................... 1070

Call Coverage Example 1............................................................................................. 1070
Call Coverage Example 2............................................................................................. 1071
How is Call Coverage Treated?.................................................................................... 1072
Call Coverage Button Indication.................................................................................... 1073
Line Appearance Buttons................................................................................................... 1074
Line Appearance Example 1........................................................................................ 1075
Line Appearance Example 2........................................................................................ 1075
How are Line Appearances Treated?............................................................................ 1076
Line Appearance Button Indication................................................................................ 1077
T3 Phone Line Appearances........................................................................................ 1078
Selected Button Indication.................................................................................................. 1080
Idle Line Preference.......................................................................................................... 1081
Ringing Line Preference..................................................................................................... 1083
Answer Pre-Select............................................................................................................. 1085
Auto Hold......................................................................................................................... 1087
Ring Delay........................................................................................................................ 1087
Delayed Ring Preference................................................................................................... 1089
Collapsing Appearances.................................................................................................... 1091
Joining Calls..................................................................................................................... 1092
Multiple Alerting Appearance Buttons.................................................................................. 1094
Twinning........................................................................................................................... 1095
Busy on Held.................................................................................................................... 1096
Reserving a Call Appearance Button................................................................................... 1096
Logging Off and Hot Desking.............................................................................................. 1096
Applications...................................................................................................................... 1097
Programming Appearance Buttons..................................................................................... 1097
Appearance Function System Settings.......................................................................... 1099
Appearance Function User Settings.............................................................................. 1100
Programming Line Appearance ID Numbers.................................................................. 1101
Outgoing Line Programming......................................................................................... 1103
Chapter 22: Overview of Data Routing.............................................................................. 1104
Network Address Translation (NAT)..................................................................................... 1105
Dynamic Host Configuration Protocol (DHCP)...................................................................... 1105
Simple ISDN Internet Connection........................................................................................ 1106
ISDN Link Between IP Offices............................................................................................ 1106
Using a Dedicated T1/PRI ISP Link..................................................................................... 1107
Tasks for Using a Dedicated T1/PRI ISP Link................................................................. 1107
Create a New WAN Service.......................................................................................... 1107
Create the Virtual WAN Port......................................................................................... 1109
Create an IP Route...................................................................................................... 1109
T1 PRI Trunk............................................................................................................... 1110
Remote Access................................................................................................................. 1110

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Contents

Creating a VoIP Link via the WAN Port Using PPP............................................................... 1113
Chapter 23: Appendix: SMDR............................................................................................ 1115
SMDR Fields..................................................................................................................... 1116
SMDR Examples............................................................................................................... 1121
Chapter 24: Documentation resources............................................................................. 1129
Finding documents on the Avaya Support website................................................................ 1129
Chapter 24: Support............................................................................................................ 1130
Chapter 24: Using the Avaya InSite Knowledge Base..................................................... 1131
Chapter 24: Viewing Avaya Mentor videos....................................................................... 1132
Chapter 24: Additional IP Office resources...................................................................... 1133

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Chapter 1: Introduction

Related links
Purpose on page 25
New in Release 11.0 FP4 on page 25

Purpose
This document contains descriptions of the configuration fields and the configuration procedures
for administering Avaya IP Office Platform using the IP Office Manager application. This document
principally covers Release 11.0 Feature Pack 4 of those products.
Intended audience
The primary audience for the Administering Avaya IP Office using IP Office Manager is the
customer system administrator. Implementation engineers and support and services personnel
may also find this information helpful, however, they are not the primary audience.
Related links
Introduction on page 25

New in Release 11.0 FP4

Default extension password
The default extension password or PIN is different than the user password and is much stronger
with a minimum of nine digits. IP Office R11.0 FP4 deployments provide for an auto-generated
default extension PIN of 10 digits, which can be viewed and modified later using the System >
VoIP > VoIP Security page. The eye icon next to the Default Extension Password field in the
user interface can be used to view the existing default extension password. The feature is
available in R11.0 FP4 deployments including new installs and upgrades.
Fallback twinning
With this feature, IP Office redirects calls to the users twinned mobile number when the primary
extensions are unreachable even if mobile twinning is disabled. The settings can be located at

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Introduction

User > Mobility page. The following two short codes are available for disabling and enabling
mobile Fallback twinning:
• Set Fallback Twinning Off: To disable Fallback twinning
• Set Fallback Twinning On: To enable Fallback twinning
IP Office Media Manager enhancements
The following features have been added to IP Office Media Manager:
• Delete recordings
• Audit trail: Administrators can keep track of the usage of recording files in IP Office Media
Manager. The following type of usage of a recording can be tracked using this feature:
- Delete
- Download
- Replay
- Search
The audit trail displays the User name, Timestamp, User Action, and Details. The audit trail can be
stored up to a duration of one year. The settings are available in Web Manager at Applications >
Media Manager.
Security enhancements for registration of SIP devices
The new security enhancements enable administrators to allow or disallow registration of SIP
devices in IP Office based on their User Agent strings. Administrators can use the settings in
System > VoIP > Access Contro Lists to add, modify, or remove SIP User Agent strings to SIP
UA Blacklist, SIP UA Whitelist, and IP Whitelist. Subsequently, the Allowed SIP User Agents
drop-down menu in System > LAN1 > VoIP can be used to select which SIP User Agents are
allowed for registering with IP Office.
Automatic user synchronization with Avaya Spaces
IP Office R 11.0 FP4 system users and user details created for Avaya Equinox™ can be
automatically synchronized with Avaya Spaces server. To use and receive additional Avaya
Spaces features, the details of users configured for Avaya Equinox™ need matching users
configured on the Avaya Spaces server. The synchronization can be done manually or
automatically. The settings are located at System > Avaya Cloud Services.
Avaya Equinox™ support on Avaya Vantage™
Avaya Equinox™ is supported on Avaya Vantage™ phones in IP Office R 11.0 FP4 deployments.
The option to select an application on Avaya Vantage™ is available at System > Telephony > TUI.
The following applications are available to select:
• Avaya Equinox™
• Vantage Basic/Vantage Connect
Route incoming SIP trunk calls based on an optional SIP header
This feature enables IP Office to route incoming SIP trunk calls based on optional SIP header P-
Called-Party. IP Office reads the P-Called-Party ID header in the SIP message and routes
the incoming SIP calls based on it. The feature can be enabled from Line > SIP Line > SIP
Advanced

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 New in Release 11.0 FP4

SIP and H.323 Registrars disabled by default

To secure IP Office R 11.0 FP4 from vulnerabilities, the H.323 Gatekeeper Enable and SIP
Registrar Enable fields in System > LAN1/LAN2 > VoIP are disabled by default. Whenever a
new H,323 or SIP extension is being added and the corresponding registrar is not enabled, IP
Office system displays the following error message:
System is configured with IP extensions. Registration requires the
corresponding registrar to be enabled
When resiliency support is enabled on an IP Office Line in systems having IP extensions and the
systems do not have the corresponding registrars enabled, IP Office system displays the following
error message:
System is configured to support resiliency. Registration of IP
extensions in failover requires the corresponding registrar to be
enabled
Avaya Equinox™ shared control
Avaya Equinox™ can now be used in shared control mode with desk phones. The shared control
feature is available even when the Avaya Equinox™ client and desk phones are registered on
different systems within the same network.
Join two Conferences
Conference Join feature allows two separate conferences to be joined into one single conference
that contains all the previous participants of both the earlier conferences. Once the conferences
have been joined it is not possible to revert to the two separate conferences again.
JEM 24 button module support on Avaya J169/J179 IP Phone
Avaya J169/J179 IP Phone can support up to three JEM24 button modules. Each JEM24 has 24
dual-LED buttons with adjacent button label display. The button LEDs are used to indicate the
status of the button feature whilst pressing the button is used to access the feature. A single
JEM24 supports 72 programmable button slots. These are arranged in 3-pages, with pages
accessed using the module's page scroll button. When multiple modules are connected to a
phone, each module only supports a single page of 24 programmable button slots.
The modules automatically match the display settings of the phone to which they are connected -
color with Avaya J179 IP Phone or greyscale with Avaya J169 IP Phone, font size, background
image, screen saver. The button modules are powered though the phone. For more information,
see the IP Office Platform R11.0 J100 Series Telephone User Guide.
Avaya J139 IP Phone
IP Office support for Avaya J139 IP Phone. Avaya J139 IP Phone is an advanced SIP desk phone
that supports IP Office interactive menus and button programming. For more information, see the
IP Office Platform R11.0 J100 Series Telephone User Guide.
Related links
Introduction on page 25

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Chapter 2: Overview

This documentation covers the use of the Avaya IP Office Manager. Manager runs on a Windows
PC and connects to the IP Office system via Ethernet LAN or WAN connections.
Important:
Manager is an off-line editor. It receives a copy of the system's current configuration settings.
Changes are made to that copy and it is then sent back to the system for those changes to
become active. This means that changes to the active configuration in the system that occur
between Manager receiving and sending back the copy may be overwritten. For example, this
may affect changes made by a user through their phone or voicemail mailbox after the copy of
the configuration is received by Manager.
Related links
Manager Modes on page 28

Manager Modes
The menus and options displayed by Manager vary depending on the actions you are performing.
Manager runs in the following modes:
Basic Edition Mode
This is the mode used when a Basic Edition configuration is opened. Basic Mode includes
systems running Partner, Norstar, or Quick Mode. For information on administering a Basic Edition
system, see the IP Office Basic Edition Manager.
Security Configuration Mode
Manager can be used to edit the security settings of IP Office systems.
Standard Mode Configuration Mode
This is the mode used when a Standard Mode configuration is opened. Standard Mode includes
systems running Standard, Preferred, or Advanced Edition.
Server Edition Configuration Mode
This is the mode used when an IP Office Server Edition network configuration is opened.

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 Manager Modes

Small Community Network Management

Manager supports loading the combined configurations from systems in a Small Community
Network.
IP Office Shell Server Mode
The IP Office Shell Server is a single installation of selected IP Office applications running on
Linux. You can use Manager to administer an IP Office Shell Server.
Embedded File Management
For systems with a memory card installed, Manager can be used to view and manage the files
stored on the card. Embedded File Management can be accessed by selecting File | Advanced |
Embedded File Management.
Upgrade Wizard
The Upgrade Wizard is a component of Manager used to upgrade the firmware run by the system.
Related links
Overview on page 28
Security Configuration Mode on page 29
Standard Mode Configuration Mode on page 30
Server Edition Configuration Mode on page 32
Shell Server Mode on page 33
Backward Compatibility on page 34

Security Configuration Mode

When Manager is in Security Mode, the screen elements shown are available.

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Overview

Related links
Manager Modes on page 28

Standard Mode Configuration Mode

When Manager is in configuration mode, the screen elements shown are available. Some of these
elements can be customized, moved and hidden.

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 Manager Modes

Manager Configuration Mode Screen Elements

1 Title Bar
In addition to the application name, when configuration settings are loaded from a system, the
title bar displays the user name used to load the settings and the operator view applied.
2 Menu Bar
The options available with the drop down menus provided here change according to whether
Manager has a set of configuration or security settings loaded or not.
3 Main Toolbar
This toolbar provides icon shortcuts to the most frequently required configuration setting actions.
4 Navigation Pane
This pane shows icons for the different types of record that the configuration can contain. Each
type is followed by the number of records of that type already in the configuration. Selecting an
icon displays the matching records in the group pane and navigation toolbar.
5 Group Pane
This pane lists all the records that match the type selected in the navigation pane or navigation
toolbar. The list can be sorted by clicking on column heading. Selecting a record in this pane
displays its details in the details pane.
Table continues…

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Overview

6 Details Pane
This pane shows the configuration settings for a particular record within the configuration. The
record is selected using the navigation toolbar or using the navigation pane and group pane.
7 Navigation Toolbar
This toolbar provides a set of drop downs which can be used to navigate to particular records in
the configuration settings. The selected options in the navigation pane, the group pane and the
details pane are synchronized with the navigation toolbar and vice versa. This toolbar is
particularly useful if you want to work with the group pane and or navigation pane hidden in order
to maximize the display space for the details pane.
8 Error Pane
This pane shows errors and warnings about the configuration settings. Selecting an item here
loads the corresponding record into the details pane.
9 Status Bar This bar display messages about communications between Manager and systems. It
also displays the security level of the communications by the use of a padlock icon.

Related links
Manager Modes on page 28

Server Edition Configuration Mode

When the configuration from a Server Edition solution is loaded into Manager, Manager switches
to Server Edition mode operation.

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 Manager Modes

Related links
Manager Modes on page 28

Shell Server Mode

An IP Office Shell Server is a single installation of selected IP Office applications running on Linux.
You can use Manager to configure and administer a Shell Server. Application Servers and Unified
Communications Modules (UCM) run on an IP Office Shell Server.
Since a Shell Server does not provide telephony, when you open a Shell Server configuration in
Manager, all telephony functions are disabled. The following Manager functions are supported for
Shell Servers:
• Discovery
• Initial configuration utility.
• System status.
• Load, edit and save security settings.

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Overview

• Load, edit, and save the configuration.

• Erase configuration and security settings.
• Audit trail display.
• Web Control.
For more information on the management of an IP Office Shell Server, see Installing and
Maintaining Avaya IP Office™ Platform Application Server and Installing Avaya IP Office™
Platform Unified Communications Module.
Related links
Manager Modes on page 28

Backward Compatibility
Manager is part of the IP Office Admin Suite of programs. The Manager application can be used to
manage configurations from systems running earlier software releases. Manager adjusts the
settings and fields that it shows to match the core software level of the system.
Manager is able display systems with software levels it does not support in the Select IP Office
discovery menu, however those systems are indicated as not supported.
Backwards compatibility is only supported for General Availability releases of IP Office software. It
is not supported for private builds.
Note that this document describes the current release. If you are running an earlier software
release, obtain the Manager document for the specific release from the Avaya support site.
Related links
Manager Modes on page 28

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Chapter 3: Getting Started

Related links
PC requirements on page 35
Installing Manager on page 36
Starting Manager on page 37
Opening a Configuration on page 38
Changing the Manager Language on page 41

PC requirements
Supported Operating Systems
• Windows 7
• Windows 8.1
• Windows 10
• Windows Server 2012 R2
• Windows Server 2016
Note:
IP Office Manager supports Windows only with the display font size set to 100%.
Minimum PC Requirements
IP Office System RAM Available Minimum free Processor Network size
System (minimum or memory hard disk (similar or supported
higher) required for space higher)
Manager
operations
Standard Mode 4 GB 2 GB 6 GB Intel® Core™ i3 Not applicable.
or equivalent, 2
GHz minimum
Server Edition 4 GB (32 bit 2 GB 6 GB Intel® Core™ i3 Up to 32 nodes
OS) or equivalent, 2
GHz minimum
Table continues…

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Getting Started

IP Office System RAM Available Minimum free Processor Network size

System (minimum or memory hard disk (similar or supported
higher) required for space higher)
Manager
operations
Server Edition 8 GB (64 bit 4 GB 6 GB Intel® Core™ i5 Up to 150
OS) or equivalent, 2 nodes
GHz minimum

Applications
If not already present, the required version of .NET Framework is installed as part of the IP Office
Manager installation.
Ports
For information on port usage see the IP Office Port Matrix document on the Avaya support site at
https://support.avaya.com/helpcenter/getGenericDetails?detailId=C201082074362003
Related links
Getting Started on page 35

Installing Manager
Manager is a component of the IP Office Admin suite of applications. This suite is supplied on the
Software DVD (Disk 1). Alternatively, the IP Office Admin Suite can be downloaded from Avaya's
support website http://support.avaya.com.
In addition to Manager, the Admin suite includes options to install the following applications:
• System Monitor This is a tool for system installers and maintainers. Interpreting the
information output by System Monitor requires detailed data and telecoms knowledge.
• System Status Application This is a Java application that can be used to monitor the status
of the system such as extension, trunks and other resources. It displays current alarms and
most recent historical alarms.
Note:
This installation process will install the required version of Windows .NET if not already
present. This may require some systems to restart and the installation process to then be
restarted.
Procedure
1. If installing from the Admin DVD, insert the DVD and when the page is displayed click on
the link for the Admin suite. This will open a file windows showing the installation files for
the suite.
2. Locate and right-click on the setup.exe file. Select Run as Administrator.

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 Starting Manager

3. Select the language you want to use for the installation process. This does not affect the
language used by Manager when it is run. Click Next >.
4. If an upgrade menu appears, it indicates that a previous installation has been detected.
Select Yes to upgrade the existing installed applications.
5. If required select the destination to which the applications should be installed. We
recommend that you accept the default destination. Click Next >.
6. The next screen is used to select which applications in the suite should be installed.
Clicking on each will display a description of the application. Click on the next to each
application to change the installation selection. When you have selected the installations
required, click Next >.
7. The applications selected are now ready to be installed. Click Next >.
8. Following installation, you will be prompted whether you want to run Manager. Selecting
Yes runs Manager.
9. On some versions of Windows, you may be required to restart the PC. Allow this to happen
if required.
Related links
Getting Started on page 35

Starting Manager
No name or password is required to start Manager. A name and password is only required when
connecting with a system.
When started, by default Manager will attempt to discover any systems on the network. If it finds
any it will display a list from which you can select the system required.
1. Select Start and then Programs or All Programs depending on the version of Windows.
Select the IP Office program group.
2. Select Manager. If a Windows Security Alert appears select Unblock to allow Manager
to run.
3. By default Manager will scan the network for any systems. What appears next depends on
whether it finds any systems.
• If Manager finds multiple systems, the Select IP Office window displays a list of those
systems from which you can select the one whose configuration you want to edit. If you want
to open a configuration go to Opening a Configuration. If you don't want to load a
configuration click on Cancel.
• If it finds a single system, it will attempt to open the configuration of that system by displaying
the Configuration Service User Login window..
• If no systems are found or you cancel the steps above, the Manager simplified view is
displayed.

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Getting Started

Use the simplified view to select one of the following action.

• Create an Offline Configuration
• Open a Configuration from a System
• Read a Configuration from a File
Related links
Getting Started on page 35

Opening a Configuration
The initial IP address ranges in which Manager searches for systems is set through the File |
Preferences | Discovery. By default, Manager scans the local network of the Manager PC.
1. Start Manager. If Manager is already started and a configuration is open in it, that
configuration must be closed first.
If Manager is set to Auto Connect on start up, it will scan for systems automatically and
either display the list of systems discovered or automatically start login to the only system
discovered.
Otherwise, select File | Open Configuration.
2. The Select IP Office window opens, listing those systems that responded.
• If Server Edition systems are detected, they are grouped together. By default the
configuration of those systems cannot be opened using Manager in Advanced View
mode and the configuration of a Primary Server can only be opened if the Open with
Server Edition Manager option is also selected.
• If Manager has been set with SCN Discovery enabled, systems in a Small Community
Network are grouped together. The checkbox next to the network name can be used to
load the configurations of all the configurations into Small Community Network
management mode.
• If the system required was not found, the Unit/Broadcast Address used for the search
can be changed. Either enter an address or use the drop-down to select a previously
used address. Then click Refreshto perform a new search.
• A list of known systems can be stored using Known System Discovery.
• Manager can be configured to search using DNS names.
• Systems found but not supported by the version of Manager being used will be listed as
Not Supported.
• If the system detected is running software other than from its primary folder, a
warning icon will be shown next to it. The configuration can still be opened but only as a
read-only file.

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 Opening a Configuration

3. When you have located the system required, check the box next to the system and click
OK.
If the system selected is a Server Edition system and Manager is not running in Server
Edition mode, an Open with Server Edition Manager checkbox is shown and pre-
selected. Clicking OK will switch Manager to its Server Edition mode before loading the
configuration.
4. The system name and password request is displayed. Enter the required details and click
OK.
The name and password used must match a service user account configured within the
system's security settings.
5. Additional messages will inform you about the success or failure of opening the
configuration from the system.
The method of connection, secure or insecure, attempted by Manager is set the
applications Secure Communications preferences setting.
• When Secure Communications is set to On, a padlock icon is displayed at all times
in the lower right Manager status field.
• New installations of Manager default to having Secure Communications enabled. This
means Manager by default attempts to use secure communications when opening a
configuration.
• For Server Edition systems, Manager will always attempt to use secure communications
regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds, Manager
will offer to fallback to using unsecured communications.
6. Following a successful log in, the configuration is opened in Manager. The menus and
options displayed will depend on the type of system configuration loaded.
Related links
Getting Started on page 35
Login messages on page 39

Login messages
While attempting to login to a system, various messages may be displayed.
Configuration Not Loaded Messages
Access Denied
Displayed as the cause if the service user name/password were incorrect, or the service user has
insufficient rights to read the configuration. The Retry option can be used to log in again but
multiple rejections in a 10 minute period may trigger events, such as locking the user account, set
by the Password Reject Limit and Password Reject Action options in the systems security
settings.

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Getting Started

Failed to communicate with system

Displayed as the cause if the network link fails, or the secure communication mode is incorrect (for
example Manager is set to unsecured, but the system is set to secure only).
Account Locked
The account of the service user name and password being used is locked. This can be caused by
a number of actions, for example too many incorrect password attempts, passing a fixed expiry
date, etc. The account lock may be temporary (10 minutes) or permanent until manually unlocked.
An account can be enabled again through the system's security settings.
Additional Messages
Your service user account will expire in X days
Indicates that an Account Expiry date has been set on the system service user account and that
date is approaching. Someone with access to the system's security settings will be required
unlock the account and set a new expiry date.
Your password will expire in X days. Do you wish to change it now?
Indicates that password ageing has been configured in the system's security settings. If your
password expires, someone with access to the system's security settings will be required to unlock
the account.
Limit of concurrent sessions per user exceeded: Indicates that the administrator account has
been used for more than five concurrent sessions. IP Office allows five concurrent sessions using
one administrator account. If five sessions are already on, logging in for the sixth session fails and
Web Manager displays an error message Limit of concurrent sessions per user
exceeded. Note that the following are also considered as a session:
• If Manager is connected with IP Office Server Edition through SE Central Access.
• If the same administrator account is used to log in to any of the IP Office third party
applications developed using the Management SDK client.
Change password
Through the system's security settings, a service user account can be required to change their
password when logging in. The menu provides fields for entering the old password and new
password.
Contact Information Check - This configuration is under special control
This message displays if a Manager user with administrator rights has entered their contact
information into the configuration. For example to indicate that they do not want the configuration
altered while a possible problem is being diagnosed. The options available are:
• Cancel Select this option to close the configuration without making any changes.
• Set configuration alteration flag Select this option if the configuration is being opened
because some urgent maintenance action. When the configuration is next opened, the fact
that it has been altered will be indicated on the System | System tab.
• Delete Contact Information Select this option to take the system out of special control.
• Leave contact information and flags unchanged (Administrators only) This option is
only available to service users logging in with administrator rights.

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 Changing the Manager Language

Related links
Opening a Configuration on page 38

Changing the Manager Language

About this task
The Manager application can run in multiple languages. By default it tries to use the best match to
the PC's regional location settings, otherwise it uses UK English.
The process below can be used to run Manager in one of its supported languages. Some
components in the application such as the error pane, tooltips, and pop-up messages continue to
use the System Locale language.
Procedure
1. Create a Windows shortcut to the Manager application .exe file. By default this file is
located in C:\Program Files\Avaya\IP Office\Manager\Manager.exe.
2. Right-click on the shortcut and select Properties.
3. The Target field can be used to specify the locale setting that Manager should use.
For example, for Italian the Target should have -locale:it-IT added to the end. For
example: "C:\Program Files\Avaya\IP Office\Manager\Manager.exe" -locale:it-IT.
4. Click OK.
5. Manager should now run in the selected language when launched using the updated
shortcut.
Example
Table 1: Manager Locales

Manager Language Shortcut Locale Setting

Brazilian Portuguese -locale:pt-Br
Chinese (Simplified) -locale:zh-Hans
Dutch -locale:nl-NL
French -locale:fr-FR
German -locale:de-DE
Italian -locale:it-IT
Mexican Spanish -locale:es-MX
Russian -locale:ru-RU
US English -locale:en-US

Related links
Getting Started on page 35

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Chapter 4: Menu Bar Commands

The commands available through the Manager's menu bar change according to the mode in which
Manager is running. Commands may also be grayed out if not currently applicable. For some
commands, an arrow symbol indicates that there are sub-commands from which a selection can be
made.
The following sections outline the functions of each command. The Edit and Help menus are not
included.
Related links
File Menu on page 42
View Menu on page 74
Tools Menu on page 75
Security Mode Menus on page 81
Embedded File Management Menus on page 82

File Menu
Related links
Menu Bar Commands on page 42
File > Open Configuration on page 43
File > Close Configuration on page 43
File > Save Configuration on page 43
File > Save Configuration As on page 45
File > Change Working Directory on page 45
File > Preferences on page 46
File > Offline on page 55
File > Advanced on page 57
File > Backup/Restore on page 73
File > Import/Export on page 73
File > Exit on page 74

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 File Menu

File > Open Configuration

This command displays the Select IP Office window used to receive a systems configuration
settings.

The same action is performed by the icon in the Main Toolbar.

The Select IP Office menu is also used for other actions such as reboot and sending a
configuration. If the unit required is not found, the Unit/Broadcast Address can be changed and
then Refresh clicked. To change the TCP addresses scanned, select File | Preferences |
Discovery and enter the required addresses in the IP Search Criteria.
Known Units is not available unless configured
Related links
File Menu on page 42

File > Close Configuration

This command closes the currently loaded configuration without saving it.
Related links
File Menu on page 42

File > Save Configuration

The File > Save Configuration command saves the amended configuration.
If the configuration has been received from a system, the Send Config menu is displayed.
If the configuration file has been opened offline or created from new, the file is saved to disk only.
Related links
File Menu on page 42
Saving Server Edition Configuration Changes on page 43

Saving Server Edition Configuration Changes

After making configuration changes using Manager, the process of saving the changes performs
several actions:
• The configurations are validated for consistency.
• For those systems for which configuration changes have been made, the new configuration is
sent to that system.
Manager defaults to rebooting those systems where a reboot is necessary for the configuration
changes to come into effect.

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Menu Bar Commands

A time stamped copy of the new configuration is also stored on the Primary Server server.
• For a new Secondary Server or expansion system added to the network configuration using
the create off-line configuration option, the offline file is stored, allowing the new system to be
configured even though not yet physically present.
• When opening the configuration from a network, if the timestamp of the stored copy differs
from that of the actual system configuration, Manager will prompt for which configuration it
should load for editing.
Saving Configuration Changes
1. Click in the main toolbar or select File | Save Configuration from the menu bar.
2. The menu displayed only shows details for those systems where the system configuration
has been changed and needs to be sent back to the system.
• Select By default all systems with configuration changes are selected. If you want to exclude
a system from having its configuration updated, either deselect it or cancel the whole
process.
• Change Mode If Manager thinks the changes made to the configuration settings are
mergeable, it will select Merge by default, otherwise it will select Immediate.
• Merge Send the configuration settings without rebooting the system. This mode should only
be used with settings that are mergeable. Refer to Mergeable Settings.
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in
progress. This mode can be combined with the Incoming Call Barring and Outgoing Call
Barring options.
• Store Offline It is possible to add a reference for a Server Edition Secondary or for a Server
Edition Expansion System to create a configuration file for that system even though it is not
physically present. Store Offline saves that configuration on the Server Edition Primary in its
file store. The same file is retrieved from there until such time as the physical server is
present at which time you are prompted whether to use the stored file or the actual servers
current configuration.
• Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Incoming Call Barring and Outgoing Call Barring options.
• Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time
for the system reboot. If the time is after midnight, the system's normal daily backup is
canceled.
• Incoming Call Barring This setting can be used when the reboot mode When Free or
Timed is selected. It bars the receiving of any new calls.
• Outgoing Call Barring This setting can be used when the reboot mode When Free or
Timed is selected. It bars the making of any new calls.
Click OK. The progress of the sending of each configuration is displayed.
Related links
File > Save Configuration on page 43

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 File Menu

File > Save Configuration As

The File > Save Configuration As command allows you to save a configuration a file on the
Manager computer. Note that dynamic configuration data, for example hunt groups advertised
from other systems in a network, are not included in a configuration file saved onto PC and then
reopened.
The command displays the Save As window. Select the drive and directory. and then enter the
new file name. Once you have entered the file name, the Save Configuration File window opens,
where you have the option to encrypt the file by entering a password. Leave the password blank if
you do not want to encrypt the file.
Important:
Encrypted configuration files can only be opened with Manager 9.1 or later. In earlier versions
of Manager, the file will open but it is empty.
Configurations saved onto the PC in this way can be reopened using the icon or the File >
Offline > Open File command. If the file has been encrypted, you must enter the password.
When Manager is running in Server Edition mode, the Save command operates differently.
Multiple files are saved, one .cfg file for each server in the network plus a single .cfi file for the
whole network.
The .cfi file can be used with the File > Offline > Open File Set command to open the whole set
of files in a single action.
Related links
File Menu on page 42

File > Change Working Directory

These settings allow you to change the default locations where Manager looks for and saves files.
These fields set the default location where Manager will look for and save files.
Directory Description
Working Directory (.cfg Sets the directory into which Manager saves .cfg files. By default this is the
files) Manager application's program directory.

Note:
On Windows 7 systems, the default folder for Manager .cfg files is C:
\Program Files (x86)\Avaya\IP Office\Manager. On some Windows 7
systems, the file is saved to the user's profile folder at C:\Users
\<user_name>\AppData\Local\VirtualStore\Program Files (x86)\Avaya\IP
Office\Manager. You must turn on Show hidden files to access this folder.
Alternatively, you can set the working directory to an alternate location.
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Menu Bar Commands

Directory Description
Binary Directory (.bin files) Sets the directory in which the Manager upgrade wizard, HTTP, TFTP and
BOOTP functions look for firmware files requested by phones and other
hardware components. That includes .bin file, .scr files and .txt files. By
default this is the Manager application's program directory.

Tip:
In the Upgrade Wizard, right-clicking and selecting Change Directory also
changes this setting.

Warning:
Historically, by default the Working Directory and Binary Directory are
the same. This is deprecated as it potentially allows remote TFTP/HTTP file
access to the folder containing copies of configuration files. Therefore it is
recommended that either of the folders is changed to an alternate location.
Known Units File Sets the file and directory into which Manager can record details of the systems
it has discovered. Once a file location has been specified, a Known Units button
becomes available on the discovery menu used for loading system configuration.
Pressing that button displays the known units file as a list from which the
required system can be selected. It also allows sorting of the list and records to
be removed.

Related links
File Menu on page 42

File > Preferences

This command displays a window for configuring various aspects of Manager's operation. The
window is divided into a number of tabs.
Related links
File Menu on page 42
File > Preferences > Preferences on page 46
File > Preferences > Directories on page 50
File > Preferences > Discovery on page 50
File > Preferences > Visual Preferences on page 51
File > Preferences > Security on page 52
File > Preferences > Validation on page 55

File > Preferences > Preferences

This tab is accessed through File | Preferences and then selecting the Preferences tab.

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 File Menu

Setting Description
Edit Services Base TCP Default = Off
Port:
This field shows or hides the base communication port settings.
Service Base TCP Port Default = 50804.
Access to the configuration and security settings on a system requires Manager
to send its requests to specific ports. This setting allows the TCP Base Port used
by Manager to be set to match the TCP Base Port setting of the system. The
system's TCP Base Port is set through its security settings.
Service Base HTTP Port Default = 80.
Access to the HTTP server on a system requires Manager to send its requests
to specific ports. This setting allows the HTTP Base Port used by Manager to be
set to match the HTTP Base Port setting of the system. The system’s HTTP
Base Port is set through its security settings.
Enable Time Server Default = On.
This setting allows Manager to respond to RFC868 Time requests from systems.
It will provide the system with both the UTC time value and the local time value
of the PC on which it is running.
Enable BootP and TFTP Default = Off.
Servers
This setting allows Manager to respond to BOOTP request from systems for
which it also has a matching BOOTP record. It also allows Manager to respond
to TFTP requests for files.
Auto Connect on start up Default = On
If on, when Manager is started it will automatically launch the Select IP Office
menu and display any discovered systems. If only one system is discovered,
Manager will automatically display the login request for that system or load its
configuration if the security settings are default.
Set Simplified View as Default = Off
default
If on, the Manager will start in simplified view mode if no configuration is loaded.
Default to Standard Mode Default = Off
If on, when a configuration from a new or defaulted system running in Basic
mode is loaded, Manager will automatically convert the configuration to
Standard mode. Sending the configuration back to the system will restart it in
Standard mode. Only select this option if the only systems you expect to install
are Standard systems.
This setting does not affect existing systems with non-default configurations.
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Menu Bar Commands

Setting Description
Use Remote Access Default = Off.
If selected, access to all the configurations of a multi-site network is allowed via
remote access to the primary server on the multi-site network. When selected,
an additional Use Remote Access check box option is displayed on the Select
IP Office menu when the Open with Server Edition Manager check box option
is selected or if Manager is already running in Server Edition mode.

Note:
To enable remote access, you must first configure an SSL VPN service
between each Server Edition system and the Avaya VPN Gateway (AVG).
For information, see Deploying Avaya IP Office™ Platform SSL VPN
Services.
Consolidate Solution to This setting is used by Manager when in Server Edition mode.
Primary Settings
If Consolidate Network to Primary Settings is selected:
• Entry and administration of consolidated records is performed only at the
solution level except for the Emergency ARS and Fallback System field
settings of location records.
• Those records are then automatically replicated in the configurations of all the
systems in the solution but, except for locations, are still only visible and
editable at the solution level.
• When the configurations are loaded or when this setting is changed to become
selected, if any inconsistency between records are found, a Consolidation
Report is displayed. This report allows selection of whether to update the
system to match the primary or to update the primary to match.
If Consolidate Network to Primary Settings is not selected:
• Entry and administration of consolidated records can be performed at both the
solution and individual system levels.
• Records entered and edited at the solution level are automatically replicated in
the configurations of all the systems in the solution. Each record displays a
label on the record indicating that it is a record that is shared across the
solution.
• If a shared record is edited at the individual system level, that copy of the
record is no longer shared with the other systems. It will not be updated by any
changes to the solution level version of the same record.
• No consolidation checking for inconsistencies is done when the configurations
are loaded.
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 File Menu

Setting Description
SE Central Access Default = Off. Applies to Server Edition systems only.
If On, all Server Edition systems in the network obtain their configuration data
from a central location on the Primary Server. As a result, the display of
configuration changes is delayed until a synchronization process runs. The
synchronization process runs every 40 seconds. If the configuration change
requires a system restart, a refreshed configuration display is delayed until 40
seconds after system restart.
This setting can be used to drive configuration changes into expansion systems
when the expansion systems are not reachable through Manager and the only
accessible system is the Primary Server.

Important:
When adding a new system to the solution, if the Manager setting File >
Preferences > Preferences > SE Central Access is set to On, an IP
Office Line is not configured from the new system to the Server Edition
Primary Server. The status of the new system is Offline. You must
configure an IP Office Line from the new system to the Server Edition
Primary Server.

Note:
When SE Central Access is set to On, you cannot open configurations
with a release number of 9.0.x or earlier. To open older configurations, set
SE Central Access to Off.
When Manager is in Server Edition mode and SE Central Access is set to On,
the following File > Advanced menu options are not available:
• System Shutdown
• Memory Card Command
In addition to the above, when Manager is in Server Edition mode and SE
Central Access is set to On and no configuration is open, the following File >
Advanced menu options are not available:
• Erase Configuration (Default)
• Reboot
• System Shutdown
• Erase Security Settings (Default)
• Memory Card Command
SE Central Access Port Default = 7070.
When SE Central Access is set to On, the port used for routing HTTPS
requests for configuration synchronization.

Related links
File > Preferences on page 46

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Menu Bar Commands

File > Preferences > Directories

These settings allow you to change the default locations where Manager looks for and saves files.
These fields set the default location where Manager will look for and save files.
Directory Description
Working Directory (.cfg Sets the directory into which Manager saves .cfg files. By default this is the
files) Manager application's program directory.

Note:
On Windows 7 systems, the default folder for Manager .cfg files is C:
\Program Files (x86)\Avaya\IP Office\Manager. On some Windows 7
systems, the file is saved to the user's profile folder at C:\Users
\<user_name>\AppData\Local\VirtualStore\Program Files (x86)\Avaya\IP
Office\Manager. You must turn on Show hidden files to access this folder.
Alternatively, you can set the working directory to an alternate location.
Binary Directory (.bin files) Sets the directory in which the Manager upgrade wizard, HTTP, TFTP and
BOOTP functions look for firmware files requested by phones and other
hardware components. That includes .bin file, .scr files and .txt files. By
default this is the Manager application's program directory.

Tip:
In the Upgrade Wizard, right-clicking and selecting Change Directory also
changes this setting.

Warning:
Historically, by default the Working Directory and Binary Directory are
the same. This is deprecated as it potentially allows remote TFTP/HTTP file
access to the folder containing copies of configuration files. Therefore it is
recommended that either of the folders is changed to an alternate location.
Known Units File Sets the file and directory into which Manager can record details of the systems
it has discovered. Once a file location has been specified, a Known Units button
becomes available on the discovery menu used for loading system configuration.
Pressing that button displays the known units file as a list from which the
required system can be selected. It also allows sorting of the list and records to
be removed.

Related links
File > Preferences on page 46

File > Preferences > Discovery

These settings affect the Select IP Office menu used by Manager to discovery systems.

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 File Menu

Setting Description
TCP and HTTP Default = On.
Discovery
This setting controls whether Manager uses TCP to discover systems. The addresses
used for TCP discovery are set through the IP Search Criteria field below.
NIC IP/NIC Subnet This area is for information only. It shows the IP address settings of the LAN network
interface cards (NIC) in the PC running Manager. Double-click on a particular NIC to
add the address range it is part of to the IP Search Criteria. Note that if the address of
any of the Manager PC's NIC cards is changed, the Manager application should be
closed and restarted.
IP Search Criteria This section is used to enter TCP addresses to be used for the TCP discovery
process. Individual addresses can be entered separated by semi-colons, for example
135.164.180.170; 135.164.180.175. Address ranges can be specified using dashes,
for example 135.64.180.170 - 135.64.180.175.
UDP Discovery Default = On
This settings controls whether Manager uses UDP to discover systems.
Enter Broadcast IP Default = 255.255.255.255
Address
The broadcast IP address range that Manager should used during UDP discovery.
Since UDP broadcast is not routable, it will not locate systems that are on different
subnets from the Manager PC unless a specific address is entered.
Use DNS Selecting this option allows Manager to use DNS name (or IP address) lookup to
locate a system. Note that this overrides the use of the TCP Discovery and UDP
Discovery options above. This option requires the system IP address to be assigned
as a name on the users DNS server. When selected, the Unit/Discovery Address
field on the Select IP Office window is replaced by a Enter Unit DNS Name or IP
Address field.
SCN Discovery If enabled, when discovering systems, the list of discovered systems will group
systems in the same Small Community Network and allow them to be loaded as a
single configuration. At least one of the systems in the Small Community Network
must be running Release 6.0 or higher software. See Configuring Small Community
Networking on page 769. This does not override the need for each system in the
Small Community Network to also be reachable by the TCP Discovery and or UDP
Discovery settings above and accessible by the router settings at the Manager
location.

Related links
File > Preferences on page 46

File > Preferences > Visual Preferences

Setting Description
Icon size Sets the size for the icons in the navigation pane between Small, Medium or Large.
Multiline Tabs Default = Off.
In the details pane, for record types with more than two tabs, Manager can either use
buttons to scroll the tabs horizontally or arrange the tabs into multiple rows. This
setting allows selection of which method Manager uses.

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Menu Bar Commands

Related links
File > Preferences on page 46

File > Preferences > Security

Additional configuration information
For additional configuration information, see Security Administration on page 148. Also see Avaya
IP Office™ Platform Security Guidelines.
Configuration settings
Controls the various security settings of Manager. To control the security settings of the system,
see the information on Security mode.
All settings, except Secure Communications, can only be changed when a configuration has
been opened using a user name and password with Administrator rights or security administration
rights.
Setting Description
Request Login on Default = On
Save
By default a valid user name and password is required to receive a configuration
from a system and also to send that same configuration back to the system.
Deselecting this setting allows Manager to send the configuration back without
having to renter user name and password details. This does not apply to a
configuration that has been saved on PC and then reopened. This setting can
only be changed when a configuration has been opened using a user name and
password with Administrator rights or security administration rights.
Close Configuration/ Default = On.
Security Settings
When selected, the open configuration file or security settings are closed after
After Send
being sent back to the system. This is the normal default. This setting does not
affect multi-site network modes of Manager which always close the configuration
after saving.
Before disabling this setting, you should recall that the configuration held by a
running system can be changed by actions other than Manager, for example
changes made by users through their phone. Keeping a configuration open in
Manager for longer than necessary increases the chances that that copy of the
configuration differs from the current configuration of the running system and will
overwrite those changes when sent back to the system.
Save Configuration Default = Off.
File After Load
When selected, a copy of the configuration is saved to Manager's working
directory. The file is named using the system name and the suffix .cfg. This
setting can only be changed when a configuration has been opened using a user
name and password with Administrator rights or security administration rights.
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 File Menu

Setting Description
Backup Files on Default = Off.
Send
If selected, whenever a copy of a configuration is sent to a system, a backup
copy is saved in Manager's working directory. The file is saved using the system
name, date and a version number followed by the Backup File Extension as set
below. This setting can only be changed when a configuration has been opened
using a user name and password with Administrator rights or security
administration rights.
Backup File Default = .BAK
Extension
Sets the file extension to use for backup copies of system configurations
generated by the Backup Files on Send option above.
Number of Backup Default = Unlimited.
Files to keep
This option allows the number of backup files kept for each system to be limited.
If set to a value other then Unlimited, when that limit would be exceeded, the file
with the oldest backup file is deleted.
Enable Application Default = On.
Idle Timer (5
When enabled, no keyboard or mouse activity for 5 minutes will cause the
minutes)
Manager to grey out the application and re-request the current service user
password. This setting can only be changed when a configuration has been
opened using a user name and password with Administrator rights or security
administration rights.
Secure Default = On
Communications
When selected, any service communication from Manager to the system uses the
TLS protocol. This will use the ports set for secure configuration and secure
security access. It also requires the configuration and or security service within
the system's security configuration settings to have been set to support secure
access. Depending on the level of that secure access selected, it may be
necessary for the Manager Certificate Checks below to be configured to match
those expected by the system for configuration and or security service.
• When Secure Communications is set to On, a padlock icon is displayed at
all times in the lower right Manager status field.
• For Server Edition systems, Manager will always attempt to use secure
communications regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds,
Manager will offer to fallback to using unsecured communications.
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Menu Bar Commands

Setting Description
Manager Certificate When the Secure Communications option above is used, Manager will process
Checks and check the certificate received from the system. This setting can only be
changed when a configuration has been opened using a user name and
password with Administrator rights or security administration rights. The options
are:
• Low: Any certificate sent by the system is accepted.
• Medium: Any certificate sent by the system is accepted if it has previously
been previously saved in the Windows' certificate store. If the certificate has not
been previously saved, the user has the option to review and either accept or
reject the certificate.
• High: Any certificate sent by the system is accepted if it has previously been
previously saved in the Windows' certificate store. Any other certificate cause a
log in failure.
Certificate Offered to Default = none Specifies the certificate used to identify Manager when the
IP Office Secure Communications option is used and the system requests a certificate.
Use the Set button to change the selected certificate. Any certificate selected
must have an associated private key held within the store:
• Select from Current User certificate store - Display certificates currently in the
currently logged-in user store.
• Select from Local Machine certificate store.
• Remove Selection – do not offer a Manager certificate.

Security – Registry Settings

Warning:
Avaya accept no liability for any issues arising from the editing of a PC’s registry settings. If
you are in any doubt about how to perform this process you should not proceed. It is your
responsibility to ensure that the registry is correctly backed up before any changes are made.
Note:
Before manually editing any registry entry, the following Microsoft support articles should be
read:
• http://support.microsoft.com/kb/256986
• http://www.microsoft.com/resources/documentation/windows/xp/all/proddocs/en-us/
regedit_permit_key.mspx
Manager stores it’s security preferences in the Windows Registry. The following key affects
manager security operation; it’s values may only be changed by a configuration or security
administrator:
HKEY_CURRENT_USER\Software\Avaya\IP400\Manager\Security\
In order to prevent circumvention by manual editing of the Windows Registry, Regedt32.exe, the
native registry editor, allows an operator user (with Full Control permissions) to edit permissions
on a per key basis.

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 File Menu

To prevent a user from manually editing the security preferences, the HKEY_USERS\User GUID
\Software\Avaya\IP400\Manager\Security key permission should be set to ‘Read’ only
for that user. Ensure that all child object permissions are replaced as well by using the ‘Advanced’
button.
To allows the security policy of all local PC users to be fixed, a set of values in the key
HKEY_CURRENT_USER\Software\Avaya\IP400\Manager\Security\ may be created. This
is tested and used in preference to any value found under HKEY_CURRENT_USER\Software
\Avaya\IP400\Manager\Security\.
This key is not created by the manager application.
Related links
File > Preferences on page 46

File > Preferences > Validation

By default Manager validates the whole configuration when it is loaded and individual fields
whenever they are edited. This tab allows selection of when automatic validation should be
applied to configuration files loaded into Manager.
Setting Description
Validate configuration Automatically validate configuration files when they are opened in Manager.
on open
Validate configuration Validate the whole configuration when OK is clicked after editing a record. For large
on edit configurations, disabling this option removes the delay caused by validating the
configuration after every edit.
Prompt for If selected, when saving or sending a configuration, a prompt is displayed asking
configuration whether the configuration should be validated. If validation is selected and error are
validation on save or found, the send or save process is canceled. This option is disabled if Validate
send configuration on edit is selected.

Related links
File > Preferences on page 46

File > Offline

Related links
File Menu on page 42
File > Offline > Create New Config on page 55
File > Offline > Open File on page 56
File > Offline > Open File Set on page 56
File > Offline > Send Config on page 56
File > Offline > Receive Config on page 56

File > Offline > Create New Config

Used to create an offline configuration.

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Menu Bar Commands

This command starts a dialog that allows you to create a default offline configuration by specifying
the system locales, the type of control unit and expansion modules and the trunk cards fitted. The
same action is performed by the icon in the Main Toolbar.
Related links
File > Offline on page 55

File > Offline > Open File

This command allows a configuration file stored on PC to be opened in Manager.
Related links
File > Offline on page 55

File > Offline > Open File Set

This command is only available when manager is running in Server Edition mode. It can be used
to load a set of files previously saved offline using the File > Save Configuration As command.
When selected, browse to the location of the saved .cfi file and associated .cfg files and select
the .cfi file.
Related links
File > Offline on page 55

File > Offline > Send Config

This command is used to send an offline configuration to a system.

Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
After sending the configuration, you should receive the configuration back from the system and
note any new validation errors shown by Manager. For example, if using Embedded Voicemail,
some sets of prompt languages may need to be updated to match the new configurations locale
setting using the Add/Display VM Locales option.
Related links
File > Offline on page 55

File > Offline > Receive Config

This command displays the Select IP Office menu used to receive a systems configuration
settings.
Once the configuration has been received, you are prompted to save it on the PC.
Related links
File > Offline on page 55

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 File Menu

File > Advanced

Related links
File Menu on page 42
File > Advanced > Erase Configuration on page 57
File > Advanced > Reboot on page 58
File > Advanced > System Shutdown on page 58
File > Advanced > Upgrade on page 59
File > Advanced > Change Mode on page 61
File > Advanced > Audit Trail on page 62
File > Advanced > Security Settings on page 63
File > Advanced > Erase Security Settings (Default) on page 63
File > Advanced > Embedded File Management on page 64
File > Advanced > Format IP Office SD Card on page 64
File > Advanced > Recreate IP Office SD Card on page 65
File > Advanced > Memory Card Command on page 66
File > Advanced > Launch Voicemail Pro on page 67
File > Advanced > System Status on page 67
File > Advanced > LVM Greeting Utility on page 67
File > Advanced > Generate WebLM ID on page 68
File > Advanced > Initial Configuration on page 68
File > Advanced > Add/Display VM Locals on page 72

File > Advanced > Erase Configuration

This command returns the configuration settings of a system back to their default values. It does
not affect the system's security settings or audit trail record.
When this command is used, the Select IP Office menu is displayed. Once a system is selected,
a valid configuration user name and password are required to complete the action.
IP500 V2 systems using IP Office A-Law or IP Office U-Law System SD cards will default to
Quick mode. Loading the configuration will switch Manager to simplified view. To change the
system back to operating in Standard mode, use either of the following methods:
• Change Mode
This will change the operating mode of the system and create a default configuration
appropriate to that mode.
Related links
File > Advanced on page 57

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Menu Bar Commands

File > Advanced > Reboot

When this command is used, the Select IP Office window is displayed. Once a system is
selected, a valid user name and password are required. The type of reboot can then be selected
in the Reboot window.
When the reboot occurs can be selected as follows:
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in
progress. This mode can be combined with the Call Barring options.
• Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Call Barring options.
Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time
for the reboot. If the time is after midnight, the system's normal daily backup is canceled.
Call Barring These settings can be used when the reboot mode When Free is selected. They bar
the sending or receiving of any new calls.
Related links
File > Advanced on page 57

File > Advanced > System Shutdown

This command can be used to shutdown systems. The shut down can be either indefinite or for a
set period of time after which the system will reboot. For Linux based telephone systems, the
shutdown command is applied to the telephony service on the server and not to the whole sever.
In that case, if the system is shutdown indefinitely, it can be restarted using the server's web
control pages to either restart the service or to restart the whole server.

Warning:
A shutdown must always be used to switch off the system. Simply removing the power cord or
switching off the power input may cause the loss of configuration data.
This is not a polite shutdown, any user calls and services in operation will be stopped. Once
shutdown, the system cannot be used to make or receive any calls until restarted.
The shutdown process takes up to a minute to complete. When shutting down a system with a
Unified Communications Module installed, the shutdown can take up to 3 minutes while the card
safely closes all open files and closes down its operating system. During this period the module's
LED 1 remains green.
When shutdown, the LEDs shown on the system are as follows. Do not remove power from the
system or remove any of the memory cards until the system is in this state:
• LED1 on each IP500 base card installed will also flash red rapidly plus LED 9 if a trunk
daughter card is fitted to the base card.
• The CPU LED on the rear of the system will flash red rapidly.

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 File Menu

• The System SD and Optional SD memory card LEDs on the rear of the system are
extinguished.
To restart a system when shutdown indefinitely, or to restart a system before the timed restart,
switch power to the system off and on again.
Once you have selected the system from the Select IP Office window, the System Shutdown
Mode window opens. Select the type of shutdown required:
• If a Timed shutdown is selected, the system will reboot after the set time has elapsed.
• If Indefinite is used, the system can only be restarted by having its power switched off and
then on again. For Linux based telephone systems, the telephony service must be restarted
through the server's web control pages.
Related links
File > Advanced on page 57

File > Advanced > Upgrade

Note:
This command is not used with Linux based systems. Linux systems are updated through the
server's webcontrol menus.
This command starts the Upgrade Wizard tool. The Upgrade Wizard is used to compare the
software level of the control unit and modules within systems against the software level of the .bin
binary files Manager has available. The Upgrade Wizard can then be used to select which units to
upgrade.

Warning:
• Incorrect use of the upgrade command can halt system operation and render units in the
system unusable. You must refer to the Technical Bulletins for a specific release for full
details of performing software upgrades to that release. There may be additional steps
required such as defaulting the security settings.
• Performing any other actions on a system during an upgrade or closing the upgrade
wizard and Manager during an upgrade may render systems unusable.
• During an upgrade the system may restrict calls and services. It will reboot and
disconnect all current calls and services.
• The Validate option must remain selected wherever possible. Use of unvalidated
upgrades is subject to a number of conditions outlined in the IP Office Installation Manual
and Technical Bulletins.
The list area shows details of systems found by the Upgrade Wizard and the software currently
held by those systems. The check boxes are used to select which units should be upgraded.
Upgrading will require entry of a valid name and password for the selected system.

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Menu Bar Commands

Column Description
Name The name of the system as set in its configuration (System | System | Name) .
IP Address The IP address of the system.
Type The type of system and the names of the various firmware files used by external
expansion systems supported by the system type.
Version Details the current software each unit in the systems is running.
Edition Indicates the operation mode of the system.
Licensed Indicates the highest value software upgrade license present in the system's
configuration. The IP Office Release that is supported by that license is also
indicated in brackets.
Required License Indicates the software upgrade license required for the current level of software
the system is running. The IP Office Release that is supported by that license is
also indicated in brackets.
It does not refer to the software upgrade license required for the level of software
which is available for upgrade. The system must include a license for the specific
level of software it is required to run.
For IP500 V2 systems, a value of 255 indicates that the control unit is still in its
initial 90 days where it can be upgraded to a higher level without requiring an
upgrade license.
Available Shows the version of the matching firmware files that Manager has available (a –
indicates no file available) in its current working directory. Upgrading to a release
higher than that supported by the current Licensed level will leave the system
unable to support any functions until the appropriate upgrade license is added to
the system configuration.

The Upgrade Wizard includes a number of check boxes that can be used to include other actions
as part of the upgrade process:
• Validate
• The Validate option should remain selected wherever possible. When selected, the upgrade
process is divided as follows: transfer new software, confirm transfer, delete old software,
restart with new software. If Validate is not selected, the old software is deleted before the
new software is transferred.
• Backup System Files
• For any IP500 V2 systems being upgraded, the Backup system files option will cause the
system to backup its memory card files as part of the upgrade.
• Upload System File
• For any IP500 V2 system being upgraded, the Upload system files option will upload
various files:
- It copies the binary files for the system control unit and possible external expansion
modules.
- It copies the firmware files used by phones supported by the system.

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 File Menu

- It copies the files for Web Manager.

- For systems configured to run Embedded Voicemail, the Embedded Voicemail prompts for
those supported languages set as the system locale, user locales, incoming call route
locales and short code locales are upgraded.
• Restart IP Phones For Manager 8.1 and higher the Restart IP Phones option can be used.
This will cause those phone to load any upgrade phone firmware included in the system
upgrade (if using the system's memory card as their firmware file source).
Related links
File > Advanced on page 57

Searching for Systems

The default address used by the Upgrade Wizard is the address shown in the Manager title bar,
which is selected through File | Preferences. If the unit required is not found, the address used can
be changed.
Procedure
1. Enter or select the required address in the Unit/Broadcast Address field.
2. Click Refreshto perform a new search.
Changing the .bin File Directory
The directory in which the Upgrade Wizard looks for .bin files is set through Manager's Binary
Directory setting. This can be changed using Files | Change Working Directory or File |
Preferences | Directories.
Use this procedure to change it directly from the Upgrade Wizard.
Procedure
1. Right-click on the list area.
2. Select Select Directory.
3. Browse to and highlight the folder containing the .bin files. Click OK.
4. The list in the Available column will be updated to show the .bin files in the selected
directory that match units or modules listed.

File > Advanced > Change Mode

This command can be used to change the operating mode of an IP500 V2 system from Basic
Edition to either standard IP Office or Server Edition expansion. To convert an existing system to
Basic Edition mode, use the default configuration options.
Important:
Using this command will default the configuration. Therefore ensure that you have a backup
copy of the configuration before using this command in case it is necessary to return to the
previous mode.

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Do not use this command if the Default to Standard Mode option is enabled in the IP Office
Manager preferences. Disable the option first.
Note that if the system includes components not supported by the mode to which it is switched,
they will not work in the new mode. For example, ETR cards which are only supported in Basic
Edition.
In order to use this command, the system security settings must be at their default settings. The
current setting can be defaulted using the Erase Security Settings (Default) command.
After a mode change, the system restarts. If the system does not restart, the most likely cause is
that the systems security settings were not at their default settings.
Related links
File > Advanced on page 57

File > Advanced > Audit Trail

The audit trail lists the last 16 actions performed on the system from which the configuration
loaded into Manager was received. It includes actions by service users such as sending a
configuration back, reboots, upgrades and defaulting the system.
Audit trail events can be output to a Syslog server through the system's System | System Events
settings.
The last failed action is always recorded and shown in red. It is kept even if there have been 16
subsequent successful actions.
The Audit Trail is part of the system configuration file received from the system. If the configuration
is kept open between send and reboot operations (ie. if Close Configuration/Security Setting After
Send is not selected), the Audit Trail will not show details of those operations. It will only show
details of those operations if the configuration is closed and then a new copy of the configuration
is received from the system.
Audit Details
When a specific access event is selected from the list, the following information is shown in the
Audit Details section:
• The Security User shows the service user name used for the access action.
• The Date and Time of Access indicate the local system time when the recorded event
occurred.
• The PC Login is the computer name of the PC used for the access.
• The PC IP Address and PC MAC Address are the IP address and MAC address of the PC
used for access.
• The Access Type details the type of action that was performed.
• The Outcome shows the system's response to the access. The outcome Success
(Warning) refers to the sending of a configuration that contains fields marked as errors or
warnings by Manager's validation function. Success (Clean) refers to the sending of a
configuration that does not contain any validation errors or warnings.

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 File Menu

• The IP Office Firmware indicates the IP Office Release version.

Items Changed
The Items Changed area summarizes the changes contained in a sent configuration. Where
changes to a single record of a particular type are made, the Item Name field lists the individual
record changed. Where changes are made to several records of the same type, the Item Name
field displays Multiple items.
Related links
File > Advanced on page 57

File > Advanced > Security Settings

This command is used to switch the Manager application to security mode. In that mode, Manager
is used to edit the security settings of a system.
Related links
File > Advanced on page 57

File > Advanced > Erase Security Settings (Default)

This command returns the security settings of a system back to their default values. This action
does not affect the system's configuration or audit trail record. When this command is used, the
Select IP Office menu is displayed. Once a system is selected, a valid security user name and
password are required to complete the action.
The system's security settings are returned to their defaults as indicated in the Security Mode
chapter.
Note that any security certificates stored and being used by the system are deleted. Any services
currently using those certificates are disconnected and disabled until the appropriate certificates
are added back to the system's security configuration. That includes SSL VPN connections being
used to perform system maintenance.
The name and password used for this command are those required for security configuration
access which are different from those used for normal configuration access.
For IP500 V2 control units, if the security settings cannot be defaulted using this command, they
can be defaulted using a DTE cable connection to the system. Refer to the IP Office Installation
manual for details.

Warning:
Service Disruption.
Whilst defaulting the security settings does not require a system reboot, it may cause service
disruption for several minutes while the system generates a new default security certificate.
Related links
File > Advanced on page 57

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Menu Bar Commands

File > Advanced > Embedded File Management

For control units with a memory card installed, the contents of the card can be viewed using
Manager. This view can also be used to add and remove files from the card. This may be useful
when the memory card is being used to store Music on Hold or IP phone firmware files.
• For non-Linux systems, the folders accessed using embedded file management are those on
the SD cards installed in the control unit.
• For Linux based systems, the folder path displays as disk in the embedded file manager.
That path maps onto /opt/ipoffice on the server.
Related links
File > Advanced on page 57

File > Advanced > Format IP Office SD Card

This command allows suitable SD cards to be formatted by the Manager PC. The system supports
SD cards with the following format: SDHC minimum 4GB FAT32 format (Single partition, SDHC,
class2+, FAT32, SPI & SD bus). Non-Avaya supplied cards of the same format can be used a
system's Optional SD slot for additional actions such as backup.

Warning:
• Do not re-purpose a Enterprise Branch SD card for use with any other IP Office mode.
Doing so may damage the SD card and make it unusable for your Enterprise Branch
system.
• All File Will Be Erased Note that this action will erase any existing files and folders on
the card. If the requirement is just to update the card, use Recreate IP Office SD Card
without reformatting. Once a card has been formatted, the folders and files required for
operation can be loaded onto the card from the Manager PC using the Recreate IP Office
SD Card command.
• Avaya supplied SD cards should not be formatted using any other method than the
format commands within Manager and System Status Application. Formatting the cards
using any other method will remove the feature key used for system licensing from the
card.
Related links
File > Advanced on page 57
Formating the SD card on page 64

Formating the SD card

Procedure
1. Insert the SD card into a reader slot on the Manager computer.
2. Using Manager, select File | Advanced | Format IP Office SD Card.
3. Select the type of card.

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 File Menu

This selection just sets the card label shown when viewing the card details. It does not
affect the actual formatting. Select the label that matches the file set you will be placing on
the card.
• IP Office A-Law A system fitted with this type of card will default to A-Law telephony.
• IP Office U-Law A system fitted with this type of card will default to U-Law telephony.
• Enerprise Branch Use this option for an SD card intended to be used with an IP Office
system running in Enterprise Branch Mode. There is a separate SD card for Enterprise
Branch. The Enterprise Branch SD card can only be used for Enterprise Branch
operation and cannot be used to change modes to IP Office. You also cannot use or
change an IP Office SD card for use with an Enterprise Branch system.

Warning:
Do not re-purpose a Enerprise Branch card for use with any other IP Office mode.
Doing so may damage the SD card and make it unusable for your Enterprise Branch
system.
4. Browse to the card location and click OK.
5. The status bar at the bottom of Manager will display the progress of the formatting
process.
6. When the formatting is complete, you can use the Recreate IP Office SD Card command to
load the system folders and files onto the card from the Manager PC.
Related links
File > Advanced > Format IP Office SD Card on page 64

File > Advanced > Recreate IP Office SD Card

This command can be used with the System SD cards used by IP500 V2 control units. It allows
Manager to copy all the files and folders used by a system when starting onto the card that has
been place into the card slot of the PC running Manager. It updates the card with the version of
those files installed with the Manager application. It includes the binary files for the system,
external expansion modules and phones. The command also copies all language prompt sets
used by Embedded Voicemail.
If the card contains dynamic system files such as SMDR records, they are temporarily backed up
by Manager and then restored after the card is recreated. For the card to be used in a system's
System SD slot the card must be Avaya SD Feature Key card. The card must be correctly
formatted, however a reformat of an existing working card is not necessary before using recreate
to update the card contents.
The source for the files copied to the SD card are the sub-folders of the \Memory Cards folder
under Manager's Working Directory (normally C:\Program Files\Avaya\IPOffice
\Manager). However, if the Working Directory is changed to a location without an appropriate set
of \Memory Cards sub-folders, the required set of files will not be copied onto the SD card.
Related links
File > Advanced on page 57

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Menu Bar Commands

Recreating the IP Office SD Card on page 66

Recreating the IP Office SD Card

About this task
Note:
This process can take up to 20 minutes depending on the PC. Once started, the process
should not be interrupted.
Procedure
1. Insert the SD card into a reader slot on the Manager computer.
2. Using Manager, select File | Advanced | Recreate IP Office SD Card.
3. Select the type of system for which the card is intended.
This selection will affect how the system operates when defaulted with this card present in
its System SD card slot.
• IP Office A-Law A system fitted with this type of card will default to A-Law telephony.
• IP Office U-Law A system fitted with this type of card will default to U-Law telephony.
• Enterprise Branch Use this option for an SD card intended to be used with an IP Office
system running in Enterprise Branch mode. There is a separate SD card for IP Office.
The Enterprise Branch SD card can only be used for IP Office operation and cannot be
used tochange modes to IP Office. You also cannot use or change an IP Office SD card
for use withan Enterprise Branch system.

Warning:
Do not re-purpose a Enterprise Branch SD card for use with any other IP Office
mode. Doing so may damage the SD card and make it unusable for your Enterprise
Branch system.
4. Browse to the card location and click OK.
5. For all systems, these files are necessary if you want to go through the process of on-
boarding registration.
6. Manager will start creating folders on the SD card and copying the required files into those
folders.
7. Do not remove the card until the process is completed and Manager displays a message
that the process has been completed.
Related links
File > Advanced > Recreate IP Office SD Card on page 65

File > Advanced > Memory Card Command

These commands are used with the memory cards installed in IP500 V2 control units.

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 File Menu

Shutdown
This command can be used to shutdown the operation of IP500 V2 unit memory cards.
This action or a system shutdown must be performed before a memory card is removed from the
unit. Removing a memory card while the system is running may cause file corruption. Card
services can be restarted by either reinserting the card or using the Start Up command.
Shutting down the memory card will disable all services provided by the card including Embedded
Voicemail if being used. Features licensed by the memory card will continue to operate for up to 2
hours.
Start Up
This command can be used to restart operation of an IP500 V2 memory card that has been shut
down. >The command will start the Select IP Office discovery process for selection of the system.
Related links
File > Advanced on page 57

File > Advanced > Launch Voicemail Pro

If the Voicemail Pro client is installed on the same PC as Manager, this link can be used to launch
the Voicemail Pro client. This can also be done by clicking on the icon in the Manager toolbar.
Related links
File > Advanced on page 57

File > Advanced > System Status

System Status is an application that can be used to monitor and report on the status of a system.
This is a separate application from Manager but if installed on the same PC, it can be started
using the File | Advanced | System Status link within Manager. Use of the application requires a
service user name and password configured on the system for System Status Access within the
system's security settings.
Related links
File > Advanced on page 57

File > Advanced > LVM Greeting Utility

This command launches a utility that can be used to convert .wav files to the formats used by
Embedded Voicemail (c11). The source file must be in the standard format used for all system
applications: PCM, 8kHz 16-bit, mono.
The resulting named greeting files can then be transferred to the Embedded Voicemail memory
card and selected as auto attendant greetings. That is done using the Recording Name field on
the Auto Attendant | Auto Attendant tab. The same named greeting file can be used in several
auto attendants.
The utility can be run separately using the file LVMGreeting.exe found in the LVMGreeting sub-
folder of the Manager application.

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Menu Bar Commands

Note:
The LVM Greeting Utility option is not selectable (grayed out) when Voicemail Pro is selected
as the system's voicemail type.
Related links
File > Advanced on page 57

File > Advanced > Generate WebLM ID

This menu is only used for Linux based systems. Any system being upgrading from pre-Release
10 ADI licenses must be migrated to PLDS licenses before upgrading. This is done using the files
created by the license migration tool. However, that tool assumes that the system will also be the
licence host. If instead the system is going to use PLDS license hosted by a WebLM server, the
system's web license server host ID is required in addition to the files created by the license
migration tool. The Generate WebLM ID tool provides that additional ID.
To generate the server’s Web License Server Host ID:
1. Click File | Advanced | Generate WebLM ID. The menu displayed varies depending on
whether the server is virtualized or not.
2. Enter the details of the server. For a virtualized server, the UUID can be obtained as
follows:
• Using the command line command: dmidecode -s system-uuid
• From the uuid.bios line of the virtual machines vmx file.
• From the VSphere client. See http://www-01.ibm.com/support/docview.wss?
uid=swg21682150. .
3. Click Generate.
Related links
File > Advanced on page 57

File > Advanced > Initial Configuration

Note:
The Initial Configuration utility changes the security settings. Therefore, the user running the
utility must have security read/write rights.
Standard Mode Initial Configuration
The Initial Configuration menu is displayed for all new or fully defaulted IP500 V2 systems. It
allows the required operating mode for the system to be selected.
.
For a system that you want to run in Essential Edition, Preferred Edition or Advanced Edition
modes, select IP Office Standard Mode.
For an IP500 V2 system to run in Standard Mode, its configuration must include an Essential
Edition license. A Standard Mode system without this license will not allow any telephony
functions.

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 File Menu

For a system that is being installed as an expansion server for a Server Edition solution, select
Server Edition Expansion.
Server Edition Initial Configuration
On a Standard Mode system, use the Initial Confguration option to convert the existing system
configuration into a Server Edition system configuration. It will effectively default the configuration
and reload it in Manager in Server Edition mode. Once Server Edition Expansion is selected as
the System Type, the Initial Configuration menu is displayed. If Server Edition Expansion is
selected in that menu, following selection of the various menu options, the system is rebooted as a
Expansion System (V2) for a Server Edition network.
For systems being configured for operation in a Server Edition solution, the Initial Configuration
menu is used to set or confirm a range of settings. The field shown and accessible in the form
depend on the selected System Type.
Once the menu is completed and Save is clicked, the values entered are written into the system
configuration and the system is restarted. The menu is also displayed when creating an offline
configuration for a Server Editionsystem. The configuration of an existing non-Server Edition
system can be converted to a Server Edition configuration, invoking this menu, using the File |
Advanced | Initial Configuration menu option.
System Type Indicate the type of sever role the system will perform.
Retain Configuration Data This option is shown for IP500 V2 units being converted to become
Expansion System (V2)s in a Server Edition solution.
If left unselected, the default, the existing configuration of the system is defaulted as per a
standard Server Edition expansion system.
If selected, the existing configuration is retained. However, some elements of that configuration
may be invalid or ignored in a Server Edition solution. It is the installers responsibility to ensure
that the final configuration is valid for use in the solution. For more information on IP500 V2
conversion, see Deploying Avaya IP Office™ Platform Server Edition.
Option Description
Hosted The option to select when deploying IP Office in a hosted environment. Selecting this
Deployment option sets HTTP directory to HTTPS. The default is unchecked.
System Name A name to identify this system. This is typically used to identify the configuration by the
location or customer's company name. Some features such as Gatekeeper require the
system to have a name. This field is case sensitive and within any network of systems
must be unique. Do not use <, >, |, \0, :, *, ?, . or /.
Locale This setting sets default telephony and language settings based on the selection. It also
sets various external line settings and so must be set correctly to ensure correct
operation of the system. See Avaya IP Office™ Platform Locale Settings. For individual
users the system settings can be overridden through their own locale setting (User | User
| Locale).
Table continues…

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Menu Bar Commands

Option Description
Services Device ID Set a Device ID for the system. This ID is displayed on the Solution View and System
Inventory pages and on the System | System tab in the configuration. The value can be
changed using the Device ID field on the System | System Events | Configuration tab. If
an SSL VPN is configured, , Avaya recommends that the Device ID match an SSL VPN
service Account Name. Each SSL VPN service account name has an associated SSL
VPN tunnel IP address. Having the displayed Device ID match an SSL VPN service
account name helps identify a particular SSL VPN tunnel IP address to use for remotely
managing IP Office.
LAN Interface This IP Address, IP Mask, Gateway and DHCP Mode settings can be set for the systems
two LANs, LAN1 and LAN2. These radio buttons are used to switch between displaying
the LAN1 details or the LAN2 details.
IP Address LAN1 Default = 192.168.42.1. LAN2 Default = 192.168.43.1.
This is the IP address of the Control Unit on LAN1. If the control unit is also acting as a
DHCP server on the LAN, this address is the starting address for the DHCP address
range.
IP Mask Default = 255.255.255.0. This is the IP subnet mask used with the IP address.
Gateway The address of the default gateway for routing traffic not in the same subnet address
range of the IP Address/IP Mask set above. A default IP Route for this address is added
to the systems configuration.
DHCP Mode: Default = Server.
This controls the control unit's DHCP mode for the LAN. When doing DHCP:
• LAN devices are allocated addresses from the bottom of the available address range
upwards.
• Dial In users are allocated addresses from the top of the available range downwards.
• If the control unit is acting as a DHCP server on LAN1 and LAN2, Dial in users are
allocated their address from the LAN1 pool of addresses first.
• Server When this option is selected, the system will act as a DHCP Server on this LAN,
allocating address to other devices on the network and to PPP Dial in users.
• Disabled When this option is selected, the system will not use DHCP. It will not act as a
DHCP server and it will not request an IP address from a DHCP server on this LAN.
• Dial In When this option is selected, the system will allocate DHCP addresses to PPP
Dial In users only. On systems using DHCP pools, only addresses from a pool on the
same subnet as the system's own LAN address will be used.
• Client When this option is selected, the system will request its IP Address and IP Mask
from a DHCP server on the LAN.
Server Edition The IP address of the Primary Server. This address is used to add an IP line to the
Primary Server Primary Server to the configuration.
Server Edition The IP address of the Secondary Server. This address is used to add an IP line to the
Secondary Server Secondary Server to the configuration.
Table continues…

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 File Menu

Option Description
DNS Server This is the IP address of a DNS Server. If this field is left blank, the system uses its own
address as the DNS server for DHCP client and forwards DNS requests to the service
provider when Request DNS is selected in the service being used (Service | IP).

Enterprise Branch Initial Configuration

The Initial Installation utility provides a default configuration and security settings that minimize
initial installation activities and maximize security. The system must be configured with the default
settings before the system can be administered by System Manager. This utility is used for new
installations and after an upgrade to enable System Manager administration of the IP Office.
1. Select File > Advanced > Launch Initial Installation Utility.
2. In the System Name field, enter the appropriate system name.
3. For the WAN Interface, select LAN1 or LAN2. If you select LAN1, the DHCP Mode is
disabled.
4. In the IP Address field, enter the appropriate IP address.
5. In the IP Mask field, enter the appropriate IP mask.
6. In the Gateway field, enter the appropriate gateway. Manager will create an IP route using
this gateway with the selected WAN as the destination.
7. In the DHCP Mode section, if you selected LAN1, select the appropriate DHCP option. If
you selected LAN2, DHCP Mode is disabled.
8. Select the Under Centralized Management? check box if you want the IP Office system
to be managed by System Manager.
9. If you selected the Under Centralized Management? check box, a number of additional
fields are shown, configure these additional fields as appropriate:
• SMGR Address - the IP address of the server running System Manager
• SNMP Community
• SNMP Device ID
• Trap Community
• SCEP Domain Certificate Name
• Certificate Enrollment (SCEP) Password
Select Save.
When you run the Initial Installation Utility, the Initial Installation utility also configures the following:
• System Status Interface (SSA) service security level – Unsecure only
• Configuration service security level –Secure, Medium
• Security Administration service security level – Secure, Medium
• OAMP Web Services service security level – Secure, Low (if locally administered)
• OAMP Web Services service security level – Secure, High (if administered by System
Manager)

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Menu Bar Commands

• Admin Client Certificate checks:-- High (if administered by System Manager)

• SCEP client active (if administered by System Manager)
• SCEP server IP address from SMGR IP address (if administered by System Manager)
• Legacy Program Code – Active (if locally administered)
If the system is administered by System Manager, the following is automatically configured:
• SNMP enabled
• SNMP trap destination 1 from System Manager IP address
• All SNMP traps active
• WebLM client active
• WebLM service address from System Manager IP address
• Remove all default extension users, leaving “NoUser” and “RemoteManager”
Related links
File > Advanced on page 57

File > Advanced > Add/Display VM Locals

This option is only displayed displayed when the configuration from an IP500 V2 systems with its
Voicemail Type set to Embedded Voicemail is received in Manager. It is not shown for off-line
configuration or configurations loaded from a PC file.
Selecting this option displays a list of the Embedded Voicemail prompt languages. Those
languages already present on the System SD card or not supported are greyed out. Additional
languages can be selected and then uploaded from Manager to the system.
When editing the system configuration in Manager, if the locale language selected for the system,
a user, a short code or an incoming call route is not already present on the System SD card,
Manager will display an error. Add/Display VM locales can then be used to upload the prompts
for the required language in order to correct the error.
You can reload languages that are already installed on the System SD card. For example, you
may want to reload the languages if new prompts have been added in a maintenance release. To
reload existing languages, upgrade the system (File | Advanced | Upgrade) with the Upload
System Files option checked. You can also choose Upload System Files from the Embedded File
Management utility (File | Advanced | Embedded File Management).
The Recreate IP Office SD Card command can be used to locally load all available languages
onto an SD card.
Related links
File > Advanced on page 57

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 File Menu

File > Backup/Restore

Backup Binaries and Configurations
This command copies all configuration files (.cfg) and software binary files (.bin) stored in
Manager's working directory to a selected folder.
Restore Binaries and Configurations
This command copies all configuration files (.cfg) and software files (.bin) stored in a selected
folder to the Manager's working directory.
Related links
File Menu on page 42

File > Import/Export

Export
This command allows you to export the selected parts of the configuration to either a set of CSV
text files (.csv) or a single binary file (.exp).
The display shows those exportable record types for which the configuration contains records. The
File Type and the Save In path can be selected at the base. The default location used is sub-
directory of the Manager application directory based on system name of the currently loaded
system.
Manager imports and exports CSV files using UTF8 character encoding which uses a double byte
to support characters with diacritic marks such as ä. Other applications such as Excel, depending
on the user PC settings, may use different single-byte encoding which will cause such characters
to be removed. Care should be taken to ensure that any tool used to create or edit a CSV
supports all the characters expected and is compatible with UTF8.
Import
This command allows you to import configuration settings. Two formats are supported. Binary files
(.exp) are settings previously exported from a system using File | Import /Export | Export. CSV text
files (.csv) can also be exported from a system or can be created using a plain text editor.
For the selected File Type and the Look In path, the window displays the file or files found. The
default location used is sub-directory of the Manager application directory based on system name
of the currently loaded system.
Manager imports and exports CSV files using UTF8 character encoding which uses a double byte
to support characters with diacritic marks such as ä. Other applications such as Excel, depending
on the user PC settings, may use different single-byte encoding which will cause such characters
to be removed. Care should be taken to ensure that any tool used to create or edit a CSV
supports all the characters expected and is compatible with UTF8.
Related links
File Menu on page 42

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Menu Bar Commands

File > Exit

The File | Exit command exits the Manager application.
Related links
File Menu on page 42

View Menu
View > Toolbars
Allows selection of which toolbars should be shown or hidden in configuration mode. A tick mark is
displayed next to the name of those toolbars that are currently shown.
View > Navigation Pane
Shows or hides the Navigation Pane. A tick mark appears next to the command when the pane is
shown.
View > Group Pane
Shows or hides the Group Pane. A tick mark appears next to the command when the pane is
shown.
View > Details Pane
Sets the location of the Details Pane when the Group Pane is also shown. The Details Pane can
be placed either below or to the right of the Group Pane.
View > Error Pane
Shows or hides the Error Pane. A tick mark appears next to the command when the pane is
shown.
View > Advance View
Causes Manager to switch from its simplified view to advanced view mode. Manager automatically
switches to advanced view mode if a Standard Edition configuration is loaded.
View > Simplified View
If Manager has no configuration loaded, this command switches it from advanced view to
simplified view.
View > TFTP Log
This command displays the TFTP Log window. This window shows TFTP traffic between Manager
and devices that uses TFTP to send and receive files. For example, the TFTP Log below shows
an Avaya IP phone requesting and then being sent its software files.
Related links
Menu Bar Commands on page 42

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 Tools Menu

Tools Menu
Related links
Menu Bar Commands on page 42
Tools > Extension Renumber on page 75
Tools > Line Renumber on page 75
Tools > Connect To on page 76
Tools > Export > User on page 76
Tools > SCN Service User Management on page 77
Tools > Busy on Held Validation on page 77
Tools > MSN Configuration on page 78
Tools > Print Button Labels on page 78
Tools > Import Templates on page 79
File > Advanced > Generate WebLM ID on page 79
Tools > License Migration on page 80

Tools > Extension Renumber

This command allows the extension numbering of user extensions to be changed. The existing
extension number range to be adjusted can be specified followed by the new start point for the
range after renumbering.
The command does not alter the extension number used for hunt groups but does adjust the
extension numbers of hunt group members.
Related links
Tools Menu on page 75

Tools > Line Renumber

On external trunks Line appearance ID numbers can be assigned to each channel supported in
order to allow that channel or line to be associated with a Line Appearance button on phones that
support button programming. By default all lines are automatically numbered from 701 upwards
when added to the system. This command allows the lines to be renumbered from a different
starting point.
Related links
Tools Menu on page 75

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Menu Bar Commands

Tools > Connect To

This option can be used to create H.323 IP line connections between two systems in a multi-site
network, one being the system with its configuration currently loaded in Manager, the other being
selected from a discovery dialog. This option is not available for Server Edition mode.
Important:
This process will require the systems to be rebooted.
Procedure
1. With the configuration of the first system received from that system and displayed in
Manager, clicking on or Tools | Connect To
2. A discovery menu is displayed and will list any other systems discovered.
3. Select the system to which connection is required.
4. Enter the login name and password for configuration access to that system.
5. Manager will switch to Small Community Network management mode, displaying the
configuration of both systems.
6. Click to save the new configuration back to each system.
Related links
Tools Menu on page 75

Tools > Export > User

This tool is used for systems being migrated to a enterprise environment managed from Avaya
System Manager. Refer to the “Migrating an IP Office or B5800 Branch Gateway to an IP Office
Enterprise Branch” manual for full details of the process. If you want to export a list of users and
extension numbers for other purposes use File > Import/Export.
When performing such a migration, users are not automatically created in System Manager. The
existing users must be exported to an XML file and then that file must be imported into System
Manager.
This feature allows you to export all or selected users users from the current configuration to an
XML file. The default filename is <SystemName>_Users.xml.
The Domain Name field is added to the user names as part of the user identity in the XML file.
Related links
Tools Menu on page 75

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 Tools Menu

Tools > SCN Service User Management

When managing multiple systems, it may be useful to create a common user name and password
on all the systems for configuration access. This tool can be used to create a new service user
account, SCN_Admin, for configuration access.
This tool is also available within a Server Edition network for administration of the Administrator
account.
This process requires you to have a user name and password for security configuration access to
each of the systems.
Procedure
1. The Select IP Office menu displays the list of discovered systems.
2. Select the systems for which you want to create a common configuration account.
Click OK.
3. A user name and password for security configuration access to each system is requested.
Enter the values and click OK. If the same values can be used for all systems enter those
values, select Use above credentials for all remaining, selected IPOs. If each system
requires a different security user names and password, deselect Use above credentials
for all remaining, selected IPOs.
4. The systems will be listed and whether they already have an SCN_Admin account is
shown.
5. To create the SCN_Admin account on each system and set the password for those
account click on Create Service User.
6. Enter the common password and click OK.
7. The password can be changed in future using the Change Password option.
8. Click Close.
Related links
Tools Menu on page 75

Tools > Busy on Held Validation

Busy on Held is a user feature where, when the user has a call on hold, the system indicate the
user as being busy to any further calls.
The use of Busy on Held in conjunction with multiple call appearance buttons is deprecated. This
command can be used to identify those users who have multiple call appearance buttons and for
whom Busy on Held is currently set.

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Menu Bar Commands

When run, it shows a list of the users affected and if selected their Busy on Held setting will be
switched off.
Related links
Tools Menu on page 75

Tools > MSN Configuration

Used to populate the Incoming Call Route table with a range of MSN or DID numbers.
Setting Description
MSN/DID The first number in the set of MSN numbers for which you have subscribed.
Note:
If you require to find an exact match between the MSN numbers and the
destination numbers, enter a minus (-) sign before the first MSN number.
Destination Where incoming calls with matching digits should be routed. The drop-down list
contains the extensions and groups on the system.
Line Group ID Specifies the incoming line group ID of the trunks to which the DID routing is
applied.
Presentation Digits Set to match the number of digits from the MSN/DID number that the central
office exchange will actually present to the system.
Range How many MSN or DID number routes to create in sequence using the selected
MSN/DID and Destination as start points. Only routing to user extensions is
supported when creating a range of records.

Related links
Tools Menu on page 75

Tools > Print Button Labels

This option is only enabled if a version of DESI software is also installed on the same PC as
Manager. It can then be used when a system configuration is loaded in Manager.
DESI software can be obtained from the Avaya support web site (http://support.avaya.com) or
from DESI (http//www.desi.com). Currently, though all users are shown, only ETR, M Series, T-
Series, 1400 and 1600 phones are supported by DESI templates.
The text used on the labels:
• If a text label has been added in the user's Button Programming settings, that text label is
passed to the DESI application.
• Note that the DESI application cannot import non-ASCII characters and may render them
incorrectly.

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 Tools Menu

• Manager will display a warning if it estimates that the user's current text for some buttons
may exceed the label space of the phone type.
• If no text label has been set, the default label for the action currently assigned to the button is
passed to the DESI application.
• Once the labels are shown in the DESI application, the label text can be changed.
1. Load the configuration of the system for which you want to print button labels.
2. Select Tools and then Print Button Labels.
• Name/Extn These are the user name and extension number details of the users in the
system configuration currently loaded in Manager.
• Phone Type This field shows the type of phone, if known, that the user is currently
associated with. The drop down can be used to change the selection if required.
• Expansion Modules If the phone type supports additional button modules, this drop down
can be used to select the type and number of button modules.
• Print Extn This check box is used to select whether the phone button details should be
included in the output passed to the DESI software.
• Print BM1/Print BM2/Print BM3 These check boxes are used to select whether button
module button details should be included in the output passed to the DESI software. These
button will only be selectable if the user's Expansion Modules is set to the number of button
modules.
Click Print via DESIto transfer the information to the DESI application. Within DESI, edit the labels
as required and then print the labels.
Related links
Tools Menu on page 75

Tools > Import Templates

Manager can be used to import and use templates. See Working with Templates on page 585.
Templates are stored in a default location. This command can be used to select a folder containing
other template files and copy those files into the correct default folder.
Related links
Tools Menu on page 75

File > Advanced > Generate WebLM ID

This menu is only used for Server Edition systems. Any system being upgrading from pre-Release
10 ADI licenses must be migrated to PLDS licenses before upgrading. This is done using the files
created by the license migration tool. However, that tool assumes that the system will also be the
licence host. If instead the system is going to use PLDS licenses hosted by a WebLM server, the

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Menu Bar Commands

system's web license server host ID is required in addition to the files created by the license
migration tool. This tool generates that additional ID.
To generate the server’s Web License Server Host ID:
1. Click File > Advanced > Generate WebLM ID. The menu displayed varies depending you
indicate the server is virtualized or not.
2. Enter the details of the server:
• UUID: For a virtualized server, the UUID can be obtained as follows:
- Using the command line command: dmidecode -s system-uuid
- From the uuid.bios line of the virtual machines vmx file.
- From the VSphere client, see http://www-01.ibm.com/support/docview.wss?
uid=swg21682150.
3. Click Generate.
4. The system’s host ID is displayed. Copy and paste this value to a text file.
Related links
Tools Menu on page 75

Tools > License Migration

IP Office release 10 and higher supports only the Product Licensing and Delivery System (PLDS)
to manage license files. If you are upgrading from a previous release, you must migrate all of your
pre-R10 licenses (ADI, PLDS, mix of ADI/PLDS, virtual) to R10 PLDS licenses. The license
migration tool extracts all the licensing information from an IP Office system and saves it to a file.
This file can then be used prepare a software upgrade quote in the Avaya One Source
Configurator in order to obtain the required new PLDS R10 licenses.
For Server Edition deployments, the License Migration tool collects licensing information from
every node in the solution.
For a step by step procedure, see Migrating Licenses to PLDS on page 565.
Notes
• Ensure all licenses are loaded on the system before using the license migration tool to
extract the licensing information. For Server Edition deployments, ensure all nodes are online
in order to capture the current view of systems in the solution. This step must be completed
before preparing your R10 software upgrade quote in the the Avaya One Source
Configurator.
• License migration is supported on all IP Office modes, release 6.0 and higher.
• The license migration tool can only be used with an online configuration. The Tools >
License Migration option is disabled for offline configurations.
• The license migration tool is not available on UCM and Application servers. When you run
the license migration tool on a Server Edition server, the tool collects licensing information
from every node in the solution.

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 Security Mode Menus

• The generated file can be read but must not be edited. License migration will fail if the file has
been edited.
Note:
The License Migration Tool is not used for the upgrade of an SMGR WebLM license used in
certain pre-R10 Enterprise Branch deployments. For more information see the Avaya One
Source Configurator.
Related links
Tools Menu on page 75

Security Mode Menus

These commands are available when the Manager is in security configuration mode.
Security Settings > File > Open Security Settings
Displays the Select IP Office menu to select and load a system's security settings. This requires
entry of a user name and password with rights to access security settings of the selected system.
This behavior changes when configuration settings have already be received from a system using
a service user name and password that also has security access rights for that system. In that
case, the system's security settings are automatically loaded without requiring name and
password entry.
Security Settings > File > Close Security Settings
Close the currently open set of security settings received from a system without saving those
settings.
Security Settings > File > Save Security Settings
Send edited security settings back to the system. Requires re-entry of a service user name and
password with access rights for security settings.
Security Settings > File > Reset Security Settings
Reset the security settings of the selected system to defaults. Requires entry of a service user
name and password with access rights for resetting the security settings. This option is not usable
while a set of security configuration settings is loaded.
The command File | Advanced | Erase Security Settings (Default) performs the same action
from Manager configuration mode.
Security Settings > File > Preferences
Displays a window for configuring various aspects of Manager's operation. The window is divided
into a number of tabs.
For a description of the Preferences options, see File > Preferences.
Security Settings > File > Configuration
Returns Manager to configuration mode.

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Menu Bar Commands

Security Settings > File > Exit

This command closes Manager.
Related links
Menu Bar Commands on page 42

Embedded File Management Menus

For control units with a memory card installed, the contents of the card can be viewed using
Manager. This view can also be used to add and remove files from the card. This may be useful
when the memory card is being used to store Music on Hold or IP phone firmware files.
• For non-Linux systems, the folders accessed using embedded file management are those on
the SD cards installed in the control unit.
• For Linux based systems, the folder path displays as disk in the embedded file manager.
That path maps onto /opt/ipoffice on the server.
Embedded Voicemail Files
When viewing the memory card, the files related to Embedded Voicemail are visible, however
these files are greyed out (ie. cannot be deleted, downloaded or overwritten).
• Mailbox greetings and messages are shown as .clp files.
• The language prompts for Embedded Voicemail functions are stored in separate language
sub-folders of lvmail. These are .c11 files.
• Named prompt files for use by Embedded Voicemail auto attendants are stored in the
lvmail\AAG folder and use the same .c11 or .c23 file formats as the language prompts.
These files can be created from standard .wav files before being downloaded to the memory
card by using the LVM Greeting Utility.
Avaya IP Phone Files
The memory card can be used as the source of files requested by IP Phones when rebooting. For
phones using system DHCP, once the files are loaded onto the card, the TFTP Server IP Address
and HTTP Server IP Address on the System | System tab must be set to match the system's LAN
address.
Viewing a Memory Card
When Advanced | Embedded File Management is selected, the Manager will go through normal
system discovery but will only allow selection of systems which can support a memory card. When
a system is selected, a valid service user name and password for configuration access to that
system is requested. If the system selected does not have a memory card installed, the files view
remains blank and the message TFTP:Received TFTP Error "Not Found" appears in Manager's
status bar.
Changing the Files View

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 Embedded File Management Menus

The type of display used in the Files pane can be changed by selecting from the View menu in
the toolbar.
Open File Settings
Select a system and display the contents of its memory cards if any are present and in use.
Close File Settings
Close the current memory card contents listing without exiting embedded file management mode.
Refresh File Settings
This command can be used to request a file update from the system.
Upload File
This command can be used to select and upload a file to the memory card in the system.
Upload System Files
This command is available with IP500 V2 systems. When this command is selected, Manager will
upload the software files for operation to the System SD card.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
• It copies the binary files for the system control unit and possible external expansion modules.
• It copies the firmware files used by phones supported by the system.
• For systems configured to run Embedded Voicemail, the Embedded Voicemail prompts for
those supported languages set as the system locale, user locales, incoming call route locales
and short code locales are upgraded. In addition the English language prompts are upgraded
as follows: IP Office A-Law/Norstar SD Cards - UK English, IP Office U Law/PARTNER SD
Cards - US English.
Backup System Files
This command is available with IP500 V2 systems. When selected, Manager copies the folders
and files from the System SD card's /primary folder to its /backup folder. Any matching files
and folders already present are overwritten. This action can be included as part of the system's
automatic daily backup process (System | System | Automatic Backup).
Restore System Files
This command is available with IP500 V2 systems. When selected, Manager copies the folders
and files from the System SD card's /backup folder to its /primary folder. Any matching files
and folders already present are overwritten.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
Upgrade Binaries
This command is available for IP500 V2 systems that have a system SD card and Optional SD
card installed.

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Menu Bar Commands

When this command is selected, all files except config.cfg and keys.txt files in the Optional SD
card's \primary folder are copied to the System SD card.

Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
Upgrade Configuration
This command is available for IP500 V2 systems that have a system SD card and Optional SD
card installed.
When this command is selected, any config.cfg and keys.txt files in the Optional SD card's
\primary folder are copied to the System SD card.

Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
Upload Phone Files
This command is available for IP500 V2 control units. When this command is selected, Manager
copies the software files relating to phone firmware to the memory card. For IP500 V2 control
units, use Upload System Files.
Copy System Card
This command is available for IP500 V2 systems that have an Optional SD card installed in
addition to the mandatory System SD card. When this command is selected, the system will copy
the folders and files on its System SD card to the Optional SD card. Any matching files and
folders already present on the Optional SD card are overwritten.
This process takes at least 90 minutes and can take longer.
Configuration
This command will exit Embedded File Management and return Manager to configuration editing
mode.
Related links
Menu Bar Commands on page 42

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Chapter 5: Manager User Interface

This section of the documentation covers the operation of Manager when being used to edit the
configuration of a system running in Standard Mode. Much of it is also applicable for when also
editing the configuration of systems running in Server Edition mode. Additional Server Edition Mode
functions are detailed in the next chapter.
Related links
Title Bar on page 85
Toolbars on page 85
The Navigation Pane on page 87
The Group Pane on page 88
The Details Pane on page 90
The Error Pane on page 92
The Status Bar on page 94

Title Bar
The Manager title bar shows the following information.
• The Manager application version.
• The system name of the system from which the currently loaded configuration was received.
• The software level of the system's control unit.
• The service user name used to receive the configuration and that user's associated operator
rights.
Related links
Manager User Interface on page 85

Toolbars
Manager displays the following toolbars:
• Main Toolbar

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Manager User Interface

• Navigation Toolbar
• Details Toolbar
Related links
Manager User Interface on page 85

The Main Toolbar

The Main toolbar is displayed at the top of the Manager window, just below the menu bar. This
toolbar is also available when Manager is in security mode. However many of the controls will not
function in security mode.

Open Configuration from a System Advertises to the address currently shown in the Manager's
title bar for any available systems. A list of responding systems is then displayed. When a system
is selected from this list, a valid user name and password must be entered. Equivalent to File |
Open Configuration.

Open Configuration File Open a configuration file stored on a PC. The button can be clicked to
display a browse window. Alternatively the adjacent arrow can be used to drop-down a list of the
last 4 previously opened configuration files. Equivalent to File | Offline | Open File.

Save Configuration File The action of this icon depends on whether the currently loaded
configuration settings were received from a system or opened from a file stored on PC. If the
former applies, the menu sending the configuration back to the system is displayed. In the latter
case, the file changes are saved to the original file. Equivalent to File | Save Configuration.

Collapse All Groups Causes all symbols in the navigation pane to be collapsed to symbols.

Show/Hide the Navigation Pane

Show/Hide the Group Pane

Show/Hide the Error Pane

Validate Configuration Runs a validation on all the currently loaded configuration settings. The
results appear in the error pane. By default the configuration is automatically validated when
loaded and changes are validated when made, however the validation preferences can be
changed through File | Preferences | Validation.

Create New Configuration Runs a series of dialogs that create a new configuration from
scratch.

Connect To For a standalone system, start the process of adding it to a multi-site network. Not
available in Server Edition mode.

Voicemail Pro Client Launch the Voicemail Pro client if also installed on the Manager PC.

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 The Navigation Pane

Server Edition Solution View Switch to the solution view. This option is only shown when
Manager is running in Server Edition mode.

The Navigation Toolbar

This toolbar provides drop down lists which can be used to navigate to particular records in the
configuration settings. The selected options in the navigation pane, group pane and the details
pane are synchronized with the navigation toolbar and vice versa. This toolbar is particularly
useful if you want to work with the group pane and or navigation pane hidden in order to maximize
the display space for the details pane.

This toolbar is not available when Manager is in security mode.

The Details Toolbar

This toolbar is shown in the top-right of the details pane. The options within the toolbar may vary
or be greyed out depending on the actions allowed for a particular configuration record.

Create a New Record The arrow is used to select the record type to be created. For
example; when adding an extension clicking may allow selection of a VoIP Extension or IP
DECT Extension.

Export as Template Save the current record as a tempate. The template can then be used to
create new records.
Delete Current Record Delete the currently displayed record.
Validate Current Record By default records are validated when opened and when edited. This
is set through the Manager application's validation settings.
< > Previous Record/Next Record Click < or > at the top-right to move to the previous or next
record.

The Navigation Pane

This pane shows icons for the different types of record that the configuration can contain. Each
type is followed by the number of records of that type already in the configuration. When Manager
is used in security mode, this pane is also used by Manager in security mode to display records
for security settings.

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Manager User Interface

Selecting an icon displays the matching records in the group pane, navigation toolbar and details
pane. Note that Manager is used to configure different types of system. Therefore the icons shown
may vary depending on the type of system you are configuring. For descriptions of the different
icons refer to Configuration Settings.
The information in the pane also depends on whether the group pane is visible or not. If the group
pane is visible, the navigation pane just shows icons for accessing which types of records should
be shown in the group pane. The group pane can then be used to select which of those records is
currently shown in the details pane. If the group pane is not visible, the navigation pane shows
icons for each type of records and under those icons for each individual record. The navigation
pane can then be used to select which of those records is currently shown in the details pane.
Related links
Manager User Interface on page 85

Expanding and Collapsing the Navigation Tree

Where or icons appear in the pane, they allow the structure to be expanded or collapsed.
When the group pane is hidden, and icons are shown for each record type and allow the
record type to be expanded to display all the existing records of that type.

The icon in the main toolbar can also be used to collapse all the expanded record types shown
in the navigation pane.

The Group Pane

This pane lists all the records that match the type selected in the navigation pane or navigation
toolbar. The list can be sorted by clicking on a column heading. Selecting a record in this pane
displays its details in the details pane.
The icons used in the pane may vary according to the state of the record. For example, some of
the users shown in this example have been configured for hot desking. This pane is also used by
Manager in security mode to display records for security settings.
Related links
Manager User Interface on page 85

Sorting the List

About this task
The records shown in the group pane can be sorted using any of the columns displayed.

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 The Group Pane

Procedure
1. To sort the list using the details in a particular column, click on the column header.
2. Clicking on the same column header again reverses the sort order.

Customizing the Columns Displayed

About this task
For each record type, which details are shown in the group pane can be customized. Also the
order of the column can be adjusted.
Procedure
1. Right-click on the pane and select Customize Columns.
2. To add a column, selects its name in the left-hand Available Columns list and click >>to
move it to the right-hand Selected Columns list.
3. To remove a column, select its name in the right-hand Selected Columns list and click <<
to move it to the left-hand Available Columns list.
4. To change the order of the Selected Columns, click on a column name and use the ^ and
V controls.
5. Click OK.

Changing the Column Widths

About this task
Procedure
1. In the column headers, place the cursor over the border between two columns.
2. When the cursor changes to a double headed arrow with a bar through it, click and hold
the cursor.
3. Drag the border to the required position and release the cursor.

Adding a New Record

About this task
The group pane can be used to add a new record of the type currently displayed.
Right-click on the pane and select New.
A arrow symbol next to New indicates that you can select a particular type of new record to
create. Click the arrow and select an option from the list.

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Manager User Interface

Procedure
1. Use the details pane to configure the new record.
2. Click OKin the details pane.

Deleting an Record
About this task
Procedure
1. Select the record to be deleted by clicking on it.
2. Right-click on the pane and select Delete.

Validating an Record
About this task
Procedure
1. Select the record to be validated by clicking on it.
2. Right-click on the pane and select Validate.

Show in Groups
About this task
This command groups the items shown in the group pane. The grouping method will vary
depending on the record type being listed. For example, short codes are grouped based on short
code feature type such as all forwarding short codes together.
Procedure
Right-click on the pane and select Show In Groups.

The Details Pane

Whenever a selection is made through the group pane or the navigation toolbar, the settings for
the matching record are shown in the details pane. This pane is also used by Manager in security
mode to display records for security settings.
The details are grouped into tabs. The tabs available may vary depending on what particular type
of record is being viewed.

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 The Details Pane

Individual settings may also be grayed out. This indicates that they are either for information only
or that they cannot be used until another setting is enabled.
The top-left icon indicates the following:
Locked Indicates that you can view the settings but
cannot change them.
Editable Indicates that you can change the settings
if required.
Changed Indicates that the settings have been
changed since the tab was opened. Click OK to
save the changes or Cancel to undo.

Various icons may appear adjacent to settings:

Locked Setting The setting cannot be changed
through this tab. This icon appears on user settings
where the user is associated with User Rights that
controls the setting.
Information Indicates a value which does not have
to be set but may be useful if set.
Warning A warning indicates a configuration setting
value that is not typical and may indicate
misconfiguration.
Error An error indicates a configuration setting
value that is not supported by the system. Such
settings may cause the system to not operate as
expected.

Related links
Manager User Interface on page 85
Managing Records on page 91

Managing Records
Procedure
1. Edit a record
a. The method of entering a record varies as different fields may use different methods.
For example text record boxes or drop down lists.
b. By default when changes are made, they are validated once another field is selected.
See File | Preferences | Validation.
c. Click on OK at the base of the details pane to accept the changes or click on Cancel
to undo the changes.

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Manager User Interface

2. Add a record.
a. Click at the top-right of the details pane.
b. Select the type of record required. For example, with extensions you can select from
H.323 Extension or SIP Extension.
3. Delete a record.
Click at the top-right of the details pane.
4. Validate a record.
Click at the top-right of the details pane.
5. Move to the previous or next record.
Click <or > at the top-right to move to the previous or next record.
6. Select a new tab.
a. To view the detail stored on a particular tab, click on the name of that tab.
b. If the tab required is not shown, use the controls if shown on the right to scroll
through the available tabs. The tabs available may vary depending on what particular
type of record is being viewed.
Related links
The Details Pane on page 90

The Error Pane

Validation is a process where Manager checks configuration records for errors or for values for
which it regards as requiring a warning. The results of this checking are shown by icons next to
the field that caused the error or warning, All errors and warnings are also listed in the Error Pane.
By default validation is performed automatically whenever a configuration file is opened and when
any field is edited. However, if required, the use of automatic validation can be controlled through
the settings on the File | Preference | Validation tab.
Icons
The icons used for errors and warnings are as follows. These are shown in the error pane and
also next to the related field in the details pane. In the details pane, the error or warning
description is shown when the cursor is hovered over the icon.

Error An error indicates a configuration setting value that is not supported by the system. Such
settings are likely to cause the system to not operate as expected.

Warning A warning indicates a configuration setting value that is not typical and may indicate
misconfiguration.

Information Typically indicates a setting which may be useful to set.

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 The Error Pane

Related links
Manager User Interface on page 85

Altering the Automatic Validation Settings

About this task
The settings for automatic validation are adjustable.
Procedure
1. Select File | Preferences.
2. Select the Validation tab.
Select the options required.
• Validate configuration on open Automatically validate configuration files when they
are opened in Manager.
• Validate configuration on edit Validate the whole configuration when OK is clicked
after editing a record. For large configurations, disabling this option removes the delay
caused by validating the configuration after every edit.
• Prompt for configuration validation on save or send If selected, when saving or
sending a configuration, a prompt is displayed asking whether the configuration should
be validated. If validation is selected and error are found, the send or save process is
canceled. This option is disabled if Validate configuration on edit is selected.
3. Click OK.

Revalidating Configuration Settings

About this task
If necessary, you can force a validation check of the whole configuration or of the current record
shown in the details pane.
Procedure
1. To validate the whole configuration, click in the main toolbar.
2. For a particular record, click in the details pane.

Viewing an Error or Warning

About this task
Procedure
1. Clicking on an error or warning in the error pane will load the matching record tab into the
details pane.

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Manager User Interface

2. The < and > can be used to move to the next error or warning in the error pane.

The Status Bar

The status bar at the base of the Manager screen is used to display icons and messages about
communications between Manager and systems. If the Manager is also acting as a BOOTP and
TFTP server it will also show BOOTP and TFTP messages.
A padlock icon is displayed whenever the Manager communications settings are set to secure.
This indicates all attempted configuration and security settings exchanged will be attempted over a
secure TLS link:
Status bar messages display information about communications the Manager application receives.
Some typical status bar messages are listed below.
Ready
This message is normally seen when Manager has just started and no configuration has been
received.
Received BOOTP request for 001125465ab2, unable to process
Manager is acting as a BOOTP server. It has received a BOOTP request that does not match a
system listed in its BOOTP records. The cause may be a device or application, other than an IP
Office, that also uses BOOTP.
TFTP: Received TFTP Error "NotFound" from 192.168.42.1
An attempt to receive settings from or send settings to the system failed. The most probable cause
is a name or password error.
TFTP: Received 17408 bytes for Marks_Test
Manager has received configuration settings from the named system using TFTP.
Sent 100% of C:\Program Files\Avaya\IP Office\Manager\b10d01b2_3.bin
Manager has sent the indicated file in response to a BOOTP request.
Related links
Manager User Interface on page 85

Configuring the Interface

The Manager configuration settings interface can be customized in a number of ways. These
changes are remembered the next time Manager is started.

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 Configuring the Interface

Related links
Manager User Interface on page 85

Resizing the Manager Window

About this task
When the Manager window is not maximized or minimized, it size can be adjusted.
Procedure
1. Place the cursor over the edge of the current window.
2. When the cursor changes to a double-headed arrow, click and hold the cursor.
3. Drag the edge to the required position and then release the cursor.

Moving the Border Between the Panes

About this task
The border between the visible panes can be adjusted. Note that this is a proportional rather than
exact position. If the whole window size is altered, the border position may also move.
Procedure
1. Place the cursor over the border between two panes.
2. When the cursor changes to a double headed arrow with a bar through it, click and hold
the cursor.
3. Drag the border to the required position and release the cursor.

Showing or Hiding Toolbars

About this task
The different toolbars can be hidden if not required.
Procedure
1. Select View and then Toolbars.
Those toolbars currently shown are indicated by a tick mark.
2. To show or hide a toolbar, click on its name.

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Manager User Interface

Moving Toolbars
About this task
The position of the Manager toolbars can be moved. Note that when moving a toolbar, the other
toolbars and panes may adjust their size or position to ensure that all the toolbar icons remain
visible.
Procedure
1. Place the cursor over the end of the toolbar.
2. When the cursor changes to a four-way arrow, click and hold the cursor.
3. Move the toolbar to the required position and release the cursor.

Showing or Hiding Panes

About this task
The details pane cannot be hidden. The navigation pane, group pane and error pane can be
shown or hidden. To do this use either of the following methods.
From the main toolbar, use the following icons:
• Hide/Show Navigation Pane.
• Hide/Show Group Pane.
• Hide/Show Error Pane.
or
Procedure
1. Select View.
Those panes currently shown are indicated by a tick mark.
2. To show or hide a pane, click on its name.

Changing the Position of the Details Pane

About this task
When the group pane is visible, the details pane is shown either below it or to its right. This
position can be adjusted.
Procedure
1. Select View and then Details Pane.

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 Configuring the Interface

2. The current position setting is indicated by a tick mark.

3. To select a position, click on it.

Changing the Size of Configuration Icons

About this task
The size of the icons used on the navigation pane and details pane can be adjusted.
Procedure
1. Select File and then Preferences.
2. Select the Visual Preferences tab.
3. Select the required icon size from Small, Medium or Large.
4. Click OK.

Changing Tab Display

About this task
For records with more than two tabs, you can select whether Manager should use controls or
arrange the tabs as multiple rows when necessary.
Procedure
1. Select Files | Preferences | Visual Preferences.
2. Select Multiline Tabs.
3. Click OK.

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Chapter 6: Working with the Server Edition
Manager User Interface

Related links
Server Edition Solution View on page 98
System Inventories on page 101
Default Settings on page 101
Record Consolidation on page 102
Telephone Features Supported Across Server Edition and SCN Networks on page 103

Server Edition Solution View

When the configuration of a Server Edition solution is loaded into Manager, Manager starts with
the Server Edition Solution View menu. This menu includes the system inventory of the servers,
links for launching various functions and a summary table of the servers and the links between the
servers.
Displaying the Server Edition Solution View
Manager normally starts with the Server Edition Solution View when the configuration for a Server
Edition network is loaded. However, if required, to return to the solution view do one of the
following.
• Click on the Server EditionSolution View icon in the toolbar.
• Click on the Solution icon in the navigation pane.
Interpreting and Using the Network Table
The table at the bottom of the solution view give a quick overview of all the servers and whether
their configuration was loaded into Manager.

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 Server Edition Solution View

Description This column describes the type of server being detailed by the row. It also includes a
status indicator for the configuration file that Manager has loaded for the server.
• Green - Configuration Loaded The configuration of the server has been
successfully retrieved and can be edited in Manager.
• Yellow - Offline Configuration Loaded The configuration loaded is an offline
configuration. This will appear for a server that has been added to the solution
when the physical server is not currently connected on the network and Create
Offline Configuration was selected. The offline configuration file is stored on and
retrieved from the primary server until it can be replaced by or replace the actual
server configuration.
• Red - Configuration Not Loaded There is no configuration for the system
loaded even though the solution configuration includes an entry for the server. This
will appear for a server that has been added to the network when the physical
server is not currently connected on the network and Create Offline
Configuration was not selected. If may also appear if the server is currently not
contactable.
• Grey - No Connection This icon is used in conjunctions with the others to
indicate that there is no current connection to the server. For example:
• In conjunction with a green icon, it indicates that the server for which a
configuration has been loaded cannot be detected on the network. This may be a
temporary issue caused by that particular server rebooting following a
configuration change.
• In conjunction with a red icon, it indicates that the server for which a configuration
has not been loaded has now been detected on the network. Saving and reloading
the solution configuration may resolve the issue.
Name This is the server name as taken from its configuration file. Offline is shown if no
configuration file is available.
Address The IP address of the server. This is the address that is used when Manager
attempts to retrieve the servers configuration when loading the solution
configuration.
Primary Link This value indicates the configuration settings of the H.323 IP trunk between the
primary server and the server indicated by the row. It should state Bothway. If it
states anything other, that indicates a mismatch in H.323 IP trunk configuration
between the system and the primary server. To correct this, right-click on the row
and select Connect to Primary.
Secondary Link This column is only shown after a secondary server has been added to the
configuration of the solution. The value indicates the configuration settings of the H.
323 IP trunk between the secondary server and the server indicated by the row. It
should state Bothway. If it states anything other, that indicates a mismatch in H.323
IP trunk configuration between the system and the secondary server. To correct this,
right-click on the row and select Connect to Secondary.
Users Configured This column summarizes the number of users (other than NoUser) configured on
the server. A total for the whole network is shown in the Solution row.
Table continues…

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Working with the Server Edition Manager User Interface

Extensions Configured This column summarizes the number of extensions configured on the server. A total
for the whole network is shown in the Solution row.

Right-clicking on a server in the table may present a number of action. The actions available vary
with the current state of the network configuration.
• Remove Remove the server from the solution configuration.
• Connect to Primary Repair the configuration of the H.323 IP trunks between the server and
the primary server.
• Connect to Secondary Repair the configuration of the H.323 IP trunks between the server
and the secondary server.
• Create Offline Configuration Create an offline configuration file for a server for which no
actual configuration has been loaded. The Offline Configuration menu will be displayed
followed by the Initial Configuration menu for the server type. The offline configuration file is
saved on the primary server.
Open...
The right side of the solution view contains links to open the following tools.
• Configuration
• System Status
• Voicemail Administration
• Resiliency Administration
• On-boarding
• IP Office Web Manager
• Help
Set All Nodes to Select
Use this command to implement Select licensing in a IP Office Server Edition Solution. All
systems in the solution must use the same licensing type.
Set All Nodes License Source
All systems in the Server Edition solution must use the same license source. The license source is
defined by the configuration setting License | License | License Source. Use this setting to set
all nodes to use the same license source.
Add...
Add a Server Edition Secondary Server or an Server Edition Expansion System.
When you add a system, IP Office Lines connecting the new system are configured with default
settings.
Important:
If the Manager setting File > Preferences > Preferences > SE Central Access is set to On,
an IP Office Line is not configured from the new system to the Server Edition Primary Server.
The status of the new system is Offline. You must configure an IP Office Line from the new
system to the Server Edition Primary Server.

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 System Inventories

Related links
Working with the Server Edition Manager User Interface on page 98

System Inventories
Manager can be used to display a system inventory for any of the servers in the Server Edition
solution. The system inventory is a quick summary of key settings and information about the
server. It can also display an overview system inventory for the whole Server Edition solution.
Displaying a Server's System Inventory
The method for displaying the system inventory depends on what is currently being displayed by
Manager.
In the Server Edition Solution View, using the table at the bottom of the menu, click on the server
for which you want to display the system inventory. Click on Network for the inventory of the
Server Edition network.
or
In the navigation pane, click on the icon of the server for which you want to display the system
inventory. Click on the Network icon for the inventory of the Server Edition network.
Related links
Working with the Server Edition Manager User Interface on page 98

Default Settings
Most of the defaults for systems in a Server Edition solution match those of individual IP Office
systems as detailed in the Configuration Settings section. The table lists some differences.
All auto-create extension and auto-create user settings for IP devices are set to off.
Settings Primary Server Secondary Server Expansion System
System Time Hidden. Time taken SNTP from the primary server.
Settings from host server.
Voicemail Voicemail Pro Centralized Voicemail to the primary server
Alarms Syslog relay all alarms Syslog relay all alarms to the primary server.
to the local host.
IP Address Specified during initial configuration menu.
Lines Physical – – Auto-created
Table continues…

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Working with the Server Edition Manager User Interface

Settings Primary Server Secondary Server Expansion System

IP Lines H.323 line to the H.323 line to the H.323 line to the
secondary and each primary and each primary and to the
expansion system. expansion system. secondary if present.
Backup to secondary. Backup to primary. No backup.
Extension Physical – – Auto-created but no
base extension setting.
IP None None None
User None None None
Hunt Group None Not allowed Not allowed
Incoming Call Route None Replicated from primary.
Directory Stored on the primary Configured to obtain system directory from the
primary server.
User Rights None Replicated from primary.
ARS 50:Main Short code to Short code to primary Short code to primary
secondary and to secondary if
present.

Related links
Working with the Server Edition Manager User Interface on page 98

Record Consolidation
By default, to maintain the configurations of the systems in a Server Edition solution in synch,
certain types of configuration records are consolidated. That is, they are replicated in the individual
configuration of each system in the network. Consolidation is applied to:
• Short Code System short codes only.
• Time Profile
• Account Code
• User Rights
• Location Though consolidated, the Emergency ARS and Fallback System field settings of
each location are configured separately at individual system level.
• Incoming Call Route For release 9.1 and higher, record consolidation is no longer applied to
Incoming Call Routes.
In Web Manager, consolidated records are shown at the top the Solutions page, under Solution
Objects. In Manager, operation of record consolidation is controlled by the File > Preferences >
Preferences setting Consolidate Solution to Primary Settings. By default that setting is
selected. The setting has the following effects.

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 Telephone Features Supported Across Server Edition and SCN Networks

If Consolidate Network to Primary Settings is selected:

• Entry and administration of consolidated records is performed only at the solution level
except for the Emergency ARS and Fallback System field settings of location records.
• Those records are then automatically replicated in the configurations of all the systems in the
solution but, except for locations, are still only visible and editable at the solution level.
• When the configurations are loaded or when this setting is changed to become selected, if
any inconsistency between records are found, a Consolidation Report is displayed. This
report allows selection of whether to update the system to match the primary or to update the
primary to match.
If Consolidate Network to Primary Settings is not selected:
• Entry and administration of consolidated records can be performed at both the solution and
individual system levels.
• Records entered and edited at the solution level are automatically replicated in the
configurations of all the systems in the solution. Each record displays a label on the record
indicating that it is a record that is shared across the solution.
• If a shared record is edited at the individual system level, that copy of the record is no longer
shared with the other systems. It will not be updated by any changes to the solution level
version of the same record.
• No consolidation checking for inconsistencies is done when the configurations are loaded.
Related links
Working with the Server Edition Manager User Interface on page 98

Telephone Features Supported Across Server Edition and

SCN Networks
Each system running IP Office in a multi-site network acts as a self-contained IP Office telephone
system. In addition to the remote systems sharing knowledge of user and hunt group extension
numbers, the following additional telephony features are supported between systems in a multi-
site network. Features not listed are not supported across the multi-site network.
• Absence Text
• Advertised Hunt Groups Hunt groups set to advertised can be dialed by users on other
systems
• Anti-tromboning Calls routed across the multi-site network and back to the originating
system are turned back into internal calls on the originating system only.
• Break Out Dialing
• Call Park / Unpark Call
• Call Pick-up Extension

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Working with the Server Edition Manager User Interface

• Call Tagging
• Callback When Free
• Centralized Call Log
• Centralized Personal Directory
• Conference
• Distributed Hunt Groups
• Distributed Voicemail Server Support When using Vociemail Pro, each system can support
its own Voicemail Pro server.
• Enable ARS / Disable ARS
• Extension Dialing Each system automatically learns the user extension numbers available
on other systems and routes calls to those numbers.
• Resiliency Options
• Fax Relay
• Follow Me Here / Follow Me To
• Forwarding
• Hold Held calls are signalled across the network.
• Internal Twining
• Intrusion Features
• Mobile Call Control Licensed mobile call control users who remote hot desk to another
system take their licensed status with them.
• Music On Hold Source Selection
• Remote Hot Desking
• Set Hunt Group Out of Service / Clear Hunt Group Out of Service
• Transfer Calls can be transferred to network extensions.
• User DSS/BLF Monitoring of user status only. The ability to use additional features such as
call pickup via a USER button will differ depending on whether the monitored user is local or
remote. Indication of new voicemail messages provided by SoftConsole user speed dial icon
is not supported.
• User Profile Resilence When a user hot desks to another system, they retain their Profile
settings and rights.
Related links
Working with the Server Edition Manager User Interface on page 98
Configuring Small Community Networking on page 769

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Chapter 7: Security Administration

The security settings are stored on the system and are separate from the system's configuration
settings. To change a system's security settings, Manager must first be switched to security mode
by selecting File | Advanced | Security Settings from the menu bar.
Security settings can only be loaded directly from a system. These settings cannot be saved as a
file on the local PC, nor do they appear as a temporary file at any time. You can optionally secure
the link between the system and Manager for configuration and security settings exchanges. By
default Manager and the system will always attempt to use the original, unsecured link.
Administration security is achieved using a number of optional cryptographic elements:
• Access control to prevent unauthorized use.
• Encryption to guarantee data remains private.
• Message Authentication ensures data has not been tampered with.
• Identity assures the source of the data.
Related links
Service Users, Application Roles, and Rights Groups on page 105
Access Control on page 119
Encryption on page 120
Message Authentication on page 121
Certificates on page 122
Implementing Security on page 123
SRTP on page 125

Service Users, Application Roles, and Rights Groups

In IP Office system, the managing, provisioning, and controlling the access to a system is based
on assigned Roles and Rights groups to a Service user. You can manage user access to the IP
Office system by assigning your Service users to various Rights groups. The Rights groups also
contain Application roles for IP Office Manager and IP Office Web Manager. The settings of these
roles determine the type of access a Service User has within the application. Roles and Rights
groups provide multiple levels of security to IP Office Manager and IP Office Web Manager. Each
Application role defines a set of permissions for a specific resource within the application. When
you assign an Application role to a Rights group and then assign Service users to that Rights

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Security Administration

group, you grant those service users all the access permissions that are defined by the Application
role.
Access to system settings is controlled by Service Users and Rights Groups stored in the
control unit's security settings. These are stored separately from the system's configuration
settings. All actions involving communications between Manager and the system require a service
user name and password. That service user must be a member of a Rights Group with
permissions to perform the required action.
Security Administrator: The security administrator can access the system's security settings
and the account cannot be removed or disabled.
In addition a further security setting can force this account to have exclusive security rights,
preventing another Service Users from security settings access.
Service Users: Each service user has a name, a password and is a member of one or more
Rights Groups. The accounts may be in one of a number of states, including enabled, disabled,
locked out and enforced password change.
IP Office supports a maximum of 64 Service Users.
Rights Groups: The Rights Groups to which a service user belongs determine what actions they
can perform. It can be thought of as a role, but has much more flexibility. Actions available to
Rights Groups include configuration, security actions and maintenance actions. Where a service
user has been configured as a member of more than one Rights Group, they combine the
functions available in the separate Rights Groups.
IP Office supports a maximum of 32 Rights Groups.
Application Roles: In addition to rights of IP Office service access, Rights Groups can also
contain ‘Roles’ for IP Office Manager and Web Manager; the settings of these roles determine
what rights of access the Service User has within that application. It allows more granularity of
access control within that application than the basic service access rights. For example the IP
Office configuration service has two basic rights of access: Read All and Write All. However the
Manager Operator roles can further constrain what can be written, viewed or edited.
Example Rights Assignment

In the above illustration:

• Service user X can read and write the configuration. However they can only edit Operator
settings and can only make changes that can be merged.
• Service user Y can read and write the configuration, edit all settings and make changes that
require reboots or merges. They can also access the Voicemail Pro settings.

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 Service Users, Application Roles, and Rights Groups

• Service user Z can read and write the configuration, edit all settings and make changes that
require reboots. They can also access the security and the Voicemail Pro settings.
• The Security Administrator can only access the security settings.
Changing Administrative Users and Rights Groups
IP Office Manager and Web Manager allow modification of Service Users and Rights Groups.
Prior to any change, the following should be considered:
• A Server Edition or multi-site IP500 V2 deployment must have consistent Service Users and
Rights Groups. IP Office Manager and IP Office Web Manager have synchronization tools to
assist.
• All changes must follow security best practices such as password policy and minimal rights of
access.
Security Settings on Upgrade
When the IP Office system is upgraded and new rights groups or services added, existing users
will only be granted the new rights if the Service Users’ accounts are at default. This prevents
unexpected changes of rights on upgrade. If access to these new rights or services are required,
they must be added manually after the upgrade process has been completed.
Default Service Users and Rights Groups
For IP Office Release 11.0 and prior releases, the default Service Users and Rights Groups
remain the same. For more information see Default Service Users and Rights Groups for IP Office
R 11.0 and earlier on page 114. A new access management is available to users on fresh
installations of Powered by R3.0.3, Powered by R3.0.4, and IP Office Release 11.0 FP4. However,
systems upgraded from previous versions to the these IP Office releases will still have the older
security settings. In fresh deployments, the only enabled account by default will be Administrator
for which the password must be changed on the first login. Administrator can then enable other
default user accounts by using IP Office Manager security settings or using the Service User
screen on IP Office Web Manager. The new access management has fewer Service user
accounts compared to the previous releases.
Related links
Security Administration on page 105
Default Service Users and Rights Groups on page 107
Default Service Users and Rights Groups for IP Office R 11.0 and earlier on page 114

Default Service Users and Rights Groups

The following information is applicable for IP Office R11.0 FP4 and Powered by Avaya Release
3.0.3 and 3.0.4.
Security Administrator Account
The following Security Administrator account is present on first startup and security settings reset.

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Security Administration

Name Default Usage Rights Group Notes

Account Membership
Status
Security Enabled, This is the default security Implied all Cannot be removed or
administration account. security rights disabled
Force
password Has all rights to all security
change management and
maintenance services

Service User Accounts

The following Service User accounts are present on first start-up and security settings reset.
Name Default Usage Rights Group Notes
Account Membership
Status
Administrator Enabled, This is the default account Administrator, This Service User should
used for system System Status, not be removed, disabled,
Force
configuration using the IP Business Partner or renamed.
password
Office and Web Manager
change
applications, including one-
X Portal/Voicemail Pro
administration.
Has all rights to all
management and
maintenance services
including security settings.
BusinessPartner Disabled Similar access rights to Business Partner This Service User can be
Administrator and can be removed or disabled if not
Force
used as a separate required.
password
account for Business
change
Partners
CustomerAdmin Disabled A Customer Administrator Customer Admin -
Service User can edit the
Force
system configuration
password
settings.
change
MCMAdmin Disabled The MCMAdmin Service MCM Admin -
User can administer IP
Office using Cloud
Operations Manager and
can write own Service User
password.
Table continues…

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 Service Users, Application Roles, and Rights Groups

Name Default Usage Rights Group Notes

Account Membership
Status
EnhTcpaService Enabled This account is used for TCPA Group Although not enforced, the
®
Avaya one-X Portal password should be
connections to the IP Office changed as soon as
system. possible in both IP Office
and Avaya one-X® Portal.
Enable this Service User
only when Avaya one-X®
Portal is deployed.
DirectoryService Enabled This account has Read and Directory Group -
Write service access rights
to Directory Read and
Directory Write in HTTP.
IPDECTService Disabled This account is used for IPDECT Group Enable only when DECT
DECT R4 system R4 deployed and
provisioning provisioning mode active

Rights Groups
The following Rights Groups are present on first start-up and security settings reset.
Name Usage Rights Group Notes
User
Administrator Allows full configuration and Administrator All IP Office Manager operations are
Group security access to the IP permitted
Office Manager and IP
Office Web Manager
application to configure the
system.
Business Partner To enable effectively Business Partner • Can only create or modify Customer
management of hosted IP Admin users
Office systems.
• Will not have any security
configuration write related rights on
fresh install or upgrading from
default.
MCM Admin Service Monitor Read using MCM Admin Used for IP Office administration
Cloud Operations Manager. through Cloud Operations Manager.
Allows writing own Service
User password and
Upgrade.
Table continues…

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Security Administration

Name Usage Rights Group Notes

User
Customer Admin Provides a targeted view to Customer Admin Customer Admin Service Users can log
end customers who have in and operate on port 8443 only and
little or no prior IP Office hence will be working through Web
experience. The Web Service service rights and not HTTP
Manager Right of Customer service right in Powered by Avaya
Admin will restrict the Web deployments. Customer Admin cannot
Manager view and access to use options that would require a restart
a subset of configuration. of the system for the new option to
apply (offline-mode) and cannot access
the security settings.
Customer Admin will not have security
configuration related rights and hence
will have access to limited configuration
screens. Lines will also not be
accessible objects. Customer Admin
can change its own password using
Web Manager and download the root
CA certificates.
IP500 V2 specific objects that will not
be part of Customer Admin account in
IP Office Powered by Avaya solution -
IP Routes, WAN Ports, Firewall
Profiles, RAS, Services, Tunnel
(applicable for IP500 V2 only)
TCPA Group This group is used by the EnhTcpaService -
one-X Portal for IP Office
application.
IPDECT Group This group is used by the IPDECTService -
DECT R4 master base
station to extract DECT
settings from IP Office.
Management API Allows limited access to the - -
Group IP Office Manager
application to configure the
system.
Directory Group Read and write service DirectoryService Used for HTTP directory access rights.
access rights to Directory If the access rights to Directory Read
Read and Directory Write in and Directory Write are disabled,
HTTP. authentication is enforced.

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 Service Users, Application Roles, and Rights Groups

Rights Group Assignment

Service Access Right Rights Group
• 1 = Administrator • 5 = TCPA Group
Group
• 6 = IPDECT Group
• 2 = Business Partner
• 7 = Management API Group
• 3 = Customer Admin
• 8 = Directory Group
• 4 = MCM Admin
1 2 3 4 5 6 7 8
Configuration Read all
configuration
Write all
configuration
Merge
configuration
Default
configuration
Reboot/Shutdown
immediately
Reboot when free
Reboot at time of
day
Configuration > Read Only
Manager Administrator
Operator
Rights Operator
Manager
User & Group Edit
User & Group
Admin
Directory &
Account Admin
Time & Attend
Admin
ICR & User Rights
Admin
Security Admin Read all security
settings
Write all security
settings
Reset all security
settings
Table continues…

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Security Administration

Service Access Right Rights Group

• 1 = Administrator • 5 = TCPA Group
Group
• 6 = IPDECT Group
• 2 = Business Partner
• 7 = Management API Group
• 3 = Customer Admin
• 8 = Directory Group
• 4 = MCM Admin
1 2 3 4 5 6 7 8
Write own service
user password
System Status System Status
Access
Read all
configuration
System Control
Sys Monitor
Telephony APIs Enhanced TSPI
DevLink3
Location API
HTTP DECT R4
Provisioning
Directory read
Directory write
Web Services WS Security Read
All
WS Security Write
All
WS Write Basic
Security
WS Security Write
Own Password
WS Config Read
All
WS Config Write
All
WS Config Read
Basic
WS Config Write
Basic
Backup
Table continues…

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 Service Users, Application Roles, and Rights Groups

Service Access Right Rights Group

• 1 = Administrator • 5 = TCPA Group
Group
• 6 = IPDECT Group
• 2 = Business Partner
• 7 = Management API Group
• 3 = Customer Admin
• 8 = Directory Group
• 4 = MCM Admin
1 2 3 4 5 6 7 8
Restore
Upgrade
Service Monitor
Read
Web Manager File Manager
Rights Service
Commands
Users, Extensions
Groups, Auto
Attendant
Incoming Call
Routes, Alternate
Route Selection,
Short Codes
System, Locations,
Time Profiles, and
Licensing
Lines
Directory,
Authorization
Codes, Account
Codes
IP Routes, WAN
Ports, Firewall
Profiles, RAS,
Services, Tunnel
(Applicable for
IP500 V2 only).
User Rights
Service Change
External Voicemail Pro
Basic
Voicemail Pro
Standard
Table continues…

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Security Administration

Service Access Right Rights Group

• 1 = Administrator • 5 = TCPA Group
Group
• 6 = IPDECT Group
• 2 = Business Partner
• 7 = Management API Group
• 3 = Customer Admin
• 8 = Directory Group
• 4 = MCM Admin
1 2 3 4 5 6 7 8
Voicemail Pro
Administrator
One-X Portal
Administrator
one-X Portal Super
User
Web Control
Administrator
Web Control
Security
WebRTC Gateway
Administrator
Management API
Read
Management API
Write
Media Manager
Administrator
Media Manager
Standard
one-X CTI API

Related links
Service Users, Application Roles, and Rights Groups on page 105

Default Service Users and Rights Groups for IP Office R 11.0 and
earlier
Security Administrator Account
The following Security Administrator account is present on first startup and security settings reset.

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 Service Users, Application Roles, and Rights Groups

Name Default Usage Rights Group Notes

Account Membership
Status
Security Enabled, This is the default security Implied all Cannot be removed or
administration account. security rights disabled
Force
password Has all rights to all security
change management and
maintenance services

Service User Accounts

The following Service User accounts are present on first start-up and security settings reset.
Name Default Usage Rights Group Notes
Account Membership
Status
Administrator Enabled, This is the default account Administrator, Should not be removed or
used for system System Status, disabled
Force
configuration using the IP Business Partner
password Should not be renamed
Office and Web Manager
change
applications, including one-
X Portal/Voicemail Pro
administration.
Has all rights to all
management and
maintenance services
including security settings.
EnhTcpaService Enabled This account is used for TCPA Group Although not enforced, the
one-X Portal for IP Office password should be
connections to the system. change as soon as possible
in both IP Office and one-X
Portal
Enable only when one-X
Portal deployed
IPDECTService Disabled This account is used for IPDECT Group Enable only when DECT
DECT R4 system R4 deployed and
provisioning provisioning mode active
BranchAdmin Disabled This account is used for SMGR Admin Enable only when SMGR
System Manager (SMGR) deployed; will be enabled
access in a branch when the Initial
deployment Configuration Utility (ICU)
run and SMGR
administration selected.
Must not be renamed
Table continues…

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Security Administration

Name Default Usage Rights Group Notes

Account Membership
Status
BusinessPartner Disabled Similar access rights to Business Partner Should be removed/
Administrator and can be disabled unless required
used as a separate
account for Business
Partners
Maintainer Disabled Maintenance account Maintainer Should be removed/
without edit configuration or disabled unless required
security access.
Can be used for Manager
(read-only), Web Manager
(read-only), System Status
Application (SSA), Backup/
Restore, System Monitor,
Upgrade

Rights Groups
The following Rights Groups are present on first start-up and security settings reset.
Name Usage Rights Group Notes
User
Administrator Allows full access to the IP Administrator All IP Office Manager operations are
Group Office Manager application permitted
to configure the system.
No security or maintenance
access
Manager Group Allows limited access to the – All IP Office Manager operations
IP Office Manager permitted except:
application to configure the
• Delete Short Code
system.
• View LAN2 Settings
Operator Group Allows limited access to the – All IP Office Manager operations
IP Office Manager permitted except:
application to configure the
• New object creations
system.
• View LAN2 Settings
• Delete Directory
• Delete ICR
System Status Allows limited access to the Administrator Sys Monitor access right only checked
Group SSA and Sys Monitor when using service users with Sys
applications. Monitor
Table continues…

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 Service Users, Application Roles, and Rights Groups

Name Usage Rights Group Notes

User
TCPA Group This group is used by the EnhTcpaService
one-X Portal for IP Office
application.
IPDECT Group This group is used by the IPDECTService
DECT R4 master base
station to extract DECT
settings from IP Office.
SMGR Admin This group is used by SMGR BranchAdmin Do not change the access rights
to configure IP Office.
Security Admin Allows access to security –
settings only
Backup Admin Allows access to all backup –
and restore services only,
including one-X Portal
Upgrade Admin Allows access to the – Allows upgrade of both IP Office
upgrade service applications and operating system
System Admin Allows configuration of IP –
Office, one-X Portal and
Voicemail Pro
Maint Admin Allows configuration of IP – Typically used for maintenance
Office, one-X Portal and personnel
Voicemail Pro along with
backup, restore and upgrade
Business Partner Full access to all Administrator,
configuration, security and BusinessPartner
maintenance services.
Customer Admin Web Management , one-X – No IP Office manager access
Portal and Voicemail Pro
administration
Maintainer Allows configuration view Typically used for maintenance
only, along with SSA, Sys personnel with no need for
Monitor backup, restore and configuration changes
upgrade
Directory Group Read and write service – Used for HTTP directory access rights.
access rights to Directory If the access rights to Directory Read
Read and Directory Write in and Directory Write are disabled,
HTTP. authentication is enforced.
MCM Admin Allows Backup, Upgrade, – Used for IP Office administration
Service Monitor Read using through Cloud Operations Manager.
Cloud Operations Manager.
Allows writing own service
user password

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Security Administration

Rights Group Assignment

Service Access Right Rights Group
• 1 = Administrator • 5 = TCPA Group
Group
• 6 = IPDECT Group
• 2 = Manager Group
• 7 = SMGR Admin
• 3 = Operator Group
• 4 = System Status
Group
1 2 3 4 5 6 7
Configuration Read all configuration
Write all configuration
Merge configuration
Default configuration
Reboot/Shutdown
immediately
Reboot when free
Reboot at time of day
Security Admin Read all security
settings
Write all security
settings
Reset all security
settings
Write own service user
password
System Status System Status Access
Read all configuration
System Control
Sys Monitor
Enhanced TSPI Enhanced TSPI
Access
HTTP DECT R4 Provisioning
Web Services Security Read All
Security Write All
Security Write Own
Password
Config Read All
Config Write All
Table continues…

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 Access Control

Service Access Right Rights Group

• 1 = Administrator • 5 = TCPA Group
Group
• 6 = IPDECT Group
• 2 = Manager Group
• 7 = SMGR Admin
• 3 = Operator Group
• 4 = System Status
Group
1 2 3 4 5 6 7
Backup
Restore
Upgrade
External Voicemail Pro Basic
Voicemail Pro Standard
Voicemail Pro
Administrator
One-X Portal
Administrator
one-X Portal Super
User
Web Control
Administrator
Web Control Security
WebRTC Administrator

Related links
Service Users, Application Roles, and Rights Groups on page 105

Access Control
Access to configuration, security settings and SSA is controlled by the use of service users,
passwords and Rights Groups. All actions involving communications between the Manager user
and the system require a service user name and password. That service user must be a member
of a Rights Group configured to perform the required action.

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Security Administration

In the example illustrated above:

• Service user X can read and write the configuration. However they can only edit Operator
settings and can only make changes that can be merged.
• Service user Y can read and write the configuration, edit all settings and make changes that
require reboots.
• Service user Z can read and write the configuration, edit all settings and make changes that
require reboots. They can also access the security settings.
• The Security Administrator can only access the security settings.
Granular access to configuration objects group in Web Manager
Using IP Office Manager, administrators can allow or restrict Service Users from accessing
configuration objects in IP Office Web Manager. Administrators can provide Read or Write access
to configuration depending on the role of Service User. If a Service User does not have read or
write access to a Configuration object, the objects do not appear on the interface when the
Service User logs in to IP Office Web Manager. When a Service user logs in to IP Office Web
Manager, system does a check on the Rights group provisioned for the Service User and shows
those menus that are enabled for the Rights group.
Related links
Security Administration on page 105

Encryption
Encryption ensures that all data sent by either the system or Manager cannot be ‘read’ by anyone
else, even another copy of Manager. Encryption is the application of a complex mathematical
process at the originating end, and a reverse process at the receiving end. The process at each
end uses the same ‘key’ to encrypt and decrypt the data:

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 Message Authentication

Any data sent may be optionally encrypted using a number of well known and cryptographically
secure algorithms:
Algorithm Effective key size (bits) Use
DES-40 40 Not supported.
DES-56 56 Not supported.
3DES 112 ‘Minimal’ security.
RC4-128 128 ‘Acceptable’ security.
AES-128 128 ‘Strong’ security.
AES-256 256 ‘Strong’ security.

In general the larger the key size, the more secure the encryption. However smaller key sizes
usually incur less processing. The system supports encryption using the Transport Layer Security
(TLS) v1.0 protocol. In addition, many cryptographic components of the TLS module have been
FIPS 140-2 certified, indicating the accuracy of implementation.
Related links
Security Administration on page 105

Message Authentication
Message authentication ensures that all data sent by either the system or Manager cannot be
tempered with (or substituted) by anyone else without detection. This involves the originator of the
data producing a signature (termed a hash) of the data sent, and sending that as well. The
receiver gets the data and the signature and check both match.

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Security Administration

Any data sent may be optionally authenticated using a number of well known and
cryptographically secure algorithms:
Algorithm Effective hash size (bits) Use
MD5 128 Not recommended.
SHA-1 160 ‘Acceptable’ security.
SHA-2 256, 384, 512 ‘Strong’ security

In general the larger the hash size, the more secure the signature. However smaller hash sizes
usually incur less processing.
IP Office supports message authentication using the Transport Layer Security (TLS) 1.0, 1.1, and
1.2 protocol. In addition, many cryptographic components of the TLS module have been FIPS
140-2 certified, indicating the accuracy of implementation.
Related links
Security Administration on page 105

Certificates
Public key cryptography is one of the ways to maintain a trustworthy networking environment. A
public key certificate (also known as a digital certificate or identity certificate) is an electronic
document used to prove ownership of a public key. The certificate includes information about the
key, information about its owner's identity, and the digital signature of an entity that has verified the
certificate's contents are correct. If the signature is valid, and the person examining the certificate
trusts the signer, then they know they can use that key to communicate with its owner.
For more information, see Certificate Management on page 566.
Related links
Security Administration on page 105

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 Implementing Security

Implementing Security
IP Office can be made a very secure. However, only a certain number of features are active by
default in order to ease the initial installation. If all Manager and system security settings are left at
default, no security mechanisms are active, other than the use of default service user names and
passwords. In addition, all legacy interfaces are active, and all configuration and security data is
sent unencrypted. Therefore, it is necessary to implement the configuration options listed here.
Additional setting may be necessary to further secure the individual deployment. Avaya is
presenting this information for guidance only; the customer is responsible for ensuring their
system is secure.
To improve IP Office security in practice, two main mechanisms are used:
• Activation of IP Office security features.
• Reduction of exposure to external or internal attack.
Minimum Security
A minimum security scenario could be where configuration data is open, but the security settings
are constrained: Any individual with the correct service user name and password can access the
configuration from any PC installation of Manager, no logging of access: Passwords can be
simple, and will never age.
• Change all default passwords of all service users and Security Administrator.
• Set the system Security Administration service security level to Secure, Low.
• Set the system service user Password Reject Action to None.
• Set the system Client Certificate Checks level to None (default).
• Set the system Minimum Password Complexity to Low (default).
• Set the system Previous Password Limit to zero (default).
• Set the system Password Change Period to zero (default).
• Set the system Account Idle Time to zero (default).
• Set certificate check level to low in Manager Security Preferences (default).
In addition, any PC installation of Manager can manage any IP Office.
Medium Security
A medium security scenario could be where both configuration and security settings are
constrained and a level of logging is required: Any individual with the correct service user name
and password can access the configuration from any PC installation of Manager: Passwords
cannot be simple, and will age.
• Change all default passwords of all service users and Security Administrator
• Set the system Security Administration service security level to Secure, Medium.
• Set the system Configuration service security level to Secure, Medium.
• Set the system service user Password Reject Action to Log to Audit Trail (default).
• Set the system Client Certificate Checks level to None (default).

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Security Administration

• Set the system Minimum Password Complexity to Medium.

• Set the system Previous Password Limit to non zero.
• Set the system Password Change Period to non zero.
• Set the system Account Idle Time to zero (default).
• Disable all the system Unsecured Interfaces.
• Set certificate check level to low in Manager Security Preferences (default).
Maximum Security
A maximum security scenario could be where both configuration and security settings are
constrained and a full level of logging is required: Certified individuals with the correct service user
name and password can access the configuration from specific PC installations of Manager:
Passwords cannot be simple, and will age: Manager can managed specific systems.
• Change all default passwords of all service users and Security Administrator
• Set the system Security Administration service security level to Secure, High.
• Set the system Configuration service security level to Secure, High.
• Set the system service user Password Reject Action to Log and Disable Account.
• Set the system Client Certificate Checks level to High.
• Set the system Minimum Password Complexity to High.
• Set the system Minimum Password Length to >8.
• Set the system Previous Password Limit to non zero (>5).
• Set the system Password Change Period to non zero.
• Set the system Account Idle Time to non zero.
• Set the system Session ID Cache to zero.
• Install valid, 1024 bits+, non self signed certificates (+private key) in all IP Office server
certificates, derived from a trusted certificate authority.
• Install the corresponding trusted CA certificate in each of the Manager’s windows certificate
stores.
• Install valid, 1024 bits+, non self signed certificate (+ private key) in all Manager Certificate
Stores.
• Install the corresponding certificates in all the system Certificate Stores of all permissible
Manager entities, and the trusted CA certificate.
• Disable all the system Unsecured Interfaces.
• Set Manager Certificate Checks level to high in Manager Security Preferences.
• Set Certificate offered to the system in Manager Security Preferences.
The above essentially locks the systems and corresponding Managers together. Only recognized
(by strong certificate) entities may communicate successfully on the service interfaces. All
services use strong encryption and message authentication.
The use of intermediate CA certificates can be used to overcome the limit of 6 maximum
certificates in each system Certificate Store.

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 SRTP

Related links
Security Administration on page 105

SRTP
Secure Real-Time Transport Protocol (SRTP) refers to the application of additional encryption and
or authentication to VoIP calls (SIP and H.323). SRTP can be applied between telephones,
between ends of an IP trunk or in various other combinations.
IP Office supports:
• Individual configuration for RTP and RTCP authentication and encryption
• HMAC SHA1 as the authentication algorithm
• AES-CM as the encryption algorithm
• 80 (default) or 32 bit authentication tag
• Master key length of 128 bits
• Master salt length of 112 bits.
Configuring the use of SRTP at the system level is done on the System | VoIP Security tab using
the Media Security setting. The options are:
• Best Effort
• Disabled (default)
• Enforced
When enabling SRTP on the system, the recommended setting is Best Effort. In this scenario, IP
Office uses SRTP if supported by the other end, and otherwise uses RTP. If the Enforced setting
is used, and SRTP is not supported by the other end, the call is not established.
The system level setting can be overridden at the trunk or extension level. This can be used for
special cases where the trunk or extension setting must be different from the system settings.
If the system level setting is Enforced, and devices that do not support SRTP are connected to
the system, their extension level configuration must be Disabled or calls will fail. This extra
configuration would typically not be required if the system level setting is Best Effort.
SRTP is supported on SIP Lines, SM Lines, and IP Office Lines. SRTP is not supported on H.323
IP trunks.
Encrypted RTCP
IP Office supports unencrypted RTCP by default. This default is compatible with most Avaya
endpoints which do not currently support encrypted RTCP. To the extent possible, any type of
endpoint using SRTP with IP Office should use unencrypted RTCP for consistency with other
endpoints to allow for direct media.
IP Office supports RTCP encryption as a configurable option. In addition to system level
configuration, it can be turned on at the trunk and extension level. Therefore, RTCP encryption
can be configured as an exception for an entity which only supports encrypted RTCP. In such case

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Security Administration

there will be no direct media SRTP between that entity and one that does not support encrypted
RTCP, and IP Office will relay the SRTP media.
Authentication
Authentication can be applied to both the voice part of calls (the RTP stream) and or to the control
signal associated with the call (the RTCP stream). By default, IP Office supports RTP encryption,
RTP authentication, RTCP authentication. Authentication is applied after encryption so that
packets can be authenticated at the remote end without having to be decrypted first.
• The method used for the initial exchange of authentication keys during call setup depends on
whether the call is using SIP or H.323. The IP Office system uses SDESC for SIP calls and
H235.8 for H.323 calls.
• SRTP is only supported when using an addition method such as TLS or a VPN tunnel to
establish a secure data path before call setup.
• A replay attack is when someone intercepts packets and then attempts to use them to for a
denial-of-service or to gain unauthorized access. Replay protection records the sequence of
packets already received. If a packed has been received previously, it is ignored. If packets
arrive outside a specified sequence range, the security device rejects them. All packets in a
stream (RTP and RTCP) have a sequential index number, however packets may not be
received in sequential order. SRTP protects against replay attacks by using a moving replay
window containing the index numbers of the last 64 authenticated packets received or
expected. Any packet received that has an index older than the current window is ignored.
Only packets with an index ahead of the window or inside the window but not already
received are accepted. Separate replay protection is used for the RTP and the RTCP
streams.
• Rekeying is the sending of new authentication keys at intervals during an secure call. This
option is not supported by the IP Office system which just sends authentication keys at the
start of the call.
SRTP sessions can use direct media between the devices or can be relayed via the IP Office
system. In some scenarios the IP Office system can be one end of the SRTP part of a call that
then continues to a non-SRTP destination.
If both the call originator and target require SRTP: A direct media is made if supported, using
SRTP. If direct media is not supported, the call is relayed via the IP Office system. In either case
SRTP parameters are negotiated end to end with the IP Office system translating and forwarding
them from one end to other end if necessary.
If only the originator or target requires SRTP: A non-direct media call is setup with with SRTP
negotiated between the IP Office system and the party which requires SRTP.
Emergency Calls
Emergency calls from an extension are not blocked even if SRTP is required but cannot be
established.
Calls using SRTP do not use any special indication on the user's telephone. Normal call functions
(conference, transfer, etc) remain available to the user. SRTP alarms and details of when SRTP is
being used are shown by the System Status Application and System Monitor.
Related links
Security Administration on page 105

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Chapter 8: Editing IP Office Security
Settings in Manager

The following conditions apply when editing the IP Office security settings.
• Editing of security settings may only be done online to a system.
No offline saving or editing is allowed for security purposes.
• No errors in the security settings are allowed to persist.
This prevents the system becoming inaccessible through operator error.
• Sets of changes to security objects may be made without the need for the OK button to be
selected every time.
This allows a coordinated set of changes to be accepted or canceled by the operator.

Loading Security Settings

About this task
The address ranges in which Manager searches for systems are set through the Manager
preferences (File | Preferences | Discovery). The security mechanism used for security settings
transfer between Manager and a system are set through the Secure Communications attribute of
Manager preferences (File | Preferences | Security).
If not already done, switch Manager to security mode by selecting File | Advanced | Security
Settings.
Note:
If the system's configuration settings have already been loaded using a service user name
and Password that also has security access, then the security settings are automatically
loaded when Manager is switched to security mode.
Procedure
1. If already in security mode, click in the main toolbar or select File | Open Security
Settings from the menu bar.
2. The Select IP Office window appears, listing those systems that responded.
The list can be sorted by clicking on the column names.

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Editing IP Office Security Settings in Manager

3. If the system required was not found, the address used for the search can be changed.
Enter or select the required address in the Unit/Broadcast Address field and then click
Refresh to perform a new search.
4. When the system required is located, check the box next to the system and click OK.
5. The user name and password request for the system is then displayed.
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.
6. If the security settings are received successfully, they appear within Manager.
• If the service user name/password is incorrect, or the service user has insufficient rights
to read the security settings, "Access Denied" is displayed.
• If the network link fails, or the secure communication mode is incorrect (for example
Manager is set to unsecured, but the system is set to secure only), "Failed to
communicate with IP Office" is displayed.

Saving Security Settings

About this task
Procedure
1. Click in the Main Toolbar or select File | Save Security Settings from the menu bar.
These options are only available when some change has been made.
2. The user name and password request for the system is then displayed.
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.

Resetting a System's Security Settings

About this task
Procedure
1. Select File | Reset Security Settings (if in security mode), or File | Advanced | Erase
Security Settings (if in configuration mode).
2. The Select IP Office window appears, listing those systems that responded.
The list can be sorted by clicking on the column names.
3. When the system required is located, check the box next to the system and click OK.
4. The user name and password request for the system is then displayed.

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 Resetting a System's Security Settings

Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.
5. Manager will indicate if the security settings are reset successfully.

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Chapter 9: Security Mode Field
Descriptions

The Manager Security Mode is used to load and edit the security settings of a system. How the
controls operate is similar to Manager in configuration mode.
To switch to Security Mode, select File | Advanced | Security Settings.
To switch back to Configuration Mode, select File | Configuration.
Security Mode Screen Elements
Table 2: Toolbar icons

Icon Action
Get the Security Settings

Save the Security Settings

Not Used in security mode

Show/Hide the Navigation Pane

Show/Hide the Group Pane

Not used in security mode

Not used in security mode

Security Settings Pane: This pane is used to select the type of security records that should be
displayed in the group pane or details pane.
•
General Defines general security controls for the system. When selected, the settings are
displayed in the details pane.
•
System Defines security settings for the system such as application access. When
selected, the settings are displayed in the details pane.
•
Services Secure services supported by the system. Currently these are access to security
settings and access to configuration settings.
•
Rights Groups Create groups with different access rights. When selected, the existing
Rights Groups are displayed in the group pane.

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 General Security Settings

•
Service Users Sets the name and password for an administrator. Also allows selection of
the Rights Groups to which the user belongs. When selected, the existing service users are
displayed in the group pane.
Group Pane: This pane is used to display the existing Right Groups or Service Users when those
options are selected in the security settings pane.
Details Pane: This pane shows the settings selected in the security settings pane or the group
pane.
Status Bar: This bar display messages about communications between Manager and systems. It
also displays the security level of the communications by the use of a padlock icon.
Related links
General Security Settings on page 131
System on page 136
Security Services Settings on page 144
Rights Groups on page 146
Service Users on page 152

General Security Settings

Related links
Security Mode Field Descriptions on page 130
General on page 131

General
Field Description
Security Administrator
The Security Administrator is a special service user who does not belong to any Rights Groups. The Security
Administrator is able to access the system's security settings but cannot access its configuration settings. By
default they are the only service user able to access to the security settings.
Unique Security Default = Off
Administrator
When selected, only the Security Administrator is able to access the system's security
settings. When this is selected, the security options for Rights Groups are disabled.
When not selected, the ability to access security settings can also be assigned to Rights
Groups.
Name: Default = 'security'. Range = 6 to 31 characters.
The name for the Security Administrator.
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Security Mode Field Descriptions

Field Description
Password Default = 'securitypwd'. Range = 8 to 31 characters.
The password for the Security Administrator. In order to change the Security
Administrator password, the current password must be known.
Minimum Password Default = Medium.
Complexity
The password complexity requirements for the Security Administrator. This setting is
active for attempted password changes on both Security Manager and the system. The
options are:
Low:
Any password characters may be used without constraint.
Medium:
The password characters used must include characters from at least 2 of the 'code
point sets' listed below. For example a mix of lower case and upper case. In addition, 3
or more consecutive identical characters of any type is not allowed.
High:
The password characters used must include characters from at least 3 of the 'code
point sets' listed below. For example a mix of lower case, upper case and numbers. In
addition, 3 or more consecutive identical characters of any type is not allowed.
Code Point Sets:
• Lower case alphabetic characters.
• Upper case alphabetical character.
• Numeric characters.
• Non-alphanumeric characters, for example # or *.
Previous Password Default = 4. Range = 0 (Off) to 10 records.
Limit (Entries)
The number of previous password to check for duplicates against when changing the
password. When set to 0, no checking of previous passwords takes place. This setting
is active for attempted password changes on both Security Manager and the system.
Phone Registration
Block Default IP Default: New systems: Checked, Upgraded systems: Clear
Phone Passcodes
If selected, existing IP phone registrations with default passcodes are not allowed in the
system. Administrators must type in passwords for registering the existing phones. If not
checked, existing IP phone registrations with default passcodes are allowed for
registration with the system. Allowing existing phones to register with default passcodes
pose a security risk as outsiders can access the system using those passcodes.
Service User Details
These settings control service user names and password/account policies. This setting is active for attempted
password changes on all administration interfaces.
Minimum Name Default = 6, Range 1 to 31 characters.
Length
This field sets the minimum name length for service user names.
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 General Security Settings

Field Description
Minimum Password Default = 8, Range 1 to 31 characters. This field sets the minimum password length for
Length service user passwords.
Password Reject Default = 3, Range 0 to 255 failures.
Limit
Sets how many times an invalid name or password is allowed within a 10 minute period
before the Password Reject Action is performed. Selecting 0 indicates never perform
the Password Reject Action.
Password Reject Default = Log and Temporary Disable.
Action
The action performed when a user reaches the Password Reject Limit. The options
are:
• No Action
• Log to Audit Trail Log to Audit Trail creates a record indicating the service user
account name and time of last failure.
• Log and Disable Account:. Log and Disable Account creates an audit trail record
and additionally permanently disables the service user account. This account can only
be enabled using the Security Manager Service User settings.
• Log and Temporary Disable: Log and Temporary Disable creates an audit trail
record and additionally temporarily disables the service user account for 60 seconds.
This account can only be enabled using the Security Manager Service User settings.
Minimum Password Default = Medium.
Complexity
The password complexity requirements for the Security Administrator. This setting is
active for attempted password changes on both Security Manager and the system. The
options are:
Low:
Any password characters may be used without constraint.
Medium:
The password characters used must include characters from at least 2 of the 'code
point sets' listed below. For example a mix of lower case and upper case. In addition, 3
or more consecutive identical characters of any type is not allowed.
High:
The password characters used must include characters from at least 3 of the 'code
point sets' listed below. For example a mix of lower case, upper case and numbers. In
addition, 3 or more consecutive identical characters of any type is not allowed.
Code Point Sets:
• Lower case alphabetic characters.
• Upper case alphabetical character.
• Numeric characters.
• Non-alphanumeric characters, for example # or *.
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Security Mode Field Descriptions

Field Description
Previous Password Default = 4. Range = 0 (Off) to 10 records.
Limit (Entries)
The number of previous password to check for duplicates against when changing the
password. When set to 0, no checking of previous passwords takes place. This setting
is active for attempted password changes on both Security Manager and the system.
Password Change Default = 0 (Off). Range 0 to 999 days.
Period
Sets how many days a newly changed password is valid. Selecting 0 indicates any
password is valid forever. This setting is active for password changes through this form
or prompted by Manager. Note that the user must be a member of a Rights Group that
has the Security Administration option Write own service user password enabled. If
this timer expires, the service user account is locked. The account may only be re-
enabled using the Service User Settings. To prompt the user a number of days before
the account is locked set a Expiry Reminder Time (see below).
Whenever this setting is changed and the OK button is clicked, the Security Manager
recalculates all existing service user password timers.
Account Idle Time Default = 0 (Off). Range 0 to 999 days.
Sets how many days a service user account may be inactive before it becomes
disabled. Selecting 0 indicates an account may be idle forever. If this timer expires, the
service user account is permanently disabled. The account may only be re-enabled
using the Service User Settings. The idle timer is reset whenever a service user
successfully logs in.
Whenever this setting is changed and the OK button is clicked, the Security Manager
recalculates all existing service user idle timers.
Expiry Reminder Default = 10. Range 0 (Off) to 999 days.
Time
Sets the period before password or account expiry during which a reminder indication if
the service user logs in. Selecting 0 prevents any reminders. Reminders are sent, for
password expiry due to the Password Change Period (above) or due to the Account
Expiry date (see Service User Settings on page 152) – whichever is the sooner.
Currently Manager displays reminders but System Status does not.

IP Office User Details

These settings control IP Office user password/account policies.
Password Default = On.
Enforcement
When enabled, password settings are enforced. When disabled, password
requirements are not enforced and the remaining settings are not editable
Minimum Password Default = 8, Range 1 to 31 characters.
Length
This field sets the minimum password length for user passwords
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 General Security Settings

Field Description
Minimum Password Default = Medium.
Complexity
The password complexity requirements for the Security Administrator. This setting is
active for attempted password changes on both Security Manager and the system. The
options are:
Low:
Any password characters may be used without constraint.
Medium:
The password characters used must include characters from at least 2 of the 'code
point sets' listed below. For example a mix of lower case and upper case. In addition, 3
or more consecutive identical characters of any type is not allowed.
High:
The password characters used must include characters from at least 3 of the 'code
point sets' listed below. For example a mix of lower case, upper case and numbers. In
addition, 3 or more consecutive identical characters of any type is not allowed.
Code Point Sets:
• Lower case alphabetic characters.
• Upper case alphabetical character.
• Numeric characters.
• Non-alphanumeric characters, for example # or *.
Password Reject Default = 5, Range 0 to 255 failures.
Limit
Sets how many times an invalid name or password is allowed within a 10 minute period
before the Password Reject Action is performed. Selecting 0 indicates never perform
the Password Reject Action.
Password Reject Default = Log and Temporary Disable. The action performed when a user reaches the
Action Password Reject Limit. The options are:
• No Action
• Log to Audit Trail Log to Audit Trail creates a record indicating the user account
name and time of last failure.
• Log and Disable Account Log and Disable Account creates an audit trail record and
additionally permanently disables the user account. The account can be enabled
using the Account Status field on the User | User page.
• Log and Temporary Disable Log and Temporary Disable creates an audit trail record
and additionally temporarily disables the user account for 60 seconds. The account
can be enabled using the Account Status field on the User | User page.

Related links
General Security Settings on page 131

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Security Mode Field Descriptions

System
Related links
Security Mode Field Descriptions on page 130
System Details on page 136
Unsecured Interfaces on page 138
Certificates on page 139

System Details
Field Description
Base Configuration
Services Base TCP Default = 50804. Range = 49152 to 65526.
Port
This is the base TCP port for services provided by the system. It sets the ports on which
the system listens for requests to access those services, using its LAN1 IP address.
Each service uses a port offset from the base port value. If this value is changed from
its default, the Manager application must be set to the same Base TCP Port through its
Services Base TCP Port setting (File | Preferences).
For information on port usage see the IP Office Port Matrix document on the Avaya
support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003
Maximum Service Default = 64.
Users
This is a fixed value for indication purposes only. This value is the maximum number of
service users that can be stored in a system's security settings
Maximum Rights Default = 32.
Groups
This is a fixed value for indication purposes only. This value is the maximum number of
Rights Groups that can be stored in a system's security settings.

System Discovery
System discovery is the processes used by applications to locate and list available systems. The IP Office can
be disabled from responding to this process if required. If this is done, access to the IP Office requires its
specific IP address to be used.
TCP Discovery Default = On.
Active
Selecting TCP Discovery Active allows the system to respond to those requests.
UDP Discovery Default = On.
Active
Selecting UDP Discovery Active allows the system to respond to those requests.

Security
These settings cover the per-system security aspects, primarily TLS settings.
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 System

Field Description
Session ID Cache Default = 10 hours, Range 0 to 100 hours.
This sets how long a TLS session ID is retained by the system. If retained, the session
ID may be used to quickly restart TLS communications between the system and a re-
connecting application. When set to 0, no caching takes place and each TLS
connection must be renegotiated.
HTTP Challenge Default = 10.
Timeout (Seconds)
For HTTP/HTTPS connection attempts, this field sets the timeout for connection
validation responses.
RFC2617 Session Default = 10.
Cache (Minutes)
For HTTP/HTTPS sessions, this field sets the allowed duration for successful logins as
per RFC2617.
Minimum Protocol Default = TLS 1.0
Version
This sets the TLS protocol version to be used in case of a TLS connection. If selected,
the TLS servers allow connections that meet the specified minimum requirement of the
selected protocol version and connections from a lower TLS version fails.
HTTP Ports
These settings set the ports for web based configuration access to the system.
HTTP Port Default = 80.
HTTPS Port Default = 443.
Web Services Port Default =8443.

WebSocket Proxy
These settings are applicable to WebSocket communication over IP Office lines.
Enabled Default = On.
When set to On , Web Manager uses the proxy server to communicate between the
Server Edition Primaryserver and other nodes.
When set to Off, the WebSocket proxy is disabled. Any IP Office line WebSocket
communication over an HTTP session is closed with “404 NotFound”.
Enforce Secure Default = On.
Applicable only when the Enabled check box is On.
When set to On, any proxy communication over IP Office line Websocket uses HTTPS.
When set to Off, any HTTPS communication over IP Office line Websocket is disallowed
and the session is closed with “403 Forbidden”.

Avaya Spaces API The API Key from Avaya Spaces. Use the Eye icon to view the key. To obtain the key,
Key log on to the Avaya Spaces account and browse to Zang Account > Manage
Companies > Company Profile > API Key > API Key.
Avaya Spaces Key The key secret from Avaya Spaces account. Use the Eye icon to view the key. To obtain
Secret the key secret, log on to the Avaya Spaces account and browse to Zang Account >
Manage Companies > Company Profile > API Key > View/Edit > Secret.

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Security Mode Field Descriptions

Related links
System on page 136

Unsecured Interfaces
These features relate to applications that access the system configuration settings using older
security methods.
Field Description
System Password Default = 'password'. Range = 0 to 31 characters.
The system password is used by Manager to upgrade IP Office IP500 V2 systems. Also
used for Monitor when the Monitor password setting is blank.
Voicemail Password Default = Blank. Range = 0 to 31 characters.
This password is required if a matching password is also set through the Voicemail Pro
client application. Typically no password is set.
Monitor Password Default = Blank. Range = 0 to 31 characters.
This password, if set, is used by the System Monitor application. If this password is not
set, those applications use the system password. If changing this password with no
previous password set, enter the system password as the old password.
Use Service User Default = Off.
Credentials
Set to On to enable log in to the System Monitor application using the service user
credentials.
Applications These check boxes control which actions the system will support for legacy
Controls applications. Different combinations are used by the different applications. A summary
of the applications affected by changes is listed in the Application Support list.
• TFTP Server: Default = On.
• TFTP Directory Read: Default = Off.
• TFTP Voicemail: Default = Off.
• Program Code: Default = On.
• DevLink: Default = On.
• TAPI/DevLink3: Default = Off.
• HTTP Directory Read: Default = On. Allow the system's current directory records to
be accessed using HTTP.
• HTTP Directory Write: Default = On. Allow HTTP import to be used to place
temporary directory records into the directory.
Application Support This panel is shown for information only. It indicates the effect on various applications of
the Application Controls selections.

Related links
System on page 136

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 System

Certificates
Additional Configuration Information
For additional information on certificates, see Certificate Management on page 566.
Services between the system and applications may, depending on the settings of the service
being used for the connection, require the exchange of security certificates. The system can either
generate its own certificate or certificates provided from a trusted source can be loaded.
Warning:
The process of 'on-boarding' (see Deploying Avaya IP Office™ Platform SSL VPN Services)
automatically adds a certificate for the SSL VPN to the system's security settings when the on-
boarding file is uploaded to the system. Care should be taken not to delete such certificates
except when advised by Avaya.
Configuration Settings
Field Description
Identity Certificate:
The Identity Certificate is an X.509v3 certificate that identifies the system to a connecting another device using
TLS, for example a PC running IP Office Manager set to Secure Communications.
By default, the system provides its own self-generated certificate, automatically generated when the system is
first installed. Alternatively, a certificate from another source can be uploaded to the system if required.
The system’s certificate is advertised (used) by Services which have their Service Security Level set to a value
other than Unsecure Only.
Offer Certificate Default = On.
This is a fixed value for indication purposes only. This sets whether the system will offer
a certificate in the TLS exchange when the IP Office is acting as a TLS server, which
occurs when accessing a secured service.
Offer ID Certificate Default = Off.
Chain
When set to On, this setting instructs IP Office to advertise a chain of certificates in the
TLS session establishment. The chain of certificates is built starting with the identity
certificate and adding to the chain all certificates it can find in the IP Office Trusted
Certificate Store based on the Common Name found in the "Issued By" Subject
Distinguished Name field in each of the certificates in the chain. If the Root CA
certificate is found in the IP Office Trusted Certificate Store, it will be included in the
chain of certificates. A maximum of six certificates are supported in the advertised chain
of certificates.
Issued to Default = IP Office identity certificate.
Common name of issuer in the certificate.
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Security Mode Field Descriptions

Field Description
Set This option can is used to load a certificate and associated private key. The certificate
and key must be a matching pair.
IP Office supports certificates with RSA key sizes of 1024, 2048 and 4096 bits. The use
of RSA key size 4096 may impact system performance. The recommended key size is
2048.
IP Office supports signature algorithms of SHA-1, SHA-256, SHA-384, and SHA-512.
Using signature size larger than SHA-256 may impact system performance. The
recommended signature algorithm is SHA-256.
The source may be:
• Current User Certificate Store.
• Local Machine Certificate Store.
• File in the PKCS#12 (.pfx) format
• Pasted from clipboard in PEM format, including header and footer text.
This method must be used for PEM (.cer) and password protected PEM (.cer) files.
The identity certificate requires both the certificate and private key. The .cer format
does not contain the private key. For these file types select Paste from clipboard and
then copy the certificate text and private key text into the Certificate Text Capture
window.
Using a file as the certificate source:
In Manager, when using the file option, the imported "p12" "pfx" or "cer" file for setting
the identity certificate can only contain the private key and identity certificate data. It
cannot contain additional Intermediate CA certificates or the Root CA certificate. The
Intermediate CA certificates or the Root CA certificate must be imported separately in
the IP Office Trusted Certificate Store.
This does not apply to Web Manager.

Note:
Web Manager does not accept the file of type "cer" with extension ".cer". This file
type can only be used in Manager.
View This command displays details of the current identity certificate. The certificate source,
details and valid dates are displayed.
The certificate view menu can also be used to install the certificate (but not its private
key) into the viewing PC’s local certificate store for use by the PC for secure connection
to the system or to export the certificate from the PC.
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 System

Field Description
Regenerate This command deletes the current identity certificate and generates a new self-signed
certificate.

Important:
Regenerating the certificate can take up to a minute during which system
performance may be impacted. Therefore it is recommended to only perform this
action during a maintenance window. The regeneration takes places after saving
the changes to security settings.
Clicking Regenerate opens the Regenerate Certificate window where you are prompted
to enter the following information:
• Signature: Default = SHA256/RSA2048.
This setting configures both the signature algorithm and the RSA key length to use
when generating the IP Office identity certificate. The options are:
- SHA256/RSA2048
- SHA1/RSA1024
If any other combinations are needed, the Security Administrator will need to
construct the IP Office identity certificate outside of Manager and use the Set action to
install it.
• Default Subject Name:
Default = Blank
Specifies a common name for the subject of this certificate. The subject is the end-
entity or system that owns the certificate (public key). Example:
ipoffice-0123456789AB.avaya.com.
If the field is blank, a system generated subject name is used.
• Subject Alternative Name(s):
Default = Blank
The Subject Alternative Name (SAN) field allows a list of alternate names to be bound
to the subject of the certificate.
The input field will allow the user to enter multiple Subject Alternate Names, each
separated by the comma “,” character. Each SAN consists of a PREFIX, followed by
the colon “:” character, followed by VALUE. The list of allowed PREFIX strings are
“DNS”, “URI”, “IP”, “SRV”, and “email”. The VALUE can be any text character except
the comma. (The comma is reserved as a field separator.) The input field has a
maximum size limit of 511 characters.
Example: ”DNS: ipoffice-0123456789AB.avaya.com, IP:
192.168.137.29, URI:http://avaya_example_url.com/,
email:jack@my_email_server.com”
Supported Subject Alternative Name types include: a DNS Name, a Uniform
Resource Identifier, an IP Address, an SRV record, or an electronic mail address.
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Security Mode Field Descriptions

Field Description
Although a PREFIX must be specified to select the type of name, no validation is
performed on the value.
If this field is blank, a system generated subject alternative name field value is used.
Certificate Expiry Default = 60, Range = 30 to 180
Warning Days
IP Office Manager can display a warning when a system’s security certificate is due to
expire. This setting is used to set the trigger for certificate warnings.

Use Different Default = None

Identity Certificate
The possible settings are None, SIP Trunks or SIP & SM Trunks, SIP Phones.
for SIP Telephony
When set to None, all secure telephony communications use the system’s default
identity certificate and settings.
When enabled, an extra set of options similar to those shown for Identity Certificate
section are displayed. These can be used to define the certificate used for secure
telephony communications. The certificate to use is uploaded to the system’s certificate
store using the Set button.
Received Certificate Default = None.
Checks
This setting is used for administration connections to the system by applications such as
(Management
Manager. When the Service Security Level of the service being used is set to High, a
Interface)
certificate is requested by the system. The received certificate is tested as follows:
• None: No extra checks are made (The certificate must be in date).
• Low: Certificate minimum key size 1024 bits, in date.
• Medium: Certificate minimum key size 1024 bits, in date, match to store.
• High: Certificate minimum key size 2048 bits, in date, match to store, no self signed,
no reflected, chain validation.
Received Certificate Default = None.
Checks (Telephony
This setting is used with IP telephony endpoints connecting to the system.
Endpoints)
This setting is used by IP Office to validate the identity certificate offered by the other
end of TLS connection. IP Office does not support mutual authentication for SIP phones
(an identity certificate is not installed in all SIP phones). Therefore, IP Office does not
require a client certificate from a SIP phone, only SIP/SM trunks.
The received certificate is tested as follows:
• None: No extra checks are made (The certificate must be in date).
• Low: Certificate minimum key size 1024 bits, in date.
• Medium: Certificate minimum key size 1024 bits, in date, match to store.
• High: Certificate minimum key size 2048 bits, in date, match to store, no self signed,
no reflected, chain validation.

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 System

Trusted Certificate Store

This section displays a list of the certificates held in the system’s trusted certificate store.
Installed Default = A set of fixed Avaya provided Intermediate CA or Root CA certificates.
Certificates
The certificate store contains a set of trusted certificates used to evaluate received
client certificates. Up to 25 X.509v3 certificates may be installed.
Add Add a trusted certificate. The source may be:
• Current User Certificate Store.
• Local Machine Certificate Store.
• File in one of the following formats:
- PEM (.cer)
- password protected PEM (.cer)
- DER (.cer)
- password protected DER (.cer)
• Pasted from clipboard in PEM format, including header and footer text.
This method must be used for PKCS#12 (.pfx) files. The PKCS#12 (.pfx) format
contains a private key and a trusted certificate cannot contain a private key. For this
file type, select Paste from clipboard and then copy the certificate text into the
Certificate Text Capture window.
IP Office supports certificates with RSA key sizes of 1024, 2048 and 4096 bits. The use
of RSA key size 4096 may impact system performance. The recommended key size is
2048.
IP Office supports signature algorithms of SHA-1, SHA-256, SHA-384, and SHA-512.
Using signature size larger than SHA-256 may impact system performance. The
recommended signature algorithm is SHA-256.
View View the currently selected certificate. The certificate (not the private key) may also be
installed into the local PC certificate store for export or later use when running the
manager in secured mode.
Delete Delete the currently selected certificate.

SCEP Settings
The Simple Certificate Enrollment Protocol is a protocol intended to ease the issuing of certificates in a network
where numerous devices are using certificates. Rather than having to individually administer the certificate
being used by each device, the devices can be configured to request a certificate using SCEP.
These settings are relevant for IP Office Branch deployments.
These settings are not used in IP Office Standard mode.
Active Default = Off.
Request Interval Default = 120 seconds. Range = 5 to 3600 seconds.
(seconds)
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Security Mode Field Descriptions

SCEP Server IP/ Default = Blank.

Name
SCEP Server Port Default = 80 for HTTP and 443 for HTTPS.
SCEP URI Default = /ejbca/publicweb/apply/scep/pkiclient.exe
SCEP Password Default = Blank.

Related links
System on page 136

Security Services Settings

This tab shows details of the services that the system runs to which service users can
communicate.
Field Description
Name The name of the service. This is a fixed value for indication purposes only.
Host System This field shows the system's name. This is a fixed value for indication purposes only.
Service Port This is the port on which the system listens for attempts to access the service. The
routing of traffic to this port may need to be enabled on firewalls and network devices
between the service users and the system. The base port (TCP or HTTP) for each
service is offset by a fixed amount from the ports set in System Settings.
For information on port usage see the IP Office Port Matrix document on the Avaya
support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003
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 Security Services Settings

Field Description
Service Security Sets the minimum security level the service will support. See File | Preferences |
Level Security for the corresponding Manager application setting, which must be changed to
match the appropriate service access security settings.

Warning:
If the system does not already have an X509 security certificate, selecting a setting
other than Unsecure Only will cause the system to stop responding for a period
(less than a minute) while the system generates its own unique security certificate.
The options are:
• Unsecure Only This option allows only unsecured access to the service. The service's
secure TCP port, if any, is disabled. This or disabled are the only options supported for
the System Status Interface and Enhanced TSPI services.
• Unsecure + Secure This option allows both unsecured and secure (Low) access. In
addition, TLS connections are accepted without encryption, just authentication.
• Secure, Low This option allows secure access to that service using TLS, and
demands weak (for example DES_40 + MD5) encryption and authentication or higher.
The service's unsecured TCP port is disabled.
• Secure, Medium This option allows secure access to that service using TLS, and
demands moderate (for example SHA-256) encryption and authentication or higher.
The service's unsecured TCP port is disabled.
• Secure, High This option allows secure access to that service using TLS and
demands strong (for example SHA-256) encryption and authentication, or higher. In
addition, a certificate is required from the client (usually Manager). See System Details
| Client Certificate Checks for tests made on the received certificate. The service's
unsecured TCP port is disabled.
• Disabled This option is available for the System Status Interface and Enhanced TSPI
services. If selected, access to the service is disabled.
Service Access For Server Edition systems, it is defaulted to Server Edition Manager. When set to
Source Server Edition Manager, the system can only be configured using Manager in its Server
Edition mode. When set to Unrestricted, the system can be configure using Manager in
its normal Simplified View or Advanced View modes.

Warning:
Opening the configuration of a Server Edition system in Manager running in any
mode other than Server Edition mode should be avoided unless absolutely
necessary for system recovery. Even in that case, Manager will not allow
renumbering, changes to the voicemail type and changes to H.323 lines.
For systems centrally managed using SMGR , it is defaulted to Avaya Aura System
Manager. When set to Avaya Aura System Manager, the system can only be
configured using SMGR in Branch Mode. When set to Unrestricted, the system can be
configured using Manager in its normal Simplified View or Advanced View modes.

Related links
Security Mode Field Descriptions on page 130

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Security Mode Field Descriptions

Rights Groups
Related links
Security Mode Field Descriptions on page 130
Group Details on page 146
Configuration on page 146
Security Administration on page 148
System Status on page 148
Telephony APIs on page 149
HTTP on page 149
Web Services on page 149
External on page 151

Group Details
This tab sets the name of the Rights Group.
Field Description
Name : Range = Up to 31 characters
The name for the Rights Group should be unique. The maximum number of rights groups
is 32.

Related links
Rights Groups on page 146

Configuration
This tab sets the configuration settings access for service user's who are members of this Rights
Group.
Field Description
IP Office Service This setting controls what action on the system can be performed by members of the
Rights Rights Group.
Manager This setting controls what types of configuration records Manager will allow members of the
Operator Rights Rights Group to viewed and what actions they can perform with those types of records.
Operator View/Edit/ New/Delete Configuration Record
Types
Administrator All View, edit create and delete
all configuration records.
Manager View View all except WAN Port.
Table continues…

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 Rights Groups

Field Description
Edit Extension, User, Hunt
New Group, Short Code, Service,
RAS, Incoming Call Route,
Directory, Time Profile,
Firewall Profile, IP Route,
Least Cost Route, Account
Code, ARS.
Delete As edit except Short Code.
Operator View View all except WAN Port.
Edit Extension, User, Hunt
Group, Short Code, Service,
RAS, Incoming Call Route,
Time Profile, Firewall Profile,
IP Route, Least Cost Route,
Account Code, License,
ARS.
New None.
Delete Delete Incoming Call Route
and Directory.
User & Group Edit View User and Hunt Group
Edit records only.

New None
Delete
User & Group Admin All User and Hunt Group
records only.
Dir & Account Admin All Directory and Account Code
records only.
Time & Attendant Admin All Time Profile and Auto
Attendant records only.
ICR & User Rights Admin All Incoming Call Route and
User Rights records only.
Read Only View View all configuration
records.
Edit None.
New
Delete

Related links
Rights Groups on page 146

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Security Mode Field Descriptions

Security Administration
This tab sets the security settings access for Service user's who are members of this Rights
Group. These settings are ignored and greyed out if a Unique Security Administrator has been
enabled in General Settings.
Field Description
Read all security Members of the Rights Group can view the system's security settings.
settings
Write all security Members of the Rights Group can edit and return changes to the system's security
settings settings.
Reset all security If selected, members of the Rights Group can reset the security settings to default values.
settings
Write own service If selected, members of the Rights Group can change their own password when requested
user password to do so by the system. That request may be the result of a Password Change Period,
Force new password or Account Expiry. The new password change is requested
automatically at login time.

Related links
Rights Groups on page 146

System Status
This tab sets whether members of the group can access the system using the System Status
Application (SSA).
Field Description
System Status If selected, members of the Rights Group can view the system's current status and
Access resources using the System Status Application (SSA).
Read all The System Status application includes tools to take a snapshot of the system for use by
configuration Avaya for diagnostics. That snapshot can include a full copy of the system's configuration
settings. This setting must be enabled for the SSA user to include a copy of the
configuration in the snapshot.
System Control If enabled, the SSA user is able to use SSA to initiate system shutdowns and memory
card shutdown/restarts.
SysMonitor If enabled, members of the Rights Group can use the System Monitor application to
Access perform detailed diagnosis of system problems.

Related links
Rights Groups on page 146

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 Rights Groups

Telephony APIs
Field Description
Enhanced TSPI If selected, applications in this rights group are able to use the system's Enhanced
Access TSPI interface. This interface is currently used by the one-X Portal application server
for its connection to the system.
DevLink3 If selected, applications in this rights group are able to use the system's DevLink3
interface.
This is a TCP based interface that streams real time call events (Delta3 records) and
is the recommended replacement to the existing DevLink windows based DLL. A new
Rights Group with a user name and password is required for external applications to
connect via the DevLink3 interface.
Location API If selected, applications in this rights group are able to use the system's Location API
interface.

Related links
Rights Groups on page 146

HTTP
This tab sets the HTTP services supported for members of the group.
Field Description
DECT R4 This service is used to allow the system to configure the DECT R4 master base
Provisioning station and to respond to handsets subscribing to the DECT R4 system. It requires
both the system and DECT R4 master base station to be configured to enable
provisioning. For full details refer to the DECT R4 Installation Manual.
Directory Read If selected, members of the Rights groups have HTTP service read access to
directory records.
Directory Write If selected, members of the Rights groups have HTTP service read and write access
to directory records.

Related links
Rights Groups on page 146

Web Services
These settings are used by users in rights groups using web services to configure and manage
the system. These are currently not used on Standard Mode systems

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Security Mode Field Descriptions

IP Office Service Rights

Field Description
Security Read All If selected, the rights group members can view system security settings.
Security Write All If selected, the rights group members can change system security settings.
Security Write If selected, members of the Rights Group can change their own password when
Own Password requested to do so by the system. That request may be the result of a Password Change
Period, Force new password or Account Expiry. The new password change is requested
automatically at login time.
Config Read All If selected, the rights group members can view system configuration settings
Config Write All If selected, the rights group members can change system configuration settings.
Backup If selected, the rights group members can initiate the system backup process.
Restore If selected, the rights group members can initiate the system restore process.
Upgrade If selected, the rights group members can initiate the system upgrade process.

Web Manager Rights

Field Description
File Manager If selected, the rights group members are assigned Read only access to embedded file
management configuration settings in Web Manager by default.
Service If selected, the rights group members are assigned Read only access to Service
Commands Commands configuration settings in Web Manager by default.
Users, Extensions If selected, the rights group members are assigned Read only access to Users and
Extension configuration settings in Web Manager by default.
Groups, Auto If selected, the rights group members are assigned Read only access to Groups, Auto
Attendant (For Attendant configuration settings in Web Manager by default.
Standard Edition
only)
Incoming Call If selected, the rights group members are assigned Read only access to Incoming Call
Routes, Alternate Routes, Alternate Route Selection, and Short Codes configuration settings in Web
Route Selection, Manage by default.
Short Codes
System, If selected, the rights group members are assigned Read only access to on System,
Locations, time Locations, Time Profiles, and Licensing configuration settings in Web Manager by default.
Profiles and
Licensing
Lines If selected, the rights group members are assigned Read only access to Lines
configuration settings in Web Manager by default.
‘Directory, If selected, the rights group members are assigned Read only access to Directory,
Authorization Authorization Codes, and Account Codes configuration settings in Web Manager by
codes, Account default.
Codes
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 Rights Groups

Field Description
IP Routes, WAN If selected, the rights group members are assigned Read only access to IP Routes, WAN
Ports, Firewall Portas, Firewall Profiles, RAS Services Users and Extension configuration settings in
Profiles, RAS, Web Manager by default.
Services, Tunnel
(Applicable for
IP500)
User Rights If selected, the rights group members are assigned Read only access to Users
configuration settings in Web Manager by default.

Related links
Rights Groups on page 146

External
These settings are used by users in rights groups for external components using web services to
configure and manage the system.
IP Office Service Rights
Field Description
Voicemail Pro If selected, the rights group members can read the configuration and perform backup,
Basic restore, and upgrade.
Voicemail Pro If selected, the rights group members can update the configuration and perform backup,
Standard restore, and upgrade.
Voicemail Pro If selected, the rights group members can update the configuration and security settings.
Administrator
one-X Portal If selected, the rights group members can update the configuration and security settings.
Administrator Does not include backup and restore.
one-X Portal If selected, the rights group members can perform backup and restore.
Super User
Web Control If selected, the rights group members can update the configuration settings.
Administrator
Web Control If selected, the rights group members can update the security settings.
Security
WebRTC Gateway If selected, the rights group members can update the configuration settings.
Administrator
Media Manager If selected, the rights group members can update Media Manager configurations and
Administrator settings. The rights group members can also access all archived recordings.
Media Manager If selected, the rights group members can have read-only access to Media Manager
Standard configurations and access to the recordings.
Table continues…

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Security Mode Field Descriptions

Field Description
Reporter If selected, the rights group members can have configuration access to Integrated Contact
Administrator Reporter.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Web Manager These rights are used with web service access to systems such as IP Office Web
Rights Manager.

Related links
Rights Groups on page 146

Service Users
These settings are displayed when Service Users is selected in the navigation pane and a
particular service user is selected in the group pane.
The maximum number of service users is 64.
Field Description
Name: Range = Up to 31 characters. Sets the service user's name.
The minimum name length is controlled through General settings.

Note:
If changing the user name and/or password of the current service user used to load
the security settings, after saving the changes Manager should be closed. Not
closing Manager will cause error warnings when attempting to send any further
changes.
Password: Range = Up to 31 characters. Sets the service user's password.
To change the current password click Change. Enter and confirm the new password. Note
that an error will be indicated if the password being entered does not meet the password
rules set through General settings.
To clear the cache of previous password details used by the password rules setting, click
Clear Cache. For example, if the rule restricting the reuse of old passwords has been
enabled, clearing the cache allows a previous password to be used again.
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 Service Users

Field Description
Account Status Default = Enabled.
Displays the current service user account status (correct at the time of reading from the
system). The options are:
• Enabled This status is the normal non-error state of a service user account. This setting
can be selected manually to re-enable an account that has been disabled or locked.
Note that re-enabling a locked account will reset all timers relating to the account such
as Account Idle Time.
• Force New Password This status can be selected manually. The service user is then
required to change the account password when they next log in. Until a password
change is successful, no service access is allowed. Note that the user must be a
member of a Rights Group that has the Security Administration option Write own
service user password enabled.
• Disabled This status prevents all service access. This setting can be selected manually.
The account can be enabled manually by setting the Account Status back to Enabled.
Account Expiry Default = <None> (No Expiry).
Not applicable to Web Manager.
This option can be used to set a calendar date after which the account will become
locked. The actual expiry time is 23:59:59 on the selected day. To prompt the user a
number of days before the expiry date, set an Expiry Reminder Time on the security
General Settings tab.
Rights Group The check boxes are used to set the Rights Groups to which the user belongs. The
Membership user's rights will be a combination of the rights assigned to the groups to which they
belong.

Related links
Security Mode Field Descriptions on page 130

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Chapter 10: Editing Configuration Settings

IP500 V2 Operation
Before editing the system's configuration settings, it is important to understand how those settings
are stored and used by the system.
The control unit holds copies of its configuration in both its internal non-volatile and RAM memory. A
copy is also held on the System SD card (IP500 V2).
The copies in non-volatile memory and System SD card, are retained even if power to the control
unit is removed. During power up, the system loads the configuration file stored on the System SD
card into its RAM memory. Other systems load the configuration stored in non-volatile memory into
RAM memory. The copy in RAM memory is then used to control the system's operation.
If the system encounters a problem using the configuration file in its System SD card's /primary
folder, it attempt to use the copy in its non-volatile memory. For fully details of the IP500 V2 boot
process and SD card usage refer to the IP Office Installation Manual.
Users actions such as changing their forward destinations or mailbox passcode are written to the
configuration in RAM memory.
Changes made using Manager are written to the configuration in non-volatile memory and then
copied into the RAM memory and System SD.
Between 00:00 and 00:30, a daily backup occurs which copies the configuration in the system's
operation RAM memory back into its non-volatile memory and, on IP500 V2 system's, the System
SD card.On IP500 V2 system, the contents of the system memory cards /primary folder can then
also be automatically copied to the /backup folder by enabling System | System | Automatic
Backup.
When the system is shutdown using the correct shutdown method, the configuration in RAM
memory is copied to the non-volatile memory and System SD card.

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 Using Manager
When using Manager to edit the configuration settings, the following need to be remembered:
• Manager receives the current configuration settings from RAM memory. Therefore the
configuration it receives includes any changes made by users up to that time. However it will
not contain any subsequent changes made by users.
• When sending the configuration settings back to the system, Manager allows two choices,
reboot or merge.
• Reboot sends the configuration to the system's non-volatile memory along with an instruction
to reboot. Following the reboot, the new configuration in non-volatile memory is copied to the
RAM memory and used.
• Merge sends the configuration to the system's non-volatile memory without rebooting. The
system then copies those changes that are mergeable into the RAM memory. A key point here
is that not all configuration settings are mergeable.
As a result of the above, it is important to bear the follow scenarios in mind:
• Changes made by users after a configuration is received by Manager may be lost when the
configuration is sent back from Manager. Therefore it is preferable to always edit a recently
received copy of the configuration rather than one that has been open for a period of time.
• If a merge is attempted with non-mergeable changes, those items will be written to the non-
volatile memory but will not be copied to RAM memory. If a daily backup occurs, they will then
be overwritten by the RAM. If a power loss reboot occurs, they will be written to RAM memory.
Related links
Mergeable Settings on page 156
Configuration Size on page 166
Setting the Discovery Addresses on page 167
Known System Discovery on page 169
Opening a Configuration from a System on page 171
Opening a Configuration Stored on PC on page 173

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Editing Configuration Settings

Creating New Records on page 174

Creating an Offline Configuration on page 175
Importing and Exporting Settings on page 177
Copying and Pasting on page 180
Saving a Configuration onto PC on page 180
Sending a Configuration on page 181
Erasing the Configuration on page 183
Default Settings on page 183

Mergeable Settings
The table below shows the configuration records for which changes can be merged and those that
require a system reboot. The Send Configuration menu shown when sending a configuration to
the system automatically indicates when the configuration is mergeable.
Configuration Setting Mergeable Notes
System | System Yes These settings are mergeable with the exception of
Locale and Favor RIP Routes over Static Routes.
Changing these settings requires a reboot of the
system.
System | LAN | LAN Settings No
System | LAN | VoIP No The following settings are mergeable:
• Auto-create Extn
• Auto-create User
• H.323 Signalling over TLS
• Remote Call Signalling Port
• Enable RTCP Monitoring on Port 5005
• RTCP collector IP address for phones
• Scope
• Initial keepalives
• Periodic timeout
• VLAN
• 1100 Voice VLAN Site Specific Option Number
(SSON)
• 1100 Voice VLAN IDs
The remaining settings are not mergeable. Changes to
these settings will require a reboot of the system.
Table continues…

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 Mergeable Settings

Configuration Setting Mergeable Notes

System | LAN | Network No
Topology
System | LAN | DHCP Pools Yes These settings are mergeable. However, the following
actions require a merge with service disruption.
• Changing the Start Address, Subnet Mask or
Default Router value for an existing DHCP Pool of
addresses.
• Decreasing Pool Size for an existing DHCP Pool of
addresses.
• Deleting an existing DHCP Pool of addresses.
When these actions are performed, the DHCP (Server
or DialIn) is re-initialized which triggers a reboot of the
Avaya DHCP Clients (H.323 and SIP) in order to force
the Avaya DHCP clients to renew their IP address lease
and apply the new settings. For the remaining Avaya
and non-Avaya DHCP clients, you must manually reboot
the devices in order to force the IP Addresses lease
renewal. Otherwise, the devices continue to use the
allocated IP addresses until the IP addresses lease time
out expires. IP address lease time out is set to three
days.
Note that IP Office supports phone reboot only for E129
and B179 SIP phone models.
System | DNS No
System | Voicemail Yes These settings are mergeable with the exception of
Voicemail Type and Voicemail IP Address. Changes
to these settings requires a reboot of the system.
System | Telephony | Yes Changes to Companding LAW and Media Connection
Telephony Preservation require a reboot.
System | Telephony | Park and Yes
Page
System | Telephony | Tones Yes • Changes to Disconnect Tone requires a reboot.
and Music
• Changes to Busy Tone Detection requires a reboot.
• Deleting any of the hold music Alternate Sources
requires a reboot.
System | Telephony | Ring Yes
Tones
System | Telephony | SM Yes
System | Telephony | Call Log Yes
System | Telephony | TUI Yes
Table continues…

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Editing Configuration Settings

Configuration Setting Mergeable Notes

System | Directory Services | Yes
LDAP
System | Directory Services | Yes
HTTP
System | System Events | No
Configuration
System | System Events | No
Alarms
System | SMTP No
System | SMDR Yes
System | VCM Yes
System | CCR Yes
System | Codecs Yes
System | VoIP No
System | VoIP Security Yes
System | Dialer Yes Changes to the Operation field or to the Trunk Range /
IP Office table require a reboot.
It is recommended that you do not change the
mergeable settings while the system is in use.
System | Contact Center Yes
Line | Analog Line | Line Yes These settings are mergeable with the exception of the
Settings Network Type setting. Changes to this setting will
require a reboot of the system.
Line | Analog Line | Analog No The Allow Analog Trunk to Trunk Connect setting is
Options mergeable. The remaining settings are not mergeable.
Changes to these settings will require a reboot of the
system.
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 Mergeable Settings

Configuration Setting Mergeable Notes

Line | BRI Line No The following settings are not mergeable. Changes to
these settings will require a reboot of the system.
• Line Sub Type
• Network Type
• TEI
• Add 'Not-end-to-end ISDN' Information Element
• Progress Replacement
• Clock Quality
• Force Number Plan to ISDN
Decreasing the Number of Channels setting requires a
a “merge with service disruption”. When the
configuration file is sent to the system, active calls on
the deleted channels are cleared.
The remaining settings are mergeable.
Line | BRI Line | Channels Yes
Line | E1 PRI Line No The following settings are not mergeable. Changes to
these settings require a system reboot.
• Line Sub Type
• Network Type
• TEI
• Channel Allocation
• CRC Checking
• Clock Quality
• Add 'Not-end-to-end ISDN' Information Element
• Progress Replacement
• Force Number Plan to ISDN
• Line Signalling
Decreasing the Number of Channels setting requires a
a “merge with service disruption”. When the
configuration file is sent to the system, active calls on
the deleted channels are cleared.
The remaining settings are mergeable.
Line | E1 Short Codes Yes
Table continues…

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Editing Configuration Settings

Configuration Setting Mergeable Notes

Line | E1 PRI Channels No The following settings are mergeable:
• Line Appearance ID (ETSI, ETSI CHI)
• Admin (ETSI CHI)
• Incoming Group ID (ETSI CHI)
• Outgoing Group ID (ETSI CHI)
The remaining settings are not mergeable. Changes to
these settings require a system reboot.
Line | E1–R2 Options No Changing the Admin setting is mergeable.
The remaining settings are not mergeable. Changes to
these settings will require a reboot of the system.
Line | E1–R2 Channels No The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Admin
The remaining settings are not mergeable. Changes to
these settings require a system reboot.
Line | E1–R2 MFC Group No
Line | E1–R2 Advanced No
Line | US T1 Line No The following settings are mergeable:
• Admin
• Prefix
The remaining settings are not mergeable. Changes to
these settings require a system reboot.
Line | T1 Channels No The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to
these settings require a system reboot.
Table continues…

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 Mergeable Settings

Configuration Setting Mergeable Notes

Line | T1 ISDN Line No The following settings are mergeable:
• Prefix
• Send Redirecting Number
• Admin
• Send original calling party for forwarded and
twinning calls
• Originator number for forwarded and twinning
calls
The remaining settings are not mergeable. Changes to
these settings require a system reboot.
Line | T1 ISDN Channels No The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to
these settings require a system reboot.
Line | T1 ISDN TNS No
Line | T1 ISDN Special No
Line | T1 ISDN Call by Call No
Line | H.323 Line | VoIP Line Yes
Line | H.323 Line | Short No For line subtypes ETSI, ETSI CHI, QSIG A, and QSIG
Codes B, these settings are mergeable. Changes to these
settings do not require a reboot of the system.
Line | H.323 Line | VoIP Yes
Settings
Line | IP DECT Line | Line Yes When creating an IP DECT line, these settings are
Line | IP DECT Line | Gateway Yes mergeable. You can also remove an IP DECT line
without rebooting. Changing an IP DECT line that has
Line | IP DECT Line | VoIP Yes been imported into the configuration is not mergeable.
Line | SIP Line | SIP Line Yes These settings are mergeable with the exception of the
Line Number setting. Changing the Line Number
setting requires a “merge with service disruption”. When
the configuration file is sent to the system, the SIP trunk
is restarted and all calls on the line are dropped.
Deleting a SIP line requires a “merge with service
disruption”.
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Editing Configuration Settings

Configuration Setting Mergeable Notes

Line | SIP Line | Transport Yes Changing these settings requires a “merge with service
disruption”. When the configuration file is sent to the
system, the SIP trunk is restarted.
Note that ITSP Proxy Address and Calls Route via
Registrar are mergeable.
Deleting a SIP line requires a “merge with service
disruption”.
Line | SIP Line | SIP URI Yes These settings are mergeable, with the following
exceptions.
• Registration
• Identity Header
These settings require a “merge with service disruption”.
When the configuration file is sent to the system, the
SIP trunk is restarted and all calls on the line are
dropped.
Line | SIP Line | VoIP Yes
Line | SIP Line | T38 Fax Yes
Line | SIP Line | SIP Yes
Credentials
Line | SIP Line | SIP Advanced Yes These settings are mergable, with the exception of the
Media Connection Preservation setting. Changing the
Media Connection Preservation setting requires a
“merge with service disruption”. When the configuration
file is sent to the system, the SIP trunk is restarted and
all calls on the line are dropped.
Line | SIP Line | Engineering Yes
Line | SIP DECT Line | SIP No
DECT Base
Line | SIP DECT Line | VoIP No
Line | SM Line | Session Yes Changing the In Service setting to Disabled (out of
Manager service) requires a system reboot. However, changing
the In Service setting to Enabled is mergeable.
Configuration changes made while the line is out of
service are also mergeable.
Line | SM Line | VoIP Yes
Line | SM Line | T38 Fax Yes
Line | IP Office Line | Line No
Line | IP Office Line | Short No For line subtypes ETSI, ETSI CHI, QSIG A, and QSIG
Codes B, these settings are mergeable. Changes to these
settings do not require a reboot of the system.
Table continues…

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 Mergeable Settings

Configuration Setting Mergeable Notes

Line | IP Office Line | VoIP Yes For line subtypes ETSI, ETSI CHI, QSIG A, and QSIG
Settings B, these settings are mergeable. Changes to these
settings do not require a reboot of the system.
Line | IP Office Line | T38 Fax Yes
Control Unit | Unit No
Extension | Extn Yes Mergeable except Base Extension, Extension ID, and
Caller Display Type.
Extension | Analog Extension | No
Analog
Extension | H323 Extension | No
VoIP
Extension | SIP Extension | No
VoIP
Extension | SIP Extension | Yes
T38 Fax
Extension | IP DECT Extension Yes These settings are mergeable with the exception of the
Reserve License setting. Changing the Reserve
License settings requires a reboot of the system.
Extension | SIP DECT Base No
User | User Yes
User | Voicemail Yes
User | DND Yes
User | Short Codes Yes
User | Source Numbers Yes
User | Telephony | Call Yes
Settings
User | Telephony | Supervisor Yes
Settings
User | Telephony | Multi-line Yes
Options
User | Telephony | Call Log Yes
User | Telephony | TUI Yes
User | Forwarding Yes
User | Dial In Yes
User | Voice Recording Yes
User | Button Programming Yes
User | Menu Programming | Yes
Hunt Group
Table continues…

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Editing Configuration Settings

Configuration Setting Mergeable Notes

User | Menu Programming | Yes
4400/6400
User | Mobility Yes
User | Group Membership Yes
User | Announcements Yes
User | SIP Yes
User | Personal Directory Yes
User | Web Self Administration Yes
Group | Group Yes
Group | Queuing Yes
Group | Overflow Yes
Group | Fallback Yes
Group | Voicemail Yes
Group | Voice Recording Yes
Group | Announcements Yes
Group | SIP Yes
Short Code | Short Code Yes
Service | Service Yes
Service | Bandwidth Yes
Service | IP Yes
Service | Autoconnect Yes
Service | Quota Yes
Service | PPP Yes
Service | Fallback Yes
Service | Dial In Yes
SSL VPN Service | Service Yes
SSL VPN Service | Session Yes
SSL VPN Service | NAPT Yes
SSL VPN Service | Fallback Yes
RAS | RAS Yes
RAS | PPP Yes
Incoming Call Route | Yes
Standard
Incoming Call Route | Voice Yes
Recording
Table continues…

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 Mergeable Settings

Configuration Setting Mergeable Notes

Incoming Call Route | Yes
Destinations
WAN Port | WAN Port No
WAN Port | Frame Relay No
WAN Port | DLCIs No
WAN Port | Advanced No
Directory | Directory Entry Yes
Time Profile | Time Profile Yes
Firewall Profile | Standard Yes
Firewall Profile | Custom Yes
Firewall Profile | Static NAT Yes
IP Route | IP Route Yes
IP Route | RIP Dynamic Yes
Routing
Account Code | Account Code Yes
Account Code | Voice Yes
Recording
License | License Yes
License | Remote Server No The Reserved Licenses setting is mergeable. The
remaining settings are not mergeable. Changes to these
settings requires a reboot of the system.
Tunnel | Tunnel (L2TP) No
Tunnel | Main (IPSec) No
Tunnel | IKE Policies (IPSec) No
Tunnel | IKE Policies (IPSec) No
Auto Attendant | Auto Yes
Attendant
Auto Attendant | Actions Yes
Authorization Codes Yes
User Rights | User Yes
User Rights | Short Codes Yes
User Rights | Button Yes
Programming
User Rights | Telephony | Call Yes
Settings
User Rights | Telephony | Yes
Supervisor Settings
Table continues…

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Editing Configuration Settings

Configuration Setting Mergeable Notes

User Rights | Telephony | Yes
Multi-line Options
User Rights | Telephony | Call Yes
Log
User Rights | User Rights Yes
Membership
User Rights | Voicemail Yes
User Rights | Forwarding Yes
ARS | ARS Yes
RAS Location Request Yes
Location Yes

Related links
Editing Configuration Settings on page 154

Configuration Size
The maximum size of the configuration file that can be loaded into an IP500 V2 control unit is 2.0
MB.
When you attempt to save a configuration that is too large, you will be prompted and the save is
canceled.
During normal operation, additional configuration records can be added to the configuration
without using Manager (for example call log records and directory records made from phones). If,
during the overnight backup to flash memory, the configuration if found to be too large, records will
be removed until the configuration is sufficiently small to be backed up. The recordsremoved are
call log records, system directory records and then personal directory in that order. Note that those
records will still exist in the configuration running the system in its RAM memory, however if the
system is restarted they will disappear as the configuration is reloaded from the Flash memory.
Figures for all individual records in the configuration cannot be given as they vary. The list below
gives typical values, in bytes, for common records:
Physical Extension: 70.
IP Extension: 70.
User: 170.
User Short Code: 40.
DSS Button: 20.
Hunt Group: 100.

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 Setting the Discovery Addresses

Hunt Group member: 10.

System Short Code: 10.
Normal Service: 220.
Intranet Service: 240.
WAN Service: 400.
RAS Service: 110.
Incoming Call Route: 30.
WAN Port (PPP): 70.
WAN Port (FR): 120.
Directory Record: 70.
Time Profile: 40.
Time Profile Record: 20.
Firewall Profile: 40.
Custom Firewall Record: 80.
IP Route (Static): 30.
License Key: 40.
Account Code: 40.
Logical LAN: 60.
Tunnel (L2TP): 200.
Tunnel (IPSec): 110.
Related links
Editing Configuration Settings on page 154

Setting the Discovery Addresses

By default, when or File | Open configuration is selected, Manager's Select IP Office window
opens. It performs a UDP broadcast to the address 255.255.255.255. This broadcast will only
locate systems that are on the same network subnet as the PC running Manager.
For systems not located on the same subnet as the Manager PC, the following options are
supported.
Specific Addressing The Unit/Broadcast Address shown on the Select IP Office menu can be
changed to the specific IP address of the required system. A single address is routable and so can
be used to discover a system on another subnet.

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Editing Configuration Settings

TCP Discovery Address Ranges A set of TCP addresses and address ranges can be specified
for use by the Select IP Office discovery process.
Known System Discovery Manager can write the details of systems it discovers to a file. The list
of systems in that file can then be used for access to those systems.
DNS Lookup Manager can be configured to locate systems using DNS name lookup. This
requires the systems on a customer network to be added as names on the customer's DNS server
and the Manager PC to be configured to use that server for DNS name resolution. The use of
DNS is configured through File | Preferences | Discovery.
Changing the Initial Discovery Settings The Discovery tab of the Preferences menu can be
used to set the UDP and TCP addresses used by the discovery process run by the Select IP
Office menu.
1. Select File | Preferences menu.
2. Select the Discovery tab.
TCP Discovery: Default = On. This setting controls whether Manager uses TCP to discover
systems. The addresses used for TCP discovery are set through the IP Search Criteria field
below.
NIC IP/NIC Subnet This area is for information only. It shows the IP address settings of the LAN
network interface cards (NIC) in the PC running Manager. Double-click on a particular NIC to add
the address range it is part of to the IP Search Criteria. Note that if the address of any of the
Manager PC's NIC cards is changed, the Manager application should be closed and restarted.
IP Search Criteria This section is used to enter TCP addresses to be used for the TCP discovery
process. Individual addresses can be entered separated by semi-colons, for example
135.164.180.170; 135.164.180.175. Address ranges can be specified using dashes, for example
135.64.180.170 - 135.64.180.175.
UDP Discovery: Default = On This settings controls whether Manager uses UDP to discover
systems.
Enter Broadcast IP Address: Default = 255.255.255.255 The broadcast IP address range that
Manager should use during UDP discovery. Since UDP broadcast is not routable, it will not locate
systems that are on different subnets from the Manager PC unless a specific address is entered.
Use DNS: Selecting this option allows Manager to use DNS name (or IP address) lookup to
locate a system. Note that this overrides the use of the TCP Discovery and UDP Discovery
options above. This option requires the system IP address to be assigned as a name on the users
DNS server. When selected, the Unit/Discovery Address field on the Select IP Office dialogue is
replaced by a Enter Unit DNS Name or IP Address field.
SCN Discovery: If enabled, when discovering systems, the list of discovered systems will group
systems in the same Small Community Network and allow them to be loaded as a single
configuration. At least one of the systems in the Small Community Network must be running
Release 6.0 or higher software. See Configuring Small Community Networking on page 769. This
does not override the need for each system in the Small Community Network to also be reachable
by the TCP Discovery and or UDP Discovery settings above and accessible by the router
settings at the Manager location.

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 Known System Discovery

Related links
Editing Configuration Settings on page 154

Known System Discovery

The Select IP Office menu normally displays systems discovered by Manager using either UDP
broadcast and or TCP requests. Manager can be configured to also record details of discovered
units and then display a list of those previously discovered ('known') systems.
Related links
Editing Configuration Settings on page 154

Configuring Manager for Known System Discovery

About this task
Use of known systems discovery is not enabled by default. The Manager must be configured for
the feature with a file location to which it can store and retrieve known system details.
Procedure
1. Select File | Change Working Directory.

2. In the Known Units File field, enter the directory path and file name for a CSV file into
which Manager can write details of the systems it discovers.
If the file specified does not exist it will be created by Manager.
3. Click OK.

Using Known System Discovery

About this task

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Editing Configuration Settings

Procedure
1. When the Select IP Office screen is displayed click on Known Units.

2. The screen displays the list of systems previously discovered and stored in the CSV file.
3. To select an control unit, highlight the row containing unit data and click OK.
The selected unit will appear in the Select IP Office window.
4. To filter displayed units, type the first few characters of the unit name in the Filter field.
Any unit whose name does not match the filter will be temporarily hidden.
5. Each discovery appends data to the known unit list.
It is possible that details of some records in the list may be out of date. Right clicking on
the leftmost (grey) column of any row will bring up a floating menu offering the options of
Refresh and Delete.
6. A new record may be manually added without having to access the system first through
normal discovery.
Enter the IP address of the new system in the IP Address column of the blank row shown
with a * and select Refresh from the floating menu. This will update the Known Units file
with data relating to the unit with the specified address.
7. Select Cancel to return to the Select IP Office menu.
Result
Note:
• The key used by the Known Systems CSV file is the IP address. The file cannot contain
records for separate systems that use the same IP address for access.
• The file can be made read only. In that case any attempts using Manager to update the
file will be ignored.

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 Opening a Configuration from a System

Opening a Configuration from a System

The initial IP address ranges in which Manager searches for systems are set through the Manager
preferences (File | Preferences | Discovery). By default it scans the local network of the Manager
PC.
Start Manager. If Manager is already started and a configuration is open in it, that configuration
must be closed first.
• If Manager is set to Auto Connect on start up, it will scan for systems automatically and either
display the list of systems discovered or automatically start login to the only system
discovered.
• Otherwise, click on or select File | Open Configuration.
The Select IP Office window appears, listing those systems that responded.
• If Server Edition systems are detected, they are grouped together. By default the
configuration of those systems cannot be opened using Manager in Advanced View mode
and the configuration of a Primary Server can only be opened if the Open with Server Edition
Manager option is also selected.
• If Manager has been set with SCN Discovery enabled, systems in a Small Community
Network are grouped together. The checkbox next to the network name can be used to load
the configurations of all the configurations into Small Community Network management
mode.
• If the system required was not found, the Unit/Broadcast Address used for the search can
be changed. Either enter an address or use the drop-down to select a previously used
address. Then click Refresh to perform a new search.
• The address ranges used by Manager for searching can be configured through the File |
Preferences | Discovery tab.
• A list of known systems can be stored and used.
• Manager can be configured to search using DNS names. See the setting File | Preferences |
Discovery | Use DNS
• .
• Systems found but not supported by the version of Manager being used will be listed as Not
Supported.
• If the system detected is running software other than from its primary folder, a warning
icon will be shown next to it. The configuration can still be opened but only as a read-only file.
When you have located the system required, check the box next to the system and click OK.
If the system selected is a Server Editionsystem and Manager is not running in Server Edition
mode, an Open with Server Edition Manager checkbox is shown and pre-selected. Clicking OK
will switch Manager to its Server Edition mode before loading the configuration.
The system name and password request is displayed. Enter the required details and click OK.

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Editing Configuration Settings

The name and password used must match a service user account configured within the system's
security settings.
Additional messages will inform you about the success or failure of opening the configuration from
the system.
The method of connection, secure or insecure, attempted by Manager is set the applications
Secure Communications preferences setting.
• When Secure Communications is set to On, a padlock icon is displayed at all times in the
lower right Manager status field.
• New installations of Manager default to having Secure Communications enabled. This
means Manager by default attempts to use secure communications when opening a
configuration.
• For Server Edition systems, Manager will always attempt to use secure communications
regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds, Manager will
offer to fallback to using unsecured communications.
Following a successful log in, the configuration is opened in Manager. The menus and options
displayed will depend on the type of system configuration loaded.
Login Messages
While attempting to login to a system, various additional messages may be displayed.
Configuration Not Loaded Messages
Access Denied This is displayed as the cause if the service user name/password were incorrect,
or the service user has insufficient rights to read the configuration. The Retry option can be used
to log in again but multiple rejections in a 10 minute period may trigger events, such as locking the
user account, set by the Password Reject Limit and Password Reject Action options in the
systems security settings.
Failed to communicate with system This is displayed as the cause if the network link fails, or
the secure communication mode is incorrect (for example Manager is set to unsecured, but the
system is set to secure only).
Account Locked The account of the service user name and password being used is locked. This
can be caused by a number of actions, for example too many incorrect password attempts,
passing a fixed expiry date, etc. The account lock may be temporary (10 minutes) or permanent
until manually unlocked. An account can be enabled again through the system's security settings.

Additional Messages
Your service user account will expire in X days This message indicates that an Account Expiry
date has been set on the system service user account and that date is approaching. Someone
with access to the system's security settings will be required to set a new expiry date.
Your password will expire in X days. Do you wish to change it now? This message indicates
that password ageing has been configured in the system's security settings. If your password

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 Opening a Configuration Stored on PC

expires, someone with access to the system's security settings will be required to unlock the
account.
Change password Through the system's security settings, a service user account can be
required to change their password when logging in. The menu provides fields for entering the old
password and new password.
Contact Information Check - This configuration is under special control This message will
appear if a Manager user with administrator rights has entered their contact information into the
configuration. For example to indicate that they do not want the configuration altered while a
possible problem is being diagnosed. The options available are:
Retain | Replace | Cancel This message appears when it is detected that the configuration of one
of the systems in a Server Editionnetwork has previously been edited directly rather than via
access to the primary system. Select Replace to replace the system’s update configuration with
the copy already held by the primary server. Select Retain to keep the already updated
configuration.
Cancel Select this option to close the configuration without making any changes.
Set configuration alteration flag Select this option if the configuration is being opened because
some urgent maintenance action. When the configuration is next opened, the fact that it has been
altered will be indicated on the System | System tab.
Delete Contact Information Select this option to take the system out of special control.
Leave contact information and flags unchanged (Administrators only) This option is only
available to service users logging in with administrator rights.
Related links
Editing Configuration Settings on page 154

Opening a Configuration Stored on PC

About this task
A configuration file previously saved on the PC can be reopened in Manager. This method of
access does not require entry of a service user name and password. All parts of the configuration
are visible.
Use either of the following processes to load a saved configuration file:
Procedure
1. Click the main toolbar or select File | Offline | Open File from the menu bar. If the files
is one that has previously been opened offline, click the symbol next to in the main
toolbar
2. An Open configuration file window appears.
Use this to browse to the required configuration file.

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Editing Configuration Settings

3. Select the file and click Open.

Related links
Editing Configuration Settings on page 154

Creating New Records

There are a number of ways in which you can add new records to the configuration currently
loaded in Manager.
Related links
Editing Configuration Settings on page 154

Adding a New Record Using the Details Pane

About this task
Procedure
1. Use the navigation pane, groups pane or navigation toolbar, select an existing record of
the type required.
2. Click at the top-right of the details pane.
3. Select the type of record required.
For example, with extensions you can select from H.323 Extension or SIP Extension.
4. Complete the settings for the new record and click OK.

Adding a New Record Using the Group Pane

About this task
Procedure
1. Using the navigation pane or navigation toolbar, select the type of record required.
2. Right-click on the group pane and select New.
3. If a list is displayed, select the specific type of record required.
4. Complete the settings for the new record and click OK.

Adding a New Record Using the Navigation Pane

About this task

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 Creating an Offline Configuration

Procedure
1. In the navigation pane, right-click on the type of record required and select New.
2. If a list is displayed, select the specific type of record required.
3. Complete the settings for the new record and click OK.

Other record creation methods

About this task
Procedure
1. Import records
New records can be created for users, hunt groups, short codes and directory entries by
importing files. See Importing and Exporting Settings on page 177.
2. Automatically create records
There are scenarios where new records are automatically added to the configuration:
• Certain default records are added to the configuration when a new system is started or
when a system configuration is erased.
• New records for extensions and lines are automatically added to match the hardware
present when a system is started or rebooted.
• For telephones connected via IP links (H.323, SIP and DECT), the system can be
configured to automatically create extension and or user records to match the
telephone.
3. Use templates to create records
Manager supports a number of methods by which new records can be created from
templates. See Working with Templates on page 585.

Creating an Offline Configuration

About this task
Manager can be used to create a new configuration without connecting to a system. This allows
the creation of a configuration prior to installation of the real system and so can be used to speed
up the installation process.
The configuration created must match the physical equipment in the system into which the
configuration will be loaded. Doing otherwise may cause the system to reset and experience other
problems.
The Create Configuration tool includes all control units, external expansion modules and trunk
cards supported. It is your responsibility to confirm what equipment is supported in your locale.

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Editing Configuration Settings

Creating a New Configuration

Procedure
1. Close or save any configuration currently open.
2. Click in the main toolbar or select File | Offline | Create New Config.
3. You should set the Configuration, Locale, Extension Number Length and System Unit
first.
Changing any of these after you start selecting other system hardware will reset the
hardware selections.
4. Select the type of Configuration you want to create.
The other options available will change depending on the selection. If the menu has been
started from Manager running in Server Edition mode, the only option is Server Edition
Edition.
5. Select the Locale for the system.
This defines a range of features such as default telephony settings.
6. The Extension Number Length setting value can be None or 3 to 15.
If a value is selected, all default extension, user and hunt group extension numbers
created by Manager will be that length. In addition Manager will display a warning if an
extension number of a different length is entered when editing the configuration.
7. Select the type of System Unit.
Select the hardware components for the system. For a Server Edition system this is only
necessary if a Expansion System (V2) is selected as the System Units option.
8. Select the additional cards to include in the control unit.
The number and type of cards selectable will depend on the control unit type.
9. Select the external expansion modules to also include in the system.
10. Click OK.
11. For non-Server Edition systems, the configuration is created and loaded into Manager.
For Server Edition systems, the Initial Configuration menu for the selected type of system
unit is displayed. Complete the menu and click Save.
12. Once this configuration has been edited as required it can be saved on the PC or sent to a
system.
13. To Save a Configuration File on the PC Use File | Save Configuration.
14. To Send the Configuration to a System If the system which you want to use the
configuration is available, use File | Offline | Send Configuration to send the configuration
to it.

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 Importing and Exporting Settings

Warning:
This action will cause the system to reboot and will disconnect all current calls and
service.
• Ensure that you have a copy of the systems existing configuration before overwriting it
with the off-line configuration.
• After sending the configuration, you should receive the configuration back from the
system and note any new validation errors shown by Manager. For example, if using
Embedded Voicemail, some sets of prompt languages may need to be updated to match
the new configurations locale setting using the Add/Display VM Locales option.
Related links
Editing Configuration Settings on page 154

Importing and Exporting Settings

Manager can import configuration settings created elsewhere. This can be useful when setting up
a new system or sharing common settings such as a directory between systems.
The system supports LDAP (System | Directory Services | LDAP) for automatic importation of
directory records (LDAP Version 2).
The system also supports HTTP (System | Directory Services | HTTP) for automatic importation
of directory records.
Note:
You can also manage configuration settings using templates. See Working with Templates on
page 585.
Settings are imported and exported in the following formats:
• Binary Files (.exp) These are non-editable files. During import and export it is possible to
select what types of records should be included in the file. During import the whole file is
imported.
• Comma Separated Variable Text Files (.csv) These are plain text files. In addition to being
exported from a system these files can be created and edited using programs such as
WordPad or Excel.
When opening a .csv file in Excel it will alter the way some data is displayed, automatically
changing the display format of dates and long numbers such as phone numbers.
UTF-8 Character Encoding Manager imports and exports CSV files using UTF-8 character
encoding which uses a multiple bytes to support characters with diacritic marks such as ä. Other
applications, such as Excel, may, depending on the user PC settings, use different encoding which

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Editing Configuration Settings

will cause such characters to be removed or corrupted. Care should be taken to ensure that any
tool used to create or edit the CSV supports all the characters expected and uses UTF-8 format.
• Importing into Manager from Excel From Excel save the file as a .csv. This file will use
ANSI character encoding. Open the file in Notepad and use the Save As option to rename
the file and select UTF-8 encoding. Import the UTF-8 version of the file into Manager.
• Exporting from Manager into Excel Do not double-click on the file exported from Manager.
Start Excel and use File | Open to select the file. Excel will recognize that the file uses UTF-8
encoding and will start its text file importation wizard. Follow the wizard instructions and
select comma as the field delimiter.
CSV File Formats
The format is CSV using commas as field separator, no text delimiters and no header row. The
simplest way to check the required format for a CSV file prior to import, is to export and a file from
an existing system configuration.
File Name Fields in Order
Configuration Proprietary format. Note that this does not contain all configuration
fields.
License The License option is only available for export and only for ADI licenses
present on the system.
Short Code Code, Telephone Number, Feature.
User Name, Extension, User Rights, Email Address, Full Name, Password,
VoiceMail Code, Login Code, UserTemplate, ExtensionTemplate.
Directory Name, Number, Speed Dial.
Group Name, Extension, Group, Hunt, Rotary, Longest Waiting, Queuing On,
Voicemail On, Broadcast, Voicemail Email.

Note:
Group: Apart from Name, Extension and Voicemail Email, the fields use a 1 or 0 value for on
or off.
The format of the system CSV is too complex to be described. It is a full export of all the system's
configuration settings. This file format should only be used for export and import between systems
and not for any offline editing.
Using the CSV Configurator spreadsheet
You can use the CSV Configurator spreadsheet to create or modify multiple configuration entries.
The CSV Configurator spreadsheet is available in the Manager application folder. By default,
Manager is installed under C:\Program Files or C:\Program Files (x86) on Windows 7.
The remaining path and file name is ...\Avaya\IP Office\Manager\IP Office User
CSV Configurator.xlsm.
Follow the procedure below to export configuration settings. You can then use the exported files
with the CSV Configurator spreadsheet. Follow the instructions in the spreadsheet.
Related links
Editing Configuration Settings on page 154

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 Importing and Exporting Settings

Exporting Settings
About this task
Procedure
1. Select File | Import/Export... from the menu bar.
2. Select Export.
3. Select the type of file.
The list of exportable record types will change to match the file type.
4. Select the types of items that should be exported.
5. Use the Save In path to select the location for the exported files.
The default location used is a subfolder in the Manager application directory based on
system name of the currently loaded system. For example, ...\Avaya\IP Office
\Manager\System_1.
6. Click OK.

Importing Settings
Importing settings will overwrite any existing records that match a record being imported
Procedure
1. Select File | Import/Export... from the menu bar.
2. Select Import.
3. Select the type of file.
The list of items will change to match the type of file selected and whether a matching file
or files is found in the current file path.
4. Use Look In to adjust the file path.
The default location used is a subfolder in the Manager application directory based on
system name of the currently loaded system. For example, ...\Avaya\IP Office
\Manager\System_1.
5. Select the types of items that should be imported.
6. Click OK.

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Editing Configuration Settings

Copying and Pasting

Manager supports the normal Windows methods of cutting, copying, pasting and deleting records
and settings. These can be accessed through the Edit menu in the menu bar or using the
standard Windows keyboard shortcuts for those actions. They can also be accessed by selecting
a record or text field and then right-clicking.
Copy and paste can be used with the navigation and group panes to create a new record with the
same settings as the original. The copy will be renamed as Copy of ... to avoid conflicting with the
original.
When using copy and paste between individual settings fields, whether on the same record or a
different record, care should be taken to ensure that the fields use the same type of data. Similarly
copying a record in the navigation or group pane and then pasting it into the details pane will
prompt Manager to paste the copied records data into the first field of the current record in the
details pane. As a general rule, cut and paste actions should be used with the same pane and
within similar record types.
For users and user rights, a number of controls have been provided to copy settings between a
user and a user right or vice versa. See Configuring User Rights on page 653.
Related links
Editing Configuration Settings on page 154

Saving a Configuration onto PC

The system configuration settings shown within Manager can be saved as a .cfg file on the
Manager PC. These files can be used as backups or sent to other persons to aid problem
diagnostics. Note however that an offline configuration file does not include the Audit Trail records
for the system.
Automatically Saving Sent Configurations
By default, Manager creates a file copy of the configuration before it is sent to the system. This
copy is stored in Manager's Working Directory using the system name and .cfg. This behavior is
controlled by the Backup File on Send (File | Preferences | Security) option.
The number of backups of each systems configuration can be limited to a set number of the most
recent copies.
Saving a Configuration Received from a System
Select File | Save Configuration as from the menu bar.
Saving a Configuration opened on the PC

Click in the main toolbar or select File | Save Configuration from the menu bar.

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 Sending a Configuration

Related links
Editing Configuration Settings on page 154

Sending a Configuration
The current configuration settings open within Manager can be sent to the system. The method
depends on whether Manager is being used to edit the configuration of a single system or a
network of systems.
Sending an Individual System Configuration
The first steps of this process depend on whether you are sending a configuration received from
the system or sending one opened offline/created new.
• A Configuration Opened from a System Click in the main toolbar or select File | Save
Configuration from the menu bar.
• A Configuration Created Offline or Opened from a PC File Select File | Offline | Send
Config from the menu bar.
The Send Configuration menu is displayed.
Configuration Reboot Mode If Manager thinks the changes made to the configuration settings
are mergeable, it will select Merge by default, otherwise it will select Immediate.
• Merge Send the configuration settings without rebooting the system. This mode should only
be used with settings that are mergeable. Refer to Mergeable Settings.
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in
progress. This mode can be combined with the Call Barring options.
• Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Call Barring options.
• Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time
for the system reboot. If the time is after midnight, the system's normal daily backup is
canceled.
• Call Barring These settings can be used when the reboot mode When Free or Timed is
selected. They bar the sending or receiving of any new calls.
Click OK. A service user name and password may be requested.
• If the service user name or password used do not valid, "Access Denied" is displayed.
• If the service user name used does not have rights to send a configuration or to request a
reboot or merge, "Insufficient service user rights" is displayed.

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Editing Configuration Settings

• If the service user name used does not have operator rights to make the changes that have
been made to the configuration, "Insufficient operator rights. Operator cannot modify
xxxx records" is displayed.
• The warning will appear if the configuration being sent contain any errors indicated by a
icon in the error pane. The configuration can still be sent by selected Yes.
• The message Failed to save the configuration data. (Internal error) may indicate that the
IP500 V2 system has booted using software other than that in its System SD card's primary
folder.
Sending Multiple Configurations
When Manager is running in Server Edition mode or SCN Management mode, it loads multiple
configurations at the same time.
1. Click in the main toolbar or select File | Save Configuration from the menu bar.
2. The menu displayed only shows details for those systems where the system configuration
has been changed and needs to be sent back to the system.

• Select By default all systems with configuration changes are selected. If you want to exclude
a system from having its configuration updated, either deselect it or cancel the whole
process.
• Change Mode If Manager thinks the changes made to the configuration settings are
mergeable, it will select Merge by default, otherwise it will select Immediate.
• Merge Send the configuration settings without rebooting the system. This mode should only
be used with settings that are mergeable. Refer to Mergeable Settings.
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in
progress. This mode can be combined with the Incoming Call Barring and Outgoing Call
Barring options.
• Store Offline It is possible to add a reference for a Server Edition Secondary or for a Server
Edition Expansion System to create a configuration file for that system even though it is not
physically present. Store Offline saves that configuration on the Server Edition Primary in its
file store. The same file is retrieved from there until such time as the physical server is
present at which time you are prompted whether to use the stored file or the actual servers
current configuration.

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 Erasing the Configuration

• Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Incoming Call Barring and Outgoing Call Barring options.
• Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time
for the system reboot. If the time is after midnight, the system's normal daily backup is
canceled.
• Incoming Call Barring This setting can be used when the reboot mode When Free or
Timed is selected. It bars the receiving of any new calls.
• Outgoing Call Barring This setting can be used when the reboot mode When Free or
Timed is selected. It bars the making of any new calls.
Click OK. The progress of the sending of each configuration is displayed.
Related links
Editing Configuration Settings on page 154

Erasing the Configuration

About this task
The system configuration settings can be erased. During this process, the system is rebooted and
starts with a set of default settings. This process does not erase the security settings of the
system.
Erasing the Configuration
Procedure
1. Select File | Advanced | Erase Configuration (Default).
2. Enter a valid user name and password.
3. The system will be rebooted.
Related links
Editing Configuration Settings on page 154

Default Settings
The following applies to new systems and those defaulted using the Erase Configuration
command. They also apply to IP500 V2 control units defaulted using the reset button on the rear
of the unit (refer to the Installation manual for details of using the reset button).
Mode
IP500 V2 control units can operate in a number of modes. The initial mode is determined by the
type of System SD card fitted and the level of software.

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Editing Configuration Settings

IP Office A-Law: A system fitted with this type of card will default to A-Law telephony.
IP Office U-Law: A system fitted with this type of card will default to U-LAW telephony.
Enterprise Branch: Use this option for an SD card intended to be used with an IP Office system
running in Enterprise Branch Mode. There is a separate SD card for Enterprise Branch. The
Enterprise Branch SD card can only be used for Enterprise Branch operation and cannot be used
to change modes to IP Office. You also cannot use or change an IP Office SD card for use with an
Enterprise Branch system.
Warning:
Do not re-purpose a Enerprise Branch card for use with any other IP Office mode. Doing so
may damage the SD card and make it unusable for your Enterprise Branch system.
Default Short Codes
For IP500 V2 control units, A-Law or U-Law operation is determined by the Feature Key dongle
installed in the system. Depending on the variant, a default system will use different sets of default
short codes. See the Default System Short Code List on page 803.
Default Data Settings

When a new or defaulted control unit is switched on, it requests IP address information from a
DHCP Server on the network. This operation will occur whether the LAN cable is plugged in or
not.
If a DHCP server responds within approximately 10 seconds, the control unit defaults to being a
DHCP client and uses the IP address information supplied by the DHCP server.
If no DHCP Server responds, the control unit still defaults to being the DHCP client but assumes
the following default LAN addresses:
• For its LAN1 it allocates the IP address 192.168.42.1 and IP Mask 255.255.255.0.
• For its LAN2 if supported, it allocates the IP address 192.168.43.1 and IP Mask
255.255.255.0.
Once a control unit has obtained IP address and DHCP mode settings, it will retain those settings
even if rebooted without a configuration file present on the System SD card. To fully remove the
existing IP address and DHCP mode setting, the system must be defaulted using Manager.
Default Security Settings
Security settings are held separately from the configuration settings and so are not defaulted by
actions that default the configuration. To return the security settings to their default values the
separate Erase Security Settings command should be used.
Default Standard Mode Telephony Configuration Settings
A hunt group Main is created with extension number 200. The first 16 extensions on the systems
are added to the group.
All physical extensions ports are numbered from extension number 201 upwards. A matching user
record for each extension is also created.
A default incoming call route for all voice calls is created, with the default hunt group Main as its
destination.

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 Default Settings

A default incoming call route for data calls is created with the default RAS record DialIn as its
destination.
All lines are defaulted to Incoming Group ID and Outgoing Group ID of 0.
Default short codes are created based on whether the system's locale is A-Law or U-Law.
Default Server Edition Telephony Configuration Settings
No users except NoUser.
All extensions are unnumbered.
No default hunt group or incoming calls routes are created.
All auto-create options are off by default.
Related links
Editing Configuration Settings on page 154

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Chapter 11: Configuration Mode Field
Descriptions

This following sections detail the configuration settings for the different record types within the
system. Depending on the type and locale of the system some settings and tabs may be hidden as
they are not applicable. Other settings may be grayed out. This indicates that the setting is either for
information only or that another setting needs to be enabled first.
Related links
Configuration field display on page 187
BOOTP Record on page 190
Operator on page 192
System on page 192
Line on page 266
Control Unit on page 382
Extension on page 383
User on page 401
Group on page 451
Short Code on page 474
Service on page 475
RAS on page 488
Incoming Call Route on page 490
WAN Port on page 500
Directory Entry on page 504
Time Profile on page 505
Firewall Profile on page 507
IP Route on page 511
Account Code on page 514
License on page 516
Tunnel on page 522
Auto Attendant on page 528
Authorization Codes on page 533
User Rights on page 534
ARS on page 543
Location on page 547

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 Configuration field display

Configuration field display

The way that the system configuration settings are grouped and displayed in the navigation pane
depends on whether Manager is running in its normal Standard mode or in Server Edition mode.
Related links
Configuration Mode Field Descriptions on page 186
Configuration field display in Standard mode on page 187
Configuration field display in Server Edition mode on page 188

Configuration field display in Standard mode

This order of display of different types of configuration record is used for Standard mode systems.

System Overall settings for the data and telephony operation of the system.

Line Settings for trunks and trunk channels within the system.

Control Unit Information summary of the system.

Extension Settings for extension ports.

User Settings for each system user. They may or may not be associated with an extension.

Hunt Group Collections of users to which calls can be directed for answer by any one of those
users.

Short Code These are numbers which when dialed trigger specific features or are translated
for external dialing. Short codes can be set at both the system wide level and locally for a
particular system.

Service Configuration settings such as user names and passwords needed for connections to
data services such as the Internet.

RAS Remote Access Service settings for connecting incoming data calls.

Incoming Call Route Records here are used to match incoming call details on external trunks
to destinations on the system.

WAN Port Configuration settings for the WAN ports provided on some units.

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Configuration Mode Field Descriptions

Directory External names and numbers. Used for matching names to incoming calls and for
dialing from user applications.

Time Profile Used to control when various functions are active.

Firewall Profile Use to control the types of data traffic that can cross into or out of the system.

IP Route These records are used to determine where data traffic on the system should be
routed.

Account Code Used for call logging and to control the dialing of certain numbers.

License License keys are used to enable system features and applications.

Tunnel Used to created IPSec and L2TP data tunnels.

User Rights Provide templates to control the settings applied to associated users.

Auto Attendant Used when an Avaya memory card is installed in the control unit.

ARS Automatic Route Selection is used by to control outgoing external calls.

Authorization Codes Authorization codes are similar to account codes. However, unlike
account codes which are usable by any user, each authorization code is only usable by a specific
user or users associated with a specific set of user rights.
Related links
Configuration field display on page 187

Configuration field display in Server Edition mode

When Manager is being used in Server Edition mode, the navigation pane works in the same way
as normal. However, the different types of configuration records are ordered and grouped
differently. This reflects the fact that some types of record are automatically shared across all
systems in the network.
Solution settings
The first 8 types are records have special behaviors that are different from normal records stored
in the configurations of individual systems in the network.

User These records show settings for system users. Each user may or may not be associated
with an extension. All the users configured on all systems are grouped here to allow easy
configuration access. The individual user records are still stored in the configuration of the

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 Configuration field display

particular system on which the user was created and can also be accessed through that system's
configuration settings. New users are created through the User settings of the system that hosts
the user.

Hunt Group These records are groups of users to which calls can be directed for answering
by any one of those users. Hunt group records are stored in the configuration of the Primary
Server but those hunt groups are advertised for use by all systems in the network.

Directory External names and numbers. These records are used to match names to incoming
calls and for making calls by name selection from the directory on phones or in applications.
These directory records are stored in the configuration of the Primary Server. By default all other
systems in the network automatically import a copy of the Primary Server system directory at
regular intervals.
By default, the following types of records are all shared and replicated by each system in the
network and cannot be set at an individual system level. That operation can be changed using the
consolidation settings.

Short Code These are numbers which when dialed trigger specific features or are translated
for external dialing. These short codes are common to all systems in the network.

Incoming Call Route Records set here are used to match incoming call details on external
trunks to destinations. These incoming call routes are shared by all systems in the network.

Time Profile Used to control when various functions are active. The time profiles set here are
shared by all systems in the network.

Account Code Used for call logging and to control the dialing of certain numbers. The account
codes set here are shared by all systems in the network.

User Rights Provide templates to control the settings applied to users associated with a
particular set of user rights. These user rights are shared and replicated on all systems in the
network.
Individual system settings
In addition to the settings above, a range of other types of record can be configured for each
individual system in the network. Visibility and configuration of Short Code, Incoming Call Route,
Time Profile, Account Code and User Rights records is dependent on the consolidation settings
of Manager.

System A system icon is shown for each system in the network. That is, one for the Primary
Server, one for the the Secondary Server if installed and one for each Expansion System (L) and
Expansion System (V2) systems. Each can be expanded to allow configuration of records that are
particular to that system.
•
Line Settings for trunks and trunk channels within the system.
•
Control Unit Information summary of the system.

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Configuration Mode Field Descriptions

•
Extension Settings for extension ports.
•
User Settings for each system user. They may or may not be associated with an
extension.
•
Short Code These are numbers which when dialed trigger specific features or are
translated for external dialing.
•
Service Configuration settings such as user names and passwords needed for
connections to data services such as the Internet.
•
RAS Remote Access Service settings for connecting incoming data calls.
•
WAN Port Configuration settings for the WAN ports provided on some units.
•
Firewall Profile Use to control the types of data traffic that can cross into or out of the
system.
•
IP Route These records are used to determine where data traffic on the system should
be routed.
•
License License keys are used to enable system features and applications.
•
Tunnel Used to created IPSec and L2TP data tunnels.
•
ARS Automatic Route Selection is used by to control outgoing external calls.
•
Authorization Codes Authorization codes are similar to account codes. However, unlike
account codes which are usable by any user, each authorization code is only usable by a
specific user or users associated with a specific set of user rights.
Related links
Configuration field display on page 187

BOOTP Record
Navigation: BOOTP | BOOTP Record
The BOOTP settings are used by the Manager application itself. They are not system
configuration settings.
BOOTP is protocol used by devices to request software when restarting. It is used when
upgrading the control unit within a system or when the core software within the control unit has
been erased. When running, Manager can respond to BOOTP requests and, if it finds a matching
BOOTP record for the system, provide the software file indicated by that record.

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 BOOTP Record

BOOTP records are not part of a system's configuration settings, they are items saved on the
Manager PC. Normally Manager automatically creates a BOOTP record for each system with
which it has communicated, up to a maximum of 50 records. However BOOTP records can be
added and edited manually when necessary.
Field Description
File Location The location from which Manager provides files in response to BOOTP is its binaries
directory. This can be changed using File > Change Working Directory or File >
Preferences > Directories.
This directory is also the directory used by Manager when providing files by TFTP.
Disabling BOOTP Manager can be disabled from providing BOOTP support for any systems. Select File >
Preferences > Preferences > Enable BOOTP and TFTP Server.
Enabled Default = Enabled
If unchecked, BOOTP | BOOTP Record support for the matching system from this
Manager PC is disabled.
System Name This field is not changeable. It shows the system name.
MAC Address The MAC address of the system. The address can be obtained and or verified in a number
of ways:
• When a system's configuration settings are loaded into Manager, it is shown as the
Serial Number on the Unit form. On defaulted systems, it is also used as the system
name.
• If the system is requesting software, the MAC address is shown as part of the request in
the status bar at the base of the Manager screen.
• If the system can be pinged, it may be possible to obtain its MAC address using the
command arp -a <ip address>.
IP Address The IP address of the system's LAN1.
Filename The name of the .bin software file used by that type of control unit. To be transferred to the
system, this file must exist in the Manager applications Working Directory.
Time Offset : Default = 0.
In addition to performing BOOTP support for systems, the Manager application can also
act as a time server (RFC868). This field sets the offset between the time on the PC
running Manager and the time sent to the system in response to its time requests. The
field is not used if a specific Time Server IP Address is set through the System form in the
system's configuration settings.
Manager can be disabled from acting as an Internet Time (RFC868) server. Select File >
Preferences > Preferences and uncheck Enable time server.

Related links
Configuration Mode Field Descriptions on page 186

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Configuration Mode Field Descriptions

Operator
Operator records are not part of a system's configuration settings. They are used when a pre-
Release 3.2 configuration is loaded to control what parts of a configuration can be edited.
Operator View Edit New Delete Configuration Record
Types
Administrator Yes Yes Yes Yes All configuration records
Manager Yes Yes Yes Yes View all. Other actions
Extension, User, Hunt Group,
Short Code, Service, RAS,
Incoming Call Route,
Directory, Time Profile,
Firewall Profile, IP Route,
Least Cost Routing, Account
Code, ARS.
Operator Yes Yes — — View all configuration
records. Edit all except
System, Line, Control Unit
and Authorization Codes.

If an invalid operator is specified while receiving a configuration from a pre-3.2 system, the
settings will be loaded using the Guest operator. This additional operator allows a read-only view.
Related links
Configuration Mode Field Descriptions on page 186

System
Navigation: System
There is one System record for each system being managed. When managing multi system
Server Edition or Small Community Network deployments, clicking on the System icon for a
particular system displays a system inventory page for that system.
Related links
Configuration Mode Field Descriptions on page 186
System on page 193
LAN1 on page 199
LAN2 on page 214
DNS on page 215
Voicemail on page 215
Telephony on page 222
Directory Services on page 240
System Events on page 244

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 System

SMTP on page 252

System | SMDR on page 253
VCM on page 254
Integrated Contact Reporter on page 256
VoIP on page 257
VoIP Security on page 259
Access Control Lists on page 261
Dialer on page 261
Contact Center on page 263
Avaya Cloud Services on page 265

System
Navigation: System | System
Additional configuration information
For additional information on time settings, see System Date and Time on page 581.
Configuration settings
These settings are mergeable with the exception of Locale and Favor RIP Routes over Static
Routes. Changing these settings requires a reboot of the system.
Field Description
Name Default: = System MAC Address.
A name to identify this system. This is typically used to identify the configuration by the
location or customer's company name. Some features such as Gatekeeper require the
system to have a name. This field is case sensitive and within any network of systems
must be unique. Do not use <, >, |, \0, :, *, ?, . or /.
Contact Default = Blank.
Information
This field is only be edited by service user with administrator rights. If Contact Information
is entered, it will set the system under 'special control'.
If the contact information is set using a standalone version of Manager, warnings that
"This configuration is under special control" are given when the configuration is opened
again. This can be used to warn other users of Manager that the system is being
monitored for some specific reason and provide them with contact details of the person
doing that monitoring.
Locale Sets default telephony and language settings based on the selection. It also sets various
external line settings and so must be set correctly to ensure correct operation of the
system. See Avaya IP Office™ Platform Locale Settings. For individual users, the system
settings can be overridden through their own locale setting Select User | User | Local.
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Configuration Mode Field Descriptions

Field Description
Location Default = None.
Specify a location to associate the system with a physical location. Associating a system
with a location allows emergency services to identify the source of an emergency call.
The drop down list contains all locations that have been defined in the Location page.
Customize Locale Settings
The Customize locale matches the Saudi Arabia locale but with the following additional controls shown below.
For other locales, these are set on System | Telephony | Tones and Music.
Tone Plan Default = Tone Plan 1
The tone plan control tones and ringing patterns. The options are:
• Tone Plan 1: United States.
• Tone Plan 2: United Kingdom.
• Tone Plan 3: France.
• Tone Plan 4: Germany.
• Tone Plan 5: Spain.
CLI Type Used to set the CLI detection used for incoming analogue trunks. The options are:
• DTMF
• FSK V23
• FSK BELL202

Device ID Server Edition Only. Displays the value set for Device ID on the System | System Events
| Configuration tab. If an SSL VPN is configured, Avaya recommends that the Device ID
match an SSL VPN service Account Name. Each SSL VPN service account name has an
associated SSL VPN tunnel IP address. Having the displayed Device ID match an SSL
VPN service account name helps identify a particular SSL VPN tunnel IP address to use
for remotely managing IP Office.
TFTP Server IP Default = 0.0.0.0 (Disabled. On Server Edition Systems, the default on Secondary and
Address Expansion servers is the Primary Server address.)
If the Phone File Server Type below is set to Custom, this address is included as the
TFTP file server address sent in the system’s DHCP response to phones.
The address 255.255.255.255 can be used to broadcast for the first available TFTP
server on the network.
Manager can act as a TFTP server and provides files from its configured binaries
directory. This requires the application setting File | Preferences | Preferences | Enable
BootP and TFTP Servers to be enabled.
On IP500 V2 systems, the LAN1 IP Address can be entered to specify the system’s own
memory card memory card as the TFTP file source. This requires the security setting
Security Settings | Unsecured Interfaces | Applications Controls | TFTP Directory
Read to be enabled.
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 System

Field Description
HTTP Server IP Default = 0.0.0.0 (Disabled).
Address
This address, if set, is used in a number of scenarios:
• DHCP Responses: If the Phone File Server Type below is set to Custom, this
address is included as the HTTP file server address sent in the system’s DHCP
response to phones.
• HTTP Redirection: If HTTP Redirection below is enabled, 96x1 H.323 phone binary
file requests sent to the system are redirected to this address.
• H175 Phones/Vantage Phones: Phone firmware file requests sent to the system from
these types of phone always redirected to this address.
Phone File Server Default = Memory Card (IP500 V2)/Disk (Linux system).
Type
For IP phones (H.323 and SIP) using the system as their DHCP server, the DHCP
response can include the address of a file server from which the phone should request
files. The setting of this field controls which address is used in the DHCP response. The
options are:
• Custom: The DHCP response the system provides to phones contains the addresses
set in the TFTP Server IP Address and HTTP Server IP Address fields.
• Disk: (Linux systems only) The system will respond to file requests from phones using
files on its own hard disk. The DHCP response the system provides to phones contains
its own LAN address as the TFTP and HTTP file server address.
• Memory Card: (IP500 V2 only) The system will respond to file requests from phones
using files on its own memory card. The DHCP response the system provides to
phones contains its own LAN address as the TFTP and HTTP file server address. This
is supported for up to 50 IP phones total.
• Manager: (IP500 V2 only) The system will forward any H.323 phone file request to the
configured Manager PC IP Address set below. The DHCP response the system
provides to phones contains the system’s LAN address as the HTTP file server
address.
- HTTP-TFTP Relay is support when using Manager as the TFTP server (not supported
by Linux based systems). This is done by setting the TFTP Server IP Address to the
address of the Manager PC and the HTTP Server IP Address to the control unit IP
address. This method is supported for up to 5 IP phones total.
HTTP Redirection Default = Off.
This setting allows 96x1 phones to use the system as the file server when requesting
their upgrade and settings files but have the requests for their large firmware files
redirected to the address set by the HTTP Server IP Address field. This field is available
when the Phone File Server Type is set to Memory Card or Disk.
• H175 and Vantage phone firmware requests are always redirected to the HTTPS
Server IP Address regardless of this and the Phone File Server Type settings.
Manager PC IP Default = 0.0.0.0 (Broadcast).
Address
This address is used when the Phone File Server Type is set to Manager.
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Configuration Mode Field Descriptions

Field Description
Avaya HTTP Default = Off.
Clients Only
When selected, the system only responds to HTTP requests from sources it identifies as
another system, an Avaya phone or application.
Enable SoftPhone Default = Off.
HTTP Provisioning
This option must be enabled if the IP Office Video Softphone is being supported.
Use Preferred Default = Off
Phone Ports
When selected, the system allows users to configure firewalls to block ports 80 and 443 if
alternate mechanism for administration is provided. Phones can use either port 411 or
8411 if supported. Legacy phones that still require 80 and 443 can continue to use those
ports through IP Office HTTP server. Where possible, HTTP requests from phones
received on ports 80 and 443 must result in the phone proceed to use 8411/411,
However, files continue to be served on the ports 80 and 443 to allow functionality of non-
compliant phones. Configuration files served to phones, that are not behind an SBC,
defines 8411 for HTTP and additionally 411 for TLS if the phone supports it and the phone
is remote, or incoming request is already secure.
When cleared, phones can continue to connect through all the four ports. DHCP provided
HTTP IP addresses are served.
Favor RIP Routes Default = Off
over Static Routes
RIP can be enabled on the system LAN1 and LAN2 interfaces and on specific Services.
When this setting is on, the RIP route to a destination overrides any static route to the
same destination in the system's IP Routes, regardless of the RIP route's metric. The only
exception is RIP routes with a metric of 16 which are always ignored.

Note:
If a previously learnt RIP route fails, the system applies a metric of 16 five minutes
after the failure. When off, any RIP route to a destination for which a static route has
been configured is ignored. This option is not supported on Linux based systems.
Automatic Backup Default = On.
This command is available with IP500 V2 systems. When selected, as part of its daily
backup process, the system automatically copies the folders and files from the System
SD card's /primary folder to its /backup folder. Any matching files and folders already
present in the /backup folder are overwritten.
Provider Default = Not visible. This field is visible only if the system has been branded by addition
of a special license for a specific equipment provider. The branding is fixed, that is it
remains even if the license is subsequently removed. The number shown is a unique
reference to the particular equipment provider for whom the system has been branded.
When branded, the equipment provider's name is displayed on idle phone displays and
other provider related features are enabled.
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 System

Field Description
Time Setting Time and date settings are only shown for IP500 V2 based systems. The time and date
Config Source for Linux based servers are set through the server’s Platform View menus (Settings |
System | Date and Time).
For IP500 V2 systems, the time is either set manually, obtained using Time protocol
(RFC868) requests or obtained using Network Time Protocol (RFC958) request. This field
is used to select which method is used and to apply ancillary settings based on the
selected method.
• None: The system to not make any time requests. The system time and date can be be
set and changed using by a user with system phone rights (see System Phone
Features on page 715). However, the system still automatically apply daylight saving
changes to the manually set time.
• Voicemail Pro/Manager: Both the Voicemail Pro service and the Manager program can
act as RFC868 Time servers for the system. Use of other RFC868 server sources is not
supported. They provide both the UTC time value and the local time as set on the PC.
The system makes a request to the specified address following a reboot and every 8
hours afterwards. This option should not be used with a Unified Communication Module
as in that scenario the voicemail server is being hosted by and getting its time from the
IP Office.
• SNTP: Use a list of SNTP servers to obtain the UTC time. The records in the list are
used one at a time in order until there is a response. The system makes a request to
the specified addresses following a reboot and every hour afterwards.
Time Settings — Voicemail Pro/Manager
These settings are shown for IP500 V2 based systems where the Time Setting Config Source has been set to
Voicemail Pro/Manager.
IP Address Default = 0.0.0.0 (Broadcast) The address to which the RFC868 request is sent. 0.0.0.0
means default operation. In this mode, following a reboot the control unit makes time
requests on its LAN interfaces. It first makes a request to the IP address set and, if it
receives no reply, then makes a broadcast request.
If you are running Manager when the voicemail server starts, voicemail does not start as
a time server. It is therefore recommended that you have no copy of Manager running
when you start or restart the voicemail server. Manager can be disabled from acting as a
RFC868 time server by deselecting the Enable Time Server option (File | Preferences |
Edit | Preferences).
Time Offset Default = 00:00. This value is not normally set as any time changes, including daylight
saving changes, that occur on the PC will be matched by the system.
Time Settings — None/SNTP
These settings are shown for IP500 V2 based systems where the Time Setting Config Source has been set to
None or SNTP.
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Configuration Mode Field Descriptions

Field Description
Time Server Default = Blank
Address
Displayed when the Time Setting Config Source is set to SNTP. Enter a list of IP
addresses, host names, or fully qualified domain names (FQDN) for the SNTP servers.
Separate each record with a space. The use of broadcast addresses is not supported.
The list is used in order of the records until a response is received.
Time Zone Select a time zone from the list. This sets the default time offset and DST to match the
chosen time zone.
Local Time Offset Default is based on the currently selected time zone.
from UTC
This setting is used to set the local time difference from the UTC time value provided by
an SNTP server. For example, if the system is 5 hours behind UTC, this field should be
configured with -05:00 to make the adjustment. The time offset can be adjusted in 15
minute increments. If also using the daylight time saving settings below, use this offset to
set the non-DST local time.
Automatic DST Default is based on the currently selected time zone.
When set to On, the system automatically corrects for daylight saving time (DST)
changes as configured in the Clock Forward/Back Settings below.
Clock Forward/ Default is based on the currently selected time zone.
Back Settings
Click Edit to configure the time and date for DST clock corrections. In the Daylight Time
(Start Date — End Settings window, you can configure the following information:
Date (DST Offset))
• DST Offset: the number of hours to shift for DST.
• Clock Forward/Back: Select Go Forward to set the date when the clock will move
forward. Select Go Backwards to set the date when the clock will move backward.
• Local Time To Go Forward: The time of day to move the clock forward or backward.
• Date for Clock Forward/Back: Set the year, month and day for moving the clock
forwards and backwards.
Once you click OK, the forward and back dates, plus the DST offset, are displayed using
the format (Start Date — End Date (DST Offset)).
File Writer IP Default = 0.0.0.0 (Disabled)
Address
This field set the address of the PC allowed to send files to the System SD card installed
in the system using HTTP or TFTP methods other than embedded file management.
• On non-Linux based systems, this field sets the address of the PC allowed to send files
to the memory card using HTTP or TFTP methods other than embedded file
management.
• For Linux based systems it is applied to non-embedded file management access to
the /opt/ipoffice folder on the server.
An address of 255.255.255.255 allows access from any address. If embedded file
management is used, this address is overwritten by the address of the PC using
embedded file management (unless set to 255.255.255.255).
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 System

Field Description
Dongle Serial Displayed for pre-Release 10.0 IP500 V2 systems using ADI licensing only. For system’s
Number using PLDS licensing, see the PLDS Host ID (License | License).
This field is for information only. It shows the serial number of the feature key dongle
against which the system last validated its licenses. Local is shown for a serial port,
Smart Card or System SD feature key plugged directly into the control unit. Remote is
shown for a parallel or USB feature key connected to a feature Key Server PC. The serial
number is printed on the System SD card and prefixed with FK.
System Displayed for Linux based systems. This field is for information only.
Identification
This is the unique system reference that is used to validate licenses issued for this
particular system. For a physical server this is a unique value based on the server
hardware. For a virtual server this value is based on several factors including the LAN1
and LAN2 IP addresses, the host name and the time zone. If any of those are changed,
the System ID changes and any existing licenses become invalid.
AVPP IP Address Default = 0.0.0.0 (Disabled)
Where Avaya 3600 Series SpectraLink wireless handsets are being used with the
system, this field is used to specify the IP address of the Avaya Voice Priority Processor
(AVPP)

Related links
System on page 192

LAN1
Navigation: System | LAN1
Used to configure the behavior of the services provided by the system's first LAN interface.
Up to 2 LAN's (LAN1 and LAN2) can be configured. The control unit has 2 RJ45 Ethernet ports,
marked as LAN and WAN. These form a full-duplex managed layer-3 switch. Within the system
configuration, the physical LAN port is LAN1, the physical WAN port is LAN2.
Configuring both interfaces with the same IP address on the same subnet is not supported.
However, no warning is issued when this configuration is implemented.
Related links
System on page 192
LAN Settings on page 199
VoIP on page 201
Network Topology on page 209
DHCP Pools on page 213

LAN Settings
Navigation: System | LAN | LAN Settings
Used to set the general LAN settings for the LAN interface such as the IP address mode.

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Configuration Mode Field Descriptions

Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
IP Address Default = 192.168.42.1 or DHCP client.
This is the IP address of the Control Unit on LAN1. If the control unit is also acting
as a DHCP server on the LAN, this address is the starting address for the DHCP
address range.
IP Mask Default = 255.255.255.0 or DHCP client.
This is the IP subnet mask used with the IP address.
Primary Trans. IP Default = 0.0.0.0 (Disabled)
Address
This setting is only available on control units that support a LAN2. Any incoming IP
packets without a service or session are translated to this address if set.
RIP Mode Default = None.
Routing Information Protocol (RIP) is a method by which network routers can
exchange information about device locations and routes. Routes learnt using RIP
are known as 'dynamic routes'. The system also supports 'static routes' though its
IP Route records. For Server Edition systems this setting is only available on
Expansion System (V2) systems. The options are:
• None: The LAN does not listen to or send RIP messages
• Listen Only (Passive): Listen to RIP-1 and RIP-2 messages in order to learn RIP
routes on the network.
• RIP1: Listen to RIP-1 and RIP-2 messages and send RIP-1 responses as a sub-
network broadcast.
• RIP2 Broadcast (RIP1 Compatibility): Listen to RIP-1 and RIP-2 messages and
send RIP-2 responses as a sub-network broadcast.
• RIP2 Multicast: Listen to RIP-1 and RIP-2 messages and send RIP-2 responses
to the RIP-2 multicast address.
Enable NAT Default = Off
This setting controls whether NAT should be used for IP traffic from LAN1 to LAN2.
This setting should not be used on the same LAN interface as a connected WAN3
expansion module.
Number of DHCP Default = 200 or DHCP client. Range = 1 to 999.
IP Addresses
This defines the number of sequential IP addresses available for DHCP clients.
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 System

Field Description
DHCP Mode Default = DHCP Client.
This controls the control unit's DHCP mode for the LAN. When doing DHCP:
• LAN devices are allocated addresses from the bottom of the available address
range upwards.
• Dial In users are allocated addresses from the top of the available range
downwards.
• If the control unit is acting as a DHCP server on LAN1 and LAN2, Dial in users
are allocated their address from the LAN1 pool of addresses first.
The options are:
• Server: When this option is selected, the system will act as a DHCP Server on
this LAN, allocating address to other devices on the network and to PPP Dial in
users.
• Disabled When this option is selected, the system will not use DHCP. It will not
act as a DHCP server and it will not request an IP address from a DHCP server
on this LAN.
• Dial In When this option is selected, the system will allocate DHCP addresses to
PPP Dial In users only. On systems using DHCP pools, only addresses from a
pool on the same subnet as the system's own LAN address will be used.
• Client When this option is selected, the system will request its IP Address and IP
Mask from a DHCP server on the LAN.

Note:
Do not use this option with a limited time lease line.
• Advanced: The system can be configured with a number of DHCP Pools from
which it can issue IP addresses.

Related links
LAN1 on page 199

VoIP
Navigation: System | LAN | VoIP
Additional configuration information
For more information on remote H.323 extensions, see Configuring Remote H.323 Extensions on
page 625.
Configuration settings
Used to set the system defaults for VoIP operation on the LAN interface.
The following settings are mergeable:
• Auto-create Extn
• Auto-create User

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Configuration Mode Field Descriptions

• H.323 Signalling over TLS

• Remote Call Signalling Port
• Auto-create Extn/User
• Enable RTCP Monitoring on Port 5005
• RTCP collector IP address for phones
• Scope
• Initial keepalives
• Periodic timeout
• VLAN
• 1100 Voice VLAN Site Specific Option Number (SSON)
• 1100 Voice VLAN IDs
The remaining settings are not mergeable. Changes to these settings requires a reboot of the
system.
Field Description
H.323 Gatekeeper Enable
Default = Off
This settings enables gatekeeper operation.
H.323 Signalling over Default = Disabled. For hosted deployments, default = Preferred.
TLS
When enabled, TLS is used to secure the registration and call signalling
communication between IP Office and endpoints that support TLS. The H.323 phones
that support TLS are 9608, 9611, 9621, and 9641running firmware version 6.6 or
higher.
When enabled, certificate information is configured in the 46xxSettings.txt file
on IP Office and automatically downloaded to the phone. When IP Office receives a
request from the phone for an identity certificate, IP Office searches it's trusted
certificate store and finds the root CA that issued it's identity certificate. IP Office then
provides the root CA as an auto-generated certificate file named Root-CA-
xxxxxxxx.pem.
For information on IP Office certificates, see Security Mode | System | Certificates.
The options are:
• Disabled: TLS is not used.
• Preferred: Use TLS when connecting to a phone that supports TLS.
• Enforced: TLS must be used. If the phone does not support TLS, the connection is
rejected.
When set to Enforced, the Remote Call Signalling Port setting is disabled.
If TLS security is enabled (Enforced or Preferred), it is recommended that you
enable a matching level of media security on System | VoIP Security.
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 System

Field Description
H.323 Remote Extn Default = Off.
Enable
The system can be configured to support remote H.323 extensions in the case where
NAT is used in the connection path. This could be the case where the IP Office is
located behind a corporate NAT/Firewall router and/or the H.323 phone is located
behind residential NAT enable router.
The use of this option and the interaction and configuration of external third party
elements is beyond the scope this help file.
In the case where the public IP address of the corporate router is unknown, the LAN's
Network Topology settings should be used to configure a STUN Server. Enabling H.
323 Remote Extn Enable allows configuration of the RTP Port number Range
(NAT) settings.
Currently, only 9600 Series phones are supported as H.323 remote extensions.
Auto-create Extn Default = Off
The field to set up auto creation of extensions for H.323 phones registering
themselves with the System as their gatekeeper. If selected, the system displays the
Auto Create Extension Password window prompting you to type a Password and
Confirm Password. This password is used for subsequent auto creation of
extensions. A message H.323 Auto-Create Extension option is active
is flashed next to the Auto Create Extension field till the option is cleared. SIP
Extensions use a separate setting, see below. This setting is not supported on
systems configured to use WebLM server licensing.
If using resilience backup to support Avaya IP phones, Auto-create Extn and Auto-
create User should not be left enabled after initial configuration or any subsequent
addition of new extensions and users. Leaving auto-create options enabled on a
system that is a failover target may cause duplicate extension/user records on the
multi-site network under multiple failure scenarios.
For security, any auto-create settings set to On are automatically set to Off after 24
hours.

SIP Trunks Enable Default = On.

This settings enables support of SIP trunks. It also requires entry of SIP Trunk
Channels licenses.
Enabling SIP Trunks Enable allows configuration of the RTP Port number Range
(NAT) settings.
SIP Registrar Enable
Default = Off.
Used to set the system parameters for the system acting as a SIP Registrar to which SIP endpoint devices can
register. Separate SIP registrars can be configured on LAN1 and LAN2. Registration of a SIP endpoint requires
an available IP Endpoints license. SIP endpoints are also still subject to the extension capacity limits of the
system.
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Configuration Mode Field Descriptions

Field Description
SIP Remote Extn Default = Off.
Enable
The system can be configured to support remote SIP extensions in the case where
NAT is used in the connection path. This could be the case where the IP Office is
located behind a corporate NAT/Firewall router and/or the SIP phone is located
behind residential NAT enable router.
This option cannot be enabled on both LAN1 and LAN2.
The use of this option and the interaction and configuration of external third party
elements is beyond the scope this help file.
In the case where the public IP address of the corporate router is unknown, the LAN's
Network Topology settings should be used to configure a STUN Server. Enabling SIP
Remote Extn Enable allows configuration of:
• the Remote UDP Port, Remote TCP Port, Remote TLS Port settings
• the Port Number Range (NAT) settings
Currently, only Avaya Equinox™ for Windows, Avaya Equinox™ for iOS, one-X Mobile
iOS and one-X Mobile Android SIP clients are supported as SIP remote extensions.
Allowed SIP User Default = Block blacklist only
agents
The drop-down menu to select which SIP devices are allowed to register with the IP
Office system. Depending on the selection, IP Office allows registration of SIP User
Agents specified using the System > VOIP > Access Control Lists tab. The options
are:
• Allow All: Do not block any devices based on the UI strings.
• Block Blacklist Only: Block devices whose UA string is listed in the SIP UA
Blacklist.
• Avaya Clients & Whitelisted: Only allow devices with an Avaya UA string or
whose UA string is listed in the SIP UA Whitelist.
• Avaya Clients Only: Only allow clients with an Avaya UA string.
• Whitelisted only: Only allow devices whose UA string is listed in the SIP UA
Whitelist.
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 System

Field Description
Auto-create Extn/User Default = Off.
The field to set up auto creation of extensions for SIP phones registering themselves
with the SIP registrar. If selected, the system displays the Auto Create Extension
Password window prompting you to type a Password and Confirm Password. This
password is used for subsequent auto creation of extensions. A message SIP
Auto-Create Extension/User option is active is flashed next to the Auto
Create Extension/User field till the option is cleared. This setting is not supported on
systems configured to use WebLM server licensing.
For security, any auto-create settings set to On are automatically set to Off after 24
hours.

Note:
This setting is not applicable to the Avaya A175 Desktop Video Device with the
Avaya Communicator.
SIP Domain Name Default = Blank
This value is used by SIP endpoints for registration with the IP Office system. SIP
endpoints register with IP Office using their SIP address that consists of their phone
number and IP Office SIP domain. Since IP Office does not allow calls from
unauthorized entities, the SIP domain does not need to be resolvable. However, the
SIP domain should be associated with FQDN (Fully Qualified Domain Name) for
security purposes. The entry should match the domain suffix part of the SIP Registrar
FQDN below, for example, example.com. If the field is left blank, registration uses
the LAN 1, LAN2, or public IP address.

Note:
For Avaya SIP telephones supported for resilience, the SIP Domain Name must
be common to all systems providing resilience.
SIP Registrar FQDN Default = Blank
This is the SIP registrar fully qualified domain name, for example,
server1.example.com, to which the SIP endpoint should send its registration
request. This address must be resolvable by DNS to the IP address of the IP Office
system or to the IP address, such as that of an Avaya SBCE, through which the SIP
endpoints reach the IP Office system.
Challenge Expiry Time Default = 10.
(secs)
The challenge expiry time is used during SIP extension registration. When a device
registers, the system SIP Registrar will send a challenge back to the device and waits
for an appropriate response. If the response is not received within this timeout the
registration is failed.
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Configuration Mode Field Descriptions

Field Description
Layer 4 Protocol Default = TCP and UDP. This field is used to select which protocols are supported for
SIP connections: TCP, UDP, or TLS.
• UDP Port: Default = 5060. The port to use for SIP UDP support if UDP is selected
as the Layer 4 Protocol above.
• TCP Port: Default = 5060. The port to use for SIP TCP support if TCP is selected
as the Layer 4 Protocol above.
• TLS Port: Default = 5061. The port to use for SIP TLS support.
• Remote UDP Port: Default = 5060. The port to use for SIP UDP support if UDP is
selected as the Layer 4 Protocol for remote SIP extension.
• Remote TCP Port: Default = 5060. The port to use for SIP TCP support if TCP is
selected as the Layer 4 Protocol for remote SIP extension.
• Remote TLS Port: Default = 5061. The port to use for SIP TLS support if TLS is
selected as the Layer 4 Protocol for remote SIP extension.

Note:
The E129 phone does not support UDP. In IP Office release 10 and higher, UDP
support has been removed from the configuration file sent to the phone. For the
E129 phone, you must enable TCP.

RTP
Port Number Range For each VoIP call, a receive port for incoming Real Time Protocol (RTP) traffic is
selected from a defined range of possible ports, using the even numbers in that
range. The Real Time Control Protocol (RTCP) traffic for the same call uses the RTP
port number plus 1, that is the odd numbers. For control units and Avaya H.323 IP
phones, the default port range used is 49152 to 53246. On some installations, it may
be a requirement to change or restrict the port range used. It is recommended that
only port numbers between 49152 and 65535 are used, that being the range defined
by the Internet Assigned Numbers Authority (IANA) for dynamic usage.

Important:
The minimum and maximum settings of the port range should only be adjusted
after careful consideration of the customer network configuration and existing
port usage. For pre-Release 8.1 systems, the gap between the minimum and
maximum port values must be at least 1024. For Release 8.1 and higher, the
gap between the minimum and maximum port values must be at least 254.
Port Range (minimum) IP500 V2 default = 46750. Range = 46750 to 50750.
Linux default = 40750. Range = 40750 to 50750
This sets the lower limit for the RTP port numbers used by the system.
Port Range IP500 V2 default = 50750. Range = 46750 to 50750.
(maximum)
Linux default = 50750. Range = 47000 to 50750
This sets the upper limit for the RTP port numbers used by the system.

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 System

Field Description
Port Number Range (NAT)
These settings are available when either H.323 Remote Extn Enable, SIP Trunks Enable, or SIP Remote
Extn Enable is set to On.
This option is not supported if System | LAN | Network Topology | Firewall/NAT Type is set to Symmetric
Firewall or Open Internet.
Port Range (minimum) IP500 V2 default = 46750. Range = 46750 to 50750.
Linux default = 40750. Range = 40750 to 50750
This sets the lower limit for the RTP port numbers used by the system.
Port Range IP500 V2 default = 50750. Range = 46750 to 50750.
(maximum)
Linux default = 50750. Range = 40750 to 50750
This sets the upper limit for the RTP port numbers used by the system.
Enable RTCP Monitor Default = On.
On Port 5005
For 1600, 4600, 5600 and 9600 Series H.323 phones, the system can collect VoIP
QoS (Quality of Service) data from the phones. For other phones, including non-IP
phones, it can collect QoS data for calls if they use a VCM channel. The QoS data
collected by the system is displayed by the System Status Application.
This setting is mergeable. However, it only affects H.323 phones when they register
with the system. Therefore, any change to this setting requires H.323 phones that
have already been registered to be rebooted. Avaya H.323 phones can be remotely
rebooted using the System Status Application.
The QoS data collected includes: RTP IP Address, Codec, Connection Type, Round
Trip Delay, Receive Jitter, Receive Packet Loss.
This setting is not the same as the RTCPMON option within Avaya H.323 phone
settings. The system does not support the RTCPMON option.
RTCP collector IP Default = Blank.
address for phones
This setting is used to manually set the destination for the RTCP Monitor data
described above Enable RTCP Monitor On Port 5005 field above. This enables you
to send the data collected to a third party QoS monitoring application.
The Enable RTCP Monitor On Port 5005 must be turned Off to enable this field.
Changes to this setting requires a reboot of the phones.
Keepalives
These settings are used to facilitate NAT traversal of RTP/RTCP packets and are applicable to all RTP/RTCP
session on the network interface. You should enable these settings on interfaces connected to NAT devices if
you are using SIP trunks and/or H323 and SIP remote workers.
Scope Default = Disabled
Select whether the sending of keepalive packets should be disabled or sent for RTP
or for both RTP and RTCP.
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Configuration Mode Field Descriptions

Field Description
Periodic timeout Default = 0 (Off). Range = 0 to 180 seconds.
Sets how long the system will wait before sending a keepalive if no other packets of
the select SCOPE are seen.
Initial keepalives Default = Disabled.
If enabled, keepalives can also been sent during the initial connection setup.

DiffServ Settings
When transporting voice over low speed links it is possible for normal data packets (1500 byte packets) to
prevent or delay voice packets (typically 67 or 31 bytes) from getting across the link. This can cause
unacceptable speech quality. Therefore it is important that all traffic routers and switches in a network to have
some form of Quality of Service mechanism (QoS). QoS routers are essential to ensure low speech latency and
to maintain sufficient audible quality.
The system applies the DiffServ settings to outgoing traffic on any SIP lines which have Line | SIP Line |
Transport | Use Network Topology Info set to match the LAN interface.
The system supports the DiffServ (RFC2474) QoS mechanism. This uses a Type of Service (ToS) field in the IP
packet header.
The hex and decimal entry fields for the following values are linked, the hex value being equal to the decimal
multiplied by 4.
DSCP (Hex) Default = B8 (Hex)/46 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The DiffServ Code Point (DSCP) setting applied to VoIP calls. By default, the same
setting is used for audio and video. If desired, you can configure separate values for
audio and video. For correct operation, especially over WAN links, the same value
should be set at both ends.
Video DSCP (Hex) Default = B8 (Hex)/46 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The DiffServ Code Point (DSCP) setting applied to video VoIP calls. For correct
operation, especially over WAN links, the same value should be set at both ends.
DSCP Mask (Hex) Default = FC (Hex)/63 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
Allows a mask to be applied to packets for the DSCP value.
SIG DSCP (Hex) Default = 88 (Hex)/34 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
This setting is used to prioritize VoIP call signaling.

DHCP Settings
Primary Site Specific Default = 176. Range = 128 to 254.
Option Number
A site specific option number (SSON) is used as part of DHCP to request additional
(4600/5600)
information. 176 is the default SSON used by 4600 Series and 5600 Series IP
phones.
Secondary Site Default = 242. Range = 128 to 254.
Specific Option
Similar to the primary SSON. 242 is the default SSON used by 1600 and 9600 Series
Number (1600/9600)
IP phones requesting installation settings via DHCP.
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 System

Field Description
VLAN Default = Not present. This option is applied to H.323 phones using the system for
DHCP support. If set to Disabled, the L2Q value indicated to phones in the DHCP
response is 2 (disabled). If set to Not Present, no L2Q value is included in the DHCP
response.
1100 Voice VLAN Site Default = 232.
Specific Option
This is the SSON used for responses to 1100/1200 Series phones using the system
Number (SSON)
for DHCP.
1100 Voice VLAN IDs Default = Blank.
For 1100/1200 phone being supported by DHCP, this field sets the VLAN ID that
should be provided if necessary. Multiple IDs (up to 10) can be added, each
separated by a + sign.

Related links
LAN1 on page 199

Network Topology
Navigation: System | LAN | Network Topology
STUN (Simple Traversal of UDP through NAT) is a mechanism used with overcome the effect of
NAT firewalls. The network address translation (NAT) action performed by this type of firewall can
have negative effects on VoIP calls.
Test packets are sent by the system to the address of the external STUN server, those packets
crossing the firewall in the process. The STUN server replies and includes copies of the packets it
received in the reply. By comparing the packet sent and received, it is possible for the system to
determine the type of NAT firewall and to modify future packets to overcome the effects of the
firewall.
These settings are used for SIP trunk connections from the LAN, H.323 and SIP remote
extensions. For further details of system SIP operation refer to the SIP Line section. The use of
STUN is unnecessary if the SIP ITSP uses a Session Border Controller (SBC). Use of SIP
requires entry of SIP Trunk Channels licenses.
The following fields can be completed either manually or the system can attempt to automatically
discover the appropriate values. To complete the fields automatically, only the STUN Server IP
Address is required. STUN operation is then tested by clicking Run STUN. If successful the
remaining fields are filled with the results.
Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
STUN Server IP Default = Blank
Address
Enter the IP address or fully qualified domain name (FQDN) of the SIP ITSP's STUN
server. The system will send basic SIP messages to this destination and from data
inserted into the replies can try to determine the type NAT changes being applied by
any firewall between it and the ITSP.
STUN Port Default = 3478.
Defines the port to which STUN requests are sent if STUN is used.
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 System

Field Description
Firewall/NAT Type Default = Unknown
The settings here reflect different types of network firewalls. The options are:
• Blocking Firewall
• Symmetric Firewall: SIP packets are unchanged but ports need to be opened
and kept open with keep-alives. If this type of NAT is detected or manually
selected, a warning ‘Communication is not possible unless the STUN server is
supported on same IP address as the ITSP’ will be displayed as part of the
manager validation.
• Open Internet: No action required. If this mode is selected, settings obtained by
STUN lookups are ignored. The IP address used is that of the system LAN
interface.
• Symmetric NAT: A symmetric NAT is one where all requests from the same
internal IP address and port, to a specific destination IP address and port, are
mapped to the same external IP address and port. If the same host sends a
packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a
packet can send a UDP packet back to the internal host. SIP Packets need to be
mapped but STUN will not provide the correct information unless the IP address
on the STUN server is the same as the ITSP Host. If this type of NAT/Firewall is
detected or manually selected, a warning ‘Communication is not possible unless
the STUN server is supported on same IP address as the ITSP’ will be displayed
as part of the manager validation.
• Full Cone NAT: A full cone NAT is one where all requests from the same internal
IP address and port are mapped to the same external IP address and port.
Furthermore, any external host can send a packet to the internal host, by sending
a packet to the mapped external address. SIP packets need to be mapped to NAT
address and Port; any Host in the internet can call in on the open port, that is the
local info in the SDP will apply to multiple ITSP Hosts. No warning will be
displayed for this type of NAT because the system has sufficient information to
make the connection).
• Restricted Cone NAT: A restricted cone NAT is one where all requests from the
same internal IP address and port are mapped to the same external IP address
and port. Unlike a full cone NAT, an external host (with IP address X) can send a
packet to the internal host only if the internal host had previously sent a packet to
IP address X. SIP packets needs to be mapped. Responses from hosts are
restricted to those that a packet has been sent to. So if multiple ITSP hosts are to
be supported, a keep alive will need to be sent to each host. If this type of NAT/
Firewall is detected or manually selected, no warning will be displayed for this type
of NAT.
• Port Restricted Cone NAT: A port restricted cone NAT is like a restricted cone
NAT, but the restriction includes port numbers. Specifically, an external host can
send a packet, with source IP address X and source port P, to the internal host
only if the internal host had previously sent a packet to IP address X and port P.
SIP packets needs to be mapped. Keep-alives must be sent to all ports that will be
the source of a packet for each ITSP host IP address. If this type of NAT/Firewall
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Configuration Mode Field Descriptions

Field Description
is detected or manually selected, no warning will be displayed for this type of NAT.
However, some Port Restricted NAT's have been found to be more symmetric in
behavior, creating a separate binding for each opened Port, if this is the case the
manager will display a warning ‘Communication is not possible unless the STUN
server is supported on same IP address as the ITSP’ as part of the manager
validation.
• Static Port Block: Use the RTP Port Number Range specified on the VoIP tab
without STUN translation. Those ports must be fixed as open on any NAT firewall
involved
• One-To-One NAT: This setting supports IP Office cloud deployments where the
Primary server is behind a NAT that performs IP address translation but not port
mappings. All required ports must be open on the NAT.
When set to One-To-One NAT, the following configuration settings are applied and
cannot be edited.
- The LAN | Network Topology | Public Port values are set to 0.
- LAN | VoIP | SIP Registrar Enable remote protocol port values are set to equal
their corresponding local protocol port values.
- The LAN | VoIP | RTP | Port Number Range (NAT) Minimum and Maximum
values are set to equal the corresponding Port Number Range values.
• Unknown
Binding Refresh Time Default = 0 (Never). Range = 0 to 3600 seconds.
(seconds)
Having established which TCP/UDP port number to use, through either automatic or
manual configuration, the system can send recurring ‘SIP OPTIONS requests’ to the
remote proxy terminating the trunk. Those requests will keep the port open through
the firewall. Requests are sent every x seconds as configured by this field.

Note:
If a binding refresh time has not been set you may experience problems
receiving inbound SIP calls as they are unable to get through the Firewall. In
these circumstances make sure that this value has been configured.
Public IP Address Default = 0.0.0.0 This value is either entered manually or discovered by the Run
STUN process. If no address is set, the system LAN1 address is used.
Public Port Default = 0
The public port value for UDP, TCP, and TLS. For each protocol, this value is either
entered manually or discovered by the Run STUN process.
Run STUN This button tests STUN operation between the system LAN and the STUN Server IP
Address set above. If successful the results are used to automatically fill the
remaining fields with appropriate values discovered by the system. Before using Run
STUN the SIP trunk must be configured.

When this option is used, a information icon is shown against the fields to
indicate that the values were automatically discovered rather than manually entered.
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Field Description
Run STUN on startup Default = Off
This option is used in conjunction with values automatically discovered using Run
STUN. When selected, the system will rerun STUN discovery whenever the system
is rebooted or connection failure to the SIP server occurs.

Related links
LAN1 on page 199

DHCP Pools
Navigation: System | LAN | DHCP Pools
DHCP pools allows for the configuration of of IP address pools for allocation by the system when
acting as a DHCP server. On an IP500 V2 system, you can configure up to 8 pools. On Server
Edition Linux systems, you can configure up to 64 pools.
By default the DHCP settings (IP Address, IP Mask and Number of DHCP IP Addresses) set on
the LAN Settings tab are reflected by the first pool here. For support of PPP Dial In address
requests, at least one of the pools must be on the same subnet as the system's LAN. Only
addresses from a pool on the same subnet as the system's own LAN address will be used for PPP
Dial In.
These settings are mergeable. However, the following actions require a merge with service
disruption:
• Changing the Start Address, Subnet Mask or Default Router value for an existing DHCP
Pool of addresses.
• Decreasing Pool Size for an existing DHCP Pool of addresses.
• Deleting an existing DHCP Pool of addresses.
When these actions are performed, the DHCP (Server or DialIn) is re-initialized which triggers a
reboot of the Avaya DHCP Clients (H.323 and SIP) in order to force the Avaya DHCP clients to
renew their IP address lease and apply the new settings. For the remaining Avaya and non-Avaya
DHCP clients, you must manually reboot the devices in order to force the IP Addresses lease
renewal. Otherwise, the devices continue to use the allocated IP addresses until the IP addresses
lease time out expires. IP address lease time out is set to three days.
The DHCP server re-initialization causes a reboot of all Avaya DHCP clients and not only of the
DHCP clients that have obtained an IP Address within the modified DHCP Pool IP range. Note
that IP Office supports phone reboot only for E129 and B179 SIP phone models.

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Configuration Mode Field Descriptions

Field Description
Apply to Avaya Default = Off.
IP Phones Only
When set to On, the DHCP addresses are only used for requests from Avaya IP
phones. Other devices connected to the system LAN will have to use static
addresses or obtain their address from another DHCP server.
In addition to the above control, Avaya IP phones will only complete DHCP against a
DHCP server configured to supports a Site Specific Option Number (SSON) that
matches that set on the phone. The SSON numbers supported by the system DHCP
are set on the VoIP sub-tab.
Once set to On and the configuration has been merged, you must manually reboot
the non-Avaya DHCP Client devices in order to force IP addresses lease renewal and
to make the settings new values effective. Otherwise the non-Avaya DHCP Client
devices will continue to use the allocated IP addresses until the IP addresses lease
time out expires. IP address lease time out is set to three days.
DHCP Pool Up to 8 pools can be added. The first pool matches the IP Address, IP Mask and
Number of DHCP IP Addresses on the LAN Settings sub-tab. When adding or editing
pools, Manager will attempt to warn about overlaps and conflicts between pools. The
options are:
• Start Address Sets the first address in the pool.
• Subnet Mask: Default = 255.255.255.0 Sets the subnet mask for addresses issued
from the pool.
• Default Router: Default = 0.0.0.0 For pools issuing IP addresses on the same
subnet as the system LAN's, 0.0.0.0 instructs the system to determined the actual
default router address to issue by matching the IP address/subnet mask being
issued in the IP Routing table. This matches the default behaviour used by systems
without multiple pools. For pools issuing addresses not on the same subnet as the
system LAN's, the default router should be set to the correct value for devices on
that subnet.
• Pool Size: Default = 0 Set the number of DHCP client addresses available in the
pool.

Related links
LAN1 on page 199

LAN2
Navigation: System | LAN2
These settings used to configure the system's second LAN interface. The fields available for LAN2
are the same as for LAN1 except for the following additional field.
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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 System

Field Description
Firewall Default = <None> (No firewall)
Allows the selection of a system firewall to be applied to traffic routed from LAN2 to
LAN1.

Related links
System on page 192

DNS
Navigation: System | DNS
DNS is a mechanism through which the URL's requested by users, such as www.avaya.com, are
resolved into IP addresses. These requests are sent to a Domain Name Server (DNS) server,
which converts the URL to an IP address. Typically the internet service provider (ISP) will specify
the address of the DNS server their customers should use.
WINS (Windows Internet Name Service) is a similar mechanism used within a Windows network
to convert PC and server names to IP addresses via a WINS server.
If the system is acting as a DHCP server, in addition to providing clients with their own IP address
settings it can also provide them with their DNS and WINS settings if requested by the client.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Related links
System on page 192

Voicemail
Navigation: System | Voicemail
Additional configuration information
For information on configuring Voicemail Pro resiliency, see Server Edition Resiliency on
page 729.
Configuration settings
The following settings are used to set the system's voicemail server type and location. Fields are
enabled or grayed out as appropriate to the selected voicemail type. Refer to the appropriate
voicemail installation manual for full details.
These settings are mergeable with the exception of Voicemail Type and Voicemail IP Address.
Changes to these settings requires a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Voicemail Type Defaults: Non-Server Edition = Embedded Voicemail, Primary Server = Voicemail Pro,
Server Edition Secondary Server with independent voicemail or for Outbound Contact
Express = Voicemail Pro, Server Edition Others: Centralized Voicemail.
Sets the type of voicemail system being used. The options are:
• None: No voicemail operation.
• Analogue Trunk MWI: Select this option to support receiving a message waiting
indicator (MWI) signal from analog trunks terminating on the ATM4U-V2 card. MWI
is a telephone feature that turns on a visual indicator on a telephone when there are
recorded messages.
• Avaya Aura Messaging: Select this option if you want to configure the system to
use Avaya Aura Messaging as the central voicemail system. If you choose this
option, you are still able to use Embedded Voicemail or Voicemail Pro at each
branch to provide auto-attendant operation and announcements for waiting calls.
When selected, access to voicemail is routed via an SM line to the numbers
specified in the AAM Number field. The optional AAM PSTN Number can be
configured for use when the SM Line is not in service.
For a setup where the voicemail box numbers configured on Avaya Aura Messaging
or Modular Messaging are same as the caller's DID, the short code to route the
PSTN call should be such that the caller-id is withheld ( "W" in the telephone-
number of the shortcode ). This is to make sure that, during rainy day - the voicemail
system does not automatically go to the voicemail box of the caller based on the
caller id.
• Call Pilot: Select this option if you want to configure the system to use CallPilot over
SIP as the central voicemail system. If you choose this option, you are still able to
use Embedded Voicemail or Voicemail Pro at each branch to provide auto-
attendant operation and announcements for waiting calls. When selected, access to
voicemail is routed via SM line to the numbers specified in the CallPilot Number
field.

Note:
The CallPilot PSTN Number field and associated Enable Voicemail
Instructions Using DTMF check box are not supported. IP Office cannot
access the CallPilot system over the PSTN when the Session Manager line is
down.

Note:
Users can access their CallPilot voicemail by dialing the Voicemail Collect short
code. Access to CallPilot voicemail from Auto Attendant cannot be enabled by
setting a Normal Transfer action to point to the Voicemail Collect short code. If
desired, it can be enabled by setting a Normal Transfer action to point to the
CallPilot number.
• Centralized Voicemail Select this option when using a Voicemail Pro system
installed and licensed on another system in a multi-site network. The outgoing line
group of the H.323 IP line connection to the system with the Voicemail Pro should
be entered as the Voicemail Destination. In a Server Edition network this option is
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 System

Field Description
used on the Secondary Server and expansion systems to indicate that they use the
Primary Server for as their voicemail server.
• Distributed Voicemail: This option can be used when additional Voicemail Pro
voicemail servers are installed in a multi-site network and configured to exchange
messages with the central voicemail server using email. This option is used if this
system should use one of the additional servers for its voicemail services rather
than the central sever. When selected, the Voicemail Destination field is used for
the outgoing H.323 IP line to the central system and the Voicemail IP Address is
used for the IP address of the distributed voicemail server the system should use.
This option is not supported by Server Edition systems.
• Embedded Voicemail On systems with an Avaya memory card, select this option to
run Embedded Voicemail which stores messages and prompts on the memory card.
It also supports internal Auto Attendant configuration through the system
configuration. The IP500 V2 supports 2 simultaneous Embedded Voicemail calls by
default but can be licensed for up to 6. The licensed limit applies to total number of
callers leaving messages, collecting messages and or using an auto attendant. This
option is not supported by Server Edition systems.
• Group Voicemail This option is used to support third-party voicemail systems
attached by extension ports in the group specified as the Voicemail Destination.
This option is not supported by Server Edition systems.
• Modular Messaging over SIP Select this option if you want to configure the system
to use Modular Messaging over SIP as the central voicemail system. If you choose
this option, you are still able to use Embedded Voicemail or Voicemail Pro at each
branch to provide auto-attendant operation and announcements for waiting calls.
When selected, access to voicemail is routed via an SM line to the numbers
specified in the MM Number field. The optional MM PSTN Number can be
configured for use when the SM Line is not in service.

Note:
Embedded Voicemail and Voicemail Pro are available only in Distributed
branch deployments. They are not available when there are centralized users
configured for a IP Officee system that is deployed as either a Centralized
branch or a mixed branch.
The Embedded Voicemail option uses the Essential Edition and the Additional
Voicemail Ports licenses to control the number of ports that can be used. These
licenses are also used to control the number of ports on systems where Embedded
Voicemail is configured to provide local Auto Attendant and announcements while
the selected option for voicemail is one of the central voicemail options through the
Session Manager (i.e. Avaya Aura Messaging, Modular Messaging, or CallPilot).
Similarly, the Voicemail Pro option uses the Preferred Edition and the Incremental
Voicemail Ports licenses to control the number of ports that can be used. These
licenses are also used to control the number of ports on systems where Voicemail
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Configuration Mode Field Descriptions

Field Description
Pro is configured to provide local Call Flow processing while the selected option for
voicemail is Avaya Aura Messaging, Modular Messaging or CallPilot.
- When the system routes a call to the voicemail server it indicates the locale for
which matching prompts should be provided if available. The locale sent to the
voicemail server by the system is determined as show below. If the required set of
prompts is not available, the voicemail will fallback to another appropriate
language and finally to English (refer to the appropriate voicemail installation
manual for details).
- Short Code Locale: The short code locale, if set, is used if the call is routed to
voicemail using the short code.
- Incoming Call Route Locale: The incoming call route locale, if set, is used if
caller is external.
- User Locale: The user locale, if set, is used if the caller is internal.
- System Locale: If no user or incoming call route locale is set, the system locale is
used unless overridden by a short code locale.
- Systems using Embedded Voicemail, if the required set of upgraded language
prompts to match the locale is not present on the system SD card, Manager will
display an error. The required prompt set can be uploaded from Manager using
the Add/Display VM Locales option.
• Remote Audix Voicemail: Select this option if using a remote Avaya Intuity Audix
or MultiMessage voicemail system. Requires entry of an Audix Voicemail license in
Licenses. This option is not supported by Server Edition systems.
• Voicemail Pro Select this option when using Voicemail Pro. The IP address of the
PC being used should be set as the Voicemail IP Address. In a Server Edition
network this option is used on the Primary Server. It can also be used on the
Secondary Server if the Secondary server is connected to its own voice mail server
or if the Secondary Server is part of an Outbound Contact Express deployment. Use
of Voicemail Pro requires licenses for the number of simultaneous calls to be
supported. Licenses are not required for an Outbound Contact Express deployment.
Voicemail Mode Default = IP Office Mode. Embedded Voicemail on IP500 V2 systems can use either
IP Office Mode or Intuity Mode key presses for mailbox functions. End users should
be provided with the appropriate mailbox user guide for the mode selected. You can
switch between modes without losing user data, such as passwords, greetings, or
messages.
The following user guides are available from the Avaya support web site:
• IP Office Essential Edition - Embedded Voicemail User Guide (Intuity Mode)
• IP Office Essential Edition - Embedded Voicemail User Guide (IP Office Mode)
• IP Office Voicemail Pro Mailbox User Guide (Intuity Mode)
• IP Office Voicemail Pro Mailbox User Guide (IP Office Mode)
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 System

Field Description
Voicemail Destination Defaults: Non-Server Edition = Blank, Server Edition = IP trunk connection to the
Primary Server.
• When the Voicemail Type is set to Remote Audix Voicemail, Centralized
Voicemail or Distributed Voicemail, this setting is used to enter the outgoing line
group of the line configured for connection to the phone system hosting the central
voicemail server.
• When the Voicemail Type is set to Group Voicemail, this setting is used to specify
the group whose user extensions are connected to the 3rd party voicemail system.
• When the Voicemail Type is set to Analogue Trunk MWI, this setting is used to
specify the phone number of the message center. All analogue trunks configured for
Analogue Trunk MWI must have the same destination.
Voicemail IP Address Defaults: Non-Server Edition = 255.255.255.255, Primary Server = Primary Server IP
Address.
This setting is used when the Voicemail Type is set to Voicemail Pro or Distributed
Voicemail. It is the IP address of the PC running the voicemail server that the system
should use for its voicemail services. If set as 255.255.255.255, the control unit
broadcasts on the LAN for a response from a voicemail server. If set to a specific IP
address, the system connects only to the voicemail server running at that address. If
the system is fitted with an Unified Communication Module hosting Voicemail Pro, the
field should be set to 169.254.0.2.
Backup Voicemail IP Defaults: Primary Server = Secondary Server IP Address, All others = 0.0.0.0 (Off).
Address
This option is supported with Voicemail Pro.
An additional voicemail server can be setup but left unused. If contact to the voicemail
server specified by the Voicemail IP Address is lost, responsibility for voicemail
services is temporarily transferred to this backup server address.
Maximum Record Default = 120 seconds. Range = 30 to 180 seconds. This field is only available when
Time Embedded Voicemail is selected as the Voicemail Type. The value sets the
maximum record time for messages and prompts.
Messages Button Default = On.
Goes to Visual Voice
Visual Voice allows phone users to check their voicemail mailboxes and perform
action such as play, delete and forward messages through menus displayed on their
phone. By default, on phones with a MESSAGES button, the navigation is via spoken
prompts. This option allows that to be replaced by Visual Voice on phones that
support Visual Voice menus. For further details see the button action Visual Voice on
page 1050 .
Enable Outcalling Default = Off (Outcalling not allowed).
This setting is used to enable or disable system support for outcalling on Embedded
Voicemail and Voicemail Pro. When not selected, all outcalling and configuration of
outcalling through mailboxes is disabled. For Voicemail Pro, outcalling can also be
disabled at the individual user mailbox level using the Voicemail Pro client.
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Configuration Mode Field Descriptions

Field Description
DTMF Breakout
Allows system defaults to be set. These are then applied to all user mailboxes unless the users own settings
differ.
The Park & Page feature is supported when the system voicemail type is configured as Embedded Voicemail
or Voicemail Pro. Park & Page is also supported on systems where Avaya Aura Messaging, Modular
Messaging over SIP, or CallPilot (for IP Office Aura Edition with CS 1000 deployments) is configured as the
central voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto attendant
operation. The Park & Page feature allows a call to be parked while a page is made to a hunt group or
extension. This feature can be configured for Breakout DTMF 0, Breakout DTMF 2, or Breakout DTMF 3.
Reception/Breakout The number to which a caller is transferred if they press 0while listening to the
(DTMF 0) mailbox greeting rather than leaving a message (*0 on Embedded Voicemail in IP
Office Mode).
For voicemail systems set to Intuity emulation mode, the mailbox owner can also
access this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave
Mail action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success
results connections depending on whether the caller pressed 0 before or after the
record tone.
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0)
setting.
• When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
- Paging Number: Displays a list of hunt groups and users (extensions). Select a
hunt group or extension to configure this option.
- Retries: The range is 0 to 5. The default setting is 0.
- Retry TimeoutProvided in the format M:SS (minute:seconds). The range can be
set in 15-second increments. The minimum setting is 15 seconds and the
maximum setting is 5 minutes. The default setting is 15 seconds
Breakout (DTMF 2) The number to which a caller is transferred if they press 2while listening to the
mailbox greeting rather than leaving a message (*2 on Embedded Voicemail in IP
Office Mode).
Breakout (DTMF 3) The number to which a caller is transferred if they press 3while listening to the
mailbox greeting rather than leaving a message (*3 on Embedded Voicemail in IP
Office Mode).

Voicemail Code Complexity

Defines the requirements for the voicemail code.
For IP Office systems that have Voicemail Type set to Centralized, the Voicemail Code Complexity settings
must be the same as the IP Office system that is connected to Voicemail Pro.
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 System

Field Description
Enforcement Default = On.
When on, a user PIN is required. The enforcement is not forced during upgrade but
after checking, it can not be cleared.
Minimum Length Default = 6. Maximum 31 digits. Older configurations can continue to have 4 digits
with a maximum of 20 digits.
Complexity Default = On.
When on, the following complexity rules are enforced.
• The user extension number cannot be used.
• A PIN consisting of repeated digits is not allowed (111111).
• A PIN consisting of a sequence, forward or reverse, is not allowed (123456,
564321).
The number of users having invalid Voicemail Code complexity is highlighted below
this field in red colored text.

SIP Settings For Enterprise Branch deployments, these settings are used for calls made or
received on a SIP line where any of the line’s SIP URI fields are set to use internal
data. For Embedded Voicemail and Voicemail Pro, for calls made or received on a SIP
line where any of the line's SIP URI fields are set to Use Internal Data, that data is
taken from these settings. These options are shown if the system has SIP trunks and
is set to use Embedded Voicemail, Voicemail Lite/Pro, Centralized Voicemail or
Distributed Voicemail.
SIP Name Default = Blank on Voicemail tab/Extension number on other tabs.
The value from this field is used when the From field of the SIP URI being used for a
SIP call is set to Use Internal Data.
SIP Display Name Default = Blank on Voicemail tab/Name on other tabs.
(Alias)
The value from this field is used when the Display Name field of the SIP URI being
used for a SIP call is set to Use Internal Data
Contact Default = Blank on Voicemail tab/Extension number on other tabs. The value from this
field is used when the Contact field of the SIP URI being used for a SIP call is set to
Use Internal Data.
Anonymous Default = On on Voicemail tab/Off on other tabs. If the From field in the SIP URI is set
to Use Internal Data, selecting this option inserts Anonymous into that field rather
than the SIP Name set above.

Voicemail Language Prompts

When the system routes a call to the voicemail server it indicates the locale for which matching
prompts should be provided if available. The locale sent to the voicemail server by the system is
determined as show below. If the required set of prompts is not available, the voicemail will
fallback to another appropriate language and finally to English (refer to the appropriate voicemail
installation manual for details).
• Short Code Locale: The short code locale, if set, is used if the call is routed to voicemail
using the short code.

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Configuration Mode Field Descriptions

• Incoming Call Route Locale: The incoming call route locale, if set, is used if caller is
external.
• User Locale: The user locale, if set, is used if the caller is internal.
• System Locale: If no user or incoming call route locale is set, the system locale is used
unless overridden by a short code locale.
Systems using Embedded Voicemail, if the required set of upgraded language prompts to match
the locale is not present on the system SD card, Manager will display an error. The required
prompt set can be uploaded from Manager using the Add/Display VM Locales option.
Related links
System on page 192

Telephony
Used to set the default telephony operation of the system. Some settings shown here can be
overridden for individual users through their User | Telephony tab. The settings are split into a
number of sub-tabs.
Related links
System on page 192
Telephony on page 222
Park and Page on page 229
Tones and Music on page 230
Ring Tones on page 234
SM on page 235
Call Log on page 237
TUI on page 238

Telephony
Navigation: System | Telephony
Additional configuration information
• The Directory Overrides Barring setting allows you to control barred numbers. For additional
configuration information, see Call Barring on page 648.
• The Inhibit Off-Switch Forward/Transfer stops any user from transferring or forwarding
calls externally. For additional information, see Off-Switch Transfer Restrictions on page 719.
• For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 628.
• For additional information on ring tones, see Ring Tones on page 604.
Configuration Settings
Used to configure a wide range of general purpose telephony settings for the whole system.
These settings are mergeable with the exception of Companding LAW and Media Connection
Preservation. Changes to these settings requires a reboot of the system.

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 System

Field Description
Analog Extensions
These settings apply only to analog extension ports provided by the system. For Server Edition this field is only
available on Expansion System (V2) systems
Default Outside Call Default = Normal
Sequence
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming external calls. For details of the ring types see System | Telephony | Ring
Tones.
This setting can be overridden by a user's User | Telephony | Call Settings setting.
Note that changing the pattern may cause fax and modem device extensions to not
recognize and answer calls.
Default Inside Call Default = Ring Type 1
Sequence
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming internal calls. For details of the ring types see System | Telephony | Ring
Tones. This setting can be overridden by a user's User | Telephony | Call Settings
setting.
Default Ring Back Default = Ring Type 2
Sequence
This setting is only used with analog extensions. It sets the ringing pattern used for
ringback calls such as hold return, park return, voicemail ringback, and Ring Back
when Free. For details of the ring types see System | Telephony | Ring Tones.
This setting can be overridden by a user's User | Telephony | Call Settings setting.
Restrict Analog Default = Off.
Extension Ringer
Supported on IP500 V2 systems only. If selected, the ring voltage on analogue
Voltage
extension ports on the system is limited to a maximum of 40V Peak-Peak. Also
when selected, the message waiting indication (MWI) settings for analog extension
are limited to Line Reversal A, Line Reversal B or None. Any analog extension
already set to another MWI setting is forced to Line Reversal A.

Dial Delay Time (secs) Default = 4 (USA/Japan) or 1 (ROW). Range = 1 to 30 seconds.

This setting sets the time the system waits following a dialed digit before it starts
looking for a short code match. In situations where there are potential short codes
matches but not exact match, it also sets the delay following the dialing of a digit
before dialing complete is assumed.
Dial Delay Count Default = 0 digits (USA/Japan) or 4 digits (ROW). Range = 0 to 30 digits.
This setting sets the number of digits dialed after which the system starts looking for
a short code match regardless of the Dial Delay Time.
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Configuration Mode Field Descriptions

Field Description
Default No Answer Default = 15 seconds. Range = 6 to 99999 seconds.
Time (secs)
This setting controls the amount of time before an alerting call is considered as
unanswered. How the call is treated when this time expires depends on the call type.
For calls to a user, the call follows the user's Forward on No Answer settings if
enabled. If no forward is set, the call will go to voicemail if available or else
continues to ring. This timer is also used to control the duration of call forwarding if
the forward destination does not answer. It also controls the duration of ringback call
alerting. This setting is overridden by the User | Telephony | Call Settings | No
Answer Time setting for a particular user if different.
For calls to hunt groups, this setting controls the time before the call is presented to
the next available hunt group member. This setting is overridden by the Hunt Group
| Hunt Group | No Answer Time setting for a particular hunt group if different.
Hold Timeout (secs) Default = Locale specific. Range = 0 (Off) to 99999 seconds.
This setting controls how long calls remain on hold before recalling to the user who
held the call. Note that the recall only occurs if the user has no other connected call.
Recalled calls will continue ringing and do not follow forwards or go to voicemail.
Park Timeout (secs) Default = Locale specific. Range 0 (Off) to 99999 seconds.
This setting controls how long calls remain parked before recalling to the user who
parked the call. Note that the recall only occurs if the user has no other connected
call. Recalled calls will continue ringing and do not follow forwards or go to
voicemail.
Ring Delay Default = 5 seconds. Range = 0 to 98 seconds. This setting is used when any of the
user's programmed appearance buttons is set to Delayed ringing. Calls received on
that button will initially only alert visually. Audible alerting will only occur after the ring
delay has expired. This setting can be overridden by a ring delay set for an
individual user (User | Telephony | Multi-line Options | Ring Delay).
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 System

Field Description
Call Priority Promotion Default = Disabled. Range = Disabled, 10 to 999 seconds.
Time (secs)
When calls are queued for a hunt group, higher priority calls are placed ahead of
lower priority calls, with calls of the same priority sort by time in queue. External calls
are assigned a priority (1-Low, 2-Medium or 3-High) by the Incoming Call Route
that routed the call. Internal calls are assigned a priority of 1-Low. This option can
be used to increase the priority of a call each time it has remained queued for longer
than this value. The calls priority is increased by 1 each time until it reaches 3-High.
In situations where calls are queued, high priority calls are placed before calls of a
lower priority. This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where
Voicemail Pro is being used to provided queue ETA and queue position messages
to callers since those values will no longer be accurate when a higher priority call
is placed into the queue. Note also that Voicemail Pro will not allow a value
already announced to an existing caller to increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt
group's Queue Length Limit, the limit is temporarily raised by 1. This means that
calls already queued are not rerouted by the addition of a higher priority call into
the queue.
Default Currency Default = Locale specific.
This setting is used with ISDN Advice of Charge (AOC) services. Note that changing
the currency clears all call costs stored by the system except those already logged
through SMDR. The currency is displayed in the system SMDR output.
Default Name Priority Default = Favor Trunk.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting determines
which method is used by default. For each SIP line, this setting can be overridden by
the line's own Name Priority setting if required. Select one of the following options:
• Favor Trunk: Display the name provided by the trunk. For example, the trunk may
be configured to provide the calling number or the name of the caller. The system
should display the caller information as it is provided by the trunk. If the trunk does
not provide a name, the system uses the Favor Directory method.
• Favor Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
the name provided by the SIP line. If no match is found, the name provided by the
line, if any, is used.
Media Connection Default = Enabled.
Preservation
When enabled, attempts to maintain established calls despite brief network failures.
Call handling features are no longer available when a call is in a preserved state.
When enabled, Media Connection Preservation applies to SCN links and Avaya H.
323 phones that support connection preservation.
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Configuration Mode Field Descriptions

Field Description
Phone Failback Default = Automatic.
Applies to H.323 phones that support resiliency. The options are:
• Automatic
• Manual
Phones are permitted to failover to the secondary gatekeeper when the IP Office
Line link to the primary gatekeeper is down.
When set to Automatic, if a phone’s primary gatekeeper has been up for more than
10 minutes, the system causes the phone to failback if the phone is not in use. If the
phone is in use, the system will reattempt failback 10 seconds after the phone
ceases to be in use.
When set to Manual, phones remain in failover until manually restarted or re-
registered, after which the phone attempts to fail back.

Note:
Manual failback is not supported on SIP phones.

Login Code Complexity

Defines the requirements for the login code.
Enforcement Default = On.
When on, a user PIN is required.
Minimum Length Default = 6. Maximum 15 digits.
The number of users with login codes less than six digits is displayed below the field
in red colored text.
Complexity Default = On.
When on, the following complexity rules are enforced.
• The user extension number cannot be used.
• A PIN consisting of repeated digits is not allowed (111111).
• A PIN consisting of forward or backward sequence are not allowed. Examples:
123456, 654321.
The number of users with invalid Code Complexity is highlighted below the field in
red colored text.

Send RTCP to an RTCP When the check box is selected, system RTCP reporting is enabled. For IP Office
Collector Release 10.0 and higher, in addition to having the individual phones send RTCP call
quality reports, the system can also send RTCP reports for calls.
Server Address This Sets the address of the third-party QoS monitoring application to which the
system sends RTCP reports.
UDP Port Number The destination port. The default for this filed is 5005.
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 System

Field Description
RTCP reporting interval This setting sets the time interval at which the system sends RTCP reports.
(secs)
Companding Law These settings should not normally be changed from their defaults. They should only
be used where 4400 Series phones (ULAW) are installed on systems which have A-
Law digital trunks.
A-Law or U-Law> PCM (Pulse Code Modulation) is a method for encoding voice as
data. In telephony, two methods of PCM encoding are widely used, A-Law and U-
Law (also called Mu-Law or µ-Law). Typically U-Law is used in North America and a
few other locations while A-Law is used elsewhere. As well as setting the correct
PCM encoding for the region, the A-Law or U-Law setting of a system when it is first
started affects a wide range of regional defaults relating to line settings and other
values.
For IP500 V2 systems, the encoding default is set by the type of Feature Key
installed when the system is first started. The cards are either specifically A-Law or
U-Law.
DSS Status Default = Off
This setting affects Avaya display phones with programmable buttons. It controls
whether pressing a DSS key set to another user who has a call ringing will display
details of the caller. When off, no caller information is displayed.
Auto Hold Default = On (Off for the United States locale).
Used for users with multiple appearance buttons. When on, if a user presses
another appearance button during a call, their current call is placed on hold. When
off, if a users presses another appearance button during a call, their current call is
disconnected.
Show Account Code Default = On This setting controls the display and listing of system account codes.
• When on: When entering account codes through a phone, the account code digits
are shown while being dialed.
• When off: When entering account codes through a phone, the account code digits
are replaced by s characters on the display.
Inhibit Off-Switch Default = On
Forward/Transfer
When enabled, this setting stops any user from transferring or forwarding calls
externally.
Restrict Network Default = Off.
Interconnect
When this option is enabled, each trunk is provided with a Network Type option that
can be configured as either Public or Private. The system will not allow calls on a
public trunk to be connected to a private trunk and vice versa, returning number
unobtainable indication instead.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
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Configuration Mode Field Descriptions

Field Description
Include location Default = Off.
specific information
When set to On, this setting is available in the trunk configuration settings when
Network Type is set to Private.
Set to On if the PBX on the other end of the trunk is toll compliant.
Drop External Only Default = On.
Impromptu Conference
If selected, when the last remaining internal user in a conference exits the
conference, the conference is ended, regardless of whether it contains any external
callers.
If not selected, the conference is automatically ended when the last internal party or
trunk that supports reliable disconnect exits the conference. The Inhibit Off-Switch
Forward/Transfer option above is no longer applied to conference calls.
Visually Differentiate Default = Off.
External Call
This setting is applied to the lamp flashing rate used for bridged appearance and call
coverage appearance buttons on 1400, 1600 and 9600 Series phones and on their
button modules. When selected, external calls alerting on those buttons will use a
slow flash (200ms on/50ms off). If not selected or if the call is internal, normal
flashing (500ms on/500ms off) is used.
Unsupervised Analog Default = Off.
Trunk Disconnect
When using analog trunks, various methods are used for trunk supervision, ie. to
Handling
detect when the far end of the trunk has disconnected and so disconnect the local
end of the call. Depending on the locale, the system uses Disconnect Clear
signalling and or Busy Tone Detection. This setting should only be enabled if it is
know that the analog trunks do not provide disconnect clear signalling or reliable
busy tone. For Server Edition this field is only available on Expansion System (V2)
systems. When enabled:
• Disconnect Clear signalling detection is disabled. Busy tone detection remains on.
• Unsupervised transfers and trunk-to-trunk transfers of analog trunk calls are not
allowed. The Allow Analog Trunk to Trunk Connect setting on analog trunks
(Line | Analog Options) is disabled.
• If Voicemail Pro is being used for external call transfers, Supervised Transfer
actions should be used in call flows rather than Transfer actions.
• All systems in the network must have this setting set to match each other.
High Quality Default = On.
Conferencing
Supports the use of the G.722 codec. IP lines and extensions using G.722 are
provided with wide band audio. If High Quality Conferencing is enabled, when
several wide band audio devices are in the same conference, the system will ensure
that the audio between them remains wide band, even if the conference also
contains other lines and devices using narrow band audio (analog devices, digital
devices and IP devices using codecs other than G.722).
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 System

Field Description
Digital/Analogue Auto Default = On. (IP500 V2 only. Default = Off for Server Edition/On for others)
Create User
When enabled, an associated user is created for each digital/analogue extension
created. Digital/analogue extension creation occurs on initial start up, reset of
configuration, or addition of new digital/analogue expansion units or plug-in
modules.
Directory Overrides Default = On.
Barring
When enabled, barred numbers are not barred if the dialed number is in the External
Directory.
Advertize Callee State Default = Off.
To Internal Callers
When enabled, for internal calls, additional status information is communicated to
the calling party.
Not supported for SIP endpoints except for J100 Series phones (not including the
J129).
• When calling another internal phone and the called phone is set to Do Not Disturb
or on another call, the calling phone displays “Do Not Disturb” or “On Another Call”
rather than “Number Busy”.
• On 9500 Series, 9600 Series and J100 Series, if a line appearance is programmed
on a button on phone A and that line is in use on phone B, then phone A displays
the name of the current user of the line along with the line number.
• If a line appearance on a phone is in use elsewhere in the system and another
extension unsuccessfully attempts to seize that line, the phone displays “In
Use:<name>” where <name> is the name of the user currently using the line.
This configuration parameter sets the system wide default. Individual users can be
configured for this feature using the setting User | Telephony | Call Settings |
Advertize Callee State To Internal Callers
Internal Ring on Default = Off.
Transfer
When enabled, the transfer enquiry calls ring with internal ring tone even if the call
that is being transferred is an external call. If the user transferring the call completes
the call when the call is ringing, the ring tone played to the target changes to the ring
tone appropriate for the call being transferred.
This feature is supported on phone series: 1400, 9500, 1600, 9600, and analog
phones.
This feature is not supported on SIP and H.323 DECT phones.

Related links
Telephony on page 222

Park and Page

Navigation: System | Telephony | Park and Page
The Park and Page tab allows for simple configuration of the of the short code and the
programmable button for the park and page function.

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Configuration Mode Field Descriptions

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Central Park Range Default = Blank. Range = nX to nnnnnnnXX The park slot ID range definition,
where n is a digit sequence from 1 to 9999999 and X represents a park slot
value from 0 to 99. The Central Park Range cannot exceed 9 characters total
length.
Examples:
• 1X defines range 10-19
• 3XX defines range 300-399
• 9876543XX defines range 987654300-987654399
Page Target Group Default = Blank. The list of paging group targets that are presented on supported
List phones if the Page action is requested after the Call Park.
On some phones, only the first three groups can be presented as Page options
(via the Softkeys on the phone). On phones that support scrolling lists, a larger
list of possible Page targets can be presented.

Related links
Telephony on page 222

Tones and Music

Navigation: System | Telephony | Tones and Music
Additional configuration information
For additional information on configuring hold music, see Music On Hold on page 605.
Configuration settings
Used to configure the various tones and music on hold sources used by the system.
The settings are mergable with the exception of Disconnect Tone and Busy Tone Detection.
Changes to these settings requires a reboot of the system.
Deleting any of the hold music Alternate Sources requires a reboot.
Field Description
Conferencing Tone Default = Entry & Exit Tones.
This settings controls how conference tones are used. The options are:
• Entry & Exit Tones: Asingle tone is heard when a new party joins a conference and
double-tone is heard when a party leaves the conference.
• Repeating ToneA conference tone is heard every 10 seconds by all conference
parties.
Note that no conference tones are played in a conference initiated by an Outbound
Contact Express agent.
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 System

Field Description
Disconnect Tone Default = Default (Use locale setting).
For digital and IP phones, when the system detects that the far end of a call has
disconnected, it can make the near end either go idle or play disconnect tone. By
default, the chosen behavior depends on the system locale. This field can be used to
override the locale's default action and force either disconnect tone or go idle. The
options are:
• Default: Use the system locale specific action for disconnected calls.
• On: Play disconnect tone when far end disconnection is detected.
• Off: Go idle when far end disconnection is detected.
Busy Tone Detection Default = Off. Enables or disables the use of busy tone detection for call clearing. This
is a system wide setting.
CLI Type This field is used to set the CLI detection used for incoming analogue trunks. Note that
the CLI Type field is shown for locales other than Customize. For the Customize
locale, it is set through the System | System form. The options are:
• DTMF
• FSK V23
• FSK BELL202
Local Dial Tone Default = On
For all normal operation this setting should be left enabled as it allows the system to
provide dial tone to users (essential for MSN working).
Local Busy Tone Default = Off
This setting should only be used when the local exchange gives a busy signal via Q.
931 but does not provide busy tone.
Beep on Listen Default = On
This setting controls whether call parties hear a repeating tone when their call is
monitored by another party using the Call Listen feature.

Warning:
The use of features to listen to a call without the other call parties being aware of
that monitoring may be subject to local laws and regulations. Before enabling the
feature you must ensure that you have complied with all applicable local laws
and regulations. Failure to do so may result in severe penalties.
GSM Silence Default = Off.
Suppression
This setting should only be selected if voice quality problems are experienced with
calls to voicemail or while recording calls. When on, the system signals silence by
generating silence data packets in periods when the voicemail system is not playing
prompts. Note that use of this option may cause some timeout routing options in
voicemail to no longer work.
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Configuration Mode Field Descriptions

Field Description
Analogue Trunk VAD Default = Off.
Select this option to enable Voice Activity Detection (VAD) for analog trunks
terminating on the ATM4U-V2 card. VAD functionality provides a Call Answer signal
triggered by voice activity. This signal can be used for:
• Mobile Twinning
• SMDR
• Call Forwarding
• Call Display
• Mobile Call Control
• Transfer Ringing Call
• TAPI
• Trunk to Trunk Call
Busy Tone Detection Default = System Frequency (Tone defined by system locale) Allows configuration of
the system's busy tone detection settings on lines that do not provide reliable
disconnect signalling. In that case, the system will use tone disconnect clearing to
disconnect such lines after 6 seconds of continuous tone. The default tone (frequency
and on/off cadence) detection used is defined by the system locale. The settings
should not be adjusted unless advised by Avaya Technical Support. Changes to this
setting require a reboot rather than a merge when the new configuration is sent to the
system. .For Server Edition this field is only available on Expansion System (V2)
systems.
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 System

Field Description
Hold Music
This section is used to define the source for the system's music on hold source. You must ensure that any MOH
source you use complies with copyright, performing rights and other local and national legal requirements.
Server Edition deployments support centralized music on hold, where the Primary Server streams music to the
Secondary Server and all expansion servers.
The WAV file properties must be:
• PCM
• 8kHz 16-bit
• mono
• maximum length 90 seconds (30 seconds on non-IP500 V2 systems, 600 seconds on Linux based systems)
If the file downloaded is in the incorrect format, it will be discarded from memory after the download.

Caution:
Copying files in the incorrect format directly into the opt/ipoffice/system/primary directory can
disable the music on hold function.
The first WAV file, for the system source, must be called HoldMusic.wav. Alternate source WAV files:
• can be up to 27 IA5 characters
• cannot contain spaces
• any extension is allowed
• case sensitive
System Source Default = WAV File.
Selects the default hold music source for most uses of music on hold. Note that
changes to the System Source requires a reboot. The options are:
• WAV: Use the WAV file HoldMusic.wav. This file is loaded via TFTP.
Note that on Linux systems, the file name is case sensitive.
• WAV (restart): Identical to WAV except that for each new listener, the file plays from
the beginning.
Not supported on IP500 V2 systems. Cannot be used as a centralized source.
• External: Applicable to IP500 V2 systems. Use the audio source connected to the
back of the control unit.
• Tone: The use of a double beep tone (425Hz, 02./0.2/0.2/3.4 seconds on/off) can be
selected as the system source. The hold music tone is automatically used if the
system source is set to WAV File but the HoldMusic.wav file has not yet been
successfully downloaded.
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Configuration Mode Field Descriptions

Field Description
Alternate Sources This is just a summary, for more details see Alternate Source on page 607. The
available options depends on the system type. On IP500 V2 systems, up to 3
additional sources can be specified. On Linux systems, up to 31 alternate sources can
be specified. Note that adding and changing a source can be done using a merge but
deleting a source requires a reboot.
• Number: Assigned automatically by the system.
• Name: Up to 31 characters This field is used to associate a name with the alternate
source. That name is then used in the Hold Music Source field on Incoming Call
Routes and Hunt Groups.
• Source: Up to 31 characters. Defines the source for the music on hold.
The options are listed below with a brief description. For more information, see
Alternate Source on page 607.
- WAV: To specify a wav file, enter WAV: followed by the file name.
- XTN: Any analog extension. Not applicable to Linux systems.
- WAVRST: To specify a wav file, enter WAVRST: followed by the file name.
Playback is started every time from the beginning. Not applicable to IP500 V2
systems.
- WAVDIR: Multiple WAV file source. The WAV files must be stored in the
directory /disk/tones/mohwavdir (file manager access) or /opt/ipoffice/
tones/mohwavdir/ (SSH access). Playback resumes from where it left off the
last time. Not applicable to IP500 V2 systems.
- WAVDIRRST: As per WAVDIR above, however playback is always started from
the beginning. Not applicable to IP500 V2 systems.
- USB: Supports multiple USB inputs. Enter USB:<number>. Not applicable to
IP500 V2 systems.
- LINE: In Server Edition networks, setting the Secondary Server and Expansion
Server Alternate Source to Line allows the server to receive streamed audio
from a source on the Primary Server. On the Secondary Server and Expansion
Server, enter Line:x,ywhere x is the line number to the Primary Server and y is
the MOH source number on the Primary Server.

Related links
Telephony on page 222

Ring Tones
Navigation: System | Telephony | Ring Tones
Additional configuration information
For additional ring tone configuration information, see Ring Tones. on page 604

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 System

Configuration settings
Used to configure distinct ring tones for groups and incoming call routes. Ring tone override
features are only supported on 1400 Series, 9500 Series and J100 Series (except J129) phones.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Available Ring In this table, the Number, Name, and Source values are system supplied. The
Tones Name value is used to create a ring tone plan.
Ring Tone Plan Use this table to specify available ring tones. Ring tones in this table can be
applied to hunt groups and incoming call routes and by short codes.
• Number: System supplied.
The Number can be used in a short code by adding r(x) to the Telephone
Number field, where x = 1 to 8 and specifies which ring tone plan to use.
• Name: A descriptive name for where this ring tone is used. For example, the
name of a hunt group. Each name in the table must be unique. Once configured
in this table, ring tone names can be selected from the Ring Tone Override field
at:
- Group | Group
- Incoming Call Route | Standard
• Ring Tone: The list of ring tone names from the Available Ring Tones table.

Related links
Telephony on page 222

SM
Navigation: System | Telephony | SM
Used to configure settings that apply to both SM lines.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
Short Form Dialing Default = 0. Range = 0 to 14.
Length
This number specifies the short-form dialing length for all Centralized users and
Groups. Configuration of this field allows IP Office to treat the last N digits (where
N is the number entered in this field) of each Centralized user’s extension number
as an alias to that user’s extension number. For example, if a Centralized user’s
extension number is 5381111 and the Short Form Dialing Length is 4, the system
will match calls to 1111 with this extension. When 1111 is dialed by another user
on the system, entered from the autoattendant, or comes from the ICR, then in
sunny-day that call will be sent to Session Manager with the number converted to
5381111 and in rainy-day it will target the extension 5381111 locally.
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Configuration Mode Field Descriptions

Field Description
Branch Prefix Default = Blank. Maximum range = 15 digits.
This number is used to identify the IP Office system within the Avaya Aura®
network. The branch prefix of each IP Office system must be unique and must not
overlap. For example 85, 861 and 862 are okay, but 86 and 861 overlap. On calls
routed via an SM Line, the branch prefix is added to the caller's extension number.
You have the option to leave the Branch Prefix field blank. If you do not configure
the branch prefix, the IP Officeuser extensions must be defined with the full
enterprise number.
Local Number Default = Blank (Off). Range = Blank or 3 to 9 in deployments with IP Office users
Length and blank or 3 to 15 in deployments with only centralized users.
This field sets the default length for extension numbers for extensions, users, and
hunt groups added to the IP Office configuration. Entry of an extension number of
a different length will cause an error warning by Manager.
The number of digits entered in the Branch Prefix field plus the value entered in
the Local Number Length field must not exceed 15 digits. You have the option to
leave the Local Number Length field blank.
Proactive Default = 60 seconds. Range = 60 seconds to 100000 seconds.
Monitoring
The Enterprise Branch system sends regular SIP OPTIONS messages to the SM
line in order to check the status of line. This setting controls the frequency of the
messages when the SM line is currently in service. Centralized SIP phones use
their own settings.
Monitoring Retries Default = 1. Range = 0 to 5.
The number of times the Enterprise Branch system retrys sending an OPTIONS
request to Session Manager before the SM Line is marked out-of-service.
Reactive Monitoring Default 60 seconds. Range = 10 to 3600 seconds.
The Enterprise Branch system sends regular SIP OPTIONS messages to the SM
line in order to check the status of line. This setting controls the frequency of the
messages when the SM line is currently out of service. Centralized SIP phones
use their own settings.
Failback Policy Default = Auto.
This field allows the administrator to choose between an automatic or manual
failback policy on the IP Office. In deployments with Centralized phones, this field
must be set consistently with the Failback Policy of the phones, which is
configured via the Session Manager global settings in System Manager. The
options are:
• Auto: IP Office automatically brings the SM Line to ‘In Service’ status as soon
as it detects via the Reactive Monitoring that the Session Manager is reachable
• Manual: When an SM line is in "Out of Service" state, IP Office does not bring it
back to "In Service" status based on automatic detection. IP Office keeps the
SM Line in "Out of Service" state until the administrator manually initiates
Failback of IP Office from Session Manager.

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 System

Related links
Telephony on page 222

Call Log
Navigation: System | Telephony | Call Log
The system can store a centralized call log for users. Each users' centralized call log can contain
up to 30 call records for user calls. When this limit is reached, each new call records replaces the
oldest previous record.
On Avaya phones with a fixed Call Log or History button (1400, 1600, 9500 and 9600 Series),
that button can be used to display the user's centralized call log. The centralized call log is also
used for M-Series and T-Series phone. The user can use the call log to make calls or to store as a
personal speed dial. They can also edit the call log to remove records. The same call log is also
used if the user logs into one-X Portal for IP Office.
The centralized call log moves with the user if they log on and off from different phones. This
includes if they hot desk within a network.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Default Centralized Default = On.
Call Log On
When selected, each user is defaulted to have the system store a call log of their
calls. This call log is accessible on the phone when the user is using a phone with
a Call Log or History button. The use of centralized call logging can be enabled/
disabled on a per user basis using the setting User | Telephony | Call Log |
Centralized Call Log.
Log Missed Calls Default = Off.
Answered at
This setting controls how calls to a user, that are answered by a covering user
Coverage
should be logged in the centralized call log. This option applies for calls answered
elsewhere (covered) by pickup, call coverage (call coverage buttons or coverage
group), bridged appearance button, user BLF, voicemail, etc.
Setting Targeted User Covering User
Off Nothing Answered Call
On Missed Call Answered Call
Log Missed Hunt Default = Off. By default, hunt group calls are not included in any user's
Group Calls centralized call log unless answered by the user. If this option is selected, a
separate call log is kept for each hunt group of calls that are not answered by
anyone. It includes hunt group calls that go to voicemail.
If missed hunt group calls are also being logged, the system stores up to 10 call
records for each hunt group. When this limit is reached, new call records replace
the oldest record.
Within the user call log settings (User | Telephony | Call Log), the list of hunt
groups allows selection of which hunt groups' missed call records should be
displayed as part of the user's centralized call log.

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Configuration Mode Field Descriptions

Related links
Telephony on page 222

TUI
Navigation: System | Telephony | TUI
Used to configure system wide telephony user interface (TUI) options for 1400, 1600, 9500 and
9600 Series phones.
Default phone display options:
Use these settings to define the default phone display when feature menus are disabled. Note that
for new users, the default phone display options are set to the system default values.
Feature menus can be disabled in one of two ways.
• Set System | Telephony | TUI | Features Menu to Off. Set User | Telephony | TUI | User
Setting to Same as System.
• On User | Telephony | TUI, set User Setting to Custom and set Features Menu to Off.
Configuration settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Phone Type Variable Description
1400 Display Name Preference Defines the default value of the User’s
Features > Phone User > Phone
1600
Screen Settings > Display Name
setting.
Default = Off
When enabled, displays the user name.
9500 Column View Preference Defines the default value of the User’s
Features > Phone User > Phone
9608
Screen Settings > Display Mode setting.
9611
Default = Dual
Column view can be Single or Dual.
9621 Quick Touch Panel Lines Defines the default value of the User’s
Features > Phone User > Phone
9641
Screen Settings > Quick Touch Lines
setting.
Default = Optimize
Sets the Quick Touch Panel number.
The options are 1, 2, and Optimize.
When set to Optimize:
• 9621 = 1
• 9641 = 2

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 System

Field Description
Time Format Default = Locale Defined.
Set the system time format display. The default time format is defined by the
Locale setting. You can override the default and set the time format to a 12- hour
or 24-hour clock.
Features Menu Controls
Features Menu Default = On
When set to on, you can select to turn individual menus and features on users
phone’s on or off. The system level settings can be overridden at the individual
user settings level if required for particular users. The following feature menus are
listed:
• Basic Call Functions: If selected, users can access menu options for call
pickup, park, unpark and transfer to mobile functions.
• Advanced Call Functions: If selected, users can access the menu options for
do not disturb, account code, withhold number and internal auto-answer
functions. Note, the Account Code menu is only shown if the system has been
configured with accounts codes.
• Forwarding: If selected, users the phone's menus for forwarding and follow me
functions.
• Hot Desk Functions: If selected, users can access the menu options for
logging in and out.
• Passcode Change: If selected, users can change their login code (security
credentials) through the phone menus..
• Phone Lock: If selected, users can access the menu options for locking the
phone and for setting it to automatically lock.
• Self Administration: If selected, users can access the phone’s Self-
Administration menu options.
• Voicemail Controls: If set, users can access the Visual Voice option through
the phone's Features menu.
SIP Phone Options
Application for Default = Equinox on Vantage
Vantage
Select the application to be used on Avaya Vantage™. The system supports Avaya
Vantage™ phones running either Avaya Vantage™ Basic or Avaya Equinox™
applications as the dialer application. This field sets which application is indicated
in the auto-generated K1xxSupgrade.txt file the system provides to Avaya
Vantage™ phones. If a mix of dialer applications is required, a static
K1xxSupgrade.txt file needs to be used. The options on the interface are:
• Equinox on Vantage: Select the option to use the Avaya Equinox™ client on
Avaya Vantage™ device.
• Vantage Basic/Connect: Select the option to use the Avaya Vantage™ Basic or
Avaya Vantage™ Basic applications on Avaya Vantage™ device.

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Configuration Mode Field Descriptions

Related links
Telephony on page 222

Directory Services
Navigation: System | Directory Services
Related links
System on page 192
LDAP on page 240
HTTP on page 243

LDAP
Navigation: System | Directory Services | LDAP
Additional configuration information
For additional configuration information, see Centralized System Directory on page 589.
Configuration settings
The system supports LDAP Version 2. LDAP (Lightweight Directory Access Protocol) is a software
protocol for enabling anyone to locate organizations, individuals, and other resources such as files
and devices in a network, whether on the Internet or on a corporate intranet. LDAP is a
"lightweight" (smaller amount of code) version of DAP (Directory Access Protocol), which is part of
X.500, a standard for directory services in a network. LDAP is lighter because in its initial version,
it did not include security features.
The system supports the import of directory records from one system to another using HTTP. That
includes using HTTP to import records that another system has learnt using LDAP. HTTP import,
which is simpler to configure, can be used to relay LDAP records with LDAP configured on just
one system.
LDAP records can contain several telephone numbers. Each will be treated as a separate
directory record when imported into the system directory.
In a network, a directory tells you where in the network something is located. On TCP/IP networks,
including the Internet, the Domain Name System (DNS) is the directory system used to relate the
domain name to a specific network address. However, you may not know the domain name. LDAP
allows you to search for an individual without knowing where they're located (although additional
information will help with the search).
An LDAP directory is organized in a simple "tree" hierarchy consisting of the following levels:
• The "root" directory (the starting place or the source of the tree), which branches out to
• Countries, each of which branches out to
• Organizations, which branch out to
• Organizational units (divisions, departments, and so forth), which branches out to (includes
an entry for)
• Individuals (which includes people, files, and shared resources such as printers)

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 System

An LDAP directory can be distributed among many servers. Each server can have a replicated
version of the total directory that is synchronized periodically. An LDAP server is called a Directory
System Agent (DSA). An LDAP server that receives a request from a user takes responsibility for
the request, passing it to other DSA's as necessary, but ensuring a single coordinated response
for the user.
LDAP Directory Synchronization allows the telephone number Directory held in the Control Unit to
be synchronized with the information on an LDAP server. The feature can be configured to
interoperate with any server that supports LDAP Version 2.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
LDAP Enabled Default = Off
This option turns LDAP support on or off. The system uses LDAP Version 2. If the
server being queried is an LDAP Version 3 server, support for LDAP Version 2
requests may need to be enabled on that server (all LDAP Version 3 servers
support LDAP Version 2 but do not necessarily have it enabled by default).
User Name Default = Blank
Enter the user name to authenticate connection with the LDAP database. To
determine the domain-name of a particular Windows 2000 user look on the
"Account" tab of the user's properties under "Active Directory Users and
Computers". Note that this means that the user name required is not necessarily
the same as the name of the Active Directory record. There should be a built-in
account in Active Directory for anonymous Internet access, with prefix "IUSR_"
and suffix server_name (whatever was chosen at the Windows 2000 installation).
Thus, for example, the user name entered is this field might be:
[email protected]
Password Default = Blank
Enter the password to be used to authenticate connection with the LDAP
database. Enter the password that has been configured under Active Directory for
the above user. Alternatively an Active Directory object may be made available for
anonymous read access. This is configured on the server as follows.
In "Active Directory Users and Computers" enable "Advanced Features" under the
"View" menu. Open the properties of the object to be published and select the
"Security" tab. Click "Add" and select "ANONYMOUS LOGON", click "Add", click
"OK", click "Advanced" and select "ANONYMOUS LOGON", click "View/Edit",
change "Apply onto" to "This object and all child objects", click "OK", "OK", "OK".
Once this has been done on the server, any record can be made in the User
Name field in the System configuration form (however this field cannot be left
blank) and the Password field left blank. Other non-Active Directory LDAP servers
may allow totally anonymous access, in which case neither User Name nor
Password need be configured.
Server IP Address Default = Blank
Enter the IP address of the server storing the database.
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Configuration Mode Field Descriptions

Field Description
Server Port Default = 389
This setting is used to indicate the listening port on the LDAP server.
Authentication Default = Simple
Method
Select the authentication method to be used. The options are:
• Simple: clear text authentication
• Kerberos: Not used.
Resync Interval Default = 3600 seconds. Range = 60 to 99999 seconds.
(secs)
The frequency at which the system should resynchronize the directory with the
server. This value also affects some aspects of the internal operation.
The LDAP search inquiry contains a field specifying a time limit for the search
operation and this is set to 1/16th of the resync interval. So by default a server
should terminate a search request if it has not completed within 225 seconds
(3600/16).
The client end will terminate the LDAP operation if the TCP connection has been
up for more than 1/8th of the resync interval (default 450 seconds). This time is
also the interval at which a change in state of the "LDAP Enabled" configuration
item is checked.
Search Base/Search Default = Blank These 2 fields are used together to refine the extraction of
Filter directory records. Basically the Base specifies the point in the tree to start
searching and the Filter specifies which objects under the base are of interest.
The search base is a distinguished name in string form (as defined in RFC1779).
The Filter deals with the attributes of the objects found under the Base and has its
format defined in RFC2254 (except that extensible matching is not supported). If
the Search Filter field is left blank the filter defaults to "(objectClass=*)", this will
match all objects under the Search Base. The following are some examples
applicable to an Active Directory database.
• To get all the user phone numbers in a domain:
Search Base: cn=users,dc=acme,dc=com
Search Filter: (telephonenumber=*)
• To restrict the search to a particular Organizational Unit (eg office) and get cell
phone numbers also:
Search Base: ou=holmdel,DC=example,DC=com
Search Filter: (|(telephonenumber=*)(mobile=*))
• To get the members of distribution list "group1":
Search Base: cn=users,dc=example,dc=com
Search Filter:
(&(memberof=cn=group1,cn=users,dc=example,dc=com)
(telephonenumber=*))
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 System

Field Description
Number Attributes : Default = see below
Enter the number attributes the server should return for each record that matches
the Search Base and Search Filter. Other records could be ipPhone,
otherIpPhone, facsimileTelephoneNumber, otherfacsimileTelephone Number,
pager or otherPager. The attribute names are not case sensitive. Other LDAP
servers may use different attributes.
By default the record is
"telephoneNumber,otherTelephone,homePhone=H,otherHomePhone=H,mobile=M
,otherMobile=M", as used by Windows 2000 Server Active Directory for Contacts.
The optional "=string" sub-fields define how that type of number is tagged in the
directory. Thus, for example, a cell phone number would appear in the directory
as: John Birbeck M 7325551234

Related links
Directory Services on page 240

HTTP
Navigation: System | Directory Services | HTTP
Additional configuration information
For additional configuration information, see Centralized System Directory on page 589.
Configuration settings
The system can use HTTP to import the directory records held by another system. Note that
support for HTTP can be disabled. The setting System | System | Avaya HTTP Clients Only can
restrict a system from responding to HTTP requests. The system's Unsecured Interface security
settings also included controls for HTTP access (HTTP Directory Read and HTTP Directory
Write).
For Server Edition, on Secondary Server, Expansion System (L) and Expansion System (V2)
systems, the HTTP settings are automatically defaulted to obtain the system directory from the
Primary Server.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Directory Type Default = None (No HTTP import)/IP Office SCN on Server Edition.
Set whether HTTP import should be used and the method of importation. The options
are:
• None: Do not use HTTP import.
• IP Office: Import from the system at the IP address set in the Source field.
• IP Office SCN: Import from a system in a multi-site network. The Source field is used
to select the Outgoing Line ID that matches the H.323 line to the remote system.
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Configuration Mode Field Descriptions

Field Description
Source Default = Blank/9999 on Server Edition.
The form of this field changes according to the Directory Type selection above. For IP
Office this field requires the IP address of the other system. For IP Office SCN, the
outgoing group ID of the IP Office line to the remote system is used.
List Default = All.
This field sets what types of directory record should be imported. The options are:
• All: Import the full set of directory records from the remote system.
• Config Only: Import just directory records that are part of the remote system's
configuration. Note that these will be treated as imported records and will not be added
to the local systems own configuration records.
• LDAP Only: Import just directory records that the remote system has obtained as the
result of its own LDAP import. This allows LDAP directory records to be relayed from
one system to another.
• HTTP Only: Import just directory records that the remote system has obtained as the
result of its own HTTP import. This allows HTTP directory records to be relayed from
one system to another.
URI Default = /system/dir/complete_dir_list?sdial=true
This field is for information only and cannot be adjusted. The path shown changes to
match the List setting above.
Resync Interval Default = 3600 seconds.
(secs)
Set how often the system should request an updated import. When a new import is
received, all previously imported records are discarded and the newly imported records
are processed.
HTTPS Enabled Default = On.
Turns HTTPS support on or off for directory record import.
Port Number Default = 443.
The port used for the Directory import.
When HTTPS Enabled is set to On, the default value is 443. When HTTPS Enabled is
set to Off, the default value is 80.

Related links
Directory Services on page 240

System Events
Navigation: System | System Events
The system supports a number of methods by which events occurring on the system can be
reported. These are in addition to the real-time and historical reports available through the System
Status Application (SSA).

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 System

Related links
System on page 192
Configuration on page 245
Alarms on page 246

Configuration
Navigation: System | System Events | Configuration
This form is used for general configuration related to system alarms.
Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
SNMP Agent Configuration
SNMP Enabled Default = Off.
Enables support for SNMP. This option is not required if using SMTP or Syslog.
Community Default = Blank.
(Read-only)
The SNMP community name to which the system belongs.
SNMP Port Default = 161. Range = 161, or 1024 to 65535. The port on which the system listens for
SNMP polling.
Device ID This is a text field used to add additional information to alarms. If an SSL VPN is
configured, Avaya recommends that the Device ID match an SSL VPN service Account
Name. Each SSL VPN service account name has an associated SSL VPN tunnel IP
address. Having the displayed Device ID match an SSL VPN service account name helps
identify a particular SSL VPN tunnel IP address to use for remotely managing IP Office.
Contact This is a text field used to add additional information to alarms.
Location This is a text field used to add additional information to alarms.

QoS Parameters
These parameters are used if the setting System | LAN1 | VoIP | Enable RTCP Monitor on Port 5005 is set to
On. They are used as alarm thresholds for the QoS data collected by the system for calls made by Avaya H.323
phones and for phones using VCM channels. If a monitored call exceeds any of the threshold an alarm is sent
to the System Status application. Quality of Service alarms can also be sent from the system using Alarms.
• The alarm occurs at the end of a call. If a call is held or parked and then retrieved, an alarm can occur for
each segment of the call that exceeded a threshold.
• Where a call is between two extensions on the system, it is possible that both extensions will generate an
alarm for the call.
• An alarm will not be triggered for the QoS parameters recorded during the first 5 seconds of a call.
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Configuration Mode Field Descriptions

Field Description
Round Trip Delay Default = 350.
(msec)
Less than 160ms is high quality. Less than 350ms is good quality. Any higher delay will be
noticeable by those involved in the call. Note that, depending on the compression codec
being used, some delay stems from the signal processing and cannot be removed: G.711
= 40ms, G.723a = 160ms, G.729 = 80ms.
Jitter (msec) Default =20.
Jitter is a measure of the variance in the time for different voice packets in the same call to
reach the destination. Excessive jitter will become audible as echo.
Packet Loss (%) Default = 3.0.
Excessive packet loss will be audible as clipped words and may also cause call setup
delays.
Good Quality High Quality
Round Trip Delay < 350ms < 160ms
Jitter < 20ms < 20ms
Packet Loss < 3% < 1%

Related links
System Events on page 244

Alarms
Navigation: System | System Events | Alarms
These settings are not mergeable. Changes to these settings require a reboot of the system.
This form is used to configure what can cause alarms to be sent using the different alarm
methods.
• Up to 5 alarm traps can be configured for use with the SNMP settings on the System |
System Events | Configuration tab.
• Up to 3 email alarms can be configured for sending using the systems System | SMTP
settings. The email destination is set as part of the alarm configuration below.
• Up to 2 alarms can be configured for sending to a Syslog destination that is included in the
alarm settings.
Configuration Settings
Field Description
New Alarm This area is used to show and edit the alarm.
Destination
To use SNMP or Email the appropriate settings must be configured on the Configuration sub-tab. Note that the
Destination type is grayed out if the maximum number of configurable alarms destinations of that type has been
reached. Up to 5 alarm destinations can be configured for SNMP, 3 for SMTP email, and 2 for Syslog
Table continues…

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 System

Field Description
Trap If selected, the details required in addition to the selected Events are:
• Server Address: Default = Blank. The IP address or fully qualified domain name
(FQDN) of the SNMP server to which trap information is sent.
• Port: Default = 162. Range = 0 to 65535. The SNMP transmit port.
• Community: Default = Blank The SNMP community for the transmitted traps. Must be
matched by the receiving SNMP server.
• Format: Default = IP Office. The options are:
- IP Office SNMP event alarms format in accordance with IP Office.
- SMGR SNMP event alarms format in accordance with SMGR.
Syslog If selected, the details required in addition to the selected Events are:
• IP Address: Default = Blank. The IP address of the Syslog server to which trap
information is sent.
• Port: Default = 514. Range = 0 to 65535. The Syslog destination port.
• Protocol: Default = UDP. Select UDP or TCP.
• Format: Default = Enterprise. The options are:
- Enterprise Syslog event alarms format in accordance with Enterprise.
- IP Office Syslog event alarms format in accordance with IP Office.
Email If selected, the details required in addition to the selected Events are:
Email: The destination email address.

Minimum Security Default = Warnings.

Level
The options are:
• Warnings: All events, from Warnings to Critical, are sent.
• Minor: Minor, major, and critical events are sent. Warnings are not sent.
• Major: Major and critical events are sent. Warnings and minor events will not be sent.
• Critical: Only critical events are sent.
Events Default = None
Sets which types of system events should be collected and sent. The table below lists the
alarms associated with each type of event. Text in italics in the messages is replaced with
the appropriate data. Items in [] brackets are included in the message if appropriate. The
subject line of SMTP email alarms takes the form "System name: IP address - System
Alarm".

Type Events Event State Message

Entity Application Voicemail operation The Voicemail server is now
operational.
Voicemail Failure The Voicemail server is down.
Table continues…

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Configuration Mode Field Descriptions

Type Events Event State Message

Voicemail Event - storage The Voicemail server storage is
OK OK.
Voicemail Event - storage The Voicemail server storage is
nearly full nearly full.
Voicemail Event - storage storage full The Voicemail server
full storage is full.
Service Feature license missing Attempt to use a feature for which
no license is installed. License
Type: <name>
All licenses in use The following licenses are all in
use. License Type: <name>
Clock source changed 8kHz clock source changed.
Details will be provided.
Logon failed Logon failure reason will be
provided.
No free channels available No free channels were available.
Outgoing group ID: <number>
Hold music file failure Failed to load Hold Music source
file.
All resources in use The following system resources
are all in use: <resource type> will
be provided.
OEM card slot error System running secondary
software or error description with
OEM card will be provided.
Network interconnect Details of the network
failure interconnection failure will be
provided.
SIP message too large SIP message Rx error - too large -
ignored.
Contact Flash Change The PC card in name has
Card changed.
Expansion Operational Expansion module name link is up.
Module Failure Expansion module name link is
down.
Error Expansion module name link has a
link error.
Change Expansion module name link has
changed.
Trunk Operational Trunk number (name) [on
expansion module number] is now
operational.
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 System

Type Events Event State Message

Failure Trunk number (name) [on
expansion module number] is
down.
Trunk Trunk seize failure Seize failure: Channel [number] or
Port [number].
Incoming call outgoing Incoming call outgoing trunk:
trunk failure Channel [number] or Port
[number].
CLI not delivered CLI not delivered: Channel
[number] or Port [number].
DDI incomplete DDI incomplete. Expected Number
of digits: .
LOS LOS
OOS OOS
Red Alarm Red Alarm
Blue Alarm Blue Alarm
Yellow Alarm Yellow Alarm
IP connection failure IP connection failure. IP Trunk Line
Number: <number> or Remote end
IP address: <IP address>
Small Community Network Small Community Network invalid
invalid connection connection. IP trunk line number:
<number> or remote end IP
address: <IP address>
Link Device changed Device changed. Home Extension
Number: .
LDAP server LDAP server communication failure
communication failure
Resource down Link/resource down. Module type,
number and name will be provided.
SMTP server SMTP server communication
communication failure failure
Voicemail Pro connection Voicemail Pro connection failure
failure
Dialer connection failure The Dialer connection has been
lost.
VCM Operational VCM module name is now
operational.
Failure VCM module name has failed.
Memory Card Invalid Card
Table continues…

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Configuration Mode Field Descriptions

Type Events Event State Message

Free Capacity
Generic Generic Non-primary location boot System running backup software.
alarm
Invalid SD Card Incompatible or Invalid (System or
Optional) SD Card fitted.
Network link failure Network Interface name (ip
address) has been disconnected.
Network link operational Network Interface name (ip
address) has been connected.
System warm start System has been restarted (warm
start).
System cold start System has restarted from power
fail (cold start).
SNMP Invalid community Invalid community specified in
SNMP request.
License License Server Server Operational The license server is now
operational.
Server failure The license server is no longer
operational.
License Key License Key Failure
Failure
Loopback Loopback Near end line loopback Trunk number (name) [on
expansion module number] is in
near end loopback.
Near end payload loopback Trunk number (name) [on
expansion module number] is in
near end loopback with payload.
Loopback off Trunk number (name) [on
expansion module number] has no
loopback.
Phone Change Phone Change Phone has been unplugged The phone with id n has been
removed from extension extension
(unit, port number).
Phone has been plugged in The phone with type type (id
number) has been plugged in for
extension extension (unit, port
number).
Quality of Service QoS Monitoring If Enable RTCP Monitor on Port 5005 is selected, any monitored
calls that exceeds the set QoS Parameters causes an alarm.
Syslog Basic Audit Events as written to the system Audit Trail. Available on Syslog
output only.
Table continues…

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 System

Type Events Event State Message

Extended Audit Configuration change information. Each message contains one
configuration or security settings object attribute change, and
optionally the previous and new values.
System Monitor If selected, System monitor traces are packed into Syslog traces.
System Configuration CCR group agent not CCR Group agent not targeted as
targeted it is not an CCR Agent. Group :
<name> Agents: <name1, .....,
name n>.
Small Community Network Small Community Network dial
dial plan conflict plan conflict
No incoming call route for The following line had no Incoming
call Call Route for a call. Line:
<number> or Line Group ID:
<number>.
Installed hardware failure Installed hardware failure details
will be provided.
System
Shutdown
Running Backup
Emergency Calls Emergency call successful Successful Emergency Call |
Emergency call! Location:location
Dialled:dialled number
Called:number sent on the line
CallerID: ID Usr:user
Extn:extension
Emergency call failed Failed Emergency Call |
Emergency call! Location:location
Dialled:dialled
numberFailCause:cause Usr:user
Extn: extension

Alarm Types
Note the following.
• Voicemail Pro Storage Alarms: The alarm threshold is adjustable through the Voicemail Pro
client.
• Embedded Voicemail Storage Alarms: A disk full alarm is generated when the Embedded
Voicemail memory card reaches 90% full. In addition a critical space alarm is generated at
99% full and an OK alarm is generated when the disk space returns to below 90% full.
• Loopback: This type of alarm is only available for systems with a United States locale.
The list of IP Office alarms is available on the Admin CD in the folder \snmp_mibs\IPOffice.
Related links
System Events on page 244

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Configuration Mode Field Descriptions

SMTP
Navigation: System | SMTP
These settings are not mergeable. Changes to these settings require a reboot of the system.
Configuration Settings
SMTP can be used as the method of sending system alarms. The email destination is set as part
of the email alarms configured in System | System Events | Alarms.
SMTP can be used with Embedded Voicemail for Voicemail Email. The voicemail destination is set
by the user's Voicemail Email address.
Field Description
Server Address Default = Blank
This field sets the IP address of the SMTP server being used to forward SNMP
alarms sent by email.
Port Default = 25. Range = 0 to 65534.
This field set the destination port on the SMTP server.
Email From Address Default = Blank
This field set the sender address to be used with mailed alarms. Depending of
the authentication requirements of the SMTP server this may need to be a
valid email address hosted by that server. Otherwise the SMTP email server
may need to be configured to support SMTP relay.
Use STARTTLS Default = Off. (Release 9.0.3).
Select this field to enable TLS/SSL encryption. Encryption allows voicemail-to-
email integration with hosted email providers that only permit SMTP over a
secure transport.
Server Requires Default = Off
Authentication
This field should be selected if the SMTP server being used requires
authentication to allow the sending of emails. When selected, the User Name
and Password fields become available
User Name Default = Blank This field sets the user name to be used for SMTP server
authentication.
Password Default = Blank This field sets the password to be used for SMTP server
authentication.
Use Challenge Default = Off. This field should be selected if the SMTP uses CRAM-MD5.
Response
Authentication (CRAM-
MD5)

Related links
System on page 192

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 System

System | SMDR
Navigation: System | SMDR
Using a specified IP address, the system can send a call record for each completed call.
Note:
Outbound Contact Express does not generate SMDR records.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Output Default = No Output.
Select the type of call record that the system should output via IP. The options
are:
• No Output
• SMDR Only : Send call records using the SMDR settings below.
SMDR: Station Message Detail Recorder Communications
This fields are available when SMDR is selected as the output. For information on SMDR record details,
see the appendix.
IP Address Default = 0.0.0.0 (Listen).
The destination IP address for SMDR records. The address 0.0.0.0 puts the
control unit in listen mode on the specified TCP port. When a connection is
made on that port, all SMDR records in the buffer are provided.
TCP Port Default = 0.
The destination IP port for SMDR records.
Records to Buffer Default = 500. Range = 10 to 3000.
The system can cache up to 3000 SMDR records if it detects a
communications failure with destination address. If the cache is full, the system
will begin discarding the oldest records for each new record.
Call Splitting for Default = Off.
Diverts
When enabled, for calls forwarded off-switch using an external trunk, the
SMDR produces separate initial call and forwarded call records. This applies
for calls forwarded by forward unconditional, forward on no answer, forward on
busy, DND or mobile twinning. It also applies to calls forwarded off-switch by
an incoming call route. The two sets of records will have the same Call ID. The
call time fields of the forward call record are reset from the moment of
forwarding on the external trunk.

Related links
System on page 192

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Configuration Mode Field Descriptions

VCM
Navigation: System | VCM
This form allows adjustment of the operation of any Voice Compression Modules (VCM's) installed
in a control unit.
Calls to and from IP devices can require conversion to the audio codec format being used by the
IP device. For systems this conversion is done by voice compression channels. These support the
common IP audio codecs G.711, G.723 and G.729a. For details of how to add voice compression
resources to a system, refer to the IP Office Installation Manual.
These settings should only be adjusted under the guidance of Avaya support.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
When are Voice Compression Channels Used
IP Device to Non-IP Device: These calls require a voice compression channel for the duration of
the call. If no channel is available, busy indication is returned to the caller.
IP Device to IP Device: Call progress tones (for example dial tone, secondary dial tone, etc) do
not require voice compression channels with the following exceptions:
• Short code confirmation, ARS camp on and account code entry tones require a voice
compression channel.
• Devices using G.723 require a voice compression channel for all tones except call waiting.
When a call is connected:
• If the IP devices use the same audio codec no voice compression channel is used.
• If the devices use differing audio codecs, a voice compression channel is required for each.
Non-IP Device to Non-IP Device: No voice compression channels are required.
Music on Hold: This is provided from the system's TDM bus and therefore requires a voice
compression channel when played to an IP device.
Conference Resources and IP Devices: Conferencing resources are managed by the
conference chip which is on the system's TDM bus. Therefore, a voice compression channel is
required for each IP device involved in a conference. This includes services that use conference
resources such as call listen, intrusion and silent monitoring. They also apply to call recording.
Page Calls to IP Device: Page calls require 1 voice compression channel per audio codec being
used by any IP devices involved. The system only uses G.729a for page calls, therefore only
requiring one channel but also only supporting pages to G.729a capable devices.
Voicemail Services and IP Devices: Calls to the system voicemail servers are treated as data
calls from the TDM bus. Therefore calls from an IP device to voicemail require a voice
compression channel.
Fax Calls: These are voice calls but with a slightly wider frequency range than spoken voice
calls. The system only supports fax across IP between systems with the Fax Transport option
selected.

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 System

SIP Calls:
• SIP Line Call to/from Non-IP Devices: Voice compression channel required.
• Outgoing SIP Line Call from IP Device: No voice compression channel required.
• Incoming SIP Line Call to IP Device: Voice compression channel reserved until call
connected.
T38 Fax Calls: The system supports T38 fax on SIP trunks and SIP extensions. Each T38 fax
call uses a VCM channel.
• Within a multi-site network, an T38 fax call can be converted to a call across across an H.323
line between systems using the Fax Transport Support protocol. This conversion uses 2
VCM channels.
• In order use T38 Fax connection, the Equipment Classification of an analog extension
connected to a fax machine can be set Fax Machine. Additionally, the short code feature
Dial Fax is available.
Measuring Channel Usability
The System Status Application can be used to display voice compression channel usage. Within
the Resources section it displays the number of channel in use. It also displays how often there
have been insufficient channels available and the last time such an event occurred.
Field Description
Echo Return Loss Default = 6dB. IP500 VCM, IP500 VCM V2 and IP500 Combination Cards. This
(dB) control allows adjustment of expected echo loss that should be used for the
echo cancellation process.
Echoes are typically generated by impedance mismatches when a signal is
converted from one circuit type to another, most notably from analog to IP. To
resolve this issue, an estimated echo signal can be created from one output and
then subtracted from the input to hopefully remove any echo of the output.
The options are:
• 0dB
• 3dB
• 6dB
• 9dB
Table continues…

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Configuration Mode Field Descriptions

Field Description
Nonlinear Processor Default = Adaptive. I
Mode
A low level of comfort noise is required on digital lines during periods where
there would normally be just silence. This is necessary to reassure users that
the call is still connected. These controls allow adjustment of the comfort noise
generated by the nonlinear processor (NLP) component of the VCM. The
options are:
• Adaptive: Adaptive means the comfort noise generated by the NLP will try to
match background noise.
• Silence: Silence means the NLP will not generate comfort noise at all
• Disabled: Nonlinear processing is not applied, in which case some residual
echo may be heard.
NLP Comfort Noise Default = -9dB.
Attenuation
The options are:
• -3dB
• -6dB
• -9dB
NLP Comfort Noise Default =-30dB.
Ceiling
The options are:
• -30dB
• -55dB
Modem
For Fax relay, these settings allow adjustment of the TDM side operation applied to fax calls using VCM
channels.
Tx Level (dB) Default = -9dB. Range = 0 to -13dB.
CD Threshold Default = -43dB, Options = -26dB, -31dB or -43dB.
No Activity Timeout Default = 30 seconds. Range = 10 to 600 seconds.
(secs)

Related links
System on page 192

Integrated Contact Reporter

Navigation: System | Contact Center | Integrated Contact Reporter
Integrated Contact Reporter is a simple Contact Center application that involves five to twenty five
agents and up to five supervisors. The application collects and displays information on the current
status of hunt groups and users that have been configured for Integrated Contact Reporter
operation.

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 System

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Field Description
Busy Not Available Reason Codes
Agents who indicate that they are in a 'busy not available' state can be prompt to also indicate the reason
for being in that state. This menu allows descriptions for the possible reasons to be entered. The
descriptions are then used in menus from which the Agent's make selections when setting themselves into
busy not available state and in reports on Agent status.
Code/Reason Rows 1 to 8 can be used to contain descriptions of up to 31 characters each.
Rows 0 and 9 are fixed as Unsupported and Busy Not Available.
Default After Call Default = 10. Range = 10 to 999 seconds.
Work Time (seconds)
If an agent goes into the After Call Work (ACW) state, either automatically or
manually, this field sets the duration of that state after which it is automatically
cleared. This duration can be overridden by the Agent's own setting (User |
Telephony | Supervisor Settings | After Call Work Time). During ACW state, hunt
group calls are not presented to the user.

Related links
System on page 192

VoIP
Navigation: System | VoIP
This tab is used to set the codecs available for use with all IP (H.323 and SIP) lines and
extensions and the default order of codec preference.
• Avaya H.323 telephones do not support G.723 and will ignore it if selected.
• For systems with H.323 lines and extensions, one of the G.711 codecs must be selected and
used.
• G.723 and G.729b are not supported by Linux based systems.
• The number of channels provided by an IP500 VCM 32 or IP500 VCM 64 card, up to a
maximum of 32 or 64 respectively, depends on the actual codecs being used. This also
applies to IP500 VCM 32 V2 and IP500 VCM 64 V2 cards. The following table assumes that
all calls using the VCM use the same codec.
Codec IP500 VCM 32 IP500 VCM 32 IP500 VCM 64 IP500 VCM 64
V2 V2
G.711 32 64
G.729a 30 60
G.723 22 44
G.722 30 60

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Configuration Mode Field Descriptions

Paging from an IP device uses the preferred codec of that device. It is the system administrator's
responsibility to ensure all the target phones in the paging group support that codec.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
Ignore DTMF Mismatch Default = On.
For Phones
When set to On, the following settings are visible and configurable:
• Extension | H.323 Extension | VoIP | Requires DTMF
• Extension | SIP Extension | VoIP | Requires DTMF
When set to On, during media checks, the system ignores DTMF checks if the call is
between two VoIP phones and the extension setting Requires DTMF is set to Off.
The two phones can be located on different systems in a Server Edition or SCN
deployment.

Note:
Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, are mismatched.
Allow Direct Media Default = Off.
Within NAT Location
When set to On, the system allows direct media between devices that reside behind
the same NAT. Devices are behind the same NAT if their public IP addresses are the
same.

Note:
Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, are mismatched.
The default behavior is to allow direct media between all types of devices (H323 and
SIP remote workers and IP Office Lines behind a NAT). In the case of routers that
have H323 or SIP ALG, it can be desirable to allow direct media only between
certain categories of devices. This can be configured by adding the NoUser Source
Number MEDIA_NAT_DM_INTERNAL. For information, see User | Source
Numbers.
RFC2833 Default Default = 101. Range = 96 - 127.
Payload
This field specifies the default value for RFC2833 dynamic payload negotiation.
Service providers that do not support dynamic payload negotiation may require a
fixed value.
Table continues…

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 System

Field Description
Available Codecs This list shows the codecs supported by the system and those selected as usable.
Those codecs selected in this list are then available for use in other codec lists
shown in the configuration settings. For example the adjacent Default Selection list
and the individual custom selection list on IP lines and extensions.

Warning:
Removing a codec from this list automatically removes it from the codec lists of
any individual lines and extensions that are using it.
The supported codecs (in default preference order) are: G.711 A-Law, G.711 U-
Law, G.722, G.729 and G.723.1. The default order for G.711 codecs will vary to
match the system's default companding setting. G.723.1 and G.729b are not
supported on Linux based systems.
Default Codec By default, all IP (H.323 and SIP) lines and extensions added to the system have
Selection their Codec Selection setting set to System Default. That setting matches the
codec selections made in this list. The buttons between the two lists can be used to
move codecs between the Unused and the Selected parts of the list and to change
the order of the codecs in the selected codecs list.

Related links
System on page 192

VoIP Security
Navigation: System | VoIP Security
Use to set system level media security settings. These settings apply to all lines and extensions
on which SRTP is supported and which have their Media Security settings configured to be Same
as System. Individual lines and extensions have media security settings that can override system
level settings.
Simultaneous SIP extensions that do not have physical extensions in the configuration use the
system security settings.
SM lines and all centralized user extensions must have uniform media security settings.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Name Description
Default Extension Default = Existing default extension password
Password
The field provides you with option to view and edit the existing default extension
password. The default extension password is set up during IP Office installation either by
the administrator or is randomly generated by the system. The system generated random
password is of 10 digits. Use the Eye icon to see the existing default password. The
password must be between 9 to 13 digits. The feature is not available in IP Office Basic
Edition systems.
Table continues…

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Configuration Mode Field Descriptions

Name Description
Confirm Default If you are changing the default extension password, type the new default password.
Extension
Password
Media Security Default = Disabled.
Secure RTP (SRTP) can be used between IP devices to add additional security. These
settings control whether SRTP is used for this system and the settings used for the
SRTP. The options are:
• Same as System: Matches the system setting at System | VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.

Warning:
Selecting Enforced on a line or extension that does not support media security
will result in media setup failures.
If media security is enabled (Enforced or Preferred), it is recommended that you enable
a matching level of security using System | LAN | VoIP | H.323 Signalling over TLS.
Media Security Not displayed if Media Security is set to Disabled. The options are:
Options
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which parts
of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to
select SRTP_AES_CM_128_SHA1_32.
Strict SIPS (Enterprise Branch deployments) Default = Off.
This option provides a system-wide configuration for call restrictions based on SIPS URI.
When this option is off, calls are not rejected due to SIPS. A call is sent according to the
configuration of the outgoing trunk or line that it is routed to, regardless of the way the
call came in, even if the call came in as a SIP invite with SIPS URI and is being sent with
a SIP URI onto a non-secure SIP trunk.
When this option is on, an incoming SIP invite with SIPS URI if targeted to a SIP trunk
(SM line or SIP line) is rejected if the target trunk is not configured with SIPS in the URI
Type field.

Related links
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 System

Access Control Lists

Navigation: System | VoIP
Name Description
SIP UA Blacklist The field to add SIP User Agent (UA) strings. The strings listed here are used to
block the registration of SIP devices when the system’s Allowed SIP User Agents
setting is set to Block Blacklist only. Use the Add button to add a Blacklist Entry
Name to the list.
SIP UA Whitelist The field to add SIP UA Whitelist strings. SIP User Agent (UA) strings listed here are
used to allow the registration of SIP devices when the system’s Allowed SIP User
Agents setting is set to Avaya Clients & Whitelisted or Whitelisted Only. Use the Add
button to add a Whitelist Entry Name to the list.
IP Whitelist The system can automatically blacklist traffic from an IP address based on too many
failed registration attempts from that address. This list can be used to create a list of
addresses which should not be blacklisted. This may be useful when there are
multiple devices registering from behind the same single public IP address, a
scenario where there may be a higher incidence of unintended registration failures
during initial setup. Use the Add button to add a IP Address Entry Value to the list.

Related links
System on page 192

Dialer
Navigation: System | Dialer
Use to configure the functions required for an Outbound Contact Express deployment.
These settings are mergeable. However, changes to the Operation field or to the Trunk Range /
IP Office table require a reboot.
It is recommended that you do not change the mergeable settings while the system is in use.
Field Description
Operation Default = Off.
On the primary IP Office Server Edition server, set this field to Primary. For all other IP
Office servers, set this field to Child. When set to Off or Child, no other fields are
displayed.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Record Mode Default = Off
Defines the automatic call recording function on VMPro. The options are:
• Whole Call: The entire call is recorded.
• Agent Connected: Recording starts once the conversation begins.
• Off
Record Controls : Default = Full Defines what functions an agent can perform from WebAgent or from the
handset. The options are:
• Full
• Pause
• Off
Record Mode Record Mode and Record Controls are related. The combined configuration settings are
and Record listed below.
Controls
Note that stopping and starting the recording creates multiple recording files. Pausing and
resuming the recording keeps the recording in a single file.

Record Mode Record Controls Result

Off Off Calls are not recorded.
Agent Connected Off All calls are always recorded
from the time the agent joins
the call.
Agent Connected Pause All calls are always recorded
but the Agent can pause and
resume recording.
Agent Connected Full All calls are always recorded
from the time the Agent joins
the call. Agent has full
control on when calls get
recorded.
Whole Call Off All calls are always recorded
from the time the customer
answers.
Whole Call Pause All calls are always recorded
from the time the customer
answers but the Agent can
pause and resume the
recording.
Whole Call Full Call recording starts before
the agent is connected. All
calls are always recorded
but the Agent can pause and
resume the recording
Table continues…

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 System

Field Description
Agent Call Back Default = 60. Range = 30 - 300.
Time
The number of seconds an agent has to make a manual call after a customer hang up.
Used when a customer wants to be called on a different number.
Remote Agent Default = Blank. Maximum length = 33.
Display Text
Specify the text string displayed on the remote agent extension if that extension supports
displays and the protocol allows it to be transmitted.
Remote Agent Default = Blank. Maximum length = 31.
Confirmation
Specify the Call Flow Entry point name used to play a greeting to the remote agent when
Voice Prompt
they log in. The actual Entry Point is added as a Modules Entry point using the VMPro
Client. The entry point cannot be added as a short code, user or group entry point.
Remote Agent Default = 0. The first extension number allocated to a remote agent. It must not conflict with
First Extension the existing dialing plan. If the range contains existing user extensions, they are used when
Number assigning extensions to remote users.
Remote Agent Default = 0. Maximum = 500.
Number of
The range of extensions starting from the one above. A user is created for every extension.
Extensions
If the field is edited and the number of extensions is reduced, the number of remote agents
that can log in is reduced to the new setting. However, reducing the range does not
automatically delete previously created users. Users can only be deleted manually.
Use Custom Hold Default = unchecked. Defines system behavior when a call is placed on hold. When
Treatment unchecked the the system Hold Music setting is used for the system's music on hold
source. When checked, the music on hold source is VMPro.
Record while on Default = unchecked. When the Use Custom Hold Treatment box is checked, the Record
Hold while on Hold setting can be enabled. When unchecked, recording is paused when the
call is on hold. When checked, recording continues when the call is on hold.
Trunk Range / IP The number of trunks used by Outbound Contact Express. The default entry is Trunk
Office Range: 1-250 for the Primary (Local) server. 250 is the maximum number of trunks
configured on a single server. Use this table to define the number of trunks managed by
the Primary and Secondary systems. The trunk range must match the line numbers used
by the Proactive Contact Dialer. Enter only one range per server.

Related links
System on page 192

Contact Center
Navigation: System | Contact Center
The Contact Center tab contains the user information required by IP Office to synchronize account
information with an an Avaya Contact Center Select (ACCS) system. The information is
synchronized using the Contact Center Management Application (CCMA). These settings are only
used for the deployment of an ACCS system.
This tab is visible on the Server Edition Primary Server and Standard Mode IP500 V2 systems.

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Configuration Mode Field Descriptions

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Contact Center Default = None.
Application
The options are:
• Avaya Contact Center Select
• Avaya IP Office Contact Center
• Integrated Contact Reporter (not supported in IP Office Release 11.0)
Synchronize to Default = Off.
this System
When set to On, the CCMA fields below are enabled.
CCMA Address Default = Blank
Address of the Contact Center Management Application system.
CCMA Username Default = Blank
User name on the Contact Center Management Application system.
CCMA Password Default = Blank
Password on the Contact Center Management Application system.
Default After Call Applicable for Integrated Contact Reporter
Work Time
Default = 10 seconds, Minimum = 10 seconds, Maximum = 999 seconds
The default time set for After Call Work (ACW). If configured, ACW begins at the end of a
call. Hunt group calls are not sent to the agent during ACW.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Busy Not Applicable for Integrated Contact Reporter
Available Reason
Default = 2 codes
Codes
Maximum = 9 codes
The reasons for 0 and 9 are assigned by default and cannot be modified. You can
configure the rest.
Use the configure icon to add Busy Not Available reasons and assign them to the
available codes.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.

Related links
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 System

Avaya Cloud Services

Navigation: System > Avaya Cloud Services
The Avaya Cloud Services tab contains configuration settings for user information synchronization
with Avaya Spaces server. IP Office users created for Avaya Equinox™ must be synchronized with
the Avaya Spaces server before using Avaya Equinox™ in a cloud environment. User
synchronization can be done manually or automatically.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Profile Name Default = None
The profile name for the Avaya Cloud Services settings.
Enable Avaya Default = Selected
Cloud Account
Enables interoperability. If disabled, synchronization stops but there is no effect on the
information that is already synchronized.
Zang URL Default = accounts.zang.io
Do not change this value.
Enable Settings Default = Disabled
File URL Sync
The options are:
• Enable for IP Office current Node: The settings file URL synchronization is done for
the IP Office system currently being used.
• Enable for all IP Office Nodes: The settings file URL synchronization is done for all the
connected IP Office nodes.
Zang Domain Default = Blank
The company domain registered and verified with Avaya Spaces.
Enable User Default = Not selected
Synchronization
If enabled, the IP Office system automatically synchronizes user information with Avaya
Spaces.
Avaya Spaces API The API Key from Avaya Spaces. Use the Eye icon to view the key. To obtain the key, log
Key on to the Avaya Spaces account and browse to Zang Account > Manage Companies >
Company Profile > API Key > API Key.
Avaya Spaces The key secret from Avaya Spaces account. Use the Eye icon to view the key. To obtain
Key Secret the key secret, log on to the Avaya Spaces account and browse to Zang Account >
Manage Companies > Company Profile > API Key > View/Edit > Secret.

Related links
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Configuration Mode Field Descriptions

Line
The line settings shown in the system configuration will change according to the types of trunk
cards installed in the control unit or added using external expansion modules.
Warning:
Changing Trunk Cards Changing the trunk card installed in an control unit will result in line
settings for both the previous trunk card and the currently installed trunk card. In order to
change the trunk card type in a particular card slot, the configuration must be defaulted. This
does not apply if replacing an existing card with one of a higher capacity or fitting a trunk card
into a previously unused slot.
Trunk Incoming Call Routing
Each trunk type can be categorized as either an external trunk or internal trunk. The trunk type
affects how the system routes calls received on that trunk and the routing of calls to the trunk.
External Trunks Internal Trunks
Trunk Types Analog trunks QSIG (T1, E1 or H.323)
T1 Robbed Bit BRI So
E1R2 H.323
ISDN BRI (excluding So) SCN
ISDN PRI T1 SES
ISDN PRI E1 IP Office Line
SIP
Incoming Calls Routed by All incoming calls are routed by Incoming calls are routed by
comparison of call details for looking for a match to the
matches within the system incoming digits in the following
Incoming Call Routes. order:
Line short codes are not used. • Extension number.
• Trunk short codes (excluding ?
short code).
• System short codes
(excluding ? short code).
• Trunk ? short code.
• System ? short code.

Line Groups
Each system trunk (or in some cases individual trunk channels) can be configured with an
Incoming Group ID and an Outgoing Group ID. These group IDs are used as follows:
• Incoming Call Routes For incoming calls on external trunks, the Incoming Group ID of the
trunk is one of the factors used to match the call to one of the configured incoming call
routes.

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 Line

• Short Codes - Routing Outgoing Calls For dialing which matches a short code set to a Dial
feature, the short codes Line Group ID can indicate either an ARS form or to use a trunk
from set to the same Outgoing Group ID. If the call is routed to an ARS form, the short
codes in the ARS form will specify the trunks to use by matching Outgoing Group ID.
Removing Unused Trunks
In cases where a trunk card is installed but the trunk is not physically connected, it is important to
ensure that the trunk is disabled in the configuration. This can be done on most trunks using by
setting the line's Admin setting to Out of Service.
This is especially important with analog trunks. Failure to do this may cause the system to attempt
to present outgoing calls to that trunk. Similarly, where the number of channels subscribed is less
than those supportable by the trunk type, the unsubscribed channels should be disabled.
Clock Quality
Calls between systems using digital trunks (for example E1, E1R2, T1 PRI and BRI) require an
common clock signal. The system will try to obtain this clock signal from an exchange through one
of its digital trunks. This is done by setting the Clock Quality setting of that Line to Network. If there
are multiple trunks to public exchanges, another trunk can be set as Fallback should the primary
clock signal fail. Other trunks should be set as Unsuitable.
Related links
Configuration Mode Field Descriptions on page 186
Analog Line on page 267
BRI Line on page 276
PRI Trunks on page 281
S0 Line on page 312
H.323 Line on page 315
IP DECT Line on page 321
SIP Line on page 326
SIP DECT Line on page 360
SM Line on page 362
IP Office Line on page 371

Analog Line
Analog trunks can be provided within the systems in the following ways. In all cases the physical
ports are labeled as Analog. For full details of installation refer to the IP Office Installation manual.
Using ICLID: The system can route incoming calls using the ICLID received with the call.
However ICLID is not sent instantaneously. On analog trunks set to Loop Start ICLID, there will be
a short delay while the system waits for any ICLID digits before it can determine where to present
the call.
Line Status: Analog line do not indicate call status other than whether the line is free or in use.
Some system features, for example retrieving unanswered forwards and making twinned calls
make use of the call status indicated by digital lines. This is not possible with analog lines. Once

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Configuration Mode Field Descriptions

an analog line has been seized, the system has to assume that the call is connected and treats it
as having been answered.
Dialing Complete: The majority of North-American telephony services use en-bloc dialing.
Therefore the use of a ; is recommended at the end of all dialing short codes that use an N. This is
also recommended for all dialing where secondary dial tone short codes are being used.
Ground Start: This type of analog trunk is only supported through the Analog Trunk external
expansion module.
Related links
Line on page 266
Line Settings on page 268
Analog Options on page 269

Line Settings
Navigation: Line | Analog Line | Line Settings
Configuration Settings
These settings are mergeable with the exception of the Network Type setting. Changes to this
setting will require a reboot of the system.
Field Description
Line Number This parameter is not configurable, it is allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device
providing the line. For IP500 V2 control units: 1 to 4 match the slots on the
front of the control unit from left to right. Expansion modules are numbered
from 5 upwards, for example trunks on the module in Expansion Port 1 are
shown as 5.
Port Indicates the port on the Card/Module above to which the configuration
settings relate.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict
Network Interconnect is enabled. It allows the trunk to be set as either
Public or Private. The system will return number busy indication to any
attempt to connect a call on a Private trunk to a Public trunk or vice versa.
This restriction includes transfers, forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks,
VPNremote, application telecommuter mode.
Telephone Number Used to remember the external telephone number of this line to assist with
loop-back testing. For information only.
Incoming Group ID Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs
is used to match it to incoming call routes in the system configuration. The
matching incoming call route is then used to route incoming calls. The same ID
can be used for multiple lines.
Table continues…

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 Line

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be
used. The system will then seize a line from those available with a matching
Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network. The same ID cannot be used in the
configuration of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP
Office lines to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office
lines from the primary and secondary servers to each expansion system in
the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved
for the SM line.
Outgoing Channels Default = 1 (not changeable)
Voice Channels Default = 1 (not changeable)
Prefix Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if
all users must dial a prefix to access an outside line. The prefix is automatically
placed in front of all incoming numbers so that users can dial the number back.
For outgoing calls: The system does not strip the prefix, therefore any prefixes
not suitable for external line presentation should be stripped using short codes.
Line Appearance ID Default = Auto-assigned. Range = 2 to 9 digits. Allows a number to be
assigned to the line to identify it. On phones that support call appearance
buttons, a Line Appearance button with the same number will show the status
of the line and can be used to answer calls on the line. The line appearance ID
must be unique and not match any extension number.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance
or if the trunk is not connected.

Related links
Analog Line on page 267

Analog Options
Navigation: Line | Analog Line | Analog Options

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Configuration Mode Field Descriptions

Covers analog line specific settings. The system wide setting System | Telephony | Tones &
Music | CLI Type is used for to set the incoming CLI detection method for all analogue trunks.
The Allow Analog Trunk to Trunk Connect setting is mergeable. The remaining settings are not
mergeable. Changes to these settings will require a reboot of the system.
Field Description
Channel Set by the system. Shown for information only.
Trunk Type Default = Loop Start
Sets the analog line type. The options are:
• Ground Start: Ground Start is only supported on trunks provided by the Analog
Trunk 16 expansion module. It requires that the module and the control unit are
grounded. Refer to the IP Office installation manual.
• Loop Start
• Loop Start ICLID: As the system can use ICLID to route incoming calls, on
analog Loop Start ICLID trunks there is a few seconds delay while ICLID is
received before the call routing can be determined.
Signaling Type Default = DTMF Dialing
Sets the signaling method used on the line. The options are:
• DTMF Dialing
• Pulse Dialing
Direction Default = Both Directions
Sets the allowed direction of operation of the line. The options are:
• Incoming
• Outgoing
• Both Directions
Flash Pulse Width Default = 0. Range = 0 to 2550ms.
Set the time interval for the flash pulse width.
Await Dial Tone Default = 0. Range = 0 to 25500ms.
Sets how long the system should wait before dialing out.
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 Line

Field Description
Echo Cancellation Default = 16ms.
The echo cancellation should only be adjusted as high as required to remove echo
problems. Setting it to a higher value than necessary can cause other distortions.
Not used with external expansion module trunks. The options are (milliseconds):
• Off
• 8
• 16
• 32
• 64
• 128
Echo Reduction Default = On. (ATM4Uv2 card only)
Used when impedance matching is not required but echo reduction is.
Mains Hum Filter Default = Off.
If mains hum interference on the lines is detected or suspected, this settings can be
used to attempt to remove that interference. Useable with ATM16 trunks and IP500
ATM4U trunks. The options are:
• Off
• 50Hz
• 60Hz
Impedance Set the impedance used for the line. This field is only available for system locales
where the default value can be changed.
The value used for Default is set by the setting System | System | Locale. For
information, see Avaya IP Office™ Platform Locale Settings.
The following values are used for Automatic Impedance Matching: 600+2150nF,
600, 900+2150nF, 900, 220+820||115nF, 370+620||310nF, 270+750||150nF,
320+1050||230nF, 350+1000||210nF, 800+100||210nF.
Quiet Line This field is only available for certain system locales (see above). The setting may
be required to compensate for signal loss on long lines.
Digits to break dial tone Default = 2. Range = Up to 3 digits.
During automatic impedance testing (see below), once the system has seized a
line, it dials this digit or digits to the line. In some cases it may be necessary to use
a different digit or digits. For example, if analog trunk go via another PBX system or
Centrex, it will be necessary to use the external trunk dialing prefix of the remote
system plus another digit, for example 92.
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Configuration Mode Field Descriptions

Field Description
Automatic Default = Yes. (ATM4Uv2 card only)
When set to Yes, the Default value is used. The value used for Default is set by the
system Locale.
When set to No, the Impedance value can be manually selected from the list of
possible values:
600
900 270+(750R || 150nF) and 275R + (780R || 150nF)
220+(820R || 120nF) and 220R+ (82R || 115nF)
370+(620R || 310nF)
320+(1050R || 230nF)
370+(820R || 110nF)
275+(780R || 115nF)
120+(820R || 110nF)
350+(1000R || 210nF)
200+(680R || 100nF)
600+2.16μF
900+1μF
900+2.16μF
600+1μF Global Impedance
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 Line

Field Description
Automatic Balance These controls can be used to test the impedance of a line and to then display the
Impedance Match best match resulting from the test. Testing should be performed with the line
connected but the system otherwise idle. To start testing click Start. The system will
then send a series of signals to the line and monitor the response, repeating this at
each possible impedance setting. Testing can be stopped at any time by clicking
Stop. When testing is complete, Manager will display the best match and ask
whether that match should be used for the line. If Yes is selected, Manager will also
ask whether the match should be applied to all other analog lines provided by the
same analog trunk card or module.
Note that on the Analog Trunk Module (ATM16), there are four control devices, each
supporting four channels. The impedance is set by the control device for all four
channels under its control. Consequently, the impedance match tool only functions
on lines 1, 5, 9, and 13.
Before testing, ensure that the following system settings are correctly set:
• System | System | Locale
• System | Telephony | Telephony | Companding Law
If either needs to be changed, make the required change and save the setting to the
system before proceeding with impedance matching.
Due to hardware differences, the impedance matching result will vary slightly
depending on which type of trunk card or expansion module is being used.
Automatic Balance Impedance Matching, Quiet Line and Digits to break dial
tone are available for the Bahrain, Egypt, French Canadian, India, Kuwait, Morocco,
Oman, Pakistan, Qatar, Saudi Arabia, South Africa, Turkey, United Arab Emirates,
United States and Customize locales.
Allow Analog Trunk to Default = Not selected (Off). When not enabled, users cannot transfer or forward
Trunk Connect external calls back off-switch using an analog trunk if the call was originally made or
received on another analog trunk. This prevents transfers to trunks that do not
support disconnect clear.
If the setting System | Telephony | Telephony | Unsupervised Analog Trunk
Disconnect Handling is enabled, this setting is greyed out and trunk to trunk
connections to any analog trunks are not allowed.
BCC Default = Not selected [Brazil locale only]
A collect call is a call at the receiver's expense and by his permission. If supported
by the line provider, BCC (Block Collect Call) can be used to bar collect calls.
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Configuration Mode Field Descriptions

Field Description
Secondary Dial Tone Default = Off
Configures the use of secondary dial tone on analog lines. This is a different
mechanism from secondary dial tone using short codes. This method is used mainly
within the Russian locale. When selected, the options are:
• Await time: Default = 3000ms. Range = 0 to 25500ms. Used when secondary
dial tone (above) is selected. Sets the delay.
• After n Digits: Default = 1. Range = 0 to 10. Sets where in the dialing string, the
delay for secondary dial tone, should occur.
• Matching Digit: Default =8. Range = 0 to 9. The digit which, when first matched
in the dialing string, will cause secondary dial tone delay.
Long CLI Line Default = Off
The CLI signal on some analog lines can become degraded and is not then
correctly detected. If you are sure that CLI is being provided but not detected,
selecting this option may resolve the problem.
Modem Enabled Default = Off
The first analog trunk in a control unit can be set to modem operation (V32 with V42
error correction). This allows the trunk to answer incoming modem calls and be
used for system maintenance. When on, the trunk can only be used for analog
modem calls. The default system short code *9000* can be used to toggle this
setting.
For the IP500 ATM4U-V2 Trunk Card Modem, it is not required to switch the card's
modem port on/off. The trunk card's V32 modem function can be accessed simply
by routing a modem call to the RAS service's extension number. The modem call
does not have to use the first analog trunk, instead the port remains available for
voice calls.
MWI Standard Default = None.
This setting is only displayed for ATM4U-V2 cards.
When System | Voicemail | Voicemail Type is set to Analogue MWI, change this
setting to Bellcore FSKBellcore FSK.
Pulse Dialing These settings are used for pulse dialing.
• Mark: Default = 40ms. Range = 0 to 255. Interval when DTMF signal is kept
active during transmission of DTMF signals.
• Space: Default = 60ms. Range = 0 to 255. Interval of silence between DTMF
signal transmissions.
• Inter-Digit Pause: Default = 500ms. Range = 0 to 2550ms. Sets the pause
between digits transmitted to the line.
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 Line

Field Description
Ring Detection These settings are used for ring detection.
• Ring Persistency: Default = Set according to system locale. Range = 0 to
2550ms. The minimum duration of signal required to be recognized.
• Ring Off Maximum: Default = Set according to system locale. Range = 0 to
25500ms. The time required before signaling is regarded as ended.
Disconnect Clear Disconnect clear (also known as 'Line Break' or 'Reliable Disconnect') is a method
used to signal from the line provider that the call has cleared. The system also uses
'Tone Disconnect', which clears an analog call after 6 seconds of continuous tone,
configured through the Busy Tone Detection (System | Telephony | Tones &
Music) settings.
• Enable: Default = On Enables the use of disconnect clear.
• Units: Default = 500ms. Range = 0 to 2550ms. This time must be less than the
actual disconnect time period used by the line provider by at least 150ms.
If the setting System | Telephony | Telephony | Unsupervised Analog Trunk
Disconnect Handling is enabled, this setting is greyed out and disconnect clear
disabled.
DTMF These settings are used for DTMF dialing.
• On: Default = 80ms. Range = 0 to 255ms. The width of the on pulses generated
during DTMF dialing.
• Off: Default = 80ms. Range = 0 to 255ms. The width of the off pulses generated
during DTMF dialing.
BCC Flash Pulse Width [Brazil locale only] Default = 100 (1000ms). Range = 0 to 255.
Sets the BCC (Block collect call) flash pulse width.
Gains These settings are used to adjust the perceived volume on all calls.
• A | D: Default = 0dB. Range =-10.0dB to +6.0dB in 0.5dB steps. Sets the analog
to digital gain applied to the signal received from the trunk by the system. To
conform with the Receive Objective Loudness Rating at distances greater than
2.7km from the central office, on analog trunks a receive gain of 1.5dB must be
set.
• D | A: Default = 0dB. Range =-10.0dB to +6.0dB in 0.5dB steps. Sets the digital to
analog gain applied to the signal from the system to the trunk.
• Voice Recording: Default = Low Used to adjust the volume level of calls recorded
by voicemail. The ptions are:
- Low
- Medium
- High

Related links
Analog Line on page 267

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Configuration Mode Field Descriptions

BRI Line
BRI trunks are provided by the installation of a BRI trunk card into the control unit. The cards are
available in different variants with either 2 or 4 physical ports. Each port supports 2 B-channels for
calls. For full details of installation refer to the IP Office Installation manual.
Point-to-Point or Multipoint

BRI lines can be used in either Point-to-Point or Point-to-Multipoint mode. Point-to-Point lines are
used when only one device terminates a line in a customer's office. Point-to-Multipoint lines are
used when more than one device may be used on the line at the customer's premises. There are
major benefits in using Point-to-Point lines:-
• The exchange knows when the line/terminal equipment is down/dead, thus it will not offer
calls down that line. If the lines are Point-to-Multipoint, calls are always offered down the line
and fail if there is no response from the terminal equipment. So if you have two Point-to-
Multipoint lines and one is faulty 50% of incoming calls fail.
• You get a green LED on the Control Unit when the line is connected. With Point-to-Multipoint
lines some exchanges will drop layer 1/2 signals when the line is idle for a period.
• The timing clock is locked to the exchange. If layer 1/2 signals disappear on a line then the
Control Unit will switch to another line, however this may result in some audible click when
the switchover occurs.
The system's default Terminal Equipment Identifier (TEI) will normally allow it to work on Point-to-
Point or Point-to-Multipoint lines. However if you intend to connect multiple devices simultaneously
to an BRI line, then the TEI should be set to 127. With a TEI of 127, the control unit will ask the
exchange to allocate a TEI for operation.
Note:
When connected to some manufactures equipment, which provides an S0 interface (BRI), a
defaulted Control Unit will not bring up the ISDN line. Configuring the Control Unit to a TEI of
127 for that line will usually resolve this.
Related links
Line on page 266
BRI Line on page 276
Channels on page 281

BRI Line
Navigation: Line | BRI Line
The following settings are not mergeable. Changes to these settings will require a reboot of the
system.
• Line Sub Type
• Network Type
• TEI

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 Line

• Add 'Not-end-to-end ISDN' Information Element

• Progress Replacement
• Clock Quality
• Force Number Plan to ISDN
Decreasing the Number of Channels setting requires a a “merge with service disruption”. When
the configuration file is sent to the system, active calls on the deleted channels are cleared.
The remaining settings are mergeable.
Field Description
Card/Module Indicates the card slot or expansion module being used for the trunk device
providing the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit
from left to right. Expansion modules are numbered from 5 upwards, for example
trunks on the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings
relate.
Line Number This parameter is not configurable; it is allocated by the system.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if
the trunk is not connected.
Line Sub Type Default = NTT for Japan/ETSI for other locales.
Select to match the particular line type provided by the line provider. IP500 BRI
daughter cards can be configured for S-Bus (So) operation for connection to ISDN
terminal devices. Note that this requires the addition of terminating resistors at both
the system and remote ends, and the use of a suitable cross-over cable. For full
details refer to the Deploying Avaya IP Office Platform IP500 V2 manual.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Telephone Number Used to remember the external telephone number of this line to assist with loop-
back testing. For information only.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used
to route incoming calls. The same ID can be used for multiple lines.
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Configuration Mode Field Descriptions

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used.
The system will then seize a line from those available with a matching Outgoing
Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network. The same ID cannot be used in the
configuration of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office
lines to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office
lines from the primary and secondary servers to each expansion system in the
network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for
the SM line.
Prefix Default = Blank. The prefix is used in the following ways:
• For incoming calls: The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls: The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number
is presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added.
For example 441923000000 is converted to 00441923000000.
TEI Default = 0 The Terminal Equipment Identifier. Used to identify each device
connected to a particular ISDN line. For Point-to-Point lines this is 0. It can also be
0 on a Point to Multipoint line, however if multiple devices are sharing a Point-to-
Multipoint line it should be set to 127 which results in the exchange allocating the
TEI's to be used.
Number of Channels Default = 2. Range = 0 to 2.
Defines the number of operational channels that are available on this line.
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 Line

Field Description
Outgoing Channels Default = 2. Range = 0 to 2.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
Voice Channels Default = 2. Range = 0 to 2.
The number of channels available for voice use.
Data Channels Default = 2. Range = 0 to 2.
The number of channels available for data use. If left blank, the value is 0.
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether
the system should try to take its clock source for call synchronization and signalling
from this line. Preference should always be given to using the clock source from a
central office exchange if available by setting at least one exchange line to
Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network
source not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock
source.
• In scenarios where several systems are network via digital trunk lines, care must
be taken to ensure that all the systems use the same clock source. The current
source being used by a system is reported within the System Status Application.
Add 'Not-end-to-end Default = Never*. Sets whether the optional 'Not end-to-end ISDN' information
ISDN' Information element should be added to outgoing calls on the line. The options are Never,
Element Always or POTS (only if the call was originated by an analog extension). *The
default is Never except for the following locales; for Italy the default is POTS, for
New Zealand the default is Always.
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Configuration Mode Field Descriptions

Field Description
Progress Replacement Default = None.
Progress messages are defined in the Q.931 ISDN connection control signaling
protocol. Generally, If a progress message is sent, the caller does not get
connected and so typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages.
The options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not
hear the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
Supports Partial Default = Off.
Rerouting
Partial rerouting (PR) is an ISDN feature. It is supported on external (non-network
and QSIG) ISDN exchange calls. When an external call is transferred to another
external number, the transfer is performed by the ISDN exchange and the channels
to the system are freed. Use of this service may need to be requested from the line
provider and may incur a charge.
Force Number Plan to Default = Off.
ISDN
This option is only configurable when Support Partial Rerouting is also enabled.
When selected, the plan/type parameter for Partial Rerouting is changed from
Unknown/Unknown to ISDN/Unknown.
Send Redirecting Default = Off.
Number
This option can be used on ISDN trunks where the redirecting service is supported
by the trunk provider. Where supported, on twinned calls the caller ID of the original
call is passed through to the twinning destination. This option is only used for
twinned calls.
Support Call Tracing Default = Off. The system supports the triggering of malicious caller ID (MCID)
tracing at the ISDN exchange. Use of this feature requires liaison with the ISDN
service provider and the appropriate legal authorities to whom the call trace will be
passed. The user will also need to be enabled for call tracing and be provider with
either a short code or programmable button to activate MCID call trace. Refer to
Malicious Call Tracing in the Telephone Features section for full details.
Active CCBS Support Default = Off.
Call completion to a busy subscriber (CCBS). It allows automatic callback to be
used on outgoing ISDN calls when the destination is busy. This feature can only be
used on point-to-point trunks. Use of this service may need to be requested from
the line provider and may incur a charge.
Passive CCBS Default = Off.
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 Line

Field Description
Cost Per Charging Unit The information is provided in the form of charge units. This setting is used to enter
the call cost per charging unit set by the line provider. The values are 1/10,000th of
a currency unit. For example if the call cost per unit is £1.07, a value of 10700
should be set on the line. Refer to Advice of Charge.
Send original calling Default = Off.
party for forwarded and
Use the original calling party ID when forwarding calls or routing twinned calls.
twinning calls
This setting applies to BRI lines with subtype ETSI.
Originator number for Default = blank.
forwarded and twinning
The number used as the calling party ID when forwarding calls or routing twinned
calls
calls. This field is grayed out when the Send original calling party for forwarded
and twinning calls setting is enabled.
This setting applies to BRI lines with subtype ETSI.

Related links
BRI Line on page 276

Channels
Navigation: Line | BRI Line | Channels
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel
either double-click on it or click the channel and then select Edit.
To edit multiple channels at the same time, select the required channels using Ctrl or Shift and
then click Edit. When editing multiple channels, fields that must be unique such as Line
Appearance ID are not shown.
These settings are mergeable. Changes to these settings do not require a system reboot.
Field Description
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits.
ID
Used for configuring Line Appearances with button programming. The line appearance ID
must be unique and not match any extension number. Line appearance is not supported
for trunks set to QSIG operation and is not recommended for trunks be used for DID.

Related links
BRI Line on page 276

PRI Trunks
PRI trunks are provided by the installation of a PRI trunk card into the control unit. avThe IP500
PRI-U trunk card can be configured (see below) to one of those line types. The cards are also
available with either 1 or 2 physical ports. The number of B-channels supported by each physical
port depends on the line type of the card.
• E1: 30 B-channels and 1 D-channel per port.

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Configuration Mode Field Descriptions

• T1: 24 B-channels per port.

• US PRI: 23 B-channels and 1 D-channel per port.
• E1-R2: 30 B-channels and 1 D-channel per port.
IP500 PRI-U Trunk Card Line Type
The IP500 PRI-U card can be configured to support either E1, T1 or E1-R2 PRI line types. To
select the line type required, right-click on the line in the group or navigation pane and select
Change Universal PRI Card Line Type.
The control unit supports 8 B-channels on each IP500 PRI-U card fitted. Additional B-channels up
to the full capacity of IP500 PRI-U ports installed require licenses added to the configuration. D-
channels are not affected by licensing.
For ETSI and QSIG trunks, license instances are consumed by the number of calls in progress on
B-channels.
For T1, E1R2 and ETSI CHI trunks, licenses instances are consumed by the channels set as in
service.
Related links
Line on page 266
E1 Line on page 282
E1 R2 Line on page 291
T1 Line on page 297
T1 PRI Line on page 302

E1 Line
Related links
PRI Trunks on page 281
E1 PRI Line on page 282
E1 Short Codes on page 289
E1 PRI Channels on page 289

E1 PRI Line
Navigation: Line | E1 PRI Line
The following settings are not mergeable. Changes to these settings require a system reboot.
• Line Sub Type
• Network Type
• TEI
• Channel Allocation
• CRC Checking
• Clock Quality
• Add 'Not-end-to-end ISDN' Information Element
• Progress Replacement

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 Line

• Force Number Plan to ISDN

• Line Signalling
Decreasing the Number of Channels setting requires a a “merge with service disruption”. When
the configuration file is sent to the system, active calls on the deleted channels are cleared.
The remaining settings are mergeable.
Field Description
Line Number This parameter is not configurable; it is allocated by the system.
Line Sub Type Select to match the particular line type provided by the line provider. The options are:
• ETSI
• ETSI CHI
• QSIG A
• QSIG B
ETSI CHI is used to send the channel allocation ID (CHI) in the call setup signalling.
This is a request to use a particular B-channel rather than use any B-channel
allocated by the central office exchange.
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing
the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from
left to right. Expansion modules are numbered from 5 upwards, for example trunks
on the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings
relate.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Telephone Number Used to remember the external telephone number of this line to assist with loop-back
testing. For information only.
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Configuration Mode Field Descriptions

Field Description
Channel Allocation Default = 30|1.
For lines set to ETSI CHI, this option allows the system to select the default order in
which channels should be used for outgoing calls. Typically this is set as the opposite
of the default order in which the central office exchange uses channels for incoming
calls.
For lines set to the Line Sub Type of ETSI CHI, the Incoming Group ID is set as
part of the individual channel settings.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used. The
system will then seize a line from those available with a matching Outgoing Group
ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network. The same ID cannot be used in the
configuration of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office
lines to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for
the SM line.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number is
presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
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 Line

Field Description
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added. For
example 441923000000 is converted to 00441923000000.
TEI Default = 0 The
Terminal Equipment Identifier. Used to identify each Control Unit connected to a
particular ISDN line. For Point to Point lines this is typically (always) 0. It can also be
0 on a Point to Multi-Point line, however if multiple devices are sharing a Point to
Multi-Point line it should be set to 127 which results in the exchange deciding on the
TEI's to be used.
Number of Channels Defines the number of operational channels that are available on this line. Up to 30
for E1 PRI, 23 for T1 PRI.
Outgoing Channels This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls. Only available when the
Line Sub Type is set to ETSI.
Voice Channels The number of channels available for voice use. Only available when the Line Sub
Type is set to ETSI.
Data Channels The number of channels available for data use. Only available when the Line Sub
Type is set to ETSI.
CRC Checking Default = On
Switches CRC on or off.
Line Signalling Default = CPE This option is not used for lines where the Line SubType is set to
QSIG. Select either CPE (customer premises equipment) or CO (central office). The
CO feature is intended to be used primarily as a testing aid. It allows PRI lines to be
tested in a back-to-back configuration, using crossover cables.
The CO feature operates on this line type by modifying the way in which incoming
calls are disconnected for system configuration in Brazil and Argentina. In these
locales, the CO setting uses Forced-Release instead of Clear-Back to disconnect
incoming calls. The Brazilian Double-Seizure mechanism, used to police Collect
calls, is also disabled in CO mode.
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Configuration Mode Field Descriptions

Field Description
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from
this line. Preference should always be given to using the clock source from a central
office exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network source
not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must
be taken to ensure that all the systems use the same clock source. The current
source being used by a system is reported within the System Status Application.
Add 'Not-end-to-end Default = Never
ISDN' Information
Sets whether the optional 'Not end-to-end ISDN' information element should be
Element
added to outgoing calls on the line. The options are:
• Never
• Always
• POTS(only if the call was originated by an analog extension).
The default is Never except for the following locales; for Italy the default is POTS, for
New Zealand the default is Always.
Progress Replacement Default = None.
Progress messages are defined in the Q.931 ISDN connection control signaling
protocol. Generally, If a progress message is sent, the caller does not get connected
and so typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages.
The options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not
hear the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
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 Line

Field Description
Supports Partial Default = Off.
Rerouting
Partial rerouting (PR) is an ISDN feature. It is supported on external (non-network
and QSIG) ISDN exchange calls. When an external call is transferred to another
external number, the transfer is performed by the ISDN exchange and the channels
to the system are freed. Use of this service may need to be requested from the line
provider and may incur a charge.
Force Number Plan to Default = Off.
ISDN
This option is only configurable when Support Partial Rerouting is also enabled.
When selected, the plan/type parameter for Partial Rerouting is changed from
Unknown/Unknown to ISDN/Unknown.
Send Redirecting Default = Off.
Number
This option can be used on ISDN trunks where the redirecting service is supported
by the trunk provider. Where supported, on twinned calls the caller ID of the original
call is passed through to the twinning destination. This option is only used for
twinned calls.
Support Call Tracing Default = Off.
The system supports the triggering of malicious caller ID (MCID) tracing at the ISDN
exchange. Use of this feature requires liaison with the ISDN service provider and the
appropriate legal authorities to whom the call trace will be passed. The user will also
need to be enabled for call tracing and be provider with either a short code or
programmable button to activate MCID call trace. Refer to Malicious Call Tracing in
the Telephone Features section for full details.
Active CCBS Support Default = Off.
Call completion to a busy subscriber (CCBS). It allows automatic callback to be used
on outgoing ISDN calls when the destination is busy. This feature can only be used
on point-to-point trunks. Use of this service may need to be requested from the line
provider and may incur a charge.
Passive CCBS Default = Off.
Cost Per Charging Unit Advice of charge (AOC) information can be display on T3/T3IP phones and output in
SMDR. The information is provided in the form of charge units. This setting is used to
enter the call cost per charging unit set by the line provider. The values are
1/10,000th of a currency unit. For example if the call cost per unit is £1.07, a value of
10700 should be set on the line. Refer to Advice of Charge.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
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Configuration Mode Field Descriptions

Field Description
Send original calling Default = Off.
party for forwarded
Use the original calling party ID when forwarding calls or routing twinned calls.
and twinning calls
This setting applies to the following ISDN lines:
• PRI24 with subtypes:
- PRI
- QSIGA
- QSIGB
- ETSI
- ETSI CHI
• PRI30 with subtypes
- QSIGA
- QSIGB
- ETSI
- ETSI CHI
Originator number for Default = blank.
forwarded and
The number used as the calling party ID when forwarding calls or routing twinned
twinning calls
calls. This field is grayed out when the Send original calling party for forwarded
and twinning calls setting is enabled.
This setting applies to the following ISDN lines:
• PRI24 with subtypes:
- PRI
- QSIGA
- QSIGB
- ETSI
- ETSI CHI
• PRI30 with subtypes
- QSIGA
- QSIGB
- ETSI
- ETSI CHI

The following fields are shown for a US T1 trunk card set to ETSI or QSIG operation. These cards
have the same settings E1 PRI trunk cards set to ETSI or QSIG but only support 23 channels.
These settings are not mergeable. Changing these settings requires a system reboot.

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 Line

Field Description
CSU Operation Check this field to enable the T1 line to respond to loop-back requests from the line.
Haul Length Default = 0-115 feet
Sets the line length to a specific distance.
Channel Unit Default = Foreign Exchange This field should be set to match the channel signaling
equipment provided by the Central Office. The options are Foreign Exchange,
Special Access or Normal.

Related links
E1 Line on page 282

E1 Short Codes
Navigation: Line | E1 Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those
types of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Changes to these settings do not require a reboot of the system.
Related links
E1 Line on page 282

E1 PRI Channels
Navigation: Line | E1 PRI Channels
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel
either double-click on it or click the channel and then select Edit.
To edit multiple channels at the same time, select the required channels using Ctrl or Shift and
then click Edit. When editing multiple channels, fields that must be unique such as Line
Appearance ID are not shown.
The following settings are mergeable:
• Line Appearance ID (ETSI, ETSI CHI)
• Admin (ETSI CHI)

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Configuration Mode Field Descriptions

• Incoming Group ID (ETSI CHI)

• Outgoing Group ID (ETSI CHI)
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Line Appearance ID Default = Auto-assigned. Range = 2 to 9 digits.
Used for configuring Line Appearances with button programming. The line
appearance ID must be unique and not match any extension number. Line
appearance is not supported for trunks set to QSIG operation and is not
recommended for trunks be used for DID.
If the trunk Line Sub Type is set to ETSI CHI, outgoing line appearance calls must
use the corresponding channel.

The following additional fields are shown for lines where the Line Sub Type is set to ETSI CHI.
Field Description
Incoming Group ID Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs is
used to match it to incoming call routes in the system configuration. The matching
incoming call route is then used to route incoming calls. The same ID can be used
for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used.
The system will then seize a line from those available with a matching Outgoing
Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network. The same ID cannot be used in the
configuration of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office
lines to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office
lines from the primary and secondary servers to each expansion system in the
network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for
the SM line.
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 Line

Field Description
Direction Default = Bothways
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Bothways
Bearer Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Admin Default = Out of Service.
This field can be used to indicate whether the channel is in use or not. On trunks
where only a limited number of channels have been requested from the trunk
provider (known as sub-equipped trunks), those channels not provided should be
set as Out of Service. For channels that are available but are temporarily not
being used select Maintenance.
Tx Gain Default = 0dB. Range = -10dBb to +5dB.
The transmit gain in dB.
Rx Gain Default = 0dB. Range = -10dBb to +5dB.
The receive gain in dB.

Related links
E1 Line on page 282

E1 R2 Line
Navigation: Line | E1–R2 Line
Related links
PRI Trunks on page 281
E1-R2 Options on page 291
E1-R2 Channels on page 293
E1 R2 MFC Group on page 295
E1-R2 Advanced on page 295

E1-R2 Options
Navigation: Line | E1–R2 Options
Changing the Admin setting is mergeable. The remaining settings are not mergeable. Changes to
these settings will require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Card/Module Indicates the card slot or expansion module being used for the trunk device providing
the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from
left to right. Expansion modules are numbered from 5 upwards, for example trunks on
the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private. The
system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Line Number Allocated by the system.
Line SubType Default = E1-R2
The options are:
• E1-R2
• ETSI
• QSIGA
• QSIGB
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Channel Allocation Default = 30 | 1
The order, 30 | 1 or 1 | 30, in which channels are used.
Country (Locale) Default = Mexico. Select the locale that matches the area of usage. Note that
changing the locale will return the MFC Group settings to the defaults for the selected
locale. Currently supported locales are:
• Argentina
• Brazil
• China
• India
• Korea
• Mexico
• None
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 Line

Field Description
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
The table at the base of the form displays the settings for the individual channels
provided by the line. For details of the channel settings see the E1-R2 Channel form.
To edit a channel, either double-click on it or right-click and select Edit. This will
display the Edit Channel dialog box. To edit multiple channels at the same time select
the channels whilst pressing the Shift or Ctrl key. Then right-click and select Edit.

Related links
E1 R2 Line on page 291

E1-R2 Channels
Navigation: Line | E1–R2 Channels
The channel settings are split into two sub-tabs, E1R2 Edit Channel and Timers.
The Timers tab displays the various timers provided for E1-R2 channels. These should only be
adjusted when required to match the line provider's settings.
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel,
select the required channel or channels and click Edit.
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Admin
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Descriptions
Channel The channel or channels being edited.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming
call routes in the system configuration. The matching incoming call route is
then used to route incoming calls. The same ID can be used for multiple lines.
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Configuration Mode Field Descriptions

Field Descriptions
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be
used. The system will then seize a line from those available with a matching
Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network. The same ID cannot be used in the
configuration of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP
Office lines to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP
Office lines from the primary and secondary servers to each expansion
system in the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment,
reserved for the SM line.
Direction Default = Both Directions
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Both Directions
Bearer Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Admin Default = Out of Service.
This field can be used to indicate whether the channel is in use or not. On
trunks where only a limited number of channels have been requested from the
trunk provider (known as sub-equipped trunks), those channels not provided
should be set as Out of Service. For channels that are available but are
temporarily not being used select Maintenance.
Table continues…

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 Line

Field Descriptions
Line Signaling Type Default = R2 Loop Start
The signaling type used by the channel. Current supported options are:
• R2 Loop Start
• R2 DID
• R2 DOD
• R2 DIOD
• Tie Immediate Start
• Tie Wink Start
• Tie Delay Dial
• Tie Automatic
• WAN Service
• Out of Service
Dial Type Default = MFC Dialing
The type of dialing supported by the channel. The options are:, or .
• MFC Dialing
• Pulse Dialing
• DTMF Dialing

Related links
E1 R2 Line on page 291

E1 R2 MFC Group
Navigation: Line | E1–R2 MFC Group
These settings are not mergeable. Changes to these settings will require a reboot of the system.
These tabs show the parameter assigned to each signal in an MFC group. The defaults are set
according to the Country (Locale) on the Line tab. All the values can be returned to default by the
Default All button on the Advanced tab.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
To change a setting either double-click on it or right-click and select Edit.
Related links
E1 R2 Line on page 291

E1-R2 Advanced
Navigation: Line | E1R2 Advanced
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Zero Suppression Default = HDB3
Selects the method of zero suppression used (HDB3 or AMI).
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from
this line. Preference should always be given to using the clock source from a central
office exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network source
not be available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
Line Signaling Default = CPE
The options are:
• CPE
• CO
• CO
The feature is intended to be used primarily as a testing aid. It allows T1 and E1 lines
to be tested in a back-to-back configuration, using crossover (QSIG) cables.
The CO feature operates by modifying the way in which incoming calls are
disconnected for system configuration in Brazil and Argentina. In these locales, the
CO setting uses Forced-Release instead of Clear-Back to disconnect incoming calls.
The Brazilian Double-Seizure mechanism used to police Collect calls, is also disabled
in CO mode.
Incoming Routing Default = 4
Digits
Sets the number of incoming digits used for incoming call routing.
CRC Checking Default = On
Switches CRC on or off.
Default All Group Default the MFC Group tab settings.
Settings
Line Signaling Timers To edit one of these timers, either double-click on the timer or right-click on a timer
and select the action required.

Related links
E1 R2 Line on page 291

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 Line

T1 Line
Related links
PRI Trunks on page 281
US T1 Line on page 297
T1 Channels on page 299

US T1 Line
Navigation: Line | US T1 Line
The following settings are mergeable:
• Admin
• Prefix
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Line Number Allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device
providing the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control
unit from left to right. Expansion modules are numbered from 5 upwards,
for example trunks on the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration
settings relate.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict
Network Interconnect is enabled. It allows the trunk to be set as either
Public or Private. The system will return number busy indication to any
attempt to connect a call on a Private trunk to a Public trunk or vice versa.
This restriction includes transfers, forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems
also using any of the following other system features: multi-site networks,
VPNremote, application telecommuter mode.
Line Sub Type Default = T1
Set to T1 for a T1 line.
Channel Allocation Default = 24 | 1
The order, 24 to 1 or 1 to 24, in which channels are used.
Prefix Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is
useful if all users must dial a prefix to access an outside line. The prefix is
automatically placed in front of all incoming numbers so that users can dial
the number back.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Framing Default = ESF
Selects the type of signal framing used. The options are:
• ESF
• D4
Zero Suppression Default = B8ZS
Selects the method of zero suppression used. The options are:
• B8ZS
• AMI ZCS
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets
whether the system should try to take its clock source for call
synchronization and signalling from this line. Preference should always be
given to using the clock source from a central office exchange if available
by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are
used is described in the IP Office Installation Manual. If additional lines
are available, Fallback can be used to specify a clock source to use
should the Network source not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz
clock source.
• In scenarios where several systems are network via digital trunk lines,
care must be taken to ensure that all the systems use the same clock
source. The current source being used by a system is reported within the
System Status Application.
Haul Length Default = 0-115 feet.
Sets the line length to a specific distance.
Channel Unit Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided
by the Central Office. The options are:
• Foreign Exchange
• Special Access
• Normal
CRC Checking Default = On
Turns CRC on or off.
Table continues…

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 Line

Field Description
Line Signaling Default = CPE
This field affects T1 channels set to Loop-Start or Ground-Start. The field
can be set to either CPE (Customer Premises Equipment) or CO (Central
Office). This field should normally be left at its default of CPE. The setting
CO is normally only used in lab back-to-back testing.
Incoming Routing Digits Default=0 (present call immediately)
Sets the number of routing digits expected on incoming calls. This allows
the line to present the call to the system once the expected digits have
been received rather than waiting for the digits timeout to expire. This field
only affects T1 line channels set to E&M Tie, E&M DID, E&M Switched 56K
and Direct Inward Dial.
CSU Operation Enable this field to enable the T1 line to respond to loop-back requests
from the line.
Enhanced Called Party Number Default = Off
This option is not supported for systems set to the United States locale.
Normally the dialed number length is limited to 15 digits. Selecting this
option increases the allowed dialed number length to 30 digits.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for
maintenance or if the trunk is not connected.

Related links
T1 Line on page 297

T1 Channels
Navigation: Line | T1 Channels
The settings for each channel can be edited. Users have the option of editing individual channels
by double-clicking on the channel or selecting and editing multiple channels at the same time.
Note that the Line Appearance ID cannot be updated when editing multiple channels.
When editing a channel or channels, the settings available are displayed on two sub-tabs; T1 Edit
Channel and Timers.
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to these settings require a system reboot.

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Configuration Mode Field Descriptions

Field Description
Channel Allocated by the system.
Incoming Group Default = 0, Range 0 to 99999.
ID
The Incoming Group ID to which a line belongs is used to match it to incoming call routes
in the system configuration. The matching incoming call route is then used to route
incoming calls. The same ID can be used for multiple lines.
Outgoing Group Default = 1. Range 0 to 99999.
ID
Short codes that specify a number to dial can specify the line group to be used. The
system will then seize a line from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network. The same ID cannot be used in the configuration of
any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office lines to
the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office lines from
the primary and secondary servers to each expansion system in the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for the SM
line.
Line Appearance Default = Auto-assigned. Range = 2 to 9 digits.
ID
Used for configuring Line Appearances with button programming. The line appearance ID
must be unique and not match any extension number. Line appearance is not supported
for trunks set to QSIG operation and is not recommended for trunks be used for DID.
Direction Default = Bothway
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Bothway
Bearer Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
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Field Description
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Type Default = Loop-Start.
The T1 emulates the following connections:
• Ground-Start
• Loop-Start
• E&M - TIE
• E&M - DID
• E&M Switched 56K
• Direct Inward Dial
• Clear Channel 64K
Trunks set to E&M - DID will only accept incoming calls.
If E&M - TIE is selected and the Outgoing Trunk Type is set to Automatic, no secondary
dial tone is provided for outgoing calls on this line/trunk.
Dial Type Default = DTMF Dial
Select the dialing method required. The options are:
• DTMF Dial
• Pulse Dial
Incoming Trunk Default = Wink-Start
Type
Used for E&M types only. The handshake method for incoming calls. The options are:
• Automatic
• Immediate
• Delay Dial
• Wink-Start
Outgoing Trunk Default = Wink-Start
Type
Used for E&M types only. The handshake method for outgoing calls. The options are:
• Automatic
• Immediate
• Delay Dial
• Wink-Start
If the line Type is set to E&M-TIE and the Outgoing Trunk Type is set to Automatic, no
secondary dial tone is provided for outgoing calls on this line/trunk.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Tx Gain Default = 0dB.
The transmit gain in dB.
Rx Gain Default = 0dB.
The receive gain in dB.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.

Timer Settings
This sub-tab allows various timers relating to operation of an individual channel to be adjusted.
These should only be adjusted to match the requirements of the line provider. The following is a
list of the default values. To reset a value, click on the current value and then right click and select
from the default, minimize and maximize options displayed.
Incoming Automatic Delay: 410. Silent Interval: 1100.
Incoming Wink Delay: 100. Outgoing Seizure: 10.
Wink Signal: 200. Wink Start: 5000.
Incoming Dial Guard: 50. Wink Validated: 80.
First Incoming Digit: 15000. Wink End: 350.
Incoming Inter Digit: 5000. Delay End: 5000.
Maximum Inter Digit: 300. Outgoing Dial Guard: 590.
Flash Hook Detect: 240. Outgoing IMM Dial Guard: 1500.
Incoming Disconnect: 300. Outgoing Pulse Dial Break: 60.
Incoming Disconnect Guard: 800. Outgoing Pulse Dial Make: 40.
Disconnected Signal Error: 240000. Outgoing Pulse Dial Inter Digit: 720.
Outgoing Disconnect: 300. Outgoing Pulse Dial Pause: 1500.
Outgoing Disconnect Guard: 800. Flash Hook Generation: 500.
Ring Verify Duration: 220. Outgoing End of Dial: 1000.
Ring Abandon: 6300. Answer Supervision: 300.
Ping Verify: 600. Incoming Confirm: 20.
Long Ring Time: 1100.

Related links
T1 Line on page 297

T1 PRI Line
Related links
PRI Trunks on page 281

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 Line

T1 ISDN on page 303

T1 ISDN Channels on page 307
T1 ISDN TNS on page 309
T1 ISDN Special on page 310
Call By Call (US PRI) on page 311

T1 ISDN
Navigation: Line | T1 ISDN Line
The following settings are mergeable:
• Prefix
• Send Redirecting Number
• Admin
• Send original calling party for forwarded and twinning calls
• Originator number for forwarded and twinning calls
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Variable Description
Line Number Allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device
providing the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit
from left to right. Expansion modules are numbered from 5 upwards, for
example trunks on the module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings
relate.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or
Private. The system will return number busy indication to any attempt to connect
a call on a Private trunk to a Public trunk or vice versa. This restriction includes
transfers, forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks,
VPNremote, application telecommuter mode.
Line Sub Type : Default = PRI
Set to PRI. If set to T1 see Line Form (T1). If set to ETSI, ETSI CHI, QSIG A or
QSIG B see Line (E1).
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
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Configuration Mode Field Descriptions

Variable Description
Channel Allocation Default = 23 | 1
The order, 23 to 1 or 1 to 23, in which channels are used.
Switch Type Default = NI2
The options are
• 4ESS
• 5ESS
• DMS100
• NI2
Provider Default = Local Telco
Select the PSTN service provider (AT&T, Sprint, WorldCom or Local Telco).
Prefix Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if
all users must dial a prefix to access an outside line. The prefix is automatically
placed in front of all incoming numbers so that users can dial the number back.
Add 'Not-end-to-end ISDN' Default = Never*.
Information Element
Sets whether the optional 'Not end-to-end ISDN' information element should be
added to outgoing calls on the line. The options are
• Never
• Always
• POTS (only if the call was originated by an analog extension)
*The default is Never except for the following locales; for Italy the default is
POTS, for New Zealand the default is Always.
Progress Replacement Default = None.
Progress messages are defined in the Q.931 ISDN connection control signaling
protocol. Generally, If a progress message is sent, the caller does not get
connected and so typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to
configure alternative signaling to the ISDN line for internally generated Progress
messages. The options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not
hear the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically
will accrue call costs.
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Variable Description
Send Redirecting Number Default = Off.
This option can be used on ISDN trunks where the redirecting service is
supported by the trunk provider. Where supported, on twinned calls the caller ID
of the original call is passed through to the twinning destination. This option is
only used for twinned calls.
Send Names This option is available when the Switch Type above is set to DMS100. If set,
names are sent in the display field. The Z shortcode character can be used to
specify the name to be used.
Names Length Set the allowable length for names, up to 15 characters, when Send Names is
set above.
Test Number Used to remember the external telephone number of this line to assist with loop-
back testing. For information only.
Framing Default = ESF
Selects the type of signal framing used (ESF or D4).
Zero Suppression Default = B8ZS
Selects the method of zero suppression used (B8ZS or AMI ZCS).
Clock Quality Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether
the system should try to take its clock source for call synchronization and
signalling from this line. Preference should always be given to using the clock
source from a central office exchange if available by setting at least one
exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network
source not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock
source.
• In scenarios where several systems are network via digital trunk lines, care
must be taken to ensure that all the systems use the same clock source. The
current source being used by a system is reported within the System Status
Application.
CSU Operation Tick this field to enable the T1 line to respond to loop-back requests from the
line.
Haul Length Default = 0-115 feet
Sets the line length to a specific distance.
Table continues…

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Configuration Mode Field Descriptions

Variable Description
Channel Unit Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided by
the Central Office. The options are
• Foreign Exchange
• Special Access
• Normal
CRC Checking Default = On
Turns CRC on or off.
Line Signaling The field can be set to either CPE (Customer Premises Equipment) or CO
(Central Office). This field should normally be left at its default of CPE. The
setting CO is normally only used in lab back-to-back testing.
Incoming Routing Digits Default=0 (present call immediately)
Sets the number of routing digits expected on incoming calls. This allows the line
to present the call to the system once the expected digits have been received
rather than waiting for the digits timeout to expire. This field only affects T1 line
channels set to E&M Tie, E&M DID, E&M Switched 56K and Direct Inward
Dial.
Admin Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or
if the trunk is not connected.
Send original calling party Default = Off.
for forwarded and
Use the original calling party ID when forwarding calls or routing twinned calls.
twinning calls
This setting applies to the following ISDN lines:
• PRI24 with subtypes:
- PRI
- QSIGA
- QSIGB
- ETSI
- ETSI CHI
• PRI30 with subtypes
- QSIGA
- QSIGB
- ETSI
- ETSI CHI
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 Line

Variable Description
Originator number for Default = blank.
forwarded and twinning
The number used as the calling party ID when forwarding calls or routing twinned
calls
calls. This field is grayed out when the Send original calling party for
forwarded and twinning calls setting is enabled.
This setting applies to the following ISDN lines:
• PRI24 with subtypes:
- PRI
- QSIGA
- QSIGB
- ETSI
- ETSI CHI
• PRI30 with subtypes
- QSIGA
- QSIGB
- ETSI
- ETSI CHI

Related links
T1 PRI Line on page 302

T1 ISDN Channels
Navigation: Line | T1 ISDN Channels
This tab allows settings for individual channels within the trunk to be adjusted. This tab is not
available for trunks sets to ETSI or QSIG mode.
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field Description
Channel Allocated by the system.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used. The
system will then seize a line from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network. The same ID cannot be used in the configuration of
any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office lines
to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
Line Appearance ID Default = Auto-assigned. Range = 2 to 9 digits.
Used for configuring Line Appearances with button programming. The line appearance
ID must be unique and not match any extension number.
Direction Default = Both Directions
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Both Directions
Bearer Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
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 Line

Field Description
Service Default = None.
If the line provider is set to AT&T, select the type of service provided by the channel.
The options are:
• Call by Call
• SDN (inc GSDN)
• MegaCom 800
• MegaCom
• Wats
• Accunet
• ILDS
• I800
• ETN
• Private Line
• AT&T Multiquest
For other providers, the service options are None or No Service.
Admin Default = Out of Service
Used to indicate the channel status. The options are:
• In Service
• Out of Service
• Maintenance
Tx Gain Default = 0dB
The transmit gain in dB
Rx Gain Default = 0dB
The receive gain in dB.

Related links
T1 PRI Line on page 302

T1 ISDN TNS
Navigation: Line | T1 ISDN TNS
This tab is shown when the line Provider is set to AT&T. It allows the entry of the Network
Selection settings. These are prefixes for alternative long distance carriers. When a number dialed
matches an entry in the table, that pattern is stripped from the number before being sent out. This
table is used to set field in the TNS (Transit Network Selection) information element for 4ESS and
5ESS exchanges. It is also used to set fields in the NSF information element.
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
TNS Code The pattern for the alternate long distance carrier. For example: The pattern 10XXX is
added to this tab. If 10288 is dialed, 10 is removed and 288 is placed in the TNS and
NSF information.

Related links
T1 PRI Line on page 302

T1 ISDN Special
Navigation: Line | T1 ISDN Special
This tab is shown when the line Provider is set to AT&T. This table is used to set additional fields
in the NSF information element after initial number parsing by the TNS tab. These are used to
indicate the services required by the call. If the channel is set to Call by Call, then further parsing
is done using the records in the Call by Call tab.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Short code The number which results from the application of the rules specified in the User or
System Short code tables and the Network Selection table and the Call-by-call table to
the number dialed by the user.
Number The number to be dialed to line.
Special Default = No Operator.
The options are:
• No Operator
• Local Operator
• Presubscribed Operator
Plan Default = National.
The options are:
• National
• International

Typical values are:

Short Code Number Service
011N N No Operator, International
010N N Local Operator, International
01N N Local Operator, National
00N N Presubscribed Operator, National
0N N Presubscribed Operator, National
1N 1N No operator, National

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 Line

Related links
T1 PRI Line on page 302

Call By Call (US PRI)

Navigation: Line | T1 ISDN Call by Call
This tab is shown when the line Provider is set to AT&T. Settings in this tab are only used when
calls are routed via a channel which has its Service set to Call by Call.
It allows short codes to be created to route calls to a different services according to the number
dialed. Call By Call reduces the costs and maximizes the use of facilities. Call By Call chooses the
optimal service for a particular call by including the Bearer capability in the routing decision. This is
particularly useful when there are limited resources.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Short Code The number which results from the application of the rules specified in the User or
System Short code tables and the Network Selection table to the number dialed by the
user.
Number The number to be dialed to line.
Bearer Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Service Default = AT&T
The service required by the call. The options are:
• Call by Call
• SDN (inc GSDN)
• MegaCom 800
• MegaCom
• Wats
• Accunet
• ILDS
• I800
• ETN
• Private Line
• AT&T Multiquest

Related links
T1 PRI Line on page 302

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Configuration Mode Field Descriptions

S0 Line

These settings are used for S0 ports provided by an S08 expansion module connected a control
unit. For full details of installation refer to the IP Office Installation manual.
Though displayed as lines, these BRI ports are used for connection of ISDN2 devices such as
video conferencing units or ISDN PC cards.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
Related links
Line on page 266
S0 Line on page 312
S0 Short Codes on page 314
Line | S0 Channels on page 315

S0 Line
Navigation: Line | S0 Line
The following settings are not mergeable. Changes to these settings require a system reboot.
• Line Sub Type
• Network Type
The remaining settings are mergeable.
Field Description
Line Number This parameter is not configurable. It is allocated by the system.
Card/Module Indicates the card slot or expansion module being used for the trunk device providing
the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from left
to right. Expansion modules are numbered from 5 upwards, for example trunks on the
module in Expansion Port 1 are shown as 5.
Port Indicates the port on the Card/Module above to which the configuration settings relate.
Line Sub Type Default = ETSI Select to match the particular line type provided by the line provider.
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 Line

Field Description
Network Type Default = Public.
This option is available if Restrict Network Interconnect (System | Telephony |
Telephony) is enabled. It allows the trunk to be set as either Public or Private. The
system will return number busy indication to any attempt to connect a call on a Private
trunk to a Public trunk or vice versa. This restriction includes transfers, forwarding and
conference calls.
Due to the nature of this feature, its use is not recommended on systems also using any
of the following other system features: multi-site networks, VPNremote, application
telecommuter mode.
Telephone Number Used to remember the telephone number of this line. For information only.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the Prefix
field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable for
external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number is
presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a number
is presented from ISDN as an "international number" this prefix is added. For example
441923000000 is converted to 00441923000000.
Incoming Group ID Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used. The
system will then seize a line from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network. The same ID cannot be used in the configuration of
any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office lines
to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
TEI Default = 0
Not used. The Control Unit will ignore any entry.
Number of Default = 2
Channels
Defines the number of operational channels that are available on this line. 2 for BRI and
up to 30 for PRI - depending upon the number of channels subscribed.
Outgoing Channels Default = 2
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
Voice Channels Default = 2
The number of channels available for voice use.
Data Channels Default = 2
The number of channels available for data use. If left blank the value is 0.

Related links
S0 Line on page 312

S0 Short Codes
Navigation: Line | S0 Line | Short Codes
For BRI S0 lines , these settings are mergeable.
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those

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 Line

types of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Related links
S0 Line on page 312

Line | S0 Channels
Navigation: Line | S0 Line | Channels
For S0 channels this form is not used.
Related links
S0 Line on page 312

H.323 Line
These lines are added manually. They allow voice calls to be routed over data links within the
system. They are therefore dependent on the IP data routing between the system and the
destination having being configured and tested.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
Network Assessments
Not all data connections are suitable for voice traffic. A network assessment is required for internal
network connections. For external network connections a service level agreement is required from
the service provider. Avaya cannot control or be held accountable for the suitability of a data
connection for carrying voice traffic.
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice Networking
licenses.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Related links
Line on page 266
VoIP Line on page 316
Short Codes on page 318
VoIP Settings on page 318

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Configuration Mode Field Descriptions

VoIP Line
Navigation: Line | H.323 Line | VoIP Line
Configuration Settings
These settings are mergeable. Changes to these settings does not require a reboot of the system.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
Telephone Number Used to remember the telephone number of this line. For information only.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number is
presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added.
For example 441923000000 is converted to 00441923000000.
Table continues…

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 Line

Field Description
Location Default = Cloud.
Specify a location to associate the extension with a physical location. Associating
an extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined on Location |
Location.
Description Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Send original calling Default = Off.
party for forwarded and
Use the original calling party ID when forwarding calls or routing twinned calls.
twinning calls
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used. The
system will then seize a line from those available with a matching Outgoing Group
ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a
system must be unique within the network. The same ID cannot be used in the
configuration of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office
lines to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office
lines from the primary and secondary servers to each expansion system in the
network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for
the SM line.
Number of Channels Default = 20, Range 1 to 250.
Defines the number of operational channels that are available on this line.
Outgoing Channels Default = 20, Range 0 to 250.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
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Configuration Mode Field Descriptions

Field Description
TEI Default = 0. Range = 0 to 127.
The Terminal Equipment Identifier. Used to identify each Control Unit connected to a
particular ISDN line. For Point to Point lines this is typically (always) 0. It can also
be 0 on a Point to Multi-Point line, however if multiple devices are actually sharing a
Point to Multi-Point line it should be set to 127 which will result in the exchange
deciding on the TEI's to be used by this Control Unit.

Related links
H.323 Line on page 315

Short Codes
Navigation: Line | H.323 Line | Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those
types of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
Changes to these settings do not require a reboot of the system.
Related links
H.323 Line on page 315

VoIP Settings
Navigation: Line | H.323 Line | VoIP Settings
This form is used to configure the VoIP setting applied to calls on the H.323 line.
Configuration Settings
These settings are mergeable. Changes to these settings does not require a reboot of the system.
Field Description
Gateway IP Address Default = Blank
Enter the IP address of the gateway device at the remote end.
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 Line

Field Description
Port Default = 1720
The H.323 line is identified by the IP Address:Port value. Specifying a unique port
value for this IP address allows multiple lines to use the same IP address.
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs varies to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
Within a network of systems, it is strongly recommended that all the systems and
the lines connecting those systems use the same codecs.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Supplementary Default = H450.
Services
Selects the supplementary service signaling method for use across the H.323 trunk.
The remote end of the trunk must support the same option. The options are:
• None: No supplementary services are supported.
• H450: Use for H.323 lines connected to another PBX or device that uses H450.
• QSIG: Use for H.323 lines connected to another PBX or device that uses QSIG.
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
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Configuration Mode Field Descriptions

Field Description
VoIP Silence Default = Off.
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP
lines using G.711 between systems. On trunk's between networked systems, the
same setting should be set at both ends.
Enable FastStart for Default = Off
non-Avaya IP Phones
A fast connection procedure. Reduces the number of messages that need to be
exchanged before an audio channel is created.
Fax Transport Support Default = Off
This option is only supported on trunks with their Supplementary Services set to
IP Office SCN or IP Office Small Community Network - Fallback. Fax relay is
supported across H.323 multi-site network lines with Fax Transport Support
selected. This will use 2 VCM channels in each of the systems. Fax relay is only
supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition Linux servers.
Local Tones Default = Off
When selected, the tones are generated by the local system to which the phone is
registered. This option should not be used with lines being used for a multi-site
network.
DTMF Support Default = Out of Band
DTMF tones can be sent to the remote end either as DTMF tones within the calls
audio path (In Band) or a separate signals (Out of Band). Out of Band is
recommended for compression modes such as G.729 and G.723 compression
modes where DTMF in the voice stream could become distorted.
Allow Direct Media Path Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the
need for system resources such as voice compression channels. Both ends of the
calls must support Direct Media and have compatible VoIP settings such as
matching codec, etc. If otherwise, the call will remain routed via the system.
Enabling this option may cause some vendors problems with changing the media
path mid call.
• If disabled, the call is routed via the system. In that case, RTP relay support may
still allow calls between devices using the same audio codec to not require a
voice compression channel.
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 Line

Field Description
Progress Ends Overlap Default = Off.
Send
Some telephony equipment, primarily AT&T switches, over IP trunks send a H.323
Progress rather than H.323 Proceeding message to signal that they have
recognized the digits sent in overlap state. By default the system expects an H.323
Proceeding message. This option is not available by default. If required, the value
ProgressEndsOverlapSend must be entered into the Source Numbers tab of the
NoUser user.
Default Name From Default = Off.
Display IE
When set, the Display IE is used as the default source for the name.

Related links
H.323 Line on page 315

IP DECT Line
This type of line can be manually added. They are used to route voice calls over an IP data
connection to an Avaya IP DECT system. Only one IP DECT line can be added to a system. Refer
to the IP DECT R4 Installation manual for full details.
Currently, only one IP DECT line is supported on a system.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Related links
Line on page 266
Line | IP DECT Line on page 321
Gateway on page 322
VoIP on page 324

Line | IP DECT Line

Navigation: Line | IP DECT Line | Line
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. Changing an IP DECT line that has been imported into the configuration is
not mergeable.
Field Description
Line Number This number is allocated by the system and is not adjustable.
Associated Extensions Lists all the DECT extensions associated with the IP DECT line.
Description Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.

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Configuration Mode Field Descriptions

Related links
IP DECT Line on page 321

Gateway
Navigation: Line | IP DECT Line | Gateway
This form is used to configure aspects of information exchange between the IP Office and IP
DECT systems.
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. Changing an IP DECT line that has been imported into the configuration is
not mergeable.
Field Description
Auto-Create Default = Off.
Extension
If enabled, subscription of a handset with the DECT system causes the auto-creation of a
matching numbered extension within the system configuration if one does not already
exist. This setting is not supported on systems configured to use WebLM server licensing.
For security, auto-create is automatically disabled after 24 hours.
Auto-Create User Default = Off.
This option is only usable if Auto-Create Extension is also enabled. If enabled,
subscription of a handset with the DECT system causes the auto-creation of a matching
user within the system configuration if one does not already exist.
For security, any auto-create settings set to On are automatically set to Off after 24 hours.
Enable DHCP Default = Off
Support
This option is not supported for use with Avaya IP DECT R4. The IP DECT base stations
require DHCP and TFTP support. Enable this option if the system is being used to
provide that support, using IP addresses from its DHCP range (LAN1 or LAN2) and its
TFTP server setting. If not enabled, alternate DHCP and TFTP options must be provided
during the IP DECT installation.
• If it is desired to use the system for DHCP support of the ADMM and IP DECT base
stations only, the system address range should be set to match that number of
addresses. Those addresses are then taken during the system restart and will not be
available for other DHCP responses following the restart.
• For larger IP DECT installations, the use of a non-embedded TFTP software option
other than Manager is recommended.
Boot File Default = ADMM_RFP_1_0_0.tftp. Range = Up to 31 characters.
The name and path of the ADMM software file. The path is relative to the TFTP server
root directory.
ADMM MAC Default = 00:00:00:00:00:00
Address
This field must be used to indicate the MAC address of the IP DECT base station that
should load the ADMM software file and then act as the IP DECT system's ADMM. The
address is entered in hexadecimal format using comma, dash, colon or period separators.
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 Line

Field Description
VLAN ID Default = Blank. Range = 0 to 4095.
If VLAN is being used by the IP DECT network, this field sets the VLAN address assigned
to the base stations by the system if Enable DHCP Support is selected.
• The system itself does not apply or use VLAN marking. It is assumed that the addition
of VLAN marking and routing of VLAN traffic is performed by other switches within the
customer network.
• An ID of zero is not recommended for normal VLAN operation.
• When blank, no VLAN option is sent to the IP DECT base station.
Base Station Default = Empty
Address List
This box is used to list the MAC addresses of the IP DECT base stations, other than the
base station being used as the ADMM and entered in the ADMM MAC Address field.
Right-click on the list to select Add or Delete. or use the Insert and Delete keys. The
addresses are entered in hexadecimal format using comma, dash, colon or period
separators.
Enable Provisioning
This option can be used with DECT R4 systems. It allows the setting of several values in the system
configuration that previously needed to be set separately in the master base stations configuration. For full
details refer to the DECT R4 Installation manual. The use of provisioning requires the system security settings
to include an IPDECT Group.
SARI/PARK Default = 0
Enter the PARK (Portable Access Rights Key) license key of the DECT R4 system. DECT
handset users enter this key when subscribing to the DECT system.
Subscriptions Default = Disabled
Select the method of subscription supported for handsets subscribing to the DECT R4
system. The options are:
• Disabled:Disables subscription of handsets.
• Auto-Create: Allow anonymous subscription of handsets. Once subscribed, the
handset is assigned a temporary extension number. That extension number can be
confirmed by dialing *#. A new extension number can be specified by dialing
<Extension Number>*<Login Code>#. The Auto-Create Extension and Auto-Create
User settings above should also be enabled. While configured to this mode, Manager
will not allow the manual addition of new IP DECT extensions.
• Preconfigured: Allow subscription only against existing IP DECT extensions records in
the system configuration. The handset IPEI number is used to match the subscribing
handset to a system extension.
Authentication Default = Blank.
Code
Set an authentication code that DECT handset users should enter when subscribing to
the DECT system.

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Configuration Mode Field Descriptions

Field Description
Enable Resiliency
Default = Off.
Enables resiliency on the IP DECT Line. To configure resiliency, you must also configure an IP Office Line with
Backs up my IP Dect Phones set to On.
Status Enquiry Default = 30 seconds.
Period
The period between successive verifications on the H.323 channel. The smaller the
interval, the faster the IP DECT system recognizes that IP Office is down.
Prioritize Primary Default = Off.
Only available when Enable Provisioning is set to On.
Set to On for automatic fail-over recovery. When on, the IP DECT system switches
automatically from the backup IP Office to the "primary" IP Office.
Note that the IP DECT system does not switch back automatically from the backup IP
Office to the primary. The IP DECT system must be manually switched using Web
Manager.
Supervision Default = 120 seconds.
Timeout
Only available when Enable Provisioning is set to On.
The period of time the IP DECT system will wait between attempts to switch from the
backup IP Office to its "primary" IP Office.

Related links
IP DECT Line on page 321

VoIP
Navigation: Line | IP DECT Line | VoIP
Used to configure the VoIP setting applied to calls on the IP DECT line.
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. Changing an IP DECT line that has been imported into the configuration is
not mergeable.
Field Description
Gateway IP Address Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must
not be shared by any other IP line (H.323, SIP, SES or IP DECT).
Standby IP Address Default = Blank.
IP Address of the Standby Master IP Base Station or the second Mirror Base
Station. When the primary Mirror Base Station or Master Base Station is offline the
second Mirror or the Standby Master will take over and the system will use this IP
address.
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 Line

Field Description
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs varies to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
Within a network of systems, it is strongly recommended that all the systems and
the lines connecting those systems use the same codecs.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs.
TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
VoIP Silence Default = Off.
Suppression
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems,
the same setting should be set at both ends.
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Configuration Mode Field Descriptions

Field Description
Allow Direct Media Path Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the
need for system resources such as voice compression channels. Both ends of
the calls must support Direct Media and have compatible VoIP settings such as
matching codec, etc. If otherwise, the call will remain routed via the system.
Enabling this option may cause some vendors problems with changing the
media path mid call.
• If disabled, the call is routed via the system. In that case, RTP relay support may
still allow calls between devices using the same audio codec to not require a
voice compression channel.

Related links
IP DECT Line on page 321

SIP Line
IP Office supports SIP voice calls through the addition of SIP lines to the system configuration.
This approach allows users with non-SIP phones to make and receive SIP calls.
Deleting a SIP line requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Related links
Line on page 266
SIP Line on page 326
Transport on page 330
Call Details on page 335
VoIP on page 347
T38 Fax on page 352
SIP Credentials on page 353
SIP Advanced on page 354
Engineering on page 359

SIP Line
Navigation: Line | SIP Line | SIP Line

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 Line

Configuration Settings
These settings are mergeable with the exception of the Line Number setting. Changing the Line
Number setting requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
ITSP Domain Name Default = Blank.
This field is used to specify the default host part of the SIP URI in the From, To,
and R-URI fields for outgoing calls. For example, in the SIP URI
[email protected], the host part of the URI is example.com. When empty, the
default host is provided by the SIP Line | SIP Transport | ITSP Proxy Address
field value. If multiple addresses are defined in the ITSP Proxy Address field, then
this field must be defined.
For the user making the call, the user part of the From SIP URI is determined by
the settings of the SIP URI channel record being used to route the call (see SIP
Line | SIP URI | Local URI). This will use one of the following:
• a specific name entered in Local URI field of the channel record.
• or specify using the primary or secondary authentication name set for the line
below.
• or specify using the SIP Name set for the user making the call (User | SIP | SIP
Name).
For the destination of the call, the user part of the To and R-URI fields are
determined by dial short codes of the form 9N/N"@example.com” where N is the
user part of the SIP URI and "@example.com" is optional and can be used to
override the host part of the To and R-URI.
Local Domain Name Default = Blank.
An IP address or SIP domain name as required by the service provider.
When configured, the Local Domain Name value is used in
• the From and Contact headers
• the PAI header, when the setting Line | SIP LIne | Advanced | Use Domain for
PAI is checked
• the Diversion header
If both the ITSP Domain Name and the Local Domain Name are configured,
then Local Domain takes precedence.
Local Domain Name is not used in the Remote Party ID header.
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Configuration Mode Field Descriptions

Field Description
URI Default = SIP.
When SIP or SIPS is selected in the drop-down box, the SIP URI format is used
(for example, [email protected]).
When Tel is selected in the drop-down box, the Tel URI format is used (for
example, +1-425-555-4567). This affects the From field of outgoing calls. The To
field for outgoing calls will always use the format specified by the short codes used
for outgoing call routing. Recommendation: When SIP Secured URI is required,
the URI Type should be set to SIPS. SIPS can be used only when Layer 4 Protocol
is set to TLS.
Location Default = Cloud.
Specify a location to associate the line with a physical location. Associating a line
with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined in the System |
Location form.
Prefix Default = Blank.
This prefix is removed from the called number on outgoing calls if present.
National Prefix Default = 0.
This prefix is added to calls identified as not being international.
International Prefix Default = 00.
This prefix is added to calls identified as not being national.
Country Code Default = Blank.
Set to match the local country code of the system location.
Name Priority Default = System Default.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting
determines which method is used by the line. The options are:
• System Default: Use the system setting System | Telephony | Telephony |
Default Name Priority.
• Favor Trunk: Display the name provided by the trunk. For example, the trunk
may be configured to provide the calling number or the name of the caller. The
system should display the caller information as it is provided by the trunk. If the
trunk does not provide a name, the system uses the Favor Directory method.
• Favor Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
the name provided by the SIP line. If no match is found, the name provided by
the line, if any, is used.
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 Line

Field Description
Description Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
In Service Default = On.
When this field is not selected, the SIP trunk is unregistered and not available to
incoming and outgoing calls.
Check OOS Default = On.
If enabled, the system will regularly check if the trunk is in service using the
methods listed below. Checking that SIP trunks are in service ensures that
outgoing call routing is not delayed waiting for response on a SIP trunk that is not
currently usable.
For UDP and TCP trunks, OPTIONS message are regularly sent. If no reply to an
OPTIONS message is received the trunk is taken out of service.
For trunks using DNS, if the IP address is not resolved or the DNS resolution has
expired, the trunk is taken out of service.
Session Timers
Refresh Method Default = Auto.
The options are:
• Auto
• Reinvite
• Update
When Auto is selected, if UPDATE is in the Allow: header from the far SIP
endpoint, then it is used. Otherwise INVITE is used.
Timer (seconds) Default = On Demand. Range = 90 to 64800
This field specifies the session expiry time. At the half way point of the expiry time,
a session refresh message is sent. When set to On Demand, IP Office will not
send a session refresh message but will respond to them.

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Configuration Mode Field Descriptions

Field Description
Redirect and Transfer
Redirection and blind transfer are configured separately. By default, they are disabled.
A supervised transfer occurs when a consultation call is made and the REFER contains a Replaces: header
indicating the CallID of another call leg which the REFERing agent has already initiated with the REFER target.

Note:
Do not change these settings unless directed to by the SIP service provider.
Incoming Supervised Default = Auto.
REFER
Determines if IP Office will accept a REFER being sent by the far end. The options
are:
• Always: Always accepted.
• Auto: If the far end does not advertise REFER support in the Allow: header of
the OPTIONS responses, then IP Office will reject a REFER from that endpoint.
• Never: Never accepted.
Outgoing Supervised Default = Auto.
REFER
Determines if IP Office will attempt to use the REFER mechanism to transfer a
party to a call leg which IP Office has already initiated so that it can include the
CallID in a Replaces: header. The options are:
• Always: Always use REFER.
• Auto: Use the Allow: header of the OPTIONS response to determine if the
endpoint supports REFER.
• Never: Never use REFER.
Send 302 Moved Default = Off.
Temporarily
A SIP response code used for redirecting an unanswered incoming call. It is a
response to the INVITE, and cannot be used after the 200 OK has been sent as a
response to the INVITE.
Outgoing Blind REFER Default = Off.
When enabled, a user, voicemail system or IVR can transfer a call by sending a
REFER to an endpoint that has not set up a second call. In this case, there is no
Replaces: header because there is no CallID to replace the current one. This
directs the far end to perform the transfer by initiating the new call and release the
current call with IP Office.

Related links
SIP Line on page 326

Transport
Navigation: Line | SIP Line | Transport

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 Line

Behavior during Service unavailable

A proxy server is considered Active once the system has received a response to an INVITE,
REGISTER or OPTIONS.
In the case of the proxy server responding with 503 - Service Unavailable, it should be considered
Active - In Maintenance. In this case, the following should occur:
If the response 503 - Service Unavailable was in response to an INVITE request:
• If calls are tied to registrations (Calls Route via Registrar enabled) and there are other
proxies available, the tied registrations should issue an Un-REGISTER and try to REGISTER
with a different proxy. The call should fail with cause = Temporary Fail.
• If calls are not tied, the INVITE should be immediately tried to a different proxy.
If the response 503 - Service Unavailable was in response to a REGISTER request:
• If there are other proxies available, this registration only should issue an Un-REGISTER and
try to REGISTER with a different proxy.
• If Explicit DNS Server(s) are configured, a DNS request should be sent out to see whether
the proxy server has disappeared from those being offered.
An Active-InMaintenance proxy server should not be used for a new transactions (INVITE or
REGISTER) until:
• There is a change in DNS responses indicating the proxy has become active.
• The configuration does not leave any better option available. In this case, there should be a
throttle so that no more than 5 failures (without successes) in 1 minute should be allowed.
• A configuration merge has occurred where the ITSP Proxy Address has been changed.
• 10 minutes has expired.
Behavior during Not Responding
A proxy server that is not-responding (UDP) is indicated when 3 requests are sent and no replies
are received. This would normally occur during a single INVITE transaction.
Consideration should be given whether this is caused by a local network fault or is caused by the
Proxy being out of service. Since it is likely to be local, no action should be taken unless traffic is
received from an alternative proxy while this proxy is actually not responding. The state should be
"Possibly non responding".
If explicit DNS servers are configured, a DNS request should be sent out to see whether this
Proxy server has disappeared from those being offered.
If possible, an alternative proxy should be stimulated simultaneously with stimulating the suspect
server.
The server should be considered non-responding if it is persistently non-responding while other
proxies are responding or if it is non-responding and has disappeared from the DNS
advertisement.
While in the "possibly not responding" state, it would be better to send an INVITE to an alternative
proxy while simultaneously sending any appropriate message to this proxy. This will help to
resolve whether it is really not responding rather than there being local network problems.
However, there is no requirement to blacklist the proxy.

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Configuration Mode Field Descriptions

Once in the "definitely not responding" state:

• If there are other proxies available: this registration only issues an Un-REGISTER, and try to
REGISTER with a different proxy. Calls do not automatically clear.
• If a SIP message is received from it, the state should immediately go"Active".
• This proxy should be blacklisted unless there are no better options available. While
blacklisted, only one transaction per 10 minutes is allowed.
• Even if not blacklisted, there should be a throttle so that no more than 5 failures (without
successes) in 1 minute should be allowed.
Configuration settings
The ITSP Proxy Address and Calls Route via Registrar settings are mergeable. Changing the
remaining settings requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.

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 Line

Field Description
ITSP Proxy Address Default = Blank
This is the SIP Proxy address used for outgoing SIP calls. The address can be
specified in the following ways:
• If left blank, the ITSP Domain Name is used and is resolved by DNS resolution in
the same way as if a DNS address had been specified as below.
• An IP address.
• A list of up to 4 IP addresses, with each address separated by a comma or space.
- The addresses can include an indication of the relative call weighting of each
address compared to the others. This is done by adding a w N suffix to the
address where N is the weighting value. For example, in the list 213.74.81.102w3
213.74.81.100w2, the weighting values assigns 1.5 times the weight of calls to the
first address. The default weight if not specified is 1. A weight of 0 can be used to
disable an address. Weight is only applied to outgoing calls.
If there is more than one proxy defined, and no weight indication, then calls are
only sent to the first in the list until there is a failure at which point the next proxy is
used.
- If the Calls Route via Registrar setting below is enabled, the weighting is applied
to registrations rather than calls.
• A DNS address, for example sbc.example.com.
- The DNS response may return multiple proxy addresses (RFC 3263). If that is the
case, the system will resolve the address to use based on priority, TTL and
weighting information included with each address.
- A load balancing suffix can be added to specify that multiple proxy results should
be returned if possible, for example sbc.example.com(N). where N is the required
number of addresses from 1 to 4.
This field is mergeable. However, no more than 4 IP Addresses should be in use at
any time. So, if the combined new and old address settings exceed 4, the new
addresses are only phased into use as transactions in progress on the previous
addresses are completed.
Network Configuration
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Configuration Mode Field Descriptions

Field Description
Layer 4 Protocol Default = UDP.
The options are:
• TCP
• UDP
• TLS
• Auto
TLS connections support the following ciphers:
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_WITH_AES_256_CBC_SHA
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA
• TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Use Network Default = None.
Topology Info
This field associates the SIP line with the LAN interface System | LAN | Network
Topology settings. It also applies the System | LAN | VoIP | DiffServ Settings to the
outgoing traffic on the SIP line. If None is selected, STUN lookup is not applied and
routing is determined by the system's routing tables.
If no STUN server address is set for the interface, then the System | LAN | Network
Topology | Binding Refresh Time is ignored by SIP Lines when calculating the
periodic OPTIONS timing unless the Firewall/NAT Type is set to Open Internet.
Send Port When Network Configuration is set to TLS, the default setting is 5061. When Network
Configuration is set to TCP or UDP, the default setting is 5060.
Listen Port When Network Configuration is set to TLS, the default setting is 5061. When Network
Configuration is set to TCP or UDP, the default setting is 5060.

Explicit DNS Default = 0.0.0.0 (Off)

Server(s)
If specific DNS servers should be used for SIP trunk operation rather than the general
DNS server specified or obtained for the system, the server addresses can be
specified here. If exported or imported as part of a trunk template.
Calls Route via Default = On
Registrar
If selected, all calls are routed via the same proxy as used for registration. If multiple
ITSP proxy addresses have been specified, there is no load balancing of registrations.
Separate Registrar Default = Blank
This field allows the SIP registrar address to be specified if it is different from that of
the SIP proxy. The address can be specified as an IP address or DNS name.

Related links
SIP Line on page 326

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 Line

Call Details
Navigation: Line | SIP Line | Call Details
SIP URI
Having set up the SIP trunk to the SIP ITSP, the SIP URI's registered with that ITSP are entered
on this tab. A SIP URI (Uniform Resource Identifier) is similar to an internet email address and
represents the source or destination for SIP connection. The URI consists of two parts, the user
part (name) and the host part (example.com).
SIP URI records are used to describe content of SIP headers when working with various ITSPs,
since they use these SIP headers for various purposes and match them to IP Office interpretation
of these headers. For incoming calls, headers in the SIP message must match the headers
described in the appropriate URI. For outgoing calls, IP Office internal displays and calling or
called numbers are mapped to appropriate headers in a way provider expects them to be
formatted.
If the Auto setting is used in the SIP trunk's Local URI, Contact and Display fields, that SIP trunk
will accept any incoming SIP call. The incoming call routing is still performed by the system
incoming call routes based on matching the values received with the call or the URI's incoming
group setting. On outgoing calls, Auto passes the calling and called party numbers from IP Office
call unchanged to the SIP provider. The Auto setting has replaced the wildcard * used in previous
releases.
For outgoing calls using this SIP URI, all valid short code CLI manipulations are used
(transforming calling party number to ISDN will be ignored). For a full list of valid CLI
manipulations, see “Telephone Number Field Characters” in the chapter “Short Code Overview”.
For example, character ‘i’ is not supported since it sets calling party number plan to isdn and
number type to national.
For the system, each SIP URI acts as a set of trunk channels. Outgoing calls can then be routed
to the required URI by short codes that match that URI's Outgoing Group setting. Incoming calls
can be routed by incoming call routes that match the URI's Incoming Group setting.
Note that the system supports only up to 150 URI records on a SIP line.
SIP Line Appearances
Using this feature, Line appearances can be configured on phones that support Line appearances.
Several Line appearances can be associated with one SIP Line, specified by an ITSP. The buttons
can be used to make or receive calls. ITSPs specify how many calls can be made or received
using a particular SIP number.
If a SIP number can be used to make “n” number of calls, the system allocates “n” system wide
unique line appearances for that number. All phones that wish to use the Line Appearance to
make or receive calls must have a line appearance button configured to point to that Line ID. A
SIP number can be accessed through the Line Appearance to make calls, answer calls, or
conference with other users already on the call. Lamps on the button indicate whether the number
is in use.
In case a sip number can be used to make or receive three calls, system can be configured to
associate that number with three Line Appearance IDs. For simplicity purposes, Line Appearance
IDs should be consecutive (for example 700, 701, 702…). The direction of call allocation can also
be configured. The preferred way of configuring for outgoing calls and incoming calls are reverse

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Configuration Mode Field Descriptions

of each other to avoid confusion. For example, if the outgoing calls are allocated from 700->702,
the incoming calls are allocated from 702->700.
SIP line appearances are supported on all phones that support line appearances. They are not
supported over SCN or in resiliency.
Related links
SIP Line on page 326
SIP URI on page 336
SIP Line Appearances on page 343

SIP URI
Name Description
URI This field is for information only and cannot be edited. It shows the IP address
of the system LAN interface with which the SIP trunk is associated.
Incoming Group Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming
call routes in the system configuration. The matching incoming call route is then
used to route incoming calls. The same ID can be used for multiple lines.
Outgoing Group Default = Previous external line + 1, Range 0 to 99999.
Short codes that specify a number to dial can also specify the line group to be
used. The system will then seize a line from those with the matching Outgoing
Group ID.
In a Server Edition network, the Outgoing Group ID used must be unique
within the network. The same ID cannot be used in the configuration of any
lines on another server system in the network. For non-Server Edition
deployments, the same ID can be used for multiple lines.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition network, reserved for the IP Office lines
to the Primary Server and Secondary Server respectively.
• 99901 to 99930 In a Server Edition network, reserved for the IP Office lines
from the Primary Server to each expansion system in the network.
• 0 In a Server Edition network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved
for the SM line.
Max Sessions Default =10 This field sets the maximum number of simultaneous calls that can
use the URI before the system returns busy to any further calls. For capacity
information, see Deploying Avaya IP Office™ Platform Server Edition.
Credentials Default = 0:<None>
This field is used to select from a list of the account credentials configured on
the line's SIP Credentials tab.
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 Line

Name Description
Local URI: This field sets the 'From' field for outgoing SIP calls using this URI.
• Display: This field sets the Name value for outgoing SIP calls using this URI.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this URI. The values you type in the Content field are also reflected in
the Display field.
• Outgoing Calls: Default = Caller. You can select any one of the following
options:
- Caller
- Explicit
- None
• Forwarding/Twinning: Default = Caller. This column sets the source used
for the system's SIP URIs with incoming calls that it then redirects as
outgoing calls. For example, calls it forwards or twins for users. You can
select any one of the following options:
- Caller: Use the name and number of the user or device forwarding the call.
- Explicit: Use the values manually typed in the Display and Content
columns.
- Original Caller: Use the name and number details of the original caller who
is now being forwarded. Note that use of this value form some headers
such as P-Asserted-ID and P-Preferred-ID may not be supported if the line
provider requires the caller information to be present in that header for
billing purposes.
• Incoming Calls: Default = Called. This column sets the source for the value
used to match incoming calls from the line provider. You can select any one
of the following options:
- Called: Use the automatically determined settings appropriate to the call
destination.
- Explicit:Use the values manually typed in the Display and Content
columns as intended destination.
- None: Do not send the header.
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Configuration Mode Field Descriptions

Name Description
Contact This field sets the From field for outgoing SIP calls using this URI.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this URI.
• Content: This configurable field sets the Content of SIP headers for outgoing
SIP calls using this URI. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Caller. You can select any one of the following
options:
- Caller
- Explicit
- None
• Forwarding/Twinning: Default = Caller. This field can be used to set an
originator number for forwarded and twinned calls. You can select any one of
the following options:
- Caller
- Explicit
- Original Caller
• Incoming Calls: Default = Called. You can select any one of the following
options:
- Called
- Explicit
- None
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Name Description
P. Asserted ID Default = Disabled
When selected, identity information is provided in the P Asserted Identity
header of SIP messages. You can enable P-Asserted-Identity (PAI) headers to
assert the identity of users in outgoing SIP requests or response messages.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this URI.
• Content: This configurable field sets the Content of SIP headers for outgoing
SIP calls using this URI. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Caller. You can select any one of the following
options:
- Caller
- Explicit
- None
• Forwarding/Twinning: Default = Original Caller. This field can be used to set
an originator number for forwarded and twinned calls. You can select any one
of the following options:
- Caller
- Explicit
- Original Caller
• Incoming Calls: Default = Called. You can select any one of the following
options:
- Called
- Explicit
- None
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Configuration Mode Field Descriptions

Name Description
P Preferred ID: Default = Disabled
When selected, identity information is provided in the P Preferred Identity
header of SIP messages.
• Display: This configurable field sets the Name field for outgoing SIP calls
using this URI.
• Content: This configurable field sets the Content of SIP headers for outgoing
SIP calls using this URI. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Caller. You can select any one of the following
options:
- Called
- Explicit
- None
• Forwarding/Twinning: Default = Original Caller
This field can be used to set an originator number for forwarded and twinned
calls. You can select any one of the following options:
- Caller
- Explicit
- None
- Original Caller
• Incoming Calls: Default = Called. You can select any one of the following
options:
- Called
- Explicit
- None
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 Line

Name Description
Diversion Header Default = Disabled
When selected, information from the Diversion Header is provided in the SIP
messages.
• Display: This configurable field sets the Name field for outgoing SIP calls
using this URI.
• Content: This configurable field sets the Content of SIP headers for outgoing
SIP calls using this URI. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Caller. You can select any one of the following
options:
- Caller
- Explicit
- None
• Forwarding/Twinning: Default = Caller. This field can be used to set an
originator number for forwarded and twinned calls. You can select any one of
the following options:
- Caller
- Explicit
- None
- Original Caller
• Incoming Calls: Default = None.
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Configuration Mode Field Descriptions

Name Description
Remote Party ID Default = Disabled
When selected, it sets up the Remote Party ID header. the Diversion header for
outgoing SIP calls using this URI.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this URI.
• Content: This configurable field sets the Content of SIP headers for outgoing
SIP calls using this URI. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Caller. You can select any one of the following
options:
- Caller
- Explicit
- None
• Forwarding/Twinning: Default = Original Caller. This field can be used to set
an originator number for forwarded and twinned calls. You can select any one
of the following options:
- Caller
- Explicit
- None
- Original Caller
• Incoming Calls: Default = Called. You can select any one of the following
options:
- Called
- Explicit
- None

Display and Content fields are configurable for each SIP header separately. The value can either
be entered manually or one of the following options can be selected:
• Auto: Use the number that IP Office is using to make the call as the From field. The number
will be aligned with IP Office internal numbering schema.
• Use Internal Data: Use the value User | SIP | Display Name (Alias) for the user making the
call.
The system can also use
- Group | SIP | SIP Display Name (Alias) for a group
- System | Voicemail | SIP Display Name (Alias) for voicemail
If you have selected a Credential value other than None, the following options become available in
the drop-down list for you to select:
• Credentials User Name: Use the value Line | SIP Line | SIP Credentials | User Name for
the user making the call.

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 Line

• Credentials Authentication Name: Use the value Line | SIP Line | SIP Credentials |
Authentication Name.
• Credentials Contact: Use the value Line | SIP Line | SIP Credentials | Contact.
Related links
Call Details on page 335

SIP Line Appearances

Name Description
SIP Line Appearances Default = Disabled
When enabled, the content of the pane is displayed at the bottom half of the
page. You can use the Add, Edit, and Delete options to configure SIP Line
Appearances parameters. If you disable the SIP Line Appearances check box
and save the configuration, all the configured values are lost.
Incoming Group Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming
call routes in the system configuration. The matching incoming call route is then
used to route incoming calls. The same ID can be used for multiple lines.
Outgoing Group Default = Previous external line + 1, Range 0 to 99999.
Short codes that specify a number to dial can also specify the line group to be
used. The system will then seize a line from those with the matching Outgoing
Group ID.
In a Server Edition network, the Outgoing Group ID used must be unique
within the network. The same ID cannot be used in the configuration of any
lines on another server system in the network. For non-Server Edition
deployments, the same ID can be used for multiple lines.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition network, reserved for the IP Office lines
to the Primary Server and Secondary Server respectively.
• 99901 to 99930 In a Server Edition network, reserved for the IP Office lines
from the Primary Server to each expansion system in the network.
• 0 In a Server Edition network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved
for the SM line.
Credentials Default = 0:<None>
This field is used to select from a list of the account credentials configured on
the line's SIP Credentials tab.
Max Sessions Default =10 This field sets the maximum number of simultaneous calls that can
use the URI before the system returns busy to any further calls. For capacity
information, see Deploying Avaya IP Office™ Platform Server Edition.
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Configuration Mode Field Descriptions

Name Description
Incoming Sessions Default = 3
The number of Incoming call sessions. The range varies from 0 to Max
Sessions. If you change the Max Sessions, the number of Incoming sessions
are automatically updated to reflect the change.
Outgoing Sessions Default = 3
The number of Outgoing call sessions. The range varies from 0 to Max
Sessions. If you change the Max Sessions, the number of Outgoing sessions
are automatically updated to reflect the change.
Line Appearance ID Default = 701
The Line Appearance ID is a representation of the SIP trunk line on the IP
Office system. The indicator corresponding to the Line Appearance indicates
the activities on the line.
Incoming ID The ID on which the Incoming call activities are indicated. The range of the IDs
are from maximum to minimum and changes when the Max Sessions values
are updated.
Outgoing ID The ID on which the Outgoing call activities are indicated. This is a read-only
field. The range of the IDs are from minimum to maximum and changes when
the Max Sessions values are updated.
Local URI This field sets the 'From' field for outgoing SIP calls using this SIP Line.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this SIP Line.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this SIP Line. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Explicit.
• Incoming Calls: Default = Explicit.
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 Line

Name Description
Contact This field sets the From field for outgoing SIP calls using this SIP Line.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this SIP Line.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this SIP Line. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
• Incoming Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
P Asserted ID Default = Disabled
When selected, identity information is provided in the P Asserted Identity
header of SIP messages. You can enable P-Asserted-Identity (PAI) headers to
assert the identity of users in outgoing SIP requests or response messages.

Note:
You can enter the wildcard character “*”. Entering this value populates the
SIP PAI header with the caller information available to IP Office.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this SIP Line.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this SIP Line. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
• Incoming Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
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Configuration Mode Field Descriptions

Name Description
P Preferred ID Default = Disabled
When selected, identity information is provided in the P Preferred Identity
header of SIP messages.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this SIP Line.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this SIP Line. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
• Incoming Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
Diversion Header Default = Disabled
When selected, information from the Diversion Header is provided in the SIP
messages.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this SIP Line.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this SIP Line. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
• Incoming Calls: Default = None.
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 Line

Name Description
Remote Party ID Default = Disabled
When selected, it sets up the Remote Party ID header. the Diversion header for
outgoing SIP calls using this URI.
.
• Display: This configurable field sets the Name value for outgoing SIP calls
using this SIP Line.
• Content: This field sets the Content of SIP headers for outgoing SIP calls
using this SIP Line. The values you type in the Content field are also
reflected in the Display field.
• Outgoing Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None
• Incoming Calls: Default = Explicit. You can select any one of the following
options:
- Explicit
- None

Display and Content fields are configurable for each SIP header separately. The value can either
be entered manually or one of the following options can be selected:
• Auto: Use the number that IP Office is using to make the call as the From field. The number
will be aligned with IP Office internal numbering schema.
• Use Internal Data: Use the value User | SIP | Display Name (Alias) for the user making the
call.
The system can also use
- Group | SIP | SIP Display Name (Alias) for a group
- System | Voicemail | SIP Display Name (Alias) for voicemail
If you have selected a Credential value other than None, the following options become available in
the drop-down list for you to select:
• Credentials User Name: Use the value Line | SIP Line | SIP Credentials | User Name for
the user making the call.
• Credentials Authentication Name: Use the value Line | SIP Line | SIP Credentials |
Authentication Name.
• Credentials Contact: Use the value Line | SIP Line | SIP Credentials | Contact.
Related links
Call Details on page 335

VoIP
Navigation: Line | SIP Line | VoIP

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Configuration Mode Field Descriptions

This form is used to configure the VoIP settings applied to calls on the SIP trunk.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup. The available codecs
in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs varies to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 or IP500 Combo cards. For Server Edition, it is supported on Primary
Server, Secondary Server and Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available in this form are set through the codec list on System | VoIP.
Within a network of systems, it is strongly recommended that all the systems and the
lines connecting those systems use the same codecs.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
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 Line

Field Description
Fax Transport Default = Off.
Support
This option is only available if Re-Invite Supported is selected.
IP500 V2 systems can terminate T38 fax calls. IP Office Linux systems can route the
calls between trunks/terminals with compatible fax types. If the media is routed by IP
Office between trunks/terminals with incompatible fax types or if fax is terminated by
IP Office, IP Office will detect fax tones and renegotiate the call as needed.
This setting must be configured based on what is supported by the SIP ATA. The
options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes.
• T38 Fallback When you enable this option, T38 is used for sending and receiving
faxes on a SIP line. If the called destination does not support T38, the system will
send a re-invite to change the transport method to G.711.
DTMF Support Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The options are:
• In Band: Send DTMF digits as part of the audio path.
• RFC2833: Send DTMF digits using a separate audio stream from the voice path.
Note that use of RFC2833 is negotiated with the remote end of the call. If not agreed
or not supported, the line reverts to using in band signalling.
• Info: Send the DTMF digits in SIP INFO packets.
Media Security Default = Disabled.
These setting control whether SRTP is used for this line and the settings used for the
SRTP. The options are:
• Same as System: Matches the system setting at System | VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and
data) will be enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data)
will be enforced to use SRTP only.

Warning:
Selecting Enforced on a line or extension that does not support media security
will result in media setup failures.
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Configuration Mode Field Descriptions

Field Description
Advanced Media Not displayed if Media Security is set to Disabled. The options are:
Security Options
• Same as System: Use the same settings as the system setting configured on
System | VoIP Security.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option
to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunks between networked systems, the same
setting should be set at both ends.
Local Hold Music Default = Off.
When enabled, if the far end puts the call on HOLD, the system plays music received
from far end (SIP Line) to the other end. RTCP reports are sent towards SIP Line.
When disabled, the system plays local music to the other endpoint and no RTCP
packets are sent to SIP trunk.
Re-Invite Supported Default = Off.
When enabled, Re-Invite can be used during a session to change the characteristics
of the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also
support Re-Invite. This setting must be enabled for video support.
Codec Lockdown Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response
to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP
answer that also lists multiple codecs. This means that the user agent may switch to
any of those codecs during the session without further negotiation. The system does
not support multiple concurrent codecs for a session, so loss of speech path will occur
if the codec is changed during the session. If codec lockdown is enabled, when the
system receives an SDP answer with more than one codec from the list of offered
codecs, it sends an extra re-INVITE using just a single codec from the list and
resubmits a new SDP offer with just the single chosen codec.
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 Line

Field Description
Allow Direct Media Default = On
Path
This settings controls whether IP calls must be routed via the system or can be routed
alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the
need for system resources such as voice compression channels. Both ends of the
calls must support Direct Media and have compatible VoIP settings such as
matching codec, etc. If otherwise, the call will remain routed via the system.
Enabling this option may cause some vendors problems with changing the media
path mid call.
• If disabled, the call is routed via the system. In that case, RTP relay support may still
allow calls between devices using the same audio codec to not require a voice
compression channel.
PRACK/100rel Default = Off.
Supported
When selected, supports Provisional Reliable Acknowledgement (PRACK) on SIP
trunks. Enable this parameter when you want to ensure that provisional responses,
such as announcement messages, have been delivered. Provisional responses
provide information on the progress of the request that is in process. For example,
while a cell phone call is being connected, there may be a delay while the cell phone
is located; an announcement such as “please wait while we attempt to reach the
subscriber” provides provisional information to the caller while the request is in
process. PRACK, which is defined in RFC 3262, provides a mechanism to ensure the
delivery of these provisional responses.
Force direct media Default = On
with phones
The setting is only available when the trunk's Re-invite Supported and Allow Direct
Media Path settings are enabled and its DTMF Support option is set to RFC2833/
RFC4733. It also requires the H.323 IP extension involved in the call to also have
Allow Direct Media Path enabled. This feature is only supported with Avaya H.323 IP
telephones. For calls where the Avaya H.323 IP extension using the trunk is doing so
as a direct media call, this feature allows digits pressed on the extension to be
detected and the call changed to an indirect media call so that RFC2833 DTMF can
be sent. The call remains as an indirect media call for 15 seconds after the last digit
before reverting back to being a direct media call.
G.711 Fax ECAN Default = Off
This setting is only available on IP500 V2 systems when Fax Transport Support is
set to G.711 or T.38 Fallback. When IP Office detects a fax call, the IP Office
negotiates to G.711 (if not already in G.711) and reconfigures the connection with
echo cancellation (ECAN) based on the 'G.711 Fax ECAN field. This can be used to
avoid an ECAN mismatch with the SIP trunk service provider. Also for fax calls, the
connection’s NLP is disabled, a fixed jitter buffer is set and silence suppression is
disabled.

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Configuration Mode Field Descriptions

T38 Fax
Navigation: Line | SIP Line | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which
they both support. The options are:
• 0
• 1
• 2
• 3
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For
UDPTL, redundancy error correction is supported. Forward Error Correction
(FEC) is not supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased
redundancy increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed
V.21 T.30 fax transmissions.
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27
and V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax
equipment or is found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
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 Line

Field Description
Tx Network Timeout Default = 150.
(secs)
Scan Line Fix-up Default = On.
TFOP Enhancement Default = On.
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax
transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression

NSF Override Default = Off.

If selected, the NSF (Non-Standard Facility) information sent by the T38 device
can be overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
SIP Line on page 326

SIP Credentials
Navigation: Line | SIP Line | SIP Credentials
Used to enter the ITSP username and password for the SIP account with the ITSP. If you have
several SIP accounts going to the same ITSP IP address or domain name, you can enter up to 30
sets of ITSP account names and passwords on this tab.
Use the Add, Remove, and Edit buttons to manage the set of credentials for the SIP trunk
accounts. The settings for each account are listed below.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Descriptions
Index This number is assigned automatically and cannot be edited. If the From field on the
SIP URI being used for the call is set to Use Authentication Name , the registration
field of the SIP URI will indicate the index number of the SIP credentials to use for calls
by that SIP URI.
User Name This name must be unique and is used to identify the trunk. The name can include the
domain if necessary.
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Configuration Mode Field Descriptions

Field Descriptions
Authentication Name Default = Blank.
This field can be blank but must be completed if a Password is also specified. This
value is provided by the SIP ITSP. Depending on the settings on the Local URI tab
associated with the SIP call, it may also be used as the user part of the SIP URI. The
name can include the domain if necessary.
Contact Default = Blank.
This field is used to enter a contact and can include the domain if necessary.
Password Default = Blank.
This value is provided by the SIP ITSP. If a password is specified, the matching
Authentication Name must also be set.
Expiry Default = 60 minutes.
This setting defines how often registration with the SIP ITSP is required following any
previous registration.
Registration Default = On.
Required
If selected, the fields above above are used for registration when making calls. If
exported or imported as part of a trunk template.

Related links
SIP Line on page 326

SIP Advanced
Navigation: Line | SIP Line | SIP Advanced
Additional configuration information
For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 628.
Configuration settings
These settings are mergable, with the exception of the Media Connection Preservation setting.
Changing the Media Connection Preservation setting requires a “merge with service disruption”.
When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line
are dropped.
Field Description
Addressing
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 Line

Field Description
Association Method Default = By Source IP Address.
This setting sets the method by which a SIP line is associated with an incoming SIP
request.
The match criteria used for each line can be varied. The search for a line match for an
incoming request is done against each line in turn using each lines Association
Method. The order of line matching uses the configured Line Number settings until a
match occurs. If no match occurs the request is ignored. This method allows multiple
SIP lines with the same address settings. This may be necessary for scenarios where
it may be required to support multiple SIP lines to the same ITSP. For example when
the same ITSP supports different call plans on separate lines or where all outgoing
SIP lines are routed from the system via an additional on-site system. The options are:
• By Source IP Address: This option uses the source IP address and port of the
incoming request for association. The match is against the configured remote end of
the SIP line, using either an IP address/port or the resolution of a fully qualified
domain name.
• "From" header hostpart against ITSP domain: This option uses the host part of
the From header in the incoming SIP request for association. The match is against
the ITSP Domain Name above.
• R-URI hostpart against ITSP domain: This option uses the host part of the
Request-URI header in the incoming SIP request for association. The match is
against the ITSP Domain Name above.
• "To" header hostpart against ITSP domain: This option uses the host part of the
To header in the incoming SIP request for association. The match is against the
ITSP Domain Name above.
• "From" header hostpart against DNS-resolved ITSP domain: This option uses
the host part of the FROM header in the incoming SIP request for association. The
match is found by comparing the FROM header against a list of IP addresses
resulting from resolution of the ITSP Domain Name above or, if set, the Line | SIP
Line | Transport | ITSP Proxy Address setting.
• "Via" header hostpart against DNS-resolved ITSP domain: This option uses the
host part of the VIA header in the incoming SIP request for association. The match
is found by comparing the VIA header against a list of IP addresses resulting from
resolution of the ITSP Domain Name above or, if set, the Line | SIP Line |
Transport | ITSP Proxy Address setting.
• "From" header hostpart against ITSP proxy: This option uses the host part of the
“From” header in the incoming SIP request for association. The match is against the
Line | SIP Line | Transport | ITSP Proxy Address setting.
• "To" header hostpart against ITSP proxy: This option uses the host part of the
From header in the incoming SIP request for association. The match is against the
Line | SIP Line | Transport | ITSP Proxy Address setting.
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Configuration Mode Field Descriptions

Field Description
• R-URI hostpart against ITSP proxy: This option uses the host part of the Request-
URI in the incoming SIP request for association. The match is against the Line | SIP
Line | Transport | ITSP Proxy Address setting.
Call Routing Method Default = Request URI.
This field allows selection of which incoming SIP information should be used for
incoming number matching by the system's incoming call routes. The options are to
match either the Request URI or the To Header element provided with the incoming
call.
Use P-Called-Party Default = Off.
When enabled, IP Office reads the P-Called-Party ID header if present in the SIP
message and routes the incoming SIP calls based on it. The feature can be enabled
on public SIP trunk interfaces. The configuration should be present in SIP Trunks
templates. If not present, will be treated as disabled. If the feature is enabled and the
header is not present in the SIP message, IP Office uses the header configured in the
Call Routing method for Incoming Call Route.
Suppress DNS SRV Default = Off.
Lookups
Controls whether to send SRV queries for this endpoint, or just NAPTR and A record
queries.

Identity
Use Phone Context Default = Off.
When set to On, signals SIP enabled PBXs that the call routing identifier is a
telephone number.
Add user=phone Default = Off.
This setting is available when Use Phone Context is set to On.
This setting adds the SIP parameter user with value phone to the From and To SIP
headers in outgoing calls.
Use + for Default = Off.
International
When set to On, outgoing international calls use E.164/International format with a ‘+’
followed by the country code and then the directory number.
Use PAI for Privacy Default = Off.
When set to On, if the caller ID is withheld, the SIP message From: header is made
anonymous, and the caller’s identity, for admission control, billing, and emergency
services, is inserted into the P-Asserted-Identity header. This mechanism should only
be used in a trusted network and must be stripped out of the SIP message before it is
forwarded outside the trusted domain.
Use Domain for PAI Default = Off.
When set to Off, the DNS resolved IP address of the ITSP Proxy is used for the host
part in the P-Asserted-Identity header. When set to On, the the Domain for PAI is
used.
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 Line

Field Description
Caller ID FROM Default = Off.
Header
Incoming calls can include caller ID information in both the From field and in the PAI
fields. When this option is selected, the caller ID information in the From field is used
rather than that in the PAI fields.
Send From In Clear Default = Off.
When selected, the user ID of the caller is included in the From field. This applies
even if the caller has selected to be or is configured to be anonymous, though their
anonymous state is honored in other fields used to display the caller identity.
Cache Auth Default = On.
Credentials
When set to On, allows the credentials challenge and response from a registration
transaction to be automatically inserted into later SIP messages without waiting for a
subsequent challenge.
Add UUI header Default — Off.
When set to On, the User-to-User Information (UUI) is passed in SIP headers to
applications.
Add UUI header to Default — Off.
redirected calls
When set to On, the UUI is passed in SIP headers for calls that are redirected, for
example, forwarded calls, twinned calls. This field can be modified only if the Add UUI
header is set to On.
User-Agent and Default = Blank (Use system type and software level).
Server Headers
The value set in this field is used as the User-Agent and Server value included in SIP
request headers made by this line. If the field is blank, the type of IP Office system
and its software level used. Setting a unique value can be useful in call diagnostics
when the system has multiple SIP trunks.
Send Location Info Default = Never.
The options are:
• Never: Do not send location information.
• Emergency Calls: When set to Emergency Calls, the location defined at Location
| Address is sent as part of the INVITE message when emergency calls are made.

Media
Allow Empty INVITE Default = Off.
When set to On, allows 3pcc devices to initiate calls to IP Office by sending an INVITE
without SDP.
Send Empty re- Default = Off.
INVITE
This option is only available if Line | SIP Line | VoIP | Re-Invite Supported is
selected.
If set to On, when connecting a call between two endpoints, IP Office sends an
INVITE without SDP in order to solicit the full media capabilities of both parties.
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Configuration Mode Field Descriptions

Field Description
Allow To Tag Change Default = Off.
When set to On, allows the IP Office to change media parameters when connecting a
call to a different party than that which was advertised in the media parameters of
provisional responses, such as 183 Session Progress.
P-Early-Media Default = None.
Support
The options are:
• None: IP Office will not advertise support of this SIP header and will always take
incoming early media into account regardless of presence of this header
• Receive: IP Office will advertise support of this SIP header and will discard
incoming early media unless this header is present in the sip message.
• All: IP Office will advertise support of this sip header, will discard incoming early
media unless this header is present in the sip message and will include this sip
header when providing early media.
Send Default = Off.
SilenceSupp=off
Used for the G711 codec. When checked, the silence suppression off attribute is sent
in SDP on this trunk.
Force Early Direct Default = Off.
Media
When set to On, allows the direct connection of early media streams to IP endpoints
rather than anchoring it at the IP Office.
Media Connection Default = Disabled.
Preservation
When enabled, allows established calls to continue despite brief network failures. Call
handling features are no longer available when a call is in a preserved state.
Preservation on public SIP trunks is not supported until tested with a specific service
provider.
Indicate HOLD Default = Off.
When enabled, the system sends a HOLD INVITE to the SIP trunk endpoint.

Call Control
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
(s)
Sets how long the system should wait for a response to its attempt to initiate a call
before following the alternate routes set in an ARS form.
Call Queuing Timeout Default = 5 minutes.
(m)
For incoming calls, how many minutes to wait before dropping a call that has been
queued waiting for a free VCM resource, or has remained in the unanswered state.
For outgoing calls, how many minutes to wait for the call to be answered after
receiving a provisional response.
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 Line

Field Description
Service Busy Default = 486 - Busy Here (503 - Service Unavailable for the France2 locale).
Response
For calls that result in a busy response from IP Office, this setting determines the
response code. The options are:
• 486 - Busy Here
• 503 - Service Unavailable
on No User Default = 408-Request Timeout.
Responding Send
Specifies the cause to be used when releasing incoming calls from SIP trunks, when
the cause of releasing is that user did not respond. The options are 408-Request
Timeout or 480 Temporarily Unavailable.
Action on CAC Default = Allow Voicemail
Location Limit
When set to Allow Voicemail, the call is allowed to go to a user's voicemail when the
user's location call limit has been reached. When set to Reject Call, the call is
rejected with the failure response code configured in the Service Busy Response
field.
Suppress Q.850 Default = Off.
Reason Header
When SIP calls are released by sending BYE and CANCEL, a release reason header
is added to the message. When set to On, the Q.850 reason header is not included.
Emulate NOTIFY for Default = Off.
REFER
Use for SIP providers that do not send NOTIFY messages. When set to On, after IP
Office issues a REFER, and the provider responds with 202 ACCEPTED, IP Office will
assume the transfer is complete and issue a BYE.
No REFER if using Default = Off.
Diversion
When enabled, REFER is not sent on the trunk if the forwarding was done with 'Send
Caller ID = Diversion Header'. Applies to Forwards and Twinning.

Related links
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Engineering
Navigation: Line | SIP Line | Engineering
This page is used to enter values that apply special features to the SIP line. These are entered
using the Add, Edit and Remove buttons.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Related links
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Configuration Mode Field Descriptions

SIP DECT Line

A SIP DECT line can be manually added. SIP DECT lines are used to manage D100 Base Station
operation.
Related links
Line on page 266
SIP DECT Base on page 360
VoIP on page 361

SIP DECT Base

Navigation: Line | SIP DECT Line | SIP DECT Base
Currently, IP Office supports four D100 Base Stations.
These settings are not mergeable. Changes to these settings requires a reboot of the system.
Field Description
Line Number Default = Blank. A unique line number associated with the SIP DECT Base Station.
Associated Extensions are other extensions that can log into the base station.
Base Name Default = Blank. Maximum 16 characters.
A name assigned to the base station. Each base station provisioned on the IP Office must
have a unique name. The field cannot be blank. The format is an alphanumeric string with
no special characters.
Base MAC Default = Blank.
Address
The MAC Address of the base station. If only one base station is provisioned, the field can
remain at the default value. If multiple base stations are provisioned, the MAC address for
each base station must be entered.
Configure Base IP
Configure Base IP Default = Off.
Set to On to configure IP address attributes for the base station. When enabled, the
Configure Base IP settings are displayed.
DHCP Client Default = On.
When enabled, specifies that the base station operates as a DHCP client. When enabled,
not other IP address attributes can be configured.
IP Address Default = Blank.
The IP address of the base station. The IP address must be on the same subnet as one of
the LAN interfaces.
IP Mask Default = Blank.
IP address mask.
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 Line

Field Description
IP Gateway Default = Blank.
The default gateway address
Provisioning Default = IP Office interface address. The server address from where the Base Station
Server configuration files can be retrieved.

Description Default = Blank. Maximum 31 characters.

Use this field to enter a description of this configuration.

Related links
SIP DECT Line on page 360

VoIP
Navigation: Line | SIP DECT Line | VoIP
This form is used to configure the VoIP setting applied to calls on the SIP DECT line.
These settings are not mergeable. Changes to these settings requires a reboot of the system.
Field Description
IP Address Default = Blank.
The IP address of the SIP DECT extension.
Codec Selection Default = Custom
This field defines the codec or codecs offered during call setup. The codecs available
to be used are set through System | VoIP.
The Custom option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list and
to change the order of the selected codecs. The D100 Base Station supports only
G711 codecs.
TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
DTMF Support Default =RFC2833
The D100 Base Station supports only RFC2833.
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Configuration Mode Field Descriptions

Field Description
VoIP Silence Default = Off
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
Local Hold Music Default = Off
Allow Direct Media Default = On
Path
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure.
• If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may
cause some vendors problems with changing the media path mid call.
• If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
RE-Invite Supported Default = Off.
When enabled, Re-Invite can be used during a session to change the characteristics
of the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also
support Re-Invite.

Related links
SIP DECT Line on page 360

SM Line
This type of line is used to create a SIP connection between an IP Office and an Avaya Aura®
Session Manager. The other end of the SIP connection must be configured on the Session
Manager as a SIP Entity Link.
An SM Line can only be added to IP Office system Standard Mode or Server Edition
configurations. It is typically used in IP Office Standard mode in Enterprise Branch deployments
connected to the Avaya Aura® network. For more details about IP Office Enterprise Branch
deployments refer to Deploying Avaya IP Office™ Platform as an Enterprise Branch with Avaya
Aura® Session Manager.
An SM Line can also be used in IP Office Server Edition to connect to an Avaya Aura® Session
Manager. Through the SM Line, IP Office Server Edition supports interoperability with Avaya
Aura® Session Manager. It also supports interoperability, via the Avaya Aura® Session Manager,
with Avaya Aura® Communication Manager systems and with CS 1000 systems. Note that IP
Office Server Edition is not used as an enterprise branch product and does not support some of
the IP Office enterprise branch functionality, such as management by Avaya Aura® System
Manager, WebLM licensing, Centralized Users or voicemail over the SM Line.

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 Line

If the Avaya Aura® network has multiple Avaya Aura® Session Managers to provide redundancy,
two SM lines can be added, one configured for each Avaya Aura® Session Manager.
Related links
Line on page 266
Session Manager on page 363
VoIP on page 365
T38 Fax on page 370

Session Manager
Navigation: Line | SM Line | Session Manager
Additional configuration information
For additional information regarding the Media Connection Preservation setting, see Media
Connection Preservation on page 628.
Configuration settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Changing the In Service setting to Disabled (out of service) requires a system reboot. However,
changing the In Service setting to Enabled is mergeable. Configuration changes made while the
line is out of service are also mergeable.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
• Session Manager line prioritization: Up to two Session Manager lines can be
configured. The two Session Manager lines are prioritized based on the line
number. The lower line number is considered the primary Session Manager line.
For example, if the first Session Manager line is configured as line number 17
and the second Session Manager line is configured as line 18, then line number
17 is considered the primary Session Manager line. If you want to designate the
second Session Manager line (line 18 in this example) as the primary Session
Manager line, you must change one or both of the line numbers so that the
second Session Manager line is configured with a lower number than the current
primary line.
• Session Manager line redundancy: Based on the priority of the Session
Manager lines designated by the line number, the active line to which the IP
Office> sends all calls will always be the highest priority Session Manager line in
service. That is, if the primary Session Manager line is in service, it will be the
active line for sending calls. If the connection to the primary Session Manager
line is lost, causing the IP Office to switch to the secondary Session Manager
line, then when the primary line comes back up later, the IP Office reverts back
to the primary Session Manager line.
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Configuration Mode Field Descriptions

Field Description
In Service Default = Enabled
This option can be used to administratively disable the SM Line. It does not reflect
the dynamic state of the line. If an SM Line is administratively disabled it is not
equivalent to being in the dynamic out of service state.
SM Domain Name This should match a SIP domain defined in the Session Manager system's SIP
Domains table. Unless there are reasons to do otherwise, all the Enterprise Branch
systems in the Avaya Aura® network can share the same domain.
SM Address Enter the IP address of the Session Manager the line should use in the Avaya
Aura network. The same Session Manager should be used for the matching Entity
Link record in the Avaya Aura® configuration.
Outgoing Group ID Default = 98888
This value is not changeable. However note the value as it is used in Enterprise
Branch short codes used to route calls to the Session Manager.
Prefix Default = Blank
This prefix will be added to any source number received with incoming calls.
Max Calls Default = 10
Sets the number of simultaneous calls allowed between the Enterprise Branch and
Session Manager using this connection. Each call will use one of the available
licenses that are shared by all SIP trunks configured in the system.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Include location specific Default = Off.
information
Enabled when Network Type is set to Private. Set to On if the PBX on the other
end of the trunk is toll compliant.
URI Type Default = SIP.
When SIP or SIPS is selected in the drop-down box, the SIP URI format is used
(for example, [email protected]). This affects the From field of outgoing calls.
The To field for outgoing calls will always use the format specified by the short
codes used for outgoing call routing. Recommendation: When SIP Secured URI is
required, the URI Type should be set to SIPS. SIPS can be used only when Layer
4 Protocol is set to TLS.
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 Line

Field Description
Media Connection Default = Enabled.
Preservation
When enabled, attempts to maintain established calls despite brief network
failures. Call handling features are no longer available when a call is in a
preserved state. When enabled, Media Connection Preservation applies to Avaya
H.323 phones that support connection preservation.
Location Default = Cloud.
Specify a location to associate the extension with a physical location. Associating
an extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined in the Location
form.
Network Configuration
TLS connections support the following ciphers:
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_WITH_AES_256_CBC_SHA
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA
• TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Layer 4 Protocol Default = TCP.
Send Port When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.
Listen Port When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.

Session Timer Default = 1200. Range = 90 to 64800

This field specifies the session expiry time. At the half way point of the expiry time,
a session refresh message is sent. Setting the field to On Demand disables the
session timer.
Communication Manager supports SIP session refresh via UPDATE in
Communicaton Manger release 6.2 SP1 and later. If using an earlier release of
Communication Manager, then the Session Timer parameter must be set to On
Demand.
Description Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.

Related links
SM Line on page 362

VoIP
Navigation: Line | SM Line | VoIP

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Configuration Mode Field Descriptions

These settings are mergeable. Changes to these settings do not require a reboot of the system.
These settings can be edited online. Changes to these settings do not require a reboot of the
system.
Field Description
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs varies to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
Within a network of systems, it is strongly recommended that all the systems and
the lines connecting those systems use the same codecs.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs.
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 Line

Field Description
Fax Transport Support Default = None.
This option is only available if Re-Invite Supported is selected. When enabled,
the system performs fax tone detection on calls routed via the line and, if fax tone
is detected, renegotiates the call codec as configured below. The SIP line provider
must support the selected fax method and Re-Invite. The system must have
available VCM resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo
base card.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes. This option is not
supported by Linux based systems.
• T38 Fallback When you enable this option, T38 is used for sending and
receiving faxes on a SIP line. If the called destination does not support T38, the
system will send a re-invite to change the transport method to G.711. This option
is not supported on Linux based systems.
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
DTMF Support Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled
to the remote end. The options are:
• In Band
• RFC2833
• Info
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Configuration Mode Field Descriptions

Field Description
Media Security Default = Same as System.
These setting control whether SRTP is used for this line and the settings used for
the SRTP. The options are:
• Same as System: Matches the system setting at System | VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and
data) will be enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and
data) will be enforced to use SRTP only.

Warning:
Selecting Enforced on a line or extension that does not support media
security will result in media setup failures.
Advanced Media Not displayed if Media Security is set to Disabled. The options are:
Security Options
• Same as System: Use the same settings as the system setting configured on
System | VoIP Security.
• Encryptions: Default = RTP This setting allows selection of which parts of a
media session should be protected using encryption. The default is to encrypt
just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the
option to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off.
Suppression
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems,
the same setting should be set at both ends.
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 Line

Field Description
Allow Direct Media Path Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the
need for system resources such as voice compression channels. Both ends of
the calls must support Direct Media and have compatible VoIP settings such as
matching codec, etc. If otherwise, the call will remain routed via the system.
Enabling this option may cause some vendors problems with changing the
media path mid call.
• If disabled, the call is routed via the system. In that case, RTP relay support may
still allow calls between devices using the same audio codec to not require a
voice compression channel.
Re-Invite Supported Default = On.
When enabled, Re-Invite can be used during a session to change the
characteristics of the session. For example when the target of an incoming call or a
transfer does not support the codec originally negotiated on the trunk. Requires the
ITSP to also support Re-Invite.
Codec Lockdown Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In
response to a SIP offer with a list of codecs supported, some SIP user agents
supply a SDP answer that also lists multiple codecs. This means that the user
agent may switch to any of those codecs during the session without further
negotiation. The system does not support multiple concurrent codecs for a
session, so loss of speech path will occur if the codec is changed during the
session. If codec lockdown is enabled, when the system receives an SDP answer
with more than one codec from the list of offered codecs, it sends an extra re-
INVITE using just a single codec from the list and resubmits a new SDP offer with
just the single chosen codec.
Force direct media with Default = On
phones
The setting is only available when the trunk's Re-invite Supported and Allow
Direct Media Path settings are enabled and its DTMF Support option is set to
RFC2833/RFC4733. It also requires the H.323 IP extension involved in the call to
also have Allow Direct Media Path enabled. This feature is only supported with
Avaya H.323 IP telephones. For calls where the Avaya H.323 IP extension using
the trunk is doing so as a direct media call, this feature allows digits pressed on the
extension to be detected and the call changed to an indirect media call so that
RFC2833 DTMF can be sent. The call remains as an indirect media call for 15
seconds after the last digit before reverting back to being a direct media call.
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Configuration Mode Field Descriptions

Field Description
G.711 Fax ECAN Default = Off
This setting is only available on IP500 V2 systems when Fax Transport Support
is set to G.711 or T.38 Fallback. When IP Office detects a fax call, the IP Office
negotiates to G.711 (if not already in G.711) and reconfigures the connection with
echo cancellation (ECAN) based on the 'G.711 Fax ECAN field. This can be used
to avoid an ECAN mismatch with the SIP trunk service provider. Also for fax calls,
the connection’s NLP is disabled, a fixed jitter buffer is set and silence suppression
is disabled.

Related links
SM Line on page 362

T38 Fax
Navigation: Line | SM Line | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which
they both support. The options are:
• 0
• 1
• 2
• 3
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For
UDPTL, redundancy error correction is supported. Forward Error Correction
(FEC) is not supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased
redundancy increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed
V.21 T.30 fax transmissions.
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 Line

Field Description
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27
and V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax
equipment or is found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Timeout Default = 150.
(secs)
Scan Line Fix-up Default = On.
TFOP Enhancement Default = On.
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax
transmission.
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression

NSF Override Default = Off.

If selected, the NSF (Non-Standard Facility) information sent by the T38 device
can be overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
SM Line on page 362

IP Office Line
This line type is used to connect two IP Office systems.
In previous releases, connecting two IP Office systems was achieved using H.323 Lines
configured with Supplementary Services set to IP Office SCN. In the current release, the IP
Office line type is used to connect IP Office systems. Separating out the IP Office line type from

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Configuration Mode Field Descriptions

the H.323 line type allows for the logical grouping of features and functions available when
connecting two IP Office systems, including IP Office systems connected through the cloud.
Note:
Setting an IP Office line with Transport Type = Proprietary and Networking Level = SCN
will interwork with a previous release system configured with an H.323 SCN line.
Related links
Line on page 266
Line on page 372
Short Codes on page 377
VoIP Settings on page 377
T38 Fax on page 380

Line
Navigation: Line | IP Office Line | Line
Additional configuration information
For information on the SCN Resiliency Options, see Server Edition Resiliency on page 729.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Line Number Default = Auto-filled. Range = 1 to 249 (IP500 V2)/349 (Server Edition).
Enter the line number that you wish. Note that this must be unique. On IP500 V2
systems, line numbers 1 to 16 are reserved for internal hardware.
Transport Type Default = Proprietary.
The options are
• Proprietary: The default connection type when connecting two IP Office systems.
• WebSocket Client / Websocket Server: A WebSocket connection is an HTTP /
HTTPS initiated TCP pipe through which Call signalling and Network Signaling is
tunneled. This transport type is used to connect IP Office systems through the cloud.
Selecting one of the WebSocket options enables the Security field and the
Password fields.
Networking Level Default = SCN.
The options are
• None: No supplementary services are supported.
• SCN: This option is used to link IP Office system within a multi-site network. The
systems within a multi-site network automatically exchange information about users
and extensions, allowing remote users to be called without any additional
configuration on the local system.
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 Line

Field Description
Security Default = Unsecured.
The Security field is available when Transport Type is set to WebSocket Client or
WebSocket Server.
The options are
• Unsecured : The connection uses HTTP/TCP.
• Medium: The connection uses HTTPS/TLS.
• High: The connection uses HTTPS/TLS. The server certificate store must contain
the client identity certificate.
Network Type Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private. The
system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Include location Default = Off.
specific information
Enabled when Network Type is set to Private. Set to On if the PBX on the other end
of the trunk is toll compliant.
Telephone Number Default = Blank.
Used to remember the telephone number of this line. For information only.
Prefix Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the Prefix
field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable for
external line presentation should be stripped using short codes.
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Configuration Mode Field Descriptions

Field Description
Outgoing Group ID Default = 1. Range 0 to 99999.
Short codes that specify a number to dial can specify the line group to be used. The
system will then seize a line from those available with a matching Outgoing Group ID.
In a Server Edition/Select network, the Outgoing Group ID used for lines to a system
must be unique within the network. The same ID cannot be used in the configuration
of any lines to another server system in the network.
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition/Select network, reserved for the IP Office lines
to the primary and secondary server respectively.
• 99001 to 99148 In a Server Edition/Select network, reserved for the IP Office lines
from the primary and secondary servers to each expansion system in the network.
• 0 In a Server Edition/Select network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for the
SM line.
Number of Channels Default = 20. Range 1 to 250; 1 to 500 for Select systems.
Defines the number of operational channels that are available on this line.
Outgoing Channels Default = 20, Range 0 to 250; 0 to 500 for Select systems.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
Gateway
Address Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must not
be shared by any other IP line (H.323, SIP, SES or IP DECT).
Location Default = Cloud.
Specify a location to associate the extension with a physical location. Associating an
extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined on Location |
Location.
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 Line

Field Description
Password Default = Blank.
Confirm Password The Password field is enabled when Transport Type is set to WebSocket Server or
WebSocket Client.
WebSockets are bi-directional HTTP or HTTPS communication pipes initiated from a
client to a server. They permit clients behind local a firewall to traverse the internet to
a server by using well known ports and protocols. A matching password must be set
at each end of the line.
Port When Transport Type is set to Proprietary, the default port is 1720 and cannot be
changed.
When Transport Type is set to WebSocket Client, the default port is 80.
The Port field is not available when Transport Type is set to WebSocket Server. The
HTTP and HTTPS receive ports are defined at the system level in the security settings
System Details tab.
SCN Resiliency Options
These options are only available when the Networking Level option is set to SCN. The intention of this feature
is to attempt to maintain a minimal level of operation while problems with the local system are resolved.
Supports Resiliency Default = Off.
These fields are available when Networking Level is set to SCN. When selected, all
the available options are defaulted to On. When resiliency support is enabled on an IP
Office Line in systems having IP extensions and the resilient systems do not have the
corresponding H.323 or SIP registrars enabled, the IP Office system displays the error
message -
System is configured to support resiliency. Registration
of IP extensions in failover requires the corresponding
registrar to be enabled.
Backs up my IP Default = Off.
Phones
When selected, the local system shares information about the registered phones and
users on those phones with the backup system. If the local system is no longer visible
to the phones, the phones will reregister with the backup system. When phones have
registered with the backup system, they show an R on their display.
Note that while IP Office line settings are mergeable, changed to this setting require
the IP phones to be restarted in order to become aware of the change in their failover
destination.
If the setting System | Telephony | Telephony | Phone Failback is set to Automatic,
and the phone’s primary server has been up for more than 10 minutes, the backup
system causes idle phones to perform a failback recovery to the original system.
If using resilience backup to support Avaya IP phones, Auto-create Extn and Auto-
create User should not be left enabled after initial configuration or any subsequent
addition of new extensions and users. Leaving auto-create options enabled on a
system that is a failover target may cause duplicate extension/user records on the
multi-site network under multiple failure scenarios.
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Configuration Mode Field Descriptions

Field Description
Backs up my Hunt Default = Off.
Groups
This option is available only on the IP Office Line connecting the Server Edition
Primary server to the Server Edition Secondary server.
When selected, any hunt groups the local system is advertising to the network are
advertised from the backup system when fallback is required. The trigger for this
occurring is phones registered with the local system registering with the backup
system, ie. Backs up my IP Phones above must also be enabled.
When used, the only hunt group members that will be available are as follows:
• If the group was a distributed hunt group, those members who were remote
members on other systems are still visible within the network.
• Any local members who have hot desked to another system still visible within the
network.
When the local system becomes visible to the backup system again, the groups will
return to be advertised from the local system.
Backs up my Default = Off.
Voicemail
This option can be used if the local system is hosting the Voicemail Pro server being
used by the network. If selected, when the local system is no longer visible to the
voicemail server, the backup system acts as host for the voicemail server. In a Server
Edition network, this option is only available on the H.323 trunk from the Primary
Server to the Secondary Server. It is assumed to be on and is automatically set by the
Resilience Administration tool.
The option requires the backup system to have licenses for the Voicemail Pro features
that are required to operate during any fallback period.
Backs up my IP DECT Default = Off.
Phones
This option is used for Avaya IP DECT phones registered with the system. When
selected, it will share information about the registered phones and users on those
phones with the backup system.
If the local system is no longer visible to the phones, the phones will reregister with
the backup system. The users who were currently on those phones will appear on the
backup system as if they had hot desked. Note that when the local system is restored
to the network, the phones will not automatically re-register with it. A phone reset via
either a phone power cycle or using the System Status Application is required. When
phones have registered with the backup system, they will show an R on their display.

Note:
Only one IP Office Line can have this configuration parameter set to On.
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 Line

Field Description
Backs up my one-X Default = Off.
Portal
This option is available on Server Edition Select deployments and only on the IP
Office Line connecting the Server Edition Primary server to the Server Edition
Secondary server.
When set to On, this setting enables one-X Portal resiliency and turns on the backup
one-X Portal on the Server Edition Secondary server.

Description Default = Blank. Maximum 31 characters.

Use this field to enter a description of this configuration.

Related links
IP Office Line on page 371

Short Codes
Navigation: Line | IP Office Line | Short Codes
Incoming calls on IP Office Lines are not routed using Incoming Call Route settings.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Related links
IP Office Line on page 371

VoIP Settings
Navigation: Line | IP Office Line | VoIP Settings
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup. The available codecs
in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs varies to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 or IP500 Combo cards. For Server Edition, it is supported on Primary
Server, Secondary Server and Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available in this form are set through the codec list on System | VoIP.
Within a network of systems, it is strongly recommended that all the systems and the
lines connecting those systems use the same codecs.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
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 Line

Field Description
Fax Transport Default = None.
Support
IP500 V2 systems can terminate T38 fax calls. IP Office Linux systems can route the
calls between trunks/terminals with compatible fax types. If the media is routed by IP
Office between trunks/terminals with incompatible fax types or if fax is terminated by
IP Office, IP Office will detect fax tones and renegotiate the call as needed.
The options are:
• None Select this option if fax is not supported by the line provider.
• Fax Relay On IP Office Lines, fax relay is supported across multi-site network lines
with Fax Transport Support selected. This will use 2 VCM channels in each of the
systems. Fax relay is only supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 and or IP500 Combo cards.
Not supported on Server Edition.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes.
• T38 Fallback When you enable this option, T38 is used for sending and receiving
faxes. If the called destination does not support T38, the system will renegotiate to
change the transport method to G.711.
Call Initiation Default = 4. Range = 1 to 99 seconds.
Timeouts
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
Media Security Default = Same as System.
Secure RTP (SRTP) can be used between IP Offices to add additional security. These
settings control whether SRTP is used for this line and the settings used for the SRTP.
The options are:
• Same as System: Matches the system setting at System | VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and
data) will be enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data)
will be enforced to use SRTP only.

Warning:
Selecting Enforced on a line or extension that does not support media security
will result in media setup failures.
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Configuration Mode Field Descriptions

Field Description
Advanced Media Not displayed if Media Security is set to Disabled. The options are:
Security Options
• Same as System: Use the same settings as the system setting configured on
System | VoIP Security.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option
to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
Out Of Band DTMF Default = On.
Out of Band DTMF is set to on and cannot be changed.
Allow Direct Media Default = On
Path
This settings controls whether IP calls must be routed via the system or can be routed
alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the
need for system resources such as voice compression channels. Both ends of the
calls must support Direct Media and have compatible VoIP settings such as
matching codec, etc. If otherwise, the call will remain routed via the system.
Enabling this option may cause some vendors problems with changing the media
path mid call.
• If disabled, the call is routed via the system. In that case, RTP relay support may still
allow calls between devices using the same audio codec to not require a voice
compression channel.

Related links
IP Office Line on page 371

T38 Fax
Navigation: Line | IP Office Line | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.

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 Line

Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which
they both support. The options are:
• 0
• 1
• 2
• 3
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For
UDPTL, redundancy error correction is supported. Forward Error Correction
(FEC) is not supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased
redundancy increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed
V.21 T.30 fax transmissions.
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27
and V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax
equipment or is found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Timeout Default = 150.
(secs)
Scan Line Fix-up Default = On.
TFOP Enhancement Default = On.
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax
transmission.
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Configuration Mode Field Descriptions

Field Description
Disable EFlags For Default = Off.
First DIS
Disable T30 MR Default = Off.
Compression

NSF Override Default = Off.

If selected, the NSF (Non-Standard Facility) information sent by the T38 device
can be overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
IP Office Line on page 371

Control Unit
Navigation: Control Unit | Unit
The Control Unit form gives details of the system and some devices connected to or within the
system. This includes some modules within the control nit as well as external expansion modules.
For Server Edition systems, for the Primary Server, Secondary Server and Expansion System (L)
it shows details for the physical server platform and details for the IP Office Media service being
hosted on that server. For the Expansion System (V2) it shows details of the IP500 V2 control unit
and the cards installed into the control unit.
The New and Delete actions on this form have special functions.
• New This action is used to added a WAN3 expansion module. If when a WAN3 is added to
the system, the WAN3 is not recognized following a system reboot, New on this form can be
used to scan for the WAN3 module.
• Delete This action can only be used with external expansion modules. This action can only
be used with external expansion modules attached to a system. The action should used with
caution as deleting a module will also delete any extensions or lines associated with the
module being deleted. If the module is physically present, default records are automatically
recreated following a reboot.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Device Number This is automatically allocated by the system.
Unit Type The name of the device.
Version The version of software running on each unit.
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 Extension

Field Description
Serial Number This is the number the system uses to tie a physical Control Unit to a device
configuration (device number). For the control unit this is the MAC address. For a
device connected to an Expansion port, it is the Expansion port number plus 1.
Unit IP Address This field shows the IP address for the LAN1.
Interconnect Number For external expansion modules this is the control unit expansion port used for
connection. For other devices this is 0.
Module Number For external expansion modules this is the control unit expansion port used for
connection. For internal devices in the control unit, Control Unit is displayed.
Operating Mode This field is available when a DS16B or DS30B digital expansion module is selected
as the control unit. Select the operating mode based on the type of telephones
deployed.
• DS - 1400, 9500, 5400, 2400, 4400 Series Phones
• BST - T7000, M7000 Series Phones

Related links
Configuration Mode Field Descriptions on page 186

Extension

By default, each extension is normally associated with a user and uses that user's directory
number and other setting. Users with a log in code can move between extensions by logging in
and out, so the directory number is not a fixed property of the extension.
Non-IP Extensions
Physical extension ports are either integral to the control unit or added by the installation of an
analog or digital phone expansion module. Extension records are automatically created for each
physical extension port within the system. These ports cannot be added or deleted manually.For
Server Edition, non-IP extensions are only supported on Expansion System (V2) units.

Standard Telephone
A standard extension.

Quiet Headset
Used for analog extension devices that are permanently off-hook.
:

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Configuration Mode Field Descriptions

IVR Port
Used for analog ports connected to devices that require a specific disconnect clear signal at the
end of each call.

Paging Speaker
An analog extension port set to be used as a paging speaker connection.

FAX Machine
Indicates that the extension is connected to a FAX machine.

MOH Source
Indicates that the extension is being used as a music on hold source.
IP Extensions
These are used for IP phone devices and VoIP applications.

H.323 or SIP Extension

This icon indicates an IP extension. IP extensions are either added manually or by the automatic
detection of the phone being connected. IP extensions can also be added manually to support a
third-party IP phone device. Note that third-party IP phone devices require entry of an IP End-
Points license.

IP DECT or SIP DECT

An extension port manually added to match extensions within an Avaya IP DECT system
connected to the system via an IP DECT line.
Related links
Configuration Mode Field Descriptions on page 186
Extn on page 384
Analog on page 387
Extension VoIP on page 390
Extension T38 Fax on page 398
IP DECT Extension on page 400
SIP DECT Base Extension on page 401

Extn
Navigation: Extension | Extn
Additional configuration information
The Caller Display Type setting controls the presentation of caller display information. For
additional configuration information, see Caller Display on page 598.

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 Extension

This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Configuration Settings
These settings are mergeable except Base Extension and Caller Display Type which require a
system reboot.
Field Description
Extension ID The physical ID of the extension port. Except for IP extensions, this settings is allocated
by the system and is not configurable.
Base Extension Range = 2 to 15 digits.
This is the directory number of the extension's default associated user if one is
required. The field does not have to match a user, in which case user’s need to login to
use the extension.
The field can be left blank for digital and analogue extensions, creating and extension
where users are forced to login but the extension has no default associated user. This
option is not supported for IP and CTI extensions.
Following a restart, the system will attempt to log in the user with the same extension
number (if they are not already logged in elsewhere in the multi-site network). This does
not occur if that user is set to Force Login.
If another user logs onto an extension, when they log out, the extension returns to its
default associated user unless they have logged in elsewhere or are set to Force
Login.
Phone Password Default = Blank. Range = Up to 31 digits.
H.323 Extensions only. Does not apply to DECT phones.
The password that must be entered as part of phone registration. This password is
used to secure registration of H.323 phones when there is no matching user to
authenticate against. Required if Media Security is enabled.
Caller Display Type Default = On.
Controls the presentation of caller display information for analog extensions. For digital
and IP extensions, this value is fixed as On. The table below lists the supported
options, all others are currently not used and default to matching UK.
Type Description
Off Disables caller display.
On Enables caller display using the caller display type appropriate to
the System Locale, see Avaya IP Office™ Platform Locale Settings.
If a different setting is required it can be selected from the list of
supported options. For an analog extension connected to a fax
server or other device that requires the pass through of DTMF
tones, select DTMFF.
UK FSK before the first ring conforming to BT SIN 227. Name and
number.
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Configuration Mode Field Descriptions

Field Description
UK20 As per UK but with a maximum length of 20 characters. Name and
number.
DTMFA Caller ID in the DTMF pattern A<caller ID>C. Number only.
DTMFB Caller ID in DTMF after call connection. Number only.
DTMFC Caller ID in the DTMF pattern A<caller ID>#. Number only.
DTMFF Sends the called number in DTMF after call connection. Number
only. Used for fax servers. When calls are delivered via a hunt group
it is recommended that hunt group queuing is not used. If hunt group
queuing is being used, set the Queue Type to Assign Call on Agent
Alert.
DTMFD Caller ID in the DTMF pattern D<caller ID>C. Number only.
FSKA Variant of UK used for BT Relate 1100 phones. Name and number.
FSKB ETSI specification with 0.25 second leading ring. Name and number.
FSKC ETSI specification with 1.2 second leading ring. Name and number.
FSKD Conforms to Belcore specification. Name and number.
Reset Volume after Default = Off. Resets the phone's handset volume after each call. This option is
Calls supported on Avaya 1400, 1600, 2400, 4400, 4600, 5400, 5600, 6400, 9500 and 9600
Series phones.
Device Type This field indicates, the last known type of phone connected to the extension port.
• Analogue extension ports always report as Analog Handset since the presence or
absence of actual analog phone cannot be detected.
• Digital extension ports report the type of digital phone connected or Unknown digital
handset if no phone is detected.
• H.323 extensions report the type of IP phone registered or Unknown H.323 handset
if no phone is currently registered as that extension.
• SIP extensions report the type of SIP phone registered or Unknown SIP device if no
SIP device is currently registered as that extension. Applications such as Avaya
Communicator and one-X Mobile Preferred that do not use extension records also
display Device type as Unknown SIP device.
For some types of phone, the phone can only report its general type to the system but
not the specific model. When that is the case, the field acts as a drop-drown to allow
selection of a specific model. The value selected here is also reported in other
applications such as the System Status Application, SNMP, etc.
Default Type Possible Phone Models
T7100 M7100, M7100N, T7100, Audio Conferencing Unit.
T7208 M7208, M7208N, T7208.
M7310 M7310, M7310N, T7406, T7406E.
M7310BLF M7310BLF, T7316.
M7324 M7324, M7324N.
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 Extension

Field Description
Location Specify a location to associate the extension with a physical location. Associating an
extension with a location allows emergency call routing using settings specific to that
location. The drop down list contains all locations that have been defined in the
Location page.
The display of location based time is only supported on 1100, 1200, 1600 and 9600
Series (96x0 and 96x1) phones and D100, E129 and B179 telephones.
Fallback as Remote Default = Auto.
Worker
Determines what fallback address is used for Remote Worker phone resiliency.
The options are:
• Auto: Use the fallback address configured on the IP Office Line providing the service.
• No: Use the alternate gateway private address.
• Yes: Use the alternate gateway public address.
Module This field indicates the external expansion module on which the port is located. BP
indicates an analog phone extension port on the base or control unit. BD indicates a
digital station (DS) port on the control unit. For an IP500 V2 control unit, BD and BP is
also followed by the slot number. VoIP extensions report as 0.
Port This field indicates the port number on the Module indicated above. VoIP extensions
report as 0.
Disable Default = Off (Speakerphone enabled).
Speakerphone
When selected, disables the fixed SPEAKER button if present on the phone using this
extension port. Only supported on Avaya DS, TCM and H.323 IP phones. An audible
beep is sounded when a disabled SPEAKER button is pressed. Incoming calls such as
pages and intercom calls are still connected but the speech path is not audible until the
user goes off-hook using the handset or headset. Similarly calls made or answered
using other buttons on the phone are not audible unless the user goes off-hook using
the handset or headset. Currently connected calls are not affected by changes to this
setting.

Related links
Extension on page 383

Analog
Navigation: Extension | Analog Extension | Analog
This tab contains settings that are applicable to analog extensions. These extensions are provided
by ports marked as POT or PHONE on control units and expansion modules.
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Equipment Classification:
Default = Standard Telephone.
Only available for analog extension ports. Note that changes to this setting are mergeable.
Quiet Headset On extensions set to Quiet Headset, the audio path is disabled when the extension is
idle. Ringing is presented in the audio path. Caller ID is not supported on the phone.
This option can be used with analog extensions where the handset is replaced by a
headset since in such a scenario audio is only desired when a call is connected. Since
the audio path is disabled when idle, the Quiet Headset extension cannot dial digits to
make calls. Therefore to make and answer calls this option is typically used with the
user Offhook Station (User | Telephony | Call Settings) setting which allows the
extension user to make and answer calls using applications.
Paging Speaker Used for analog ports connected to a paging amplifier. This extension will present busy
and cannot be called or be used to make calls. It can only be accessed using Dial
Paging features.
When using a UPAM connected to an analog extension port, the extension's Equipment
Classification (Extension | Analog) should be set to IVR Port and not Paging Speaker.
Standard Telephone Use for normal analog phones.
Door Phone 1/Door These two options are currently not used and so are grayed out.
Phone 2
IVR Port Used for analog ports connected to devices that require a disconnect clear signal (ie. a
break in the loop current) at the end of each call. When selected the Disconnect Pulse
Width is used.
FAX Machine If fax Relay is being used, this setting should be selected on any analog extension
connected to an analog fax machine. This setting can also be used with SIP trunks.
MOH Source If selected, the port can be used as a music on hold source in the Tones and Music
settings. An extension set as a music on hold source cannot make or receive calls. The
audio input can be monitored through the extension music on hold controls. A suitable
interface device is required to provide the audio input to the extension port. It must look
to the system like an off-hook analog phone. For example a transformer with a 600
Ohm winding (such as a Bogen WMT1A) or a dedicated MoH device with a 600Ohm
output designed for connection to a PBX extension port which is providing loop current
can be used.

Flash Hook Pulse Width

The following options are only available for analog extension ports. They define the length of loop break that will
be considered a time break recall (TBR) signal.
Use System Default = Selected (On)
Defaults
Use the default values appropriate to the system's locale.
Minimum Width Range = 20 to 2540 milliseconds.
Minimum hook flash length used if Use System Defaults is not selected. Shorter
breaks are ignored a glitches.
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 Extension

Field Description
Maximum Width Range = 30 to 2550 milliseconds.
Maximum hook flash length used if Use System Defaults is not selected. Longer
breaks are treated as clearing.
Disconnect Pulse Default = 0ms. Range = 0 to 2550ms
Width
This setting is used with analog extensions where the Equipment Classification
above has been set to IVR Port. It sets the length of loop current break used to indicate
call clearing.

Message Waiting Default = None

Lamp Indication
Allows the selection of the message waiting indication (MWI) mode for analog and IP
Type
DECT extensions.
For control unit and Phone V1 module analog extensions, the options are:
• 101V
• 51V Stepped
• 81V
• Bellcore FSK
• Line Reversal A
• Line Reversal B
• None
• On
For Phone V2 external module extensions and IP500 Phone base cards, the additional
option 101V is available.
For IP500 V2 systems, if the option Restrict Analog Extension Ringer Voltage is
selected (System | Telephony | Telephony), the MWI options are restricted to:
• Line Reversal A
• Line Reversal B
• None
Any extensions set to another option are forced to Line Reversal A.
On defaults the message waiting indication as follows using the system locale.
Locale 'On' =
Argentina, Australia, Brazil, Canada, Chile, 51V Stepped
China, Colombia, Japan, Korea, Mexico,
New Zealand, Peru, Russia, Saudi Arabia,
South Africa, Spain, United States,
Venezuela
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Configuration Mode Field Descriptions

Field Description
Bahrain, Belgium, Denmark, Egypt, On = 101V on Phone V2 modules and
Finland, France, Germany, Greece, Hong IP500 Phone cards, otherwise 81V.
Kong, Hungary, Iceland, Italy, India,
Kuwait, Morocco, Netherlands, Norway,
Oman, Pakistan, Poland, Portugal, Qatar,
Singapore, Sweden, Switzerland, Taiwan,
Turkey, United Arab Emirates, United
Kingdom
Hook Persistency: Default = 100ms. Range = 50 to 255ms.
Defines the time frame (in milliseconds) in which the system will wait before determining
that the phone is off-hook.

Related links
Extension on page 383

Extension VoIP
This tab is only available for H.323 and SIP extensions. The settings available will vary depending
on the extension type.
Related links
Extension on page 383
Extension H.323 VoIP on page 390
SIP Extension VoIP on page 394

Extension H.323 VoIP

Navigation: Extension | H323 Extension | VoIP
These settings are shown for a H.323 IP extension.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
IP Address Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address.
For phones using DHCP, the field is not updated to show the IP address being used by
the phone.
The IP Address field can be used to restrict the the source IP address that can used by
a Remote H.323 Extension. However, it should not used in the case where there is
more than one remote extension behind the domestic router.
MAC Address Default = 0000000000000 (Grayed out)
This field is grayed out and not used.
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 Extension

Field Description
Codec Selection Default = System Default This field defines the codec or codecs offered during call
setup.
The supported codecs (in default preference order) are: G.711 A-Law, G.711 U-Law,
G.722, G.729 and G.723.1. The default order for G.711 codecs will vary to match the
system's default companding setting. G.723.1 and G.729b are not supported on Linux
based systems.
The codecs available to be used are set through the System Codec list (System |
System Codec). The options are:
• System Default: This is the default setting. When selected, the codec list below show
matches the codecs set in the system wide Default Selection list (System | Codecs).
• Custom: This option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs.
TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
Supplementary Default = H450.
Services
Selects the supplementary service signaling method for use with non-Avaya IP devices.
Options are None, QSIG and H450. For H450, hold and transfer are supported. Note
that the selected method must be supported by the remote end.
Media Security Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used
for the SRTP. The options are:
• Same as System: Matches the system setting at System | VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.

Warning:
Selecting Enforced on a line or extension that does not support media security
will result in media setup failures.
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Configuration Mode Field Descriptions

Field Description
Advanced Media Not displayed if Media Security is set to Disabled. The options are:
Security Options
• Same as System: Use the same settings as the system setting configured on
System | VoIP Security.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which parts
of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to
select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
Enable FastStart for Default = Off
non-Avaya IP
A fast connection procedure. Reduces the number of messages that need to be
Phones
exchanged before an audio channel is created.
Out of Band DTMF Default = On
When on, DTMF is sent as a separate signal ("Out of Band") rather than as part of the
encoded voice stream ("In Band"). The "Out of Band" signaling is inserted back into the
audio by the remote end. This is recommended for low bit-rate compression modes
such as G.729 and G.723 where DTMF in the voice stream can become distorted.
For Avaya 1600, 4600, 5600 and 9600 Series phones, the system will enforce the
appropriate setting for the phone type.
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 Extension

Field Description
Requires DTMF Default = Off.
This field is displayed when System | VoIP | Ignore DTMF Mismatch for Phones is
set to On. It can be used to allow direct media connections between devices despite
the devices having differing DTMF setting.
When Requires DTMF is set to Off, during the checks for direct media, the system
ignores the DTMF checks if the call is between two VoIP phones. The two phones can
be located on different systems in a Server Edition or SCN deployment. Set to On if the
extension needs to receive DTMF signals.
SIP endpoints using simultaneous login, which do not have physical extensions in the
configuration, are treated by the system as not requiring DTMF.

Note:
• Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, are mismatched.
• When the system setting is set to On, the extension setting is ignored for contact
center applications. Contact center application SIP extensions are always
treated as requiring DTMF.
Local Tones Default = Off
When selected, the H.323 phones generate their own tones.
Allow Direct Media Default = On
Path
This settings controls whether IP calls must be routed via the system or can be routed
alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the need
for system resources such as voice compression channels. Both ends of the calls
must support Direct Media and have compatible VoIP settings such as matching
codec, etc. If otherwise, the call will remain routed via the system. Enabling this
option may cause some vendors problems with changing the media path mid call.
Disabling the extension’s Requires DTMF setting above allows it to attempt direct
media even if the other phone has differing DTMF settings.
• If disabled, the call is routed via the system. In that case, RTP relay support may still
allow calls between devices using the same audio codec to not require a voice
compression channel.
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Configuration Mode Field Descriptions

Field Description
Reserve License Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each
non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these
licenses are issued in the order that devices register. This option allows this extension
to be pre-licensed before the device has registered. This helps prevent a previously
licensed phone becoming unlicensed following a system restart if unlicensed devices
are also present. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
Note that when WebLM licensing is enabled, this field is automatically set to Reserve
Avaya IP Endpoint License. The None option is not available.

Related links
Extension VoIP on page 390

SIP Extension VoIP

Navigation: Extension | SIP Extension | VoIP
These settings are shown for SIP IP extensions.
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field Description
IP Address Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address.
If an address is entered, registration is only accepted from a device with that address.
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 Extension

Field Description
Codec Selection Default = System Default
This field defines the codec or codecs offered during call setup.
The available codecs in default preference order are: G.711 A-Law, G.711 ULAW, G.
729 and G.723.1. Note that the default order for G.711 codecs will vary to match the
system's default companding setting. G.723.1 is not supported on Linux based
systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary Server,
Secondary Serverand Expansion System (L) systems and on Expansion System (V2)
systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available to be used are set through the System Codec list (System |
System Codec). The options are:
• System Default: This is the default setting. When selected, the codec list below show
matches the codecs set in the system wide Default Selection list (System |
Codecs).
• Custom: This option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs.
Fax Transport Default = Off.
Support:
This option is only available if Re-Invite Supported is selected. When enabled, the
system performs fax tone detection on calls routed via the line and, if fax tone is
detected, renegotiates the call codec as configured below. The SIP line provider must
support the selected fax method and Re-Invite. The system must have available VCM
resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes.
• T38 Fallback When you enable this option, T38 is used for sending and receiving
faxes on a SIP line. If the called destination does not support T38, the system will
send a re-invite to change the transport method to G.711.
TDM | IP Gain Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain Default = Default (0dB). Range = -31dB to +31dB. Allows adjustment of the gain on
audio from the IP connection to the system TDM interface. This field is not shown on
Linux based platforms.
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Configuration Mode Field Descriptions

Field Description
DTMF Support Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The supported options are In Band, RFC2833 or Info.
3rd Party Auto Default = None.
Answer
This setting applies to 3rd party standard SIP extensions. The options are:
• RFC 5373: Add an RFC 5373 auto answer header to the INVITE.
• answer-after: Add answer-after header.
• device auto answers: IP Office relies on the phone to auto answer calls.
Media Security Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used
for the SRTP. The options are:
• Same as System: Matches the system setting at System | VoIP Security.
• Disabled: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Preferred: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
• Enforced: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.

Warning:
Selecting Enforced on a line or extension that does not support media security
will result in media setup failures.
Advanced Media Not displayed if Media Security is set to Disabled. The options are:
Security Options
• Same as System: Use the same settings as the system setting configured on
System | VoIP Security.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which parts
of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to
select SRTP_AES_CM_128_SHA1_32.
VoIP Silence Default = Off
Suppression
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends
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 Extension

Field Description
Requires DTMF Default = Off.
This field is displayed when System | VoIP | Ignore DTMF Mismatch for Phones is
set to On. It can be used to allow direct media connections between devices despite
the devices having differing DTMF setting.
When Requires DTMF is set to Off, during the checks for direct media, the system
ignores the DTMF checks if the call is between two VoIP phones. The two phones can
be located on different systems in a Server Edition or SCN deployment. Set to On if the
extension needs to receive DTMF signals.
SIP endpoints using simultaneous login, which do not have physical extensions in the
configuration, are treated by the system as not requiring DTMF.

Note:
• Direct media may still not be possible if other settings, such as codecs, NAT
settings, or security settings, are mismatched.
• When the system setting is set to On, the extension setting is ignored for contact
center applications. Contact center application SIP extensions are always
treated as requiring DTMF.
Local Hold Music Default = Off.
When enabled, the extension plays local music when on HOLD.
If Line | SIP LIne | Advanced | Local Hold Music is enabled, the extension Local
Hold Music must be disabled to play far end music to the extension.
Allow Direct Media Default = On
Path
This settings controls whether IP calls must be routed via the system or can be routed
alternatively if possible within the network structure.
• If enabled, IP calls can take routes other than through the system, removing the need
for system resources such as voice compression channels. Both ends of the calls
must support Direct Media and have compatible VoIP settings such as matching
codec, etc. If otherwise, the call will remain routed via the system. Enabling this
option may cause some vendors problems with changing the media path mid call.
Disabling the extension’s Requires DTMF setting above allows it to attempt direct
media even if the other phone has differing DTMF settings.
• If disabled, the call is routed via the system. In that case, RTP relay support may still
allow calls between devices using the same audio codec to not require a voice
compression channel.
RE-Invite Supported Default = On.
When enabled, Re-Invite can be used during a session to change the characteristics of
the session. For example, when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also support
Re-Invite. This setting must be enabled for video support.
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Configuration Mode Field Descriptions

Field Description
Codec Lockdown Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response
to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP
answer that also lists multiple codecs. This means that the user agent may switch to
any of those codecs during the session without further negotiation. The system does
not support multiple concurrent codecs for a session, so loss of speech path will occur
if the codec is changed during the session. If codec lockdown is enabled, when the
system receives an SDP answer with more than one codec from the list of offered
codecs, it sends an extra re-INVITE using just a single codec from the list and
resubmits a new SDP offer with just the single chosen codec.
Reserve License Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each
non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these
licenses are issued in the order that devices register. This option allows this extension
to be pre-licensed before the device has registered. This helps prevent a previously
licensed phone becoming unlicensed following a system restart if unlicensed devices
are also present. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
Note the following:
• When WebLM licensing is enabled, this field is automatically set to Reserve Avaya
IP Endpoint License. The Both and None options are not available.
• When the Profile of the corresponding user is set to Centralized User, this field is
automatically set to Centralized Endpoint License and cannot be changed.

Related links
Extension VoIP on page 390

Extension T38 Fax

Navigation: Extension | SIP Extension | T38 Fax
The settings are available only on IP500 V2 since it can terminate T38 fax. On the VoIP settings
for the line type, Fax Transport Support must be set to T38 or T38 Fallback.
These settings are mergeable.
Field Description
Use Default Values Default = On.
If selected, all the fields are set to their default values and greyed out.
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 Extension

Field Description
T38 Fax Version Default = 3.
During fax relay, the two gateways will negotiate to use the highest version which
they both support. The options are:
• 0
• 1
• 2
• 3
Transport Default = UDPTL (fixed).
Only UDPTL is supported. TCP and RTP transport are not supported. For
UDPTL, redundancy error correction is supported. Forward Error Correction
(FEC) is not supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased
redundancy increases the bandwidth required for the fax transport.
Low Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for low speed
V.21 T.30 fax transmissions.
High Speed Default = 0 (No redundancy). Range = 0 to 5.
Sets the number of redundant T38 fax packets that should be sent for V.17, V.27
and V.28 fax transmissions.

TCF Method Default = Trans TCF. TCF = Training Check Frame.

Max Bit Rate (bps) Default = 14400.
Lower rates can be selected if the current rate is not supported by the fax
equipment or is found to not be reliable.
EFlag Start Timer Default = 2600.
(msecs)
EFlag Stop Timer Default = 2300.
(msecs)
Tx Network Timeout Default = 150.
(secs)
Scan Line Fix-up Default = On.
TFOP Enhancement Default = On.
Disable T30 ECM Default = Off.
When selected, disabled the T.30 Error Correction Mode used for fax
transmission.
Disable EFlags For Default = Off.
First DIS
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Configuration Mode Field Descriptions

Field Description
Disable T30 MR Default = Off.
Compression

NSF Override Default = Off.

If selected, the NSF (Non-Standard Facility) information sent by the T38 device
can be overridden using the values in the fields below.
Country Code: Default = 0.
Vendor Code: Default = 0.

Related links
Extension on page 383

IP DECT Extension
Navigation: Extension | IP DECT Extension
IP DECT extensions are created manually after an IP DECT line has been added to the
configuration or added automatically as DECT handsets subscribe to the DECT system.
These settings are mergeable with the exception of the Reserve License setting. Changing the
Reserve License settings requires a reboot of the system.
Field Description
DECT Line ID Use the drop-down list to select the IP DECT line from the system to the Avaya IP
DECT system.
Message Waiting Default = On
Lamp Indication
Allows selection of the message waiting indication to use with the IP DECT extension.
Type
The options are:
• None
• On
Reserve License Default = None.
Avaya IP phones require an Avaya IP Endpoint license in order to register with the
system. Normally licenses are issued in the order that devices register. This option
allows this extension to be pre-licensed before the device has registered. The options
are
• Reserve Avaya IP Endpoint License
• None
Note that when WebLM licensing is enabled, this field is automatically set to Reserve
Avaya IP Endpoint License and cannot be changed.

The additional fields below depend on whether the IP DECT line has Enable Provisioning
selected.

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 User

Field Description
Enable Provisioning Not Selected
Handset Type Default = Unknown
Correct selection of the handset type allows application of appropriate settings for the
handset display and buttons. Selectable handset types are 3720, 3725, 3740, 3749 or
Unknown.
Enable Provisioning Selected
IPEI Default = 0
This field, if set to a value other than 0, sets the IPEI number of the handset that is able
to subscribe to the DECT R4 system using this extension number. The IPEI for each
DECT handset is unique.
Use Handset Default = Off.
Configuration
If Use Handset Configuration. is selected, the handset user is able to set the phone
language and date/time format. If not selected, those settings will be driven by the
system or user locale settings in the system configuration.

Related links
Extension on page 383

SIP DECT Base Extension

Navigation: Extension | SIP DECT Base
This tab is displayed for SIP DECT extensions.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
DECT Line ID Use the drop-down list to select the SIP DECT Line from the system to the base
station.

Related links
Extension on page 383

User
Additional configuration information
This section provides the User field descriptions.
For additional configuration information, see Configure User Settings on page 651.
Related links
Configuration Mode Field Descriptions on page 186

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Configuration Mode Field Descriptions

User on page 402

Voicemail on page 408
User | DND on page 413
Short Codes on page 414
Source Numbers on page 415
Telephony on page 421
User | Forwarding on page 433
Dial In on page 436
Voice Recording on page 437
Button Programming on page 439
User | Menu Programming on page 440
Mobility on page 441
Group Memberships on page 445
Announcements on page 445
SIP on page 446
Personal Directory on page 447
Web Self Administration on page 449

User
Navigation: User | User
Additional configuration information
• For a summary of user management, including a description of centralized users, see User
Management Overview on page 651.

This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Users are the people who use the system or are Dial In users for data access. A system User may
or may not have an Extension Number that physical exists - this is useful if users do not require a
physical extension but wish to use system features, for example voicemail, forwarding, etc.
NoUser is used to apply settings to extensions which have no associated user. Remote Manager
is used as the default settings for dial in connections.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
— Except adding/removing centralized branch users which requires a system reboot.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.

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 User

Field Description
Name Range = Up to 15 characters.
This is the user's account name used for RAS Dial In, Caller Display and voicemail
mailbox. As the display on Caller Display telephones is normally only 16 digits long it is
useful to keep the name short. Only alphanumeric characters and space are supported in
this field. This field is case sensitive and must be unique.
Names should not start with a space. Do not use punctuation characters such as #, ?, /, ^,
> and ,.
Voicemail uses the name to match a user to their mailbox. Changing a user's name will
route their voicemail calls to a new mailbox. Note however that Voicemail Pro is not case
sensitive and will treat names such as "Steve Smith", "steve smith" and "STEVE SMITH"
as being the same.
Do not provision a user with the Name "admin". The user name "admin" is a reserved
value on the one-X Portal Instant Message (IM) and Presence server. An IP Office
"admin" user will not have IM and presence services.
For Outbound Contact Express deployments, when an agent logs in to an extension, the
user name associated with the extension is changed to the agent ID.
Full Name Default = Blank
Use this field to enter the user's full name. The recommended format is <first
name><space><last name> in order for this value to be used correctly by voicemail dial
by name features. When set, the Full Name is used in place of the Name for display by
phones and user applications. Names should not start with a space. Do not use
punctuation characters such as @, #, ?, /, ^, > and ,.
Password Default = Blank. Range = Up to 31 alphanumeric characters.
This password is used by user applications such as SoftConsole and TAPI. It is also used
for user's with Dial In access.
Note that this is not the user's voicemail mailbox password (see User | Voicemail |
Voicemail Code) or their phone log in code (see User | Telephony | Supervisor
Settings | Login Code).
Password complexity rules can be set through the General security settings. If complexity
is not met, an error is displayed. The configuration can still be saved, except if system
locale is set to France2.
Confirm Password Enter and confirm the new password. Note that an error will be indicated if the password
being entered does not meet the password rules.
Unique Identity Default = Blank.
A Google for Work account email address for the user. The address must be unique for
each user. The account is used for:
• Avaya Communicator for Web client login
• Gmail voicemail to email messages
To use Gmail for voicemail to email, set User | Voicemail | Enable Gmail API to On.
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Configuration Mode Field Descriptions

Field Description
Extension Range = 2 to 15 digits.
In general all extensions should have the same number of digits. This setting can be left
blank for users used just for dial in data connections.
• Users associated with IP phones or who may log in as such devices should not be
given extension numbers greater than 7 digits.
• Centralized users’ extension numbers can be up to 13 digits in length. Although IP
Office supports extension numbers up to 15 digits, the 13-digit length is determined by
the maximum extension number length allowed for provisioning Centralized users in
Communication Manager.
Account Status Default = Enabled.
Use this setting to Enable or Disable a user account.
You can also require a password reset by selecting Force New Password. A user can
only set a new password through the one-X Portal user interface. This option should not
be used if one-X Portal is not available.
The Account Status can also be Locked - Password Error or Locked - Temporary.
The user account enters these states automatically based on the password settings
configured in the Security Settings General tab. If a user exceeds the Password Reject
Limit, then the Password Reject Action is implemented. If the Password Reject Action
is Log and Disable Account, then the account status is changed to Locked - Password
Error. If the Password Reject Action is Log and Temporary Disable, then the account
status is changed to Locked - Temporary.
Profile Default = Basic User.
A user's profile controls whether they can be configured for a number of features.
The table below lists the different user profiles and the features accessible by each
profile. Setting a user to a particular profile enables those features by default, however
they can be manually disabled if necessary. The number of users that can be configured
for each profile is controlled by the user licenses present in the configuration.
A Non-licensed User is allowed dial in access and paging and can be used as a Music
on Hold or Analog paging port.
Except for a Basic User, a Preferred Edition system license is a pre-requisite for any
user profile licenses. In a multi-site network, the Preferred Edition license of the central
system is automatically shared with other systems in the network, enabling user profile
licenses on those other systems. However, each system supporting a Voicemail Pro
server still requires its own Preferred Edition license for Voicemail Pro operation.
Locale Default = Blank (Use system locale)
Configures the language used for voicemail prompts played to the user, assuming the
language is available on the voicemail server. See Avaya IP Office™ Platform Locale
Settings. On a digital extension it also controls the display language used for messages
from the system. Note however that some phones have their own menu options for the
selected language for the phone menus.
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 User

Field Description
Priority: Default = 5. Range = 1 (Lowest) to 5 (Highest)
This setting is used by ARS.
Login code
Confirm Login
code
Audio Conference Default = Blank. Range = Up to 15 numeric characters.
PIN
Use this field to configure PIN access for meet me conferences.
An L in this field indicates that the unscheduled meet-me conference feature is disabled
for this user.
Confirm Audio Enter and confirm audio conference PIN.
Conference PIN
System Phone Default = None.
Rights
Users set as a system phone user are able to access additional functions. The settings
are:
• None: The user cannot access any system phone options.
• Level 1: The user can access all system phone options supported on the type of phone
they are using except system management and memory card commands.
• Level 2: The user can access all system phone options supported on the type of phone
they are using including system management and memory card commands. Due to the
nature of the additional commands a login code should be set for the user to restrict
access.
Device Type This field shows the type of phone at which the user is current logged in. If the user is
logged out but is associated with a Base Extension, the device type for that extension
port is shown. If the user has been logged out and is not associated with a Base
Extension, the device type is listed as Device Type Unknown.

Feature Standard Mode Server Edition

Non- Basic Office Telewor Mobile Power Basic Office Power
licensed User Worker ker Worker User User Worker User
User [1]
one-X – – Yes Yes – Yes – Yes Yes
Portal
Services
Telecom – – – Yes – Yes – – Yes
muter
options
UMS – – Yes Yes – Yes – Yes Yes
Web
Services
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Configuration Mode Field Descriptions

Feature Standard Mode Server Edition

Non- Basic Office Telewor Mobile Power Basic Office Power
licensed User Worker ker Worker User User Worker User
User [1]
TTS for – – – – Yes Yes – – Yes
Email
Reading
Remote – – – Yes – Yes Yes Yes Yes
Worker
[2]
Avaya – – Yes Yes – Yes Yes Yes Yes
Commu
nicator
[3]
WebRTC – – Yes – – Yes – Yes Yes

User Profile Notes:

1. Non-licensed users can be created on both Standard Mode and Server Edition systems.
2. The system supports users using remote H.323 or SIP extensions. On non-Server Edition
systems, up to 4 users are supported as remote extensions without needing to be
configured and licensed for a user profile. Additional remote users are supported if
licensed and configured for either a Teleworker or Power User user profile. On Server
Edition systems, the remote worker is supported for all user profiles.
3. Supported for advanced Avaya Communicator for IP Office usage if one-X Portal and
Voicemail Pro applications are also installed. If otherwise, only basic Avaya Communicator
for IP Office usage is supported.
Note:
To upgrade an Office Worker or Mobile Worker to a Power User when no additional Office
Worker or Mobile Worker licenses are available, you must first set the user Profile to Basic
User. Once the user Profile has been set to Basic User, the Power User option is available
in the drop down menu.
Field Description
Enable Desktop/ Default = Off.
Tablet Voip client
This option allows users to use Avaya Communicator for IP Office or Avaya Equinox™ for
IP Office as their current telephone device on Windows or macOS operating systems. It
can be enabled for users whose Profile is set to Officeworker, Power User, or
Teleworker. On non-Server Editionsystems, Avaya Communicator can be enabled for
Basic User or Mobile Worker using Avaya Softphone licenses.
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 User

Field Description
Enable Remote Default = Off.
Worker
Indicates whether the user is allowed to use an H.323 or SIP remote extension.
Supported for up to 4 Basic users plus any users licensed and configured as Teleworker
and or Power User user profiles. On Server Edition systems, all user types can be
Remote Workers.
If the user's Extension Number matches the Base Extension setting of an IP
extension, the H.323 Remote Extn Enable setting of that extension is automatically
changed to match the user's Enable Remote Worker setting and vice versa.
The Enable Remote Worker option does not need to be enabled for users with SIP
phones if an Avaya Session Border Controller for Enterprise (ASBCE) is deployed in the
network to allow remote workers to register their SIP phone from a remote location.
Enable Mobile VoIP Default = Off.
Client
This option allows the users to use Avaya Communicator for IP Office or Avaya Equinox™
for IP Office as their current telephone device on Android and iOS operating systems. It
can be enabled for users whose Profile is set to Power User.
Exclude From Default = Off
Directory
When on, the user does not appear in the directory list shown by the user applications
and on phones with a directory function. For users logging on as agents in an Outbound
Contact Express deployment, Exclude From Directory must be Off.
Hunt Group
Membership
Incoming Number
User Rights
User Rights View This field affects Manager only. It allows you to switch between displaying the user
settings as affected by their associated Working Hours User Rights or Out of Hours
User Rights.
Working Hours Default = <None> (Continuous).
Time Profile
If set, the selected time profile defines when the user's Working Hours User Rights are
applied. Outside the time profile, the user's Out of Hours User Rights are applied
Working Hours Default = Blank (No rights restrictions).
User Rights
This field allows selection of user rights which may set and lock some user settings. If a
Working Hours Time Profile has been selected, the Working Hours User Rights are
only applied during the times defined by that time profile, otherwise they are applied at all
times.
Out of Hours User Default = Blank (No rights restrictions).
Rights
This field allows selection of alternate user rights that are used outside the times defined
by the user's Working Hours Time Profile.

Related links
User on page 401

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Configuration Mode Field Descriptions

Voicemail
Navigation: User | Voicemail
Additional configuration information
The Enable Gmail API setting is used to configure Gmail Integration. For additional information,
see Configuring Gmail Integration on page 657.
Configuration settings
If a voicemail server application is being used on your system, each user has use of a voicemail
mailbox. You can use this form to enable this facility and various user voicemail settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.

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 User

Field Description
Voicemail Code Default = Blank. Range = 0 (no code) to 31 digits.
A code used by the voicemail server to validate access to this mailbox. If remote access is
attempted to a mailbox that has no voicemail code set, the prompt "Remote access is not
configured on this mailbox" is played.
The mailbox access code can be set through IP Office Manager or through the mailbox
telephone user interface (TUI). The minimum password length is:
• Voicemail Pro (Manager): 0
• Voicemail Pro (Intuity TUI): 2
• Embedded Voicemail (Manager): 0
• Embedded Voicemail (Intuity TUI): 0
Codes set through the Voicemail Pro telephone user interface are restricted to valid
sequences. For example, attempting to enter a code that matches the mailbox extension,
repeat the same number (111111) or a sequence of numbers (123456) are not allowed. If
these types of code are required they can be entered through Manager.
Manager does not enforce any password requirements for the code if one is set through
Manager.
• Embedded Voicemail: For Embedded Voicemail running in IP Office mailbox mode, the
voicemail code is used if set.
• IP Office mode: The voicemail code is required when accessing the mailbox from a
location that is not set as a trusted number in the user's Source Numbers list.
• Intuity Emulation mode: By default the voicemail code is required for all mailbox
access. The first time the mailbox is accessed the user will be prompted to change the
password. Also if the voicemail code setting is left blank, the caller will be prompted to
set a code when they next access the mailbox. The requirement to enter the voicemail
code can be removed by adding a customized user or default collect call flow, refer to
the Voicemail Pro manuals for full details.
• Trusted Source Access: The voicemail code is required when accessing the mailbox
from a location that is not set as a trusted number in the user's Source Numbers list.
• Call Flow Password Request: Voicemail Pro call flows containing an action where the
action's PIN code set to $ will prompt the user for their voicemail code.
• Changing the Code: All of the voicemail interfaces, except IMS and IMAP, provide
options for the user to change the voicemail code themselves. In addition, Voicemail Pro
running in Intuity emulation mode will request that the user sets a code when they first
log in to their mailbox using the phone.
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Configuration Mode Field Descriptions

Field Description
Voicemail On Default = On.
When on, the mailbox is used by the system to answer the user's unanswered calls or
calls when the user's extension returns busy. Note that selecting off does not disable use
of the user's mailbox. Messages can still be forward to their mailbox and recordings can
be placed in it. The mailbox can also still be accessed to collect messages.
When a caller is directed to voicemail to leave a message, the system indicates the target
user or hunt group mailbox.
• The mailbox of the originally targeted user or hunt group is used. This applies even if the
call has been forwarded to another destination. It also includes scenarios where a hunt
group call overflows or is in fallback to another group.
• Voicemail Pro can be used to customize which mailbox is used separately from the
mailbox indicated by the system.
Voicemail Help Default = Off
This option controls whether users retrieving messages are automatically given an
additional prompt "For help at any time press 8." If switched off, users can still press 8 for
help. For voicemail systems running in Intuity emulation mode, this option has no effect.
On those systems the default access greeting always includes the prompt "For help at any
time, press *4" (*H in the US locale).
Voicemail Default = Off
Ringback
When enabled and a new message has been received, the voicemail server calls the
user's extension to attempt to deliver the message each time the telephone is put down.
Voicemail will not ring the extension more than once every 30 seconds.
Voicemail Email Default = Off
Reading
This option can be enabled for users whose Profile is set to Mobile Worker or Power
User. If enabled, when you log into you voicemail box, it will detect your email messages
and read them to you. This email text to speech feature is set-up through Voicemail Pro.
This option is not currently supported with Linux based Voicemail Pro.
UMS Web Default = On.
Services
For Server Edition systems this option can be enabled for users whose Profile is set to
Office Worker or Power User. For standalone systems the option can be enabled for
users whose Profile is set to Teleworker, Office Worker or Power User. When selected,
the user can use any of the Voicemail Pro UMS services to access their voicemail
messages (IMAP email client, web browser or Exchange 2007 mailbox). Note that the
user must have a voicemail code set in order to use the UMS services.
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 User

Field Description
Enable Gmail API Default = Off. Available only on Server Edition systems.
Before you can enable this setting, UMS Web Services must be set to On for the user.
When set to On:
• The Voicemail Email setting is disabled.
• The Voicemail Email Mode options (Off, Copy, Forward, Alert) are available.
All voicemail to email actions are performed using the Gmail address defined in the setting
User | User | Unique Identity.
Voicemail Email Default = Blank (No voicemail email features)
This field is used to set the user or group email address used by the voicemail server for
voicemail email operation. When an address is entered, the additional Voicemail Email
control below are selectable to configure the type of voicemail email service that should be
provided.
Use of voicemail email requires the Voicemail Pro server to have been configured to use
either a local MAPI email client or an SMTP email server account. For Embedded
Voicemail, voicemail email is supportedand uses the system's SMTP settings.
The use of voicemail email for the sending (automatic or manual) of email messages with
wav files attached should be considered with care. A one-minute message creates a
1MB .wav file. Many email systems impose limits on emails and email attachment sizes.
For example the default limit on an Exchange server is 5MB.

Note:
Unicode characters are not supported.
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Configuration Mode Field Descriptions

Field Description
Voicemail Email Default = Off
Mode
the Voicemail Email Mode options become selectable when
• A Voicemail Email email address is entered for the user or group
• The Enable Gmail API is set to On
These settings control the mode of automatic voicemail email operation provided by the
voicemail server whenever the voicemail mailbox receives a new voicemail message.
Users can change their voicemail email mode using visual voice. The ability to change the
voicemail email mode can also be provided by Voicemail Pro in a call flow using a Play
Configuration Menu action or a Generic action.
If the voicemail server is set to IP Office mode
• Users can change their voicemail email mode through the telephone prompts.
• users can manually forward a message to email.
The options are:
• Off If off, none of the options below are used for automatic voicemail email. Users can
also select this mode by dialing *03 from their extension.
• Copy If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a copy of the message is attached to an email and sent to the email
address. There is no mailbox synchronization between the email and voicemail
mailboxes. For example reading and deletion of the email message does not affect the
message in the voicemail mailbox or the message waiting indication provided for that
new message.
• Forward If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, that message is attached to an email and sent to the email address.
No copy of the voicemail message is retained in the voicemail mailbox and their is no
message waiting indication. As with Copy, there is no mailbox synchronization between
the email and voicemail mailboxes. Users can also select this mode by dialing *01 from
their extension.
Note that until email forwarding is completed, the message is present in the voicemail
server mailbox and so may trigger features such as message waiting indication.
• UMS Exchange 2007 With Voicemail Pro, the system supports voicemail email to an
Exchange 2007 server email account. For users and groups also enabled for UMS Web
Services this significantly changes their mailbox operation. The Exchange Server inbox
is used as their voicemail message store and features such as message waiting
indication are set by new messages in that location rather than the voicemail mailbox on
the voicemail server. Telephone access to voicemail messages, including Visual Voice
access, is redirected to the Exchange 2007 mailbox.
• Alert If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a simple email message is sent to the email address. This is an email
message announcing details of the voicemail message but with no copy of the voicemail
message attached. Users can also select this mode by dialing *02 from their extension.
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 User

Field Description
DTMF Breakout
When a caller is directed to voicemail to leave a message, they can be given the option to be transferred to a
different extension. The greeting message needs to be recorded telling the caller the options available. The
extension numbers that they can be transferred to are entered in the fields below.System default values can be
set for these numbers and are used unless a different number is set within these user settings. The values can
be set using User Rights.
The Park & Page feature is supported when the system voicemail type is configured as Embedded Voicemail
or Voicemail Pro. Park & Page is also supported on systems where Avaya Aura Messaging, Modular
Messaging over SIP, or CallPilot (for Enterprise Branch with CS 1000 deployments) is configured as the central
voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto attendant operation. The
Park & Page feature allows a call to be parked while a page is made to a hunt group or extension. This feature
can be configured for Breakout DTMF 0, Breakout DTMF 2, or Breakout DTMF 3.
Reception/ The number to which a caller is transferred if they press 0while listening to the mailbox
Breakout (DTMF greeting rather than leaving a message (*0 on Embedded Voicemail in IP Office mode).
0)
For voicemail systems set to Intuity emulation mode, the mailbox owner can also access
this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave Mail
action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success results
connections depending on whether the caller pressed 0 before or after the record tone.
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0) setting.
When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
• Paging Number – displays a list of hunt groups and users (extensions). Select a hunt
group or extension to configure this option.
• Retries – the range is 0 to 5. The default setting is 0.
• Retry Timeout – provided in the format M:SS (minute:seconds). The range can be set
in 15-second increments. The minimum setting is 15 seconds and the maximum setting
is 5 minutes. The default setting is 15 seconds
Breakout (DTMF The number to which a caller is transferred if they press 2while listening to the mailbox
2) greeting rather than leaving a message (*2 on Embedded Voicemail in IP Office mode).
Breakout (DTMF The number to which a caller is transferred if they press 3while listening to the mailbox
3) greeting rather than leaving a message (*3 on Embedded Voicemail in IP Office mode).

Related links
User on page 401

User | DND
Navigation: User | DND

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Configuration Mode Field Descriptions

Do not disturb prevents the user from receiving hunt group and page calls. Direct callers hear
busy tone or are diverted to voicemail if available. It overrides any call forwarding, follow me and
call coverage settings. A set of exception numbers can be added to list numbers from which the
user still wants to be able to receive calls when they have do not disturb in use.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Do Not Disturb Default = Off
When checked the user's extension is considered busy, except for calls coming from
sources listed in their Do Not Disturb Exception List. When a user has do not disturb in
use, their normal extension will give alternate dialtone when off hook. Users with DND on
are indicated as 'busy' on any BLF indicators set to that user.
Do Not Disturb Default = Blank
Exception List
This is the list of telephone numbers that are still allowed through when Do Not Disturb is
set. For example this could be an assistant or an expected phone call. Internal extension
numbers or external telephone numbers can be entered. If you wish to add a range of
numbers, you can either enter each number separately or make use of the wildcards "N"
and "X" in the number. For example, to allow all numbers from 7325551000 to
7325551099, the DND Exception number can be entered as either 73255510XX or
73255510N. Note that this list is only applied to direct calls to the user.
Calls to a hunt group of which the user is a member do not use the Do Not Disturb
Exceptions list.

Related links
User on page 401

Short Codes
Navigation: User | Short Codes
Additional configuration information
For additional configuration information on short codes, see Short Code Overview on page 788.
Configuration settings
Short codes entered in this list can only be dialed by the user. They will override any matching
user rights or system short code.
User and User Rights short codes are only applied to numbers dialed by that user. For example
they are not applied to calls forwarded via the user.

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 User

Warning:
User dialing of emergency numbers must not be blocked by the addition of short codes. If
short codes are added, the users ability to dial emergency numbers must be tested and
maintained.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
*FWD:
Short codes of this form are inserted by the system. They are used in conjunction with the User |
Forwarding settings to remember previously used forwarding numbers. They can be accessed on
that tab by using the drop-down selector on the forwarding fields.
*DCP:
Short codes of this form are often inserted by the system. They are used by some phone types to
contain settings relating to functions such as ring volume and auto answer. Deleting such short
codes will cause related phone settings to return to their defaults.
*DCP/Dial/8xxxxxxx, 0, 1, 1, 0/0:
For system's with TCM phone ports, when a phone is first connected to the port, the button
programming of the associated user is overwritten with the default button programming
appropriate for the phone model. Adding the above short code prevents that behavior if not
required, for example if a pre-built configuration including user button programming is added to the
system before the connection of phones.
Related links
User on page 401

Source Numbers
Navigation: User | Source Numbers
This page is used to enter values that have special usages. These are entered using the Add,
Edit and Remove buttons.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
User Source Numbers
The following types of records can be added to a user's source numbers:

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Configuration Mode Field Descriptions

Value Description
AT<string> Strings beginning with AT can be used with a user called DTEDefault to
configure the default settings of the control unit's DTE port.
BST_MESSAGE_FOR_YOU If set, then the BST phone user sees the top line Message for you or
Messages for you, indicating that voicemail messages are present. This source
number can be used as a NoUser source number to enable the feature for all
users.
BST_NO_MESSAGE_FOR_ If the source number above has been used as a NoUser source number to
YOU enable the feature for all BST users, this individual user source number can be
used to disable the feature for selected users. If set, the user does not see a
message indication when the NoUser setting BST_MESSAGE_FOR_YOU is
set. The user's phone presents the idle date/time in the normal fashion.
Enable_OTT Enable one touch transfer operation for the user.
H<Group_Name> Allows the user to receive message waiting indication of new group messages.
Enter H followed by the group name, for example HMain. The group is added to
the user’s Visual Voice menu.
On suitable display extensions, the hunt group name and number of new
messages is displayed. Refer to the appropriate telephone user guide.
If the user is not a member of the group, a voicemail code must be set for the
group's mailbox. See the setting Group | Voicemail | Voicemail Code .
P<Telephone Number> This record sets the destination for callback (outbound alert) calls from
voicemail. Enter Pfollowed by the telephone number including any necessary
external dialing prefix, for example P917325559876. This facility is only
available when using Voicemail Pro through which a default Callback or a user
specific Callback start point has been configured. Refer to the Voicemail Pro
documentation. This feature is separate from voicemail ringback and Voicemail
Pro outcalling.
R<Caller's ICLID> To allow Dial In/RAS call access only from a specified number prefix the number
with a "R", for example R7325551234.
U<User_Name or Allows the user to receive message waiting indication of new messages. Enter
Extension#> U followed by the user name or extension number, for example U201. The
specified user is added to the user’s Visual Voice menu.
On suitable display extensions, the user name and number of new messages is
displayed. Refer to the appropriate telephone user guide.
If the user is not a trusted source and a Voicemail Code exists, the user must
enter the Voicemail Code corresponding to the monitored mailbox.
V<Caller's ICLID> Strings prefixed with a V indicate numbers from which access to the users
mailbox is allowed without requiring entry of the mailbox's voicemail code. This
is referred to as "trusted source".
For Voicemail Pro running in Intuity mode, trusted source is used for calls from
programmable buttons set to Voicemail Collect and Visual Voice. Other controls
are prompted for the mailbox number and then password.

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 User

NoUser Source Numbers

The following source numbers can also be used on the Source Numbers tab of the NoUser user.
These affect all users on the system.
Note:
Changes to these source numbers require a system reboot to become effective.
Value Description
ATM4U_PCS7_RINGDE For some cellular or mobile interfaces connected to a IP500 ATM4U card, the card
TECT may not detect the ring signal. For PCS4 and higher card this NoUser source
number can be used activate alternate ring detection. Refer to IP Office Technical
Tip 204.
ALLOW_5410_UPGRA Previously the only control over the upgrading of 5410 phones was controlled by the
DES use of the turn_on.bat and turn_off.bat batch files installed with the Manager
application. Now in addition, this option must be present for 5410 phones to update
their firmware. Refer to the IP Office Installation manual for full details.
B_DISABLE_SIP_IPAD Disables the blacklisting of SIP device registration based on the device IP address.
DR Refer to the IP Office Security Guidelines document for more details.
BST_MESSAGE_FOR_ If set, all BST phones display the top line Message for you or Messages for you,
YOU indicating that voicemail messages are present.
DECT_REVERSE_RING By default, when this parameter is not set, calls on DECT phones associated with a
CTI application will ring as a Priority call. When this parameter is set, DECT phones
ring as a normal, external or internal, call.
DISTINCT_HOLD_RING Used to display a specific message about the call type for calls returning after timing
BACK out from being parked or held. If set, such calls display Return Call - Held or Return
Call – Parked rather than connected party name or line name.
Enable_OTT Enable one touch transfer for all users.
FORCE_HANDSFREE_ If set, when using the handsfree announced transfer process, both the transfer
TRANSFER enquiry and transfer completion calls are auto-answered. Without this setting only
the transfer enquiry call is auto-answered.
HIDE_CALL_STATE Used to hide the call status information, for example Dial, Conn, etc, on DS phones.
Used in conjunction with the LONGER_NAMES option. Not supported for 1600 and
9600 Series phones.
LONGER_NAMES Used to increase the length of names sent for display on older DS phones, i.e. 2400,
4400 and 5400 Series.
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Configuration Mode Field Descriptions

Value Description
MEDIA_NAT_DM_INTE Used in conjunction with the setting System | VoIP | Allow Direct Media Within
RNAL=X NAT Location
When Allow Direct Media Within NAT Location is set to on, The default behavior
is to allow direct media between all types of devices (H323 and SIP remote workers
and IP Office Lines behind a NAT). In the case of routers that have H323 or SIP
ALG, it can be desirable to allow direct media only between certain categories of
devices. In this case, set this NoUser user source number where X is a hex number
defined as a combination of the following flags:
• 0x01 (includes H323 phones)
• 0x02 (includes SIP phones)
• 0x04 (includes IP Office Lines)
For example, if the router has SIP ALG that can't be disabled, you might want to
disable direct media for SIP devices. To configure, set
MEDIA_NAT_DM_INTERNAL=5 to include only H323 phones and IP Office Lines.
NI2_CALLED_PARTY_ X = UNKNOWN or ISDN
PLAN=X
Forces the NI2 Called Party Numbering plan for ETSI PRI trunks.
NI2_CALLED_PARTY_ X = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER
TYPE=X
Forces the NI2 Called Party Numbering type for ETSI PRI trunks.
NI2_CALLING_PARTY_ X = UNKNOWN or ISDN
PLAN=X
Forces the NI2 Calling Party Numbering plan for ETSI PRI trunks.
NI2_CALLING_PARTY_ X = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER
PLAN=X
Forces the NI2 Calling Party Numbering type for ETSI PRI trunks.
NO_DIALLED_REF_EX On outgoing external calls made using short codes to dial the full number, only the
TERNAL short code dialed is displayed on the dialing user's phone and any directory
matching is based on that number dialled. On systems with this source number
added to the configuration, after dialing a short code the full number dialled by that
short code is shown and directory matching is based on that full number.
onex_l1=X Sets the IP address of the one-X server that can be accessed by clients registered
on the LAN1 interface.
onex_l2=X Sets the IP address of the one-X server that can be accessed by clients registered
on the LAN2 interface.
onex_port_l1=X Sets the port of the one-X server that can be accessed by clients registered on the
LAN1 interface.
onex_port_l2=X Sets the port of the one-X server that can be accessed by clients registered on the
LAN2 interface.
onex_port_r1=X Sets the port of the one-X server that can be accessed by remote clients registered
on the LAN1 interface.
onex_port_r2=X Sets the port of the one-X server that can be accessed by remote clients registered
on the LAN2 interface.
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 User

Value Description
onex_r1=X Sets the IP address of the one-X server that can be accessed by remote clients
registered on the LAN1 interface.
onex_r2=X Sets the IP address of the one-X server that can be accessed by remote clients
registered on the LAN2 interface.
PRESERVED_CONN_D X = time in minutes. Range = 1 to 120.
URATION=X
When the setting System | Telephony | Telephony | Media Connection
Preservation is enabled, preserved calls have a maximum duration of 120 minutes.
After that time, they are hung up. Use this setting to change the maximum duration
value.
PRESERVED_NO_MED X = time in minutes. Range = 1 to 120.
IA_DURATION=X
When the setting System | Telephony | Telephony | Media Connection
Preservation is enabled, preserved calls have a maximum duration of 120 minutes.
If monitoring RTP or RTCP and no speech is detected, calls are hung up after 10
minutes. Use this setting to change the default value of 10 minutes.
ProgressEndsOverlapS See Line | VoIP.
end
REPEATING_BEEP_ON By default, if you set Beep on Listen and invoke Call Listen you'll hear an entry tone
_LISTEN (3 beeps). When this parameter is set, you hear a beep every 10 seconds when you
invoke Call Listen.
RTCP_COLLECTOR_IP X = IP address of the IP Office system as configured in the Prognosis server.
=X
RW_SBC_REG=<SBC- Used for Remote Worker Session Boarder Controller Enterprise
B1-public-SIP-IPaddr> (SBCE) configuration on 11xx, 12xx, and E129 phones. The IP address is used as a
S1/S2 for 11xx and 12xx and for outbound-proxy-server for E129 sets.
RW_SBC_PROV=<SBC Used for Remote Worker Session Boarder Controller Enterprise
-B1-private-HTTP/S- (SBCE) configuration on 11xx, 12xx, and E129 phones. The IP address is used to
IPaddr> determine whether a 11xx, 12xx, or E129 set is registered as an IP Office SBCE
Remote Worker.
RW_SBC_TLS=<SBC- Used for Remote Worker Session Boarder Controller Enterprise
public-TLS-port> (SBCE) configuration on 11xx, 12xx, and E129 phones. The port is used as a S1/
S2 TLS port for 11xx and 12xx phones and as outbound-proxy-server TLS port for
E129 phones.
RW_SBC_TCP=<SBC- Used for Remote Worker Session Boarder Controller Enterprise
public-TCP-port> (SBCE) configuration on 11xx, 12xx, and E129 phones. The port is used as a S1/
S2 TCP port for 11xx and 12xx phones and as outbound-proxy-server TCP port for
E129 phones.
RW_SBC_UDP=<SBC- Used for Remote Worker Session Boarder Controller Enterprise
public-UDP-port> (SBCE) configuration on 11xx, 12xx, and E129 phones. The port is used as a S1/
S2 UDP port for 11xx and 12xx phones and as outbound-proxy-server UDP port for
E129 phones.
Table continues…

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Configuration Mode Field Descriptions

Value Description
SET_46xx_PROCPSWD X= New password
=X
When set, the new password is indicated to phones through the auto-generated
settings file.
SET_96xx_SIG=X When set, inserts the line “SET SIG X into the auto-generated settings files.
SET_HEADSYS_1 If set, alters the operation of the headset button on 96x1 phones via the auto-
generated settings file. Normally the headset goes off-hook when the far end
disconnects. When this option is set, the headset remains on-hook when the far end
disconnects.
SIP_E129_PREFER_UD When set, the auto-generated E129 configuration file is altered to set the transport
P method as UDP regardless of whether TCP or TLS are selected on the LAN1/LAN2
VoIP configuration settings.
SIP_ENABLE_HOT_DE For IP Office Release 10.1, by default the use of hot-desking on J129 and H175
SK phones is blocked. This source numbers overrides that behavior.
SIP_EXTN_CALL_Q_TI X = Number of minutes (0 (no limit) to 255).
MEOUT=X
Sets the unanswered call duration after which unanswered SIP calls are
automatically disconnected. If not set, the normal default is 5 minutes.
SIP_OPTIONS_PERIOD X = time in minutes. The system sends SIP options messages periodically to
=X determine if the SIP connection is active. The rate at which the messages are sent is
determined by the combination of the Binding Refresh Time (in seconds) set on the
Network Topology tab and the SIP_OPTIONS_PERIOD parameter (in minutes). The
frequency of sent messages is determined as follows:
If no SIP_OPTIONS_PERIOD parameter is defined and the Binding Refresh Time
is 0, then the default value of 300 seconds is used.
To establish a period less than 300 seconds, do not define a
SIP_OPTIONS_PERIOD parameter and set the Binding Refresh Time to a value
less than 300 seconds. The OPTIONS message period will be equal to the Binding
Refresh Time.
To establish a period greater than 300 seconds, a SIP_OPTIONS_PERIOD
parameter must be defined. The Binding Refresh Time must be set to a value
greater than 300 seconds. The OPTIONS message period will be the smaller of the
Binding Refresh Time and the SIP_OPTIONS_PERIOD.
SOFTPHONE_RTP_MA X = Maximum port in the range 1024 to 65534.
X=X
The maximum usable port indicated to the IP Office Video Softphone when
SOFTPHONE_RTP_RANGE_ENABLE and SOFTPHONE_RTP_MIN are set.
SOFTPHONE_RTP_MIN X = Minimum port in the range 1024 to 65534.
=X
The minimum usable port indicated to the IP Office Video Softphone when
SOFTPHONE_RTP_RANGE_ENABLE and SOFTPHONE_RTP_MAX are set.
SOFTPHONE_RTP_RA When set, the usable ports indicated to the IP Office Video Softphone are set via the
NGE_ENABLE SOFTPHONE_RTP_MIN and SOFTPHONE_RTP_MAX values.
SUPPRESS_ALARM=1 When set, suppresses the NoCallerID alarm otherwise shown in SysMonitor, SNMP
traps, email notifications, SysLog or System Status.
Table continues…

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 User

Value Description
TUI:NAME_SEARCH_M The default directory search matching behavior is to simultaneously match against
ODE=1 first and last name characters. This source number sets the system to match from
the start of the name only.
VM_TRUNCATE_TIME= X= time in seconds. Range = 0 to 7.
X
On analog trunks, call disconnection can occur though busy tone detection. When
such calls go to voicemail to be recorded or leave a message, when the call ends
the system indicates to the voicemail server how much to remove from the end of
the recording in order to remove the busy tone segment. This amount varies by
system locale, the defaults being listed below. For some systems it may be
necessary to override the default if analog call recordings are being clipped or
include busy tone. That can be done by adding a VM_TRUNCATE_TIME= setting
with the required value in the range 0 to 7 seconds.
• New Zealand, Australia, China, Saudi Arabia and Custom: 5 seconds.
• Korea: 3 seconds.
• Italy, Mexico, Chile, Colombia and Brazil: 2 seconds.
• Argentina, United States, Canada and Turkey: 0 seconds.
• All other locales: 7 seconds.
VMAIL_WAIT_DURATI The number of milliseconds to wait before cutting through the audio to Voicemail.
ON=X Some delay is required to allow for codec negotiation.
VMPRO_OOB_DTMF_O When set, disabled the sending of out-of-band digits to the Voicemail Pro voicemail
FF server.
xmpp_port_l1=X X = The port of the XMPP server that can be accessed by clients registered on the
LAN1 interface.
xmpp_port_l2=X X = The port of the XMPP server that can be accessed by clients registered on the
LAN2 interface.
xmpp_port_r1=X X = The port of the XMPP server that can be accessed by remote clients registered
on the LAN1 interface.
xmpp_port_r2=X X = The port of the XMPP server that can be accessed by remote clients registered
on the LAN2 interface.

Related links
User on page 401

Telephony
Navigation: User | Telephony
This form allows you to set telephony related features for the user. These override any matching
setting in the System | Telephony tab. The settings are grouped into a number of sub-tabs.
Related links
User on page 401

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Configuration Mode Field Descriptions

Call Settings on page 422

Supervisor Settings on page 425
Multi-line Options on page 429
Call Log on page 431
TUI on page 432

Call Settings
Navigation: User | Telephony | Call Settings
Additional configuration information
For additional information on ring tones, see Ring Tones on page 604.
Configuration settings
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Outside Call Default = Default Ring (Use system setting)
Sequence
Applies only to analog phones. Sets the ring pattern used for external calls to the user.
The distinctive ring patterns used for other phones are fixed. Note that changing the
pattern for users associated with fax and modem device extensions may cause those
devices to not recognize and answer calls.
Inside Call Default = Default Ring (Use system setting)
Sequence
Applies only to analog phones. Sets the ring pattern used for internal calls to the user.
The distinctive ring patterns used for other phones are fixed.
Ring Back Default = Default Ring (Use system setting)
Sequence
Applies only to analog phones. Sets the ring pattern used for ringback calls to the user.
The distinctive ring patterns used for other phones are fixed.
No Answer Time Default = Blank (Use system setting). Range = 6 to 99999 seconds.
Sets how long a call rings the user before following forwarded on no answer if set or
going to voicemail. Leave blank to use the system default setting.
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 User

Field Description
Wrap-up Time Default = 2 seconds, Range 0 to 99999 seconds. Specifies the amount of time after
(secs) ending one call during which the user is treated as still being busy. During this time:
• Other phones or applications monitoring the user's status will indicate the user as still
being busy (on a call).
• Hunt group calls will not be presented to the user.
• If the user is using a single line set, direct calls also receive busy treatment. If the user
is using a mutli-line set (multiple call appearances), direct calls to them will ring as
normal.
• It is recommended that this option is not set to less than the default of 2 seconds. 0 is
used to allow immediate ringing.
• For users set as a CCR Agent, use the setting User | Telephony | Supervisor
Settings | After Call Work Time.
Transfer Return Default = Blank (Off), Range 1 to 99999 seconds.
Time (secs)
Sets the delay after which any call transferred by the user, which remains unanswered,
should return to the user. A return call will continue ringing and does not follow any
forwards or go to voicemail.
Transfer return will occur if the user has an available call appearance button.
Transfer return is not applied if the transfer is to a hunt group that has queuing enabled.
Call Cost Mark-Up Default = 100.
This setting is used for ISDN advice of charge (AOC). The markup is applied to the cost
calculations based on the number of units and the line base cost per charging unit. The
field is in units of 1/100th, for example an entry of 100 is a markup factor of 1. This value
is included in the system SMDR output.
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Configuration Mode Field Descriptions

Field Description
Advertize Callee Default = System Default (Off).
State To Internal
The options are:
Callers
• System Default (Off). The system setting is System | Telephony | Telephony |
Advertize Callee State To Internal Callers.
• On
• Off
When enabled, for internal calls, additional status information is communicated to the
calling party.
Not supported for SIP endpoints except the J100 Series (excluding the J129).
• When calling another internal phone and the called phone is set to Do Not Disturb or on
another call, the calling phone displays “Do Not Disturb” or “On Another Call” rather
than “Number Busy”.
• On 9500 Series, 9600 Series and J100 Series phones, if a line appearance is
programmed on a button on phone A and that line is in use on phone B, then phone A
displays the name of the current user of the line along with the line number.
• If a line appearance on a phone is in use elsewhere in the system and another
extension unsuccessfully attempts to seize that line, the phone displays “In
Use:<name>” where <name> is the name of the user currently using the line.
Call Waiting On Default = Off
For users on phones without appearance buttons, if the user is on a call and a second
call arrives for them, an audio tone can be given in the speech path to indicate a waiting
call (the call waiting tone varies according to locale). The waiting caller hears ringing
rather than receiving busy. There can only be one waiting call, any further calls receive
normal busy treatment. If the call waiting is not answered within the no answer time, it
follows forward on no answer or goes to voicemail as appropriate. User call waiting is not
used for users on phones with multiple call appearance buttons.
Answer Call Default = On
Waiting on Hold
Applies to analog and IP DECT extension users only. If the user has a call waiting and
places their current call on hold, the waiting call is automatically connected.
Busy on Held Default = Off for users with call appearance buttons/On for other users.
If on, when the user has a call on hold, new calls receive busy treatment. They will follow
the user's forward on busy setting or are diverted to voicemail. Otherwise busy tone
(ringing for incoming analog calls) is played. This overrides call waiting when the user has
a call on hold. The use of Busy on Held for users with multiple call appearance buttons is
deprecated and Manager will prompt whether it should switch off the feature off for such a
user.
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 User

Field Description
Offhook Station Default = Off
Off-hook station allows an analog extension to be left permanently off-hook, with calls
being made and answered using an application or TAPI. When enabled, the analog
extension user is able to control calls using the application in the following ways:
Offhook station does not disable the physical off-hook on the phone. When starting with
the phone on-hook, making and answering calls is the same as normal analog extension
operation. Additionally however calls can be initiated from the application. After entering
the required number and making the call, the on-hook analog extension receives a
ringback showing the users own caller ID and when answered the outgoing call leg to the
dialed number is started. Calls to a busy destination present busy tone before being
cleared.
The application can be used to end a call with the analog extension still off-hook. Instead
of hearing disconnect tone the user hears silence and can use the application to make
another call. Though off-hook the user is indicated as idle on BLF indicators. Without off-
hook Station set the user would be indicated as busy when off-hook, whether on a call or
not.
If off-hook and idle (having cleared a previous call), incoming call alerts by presenting
ringing through the audio path. The call can be answered using the application or going
on-hook/off-hook or by pressing recall. Note that if the phone normally displays call ID,
any caller ID displayed on the phone is not updated in this mode, however the call ID in
the application will be that of the current call.
If on-hook, an incoming call alerts as normal using the phone's ringer and is answered by
going off-hook. The answer call option in the application cannot be used to answer calls
to an on-hook analog extension.
While off-hook and idle, the analog extension user will receive page calls.
If the analog extension handset is replaced with a headset, changing the Manager setting
Extension | Analog | Equipment Classification to Quiet Handset is recommended.

Related links
Telephony on page 421

Supervisor Settings
Navigation: User | Telephony | Supervisor Settings
Additional configuration information
• For additional information on the Force Authorization Code setting, see Configuring
Authorization Codes on page 645.
• For additional information on the Inhibit Off-Switch Forward/Transfers see, Off-Switch
Transfer Restrictions on page 719.
Configuration settings
These settings relate to user features normally only adjusted by the user's supervisor.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours

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Configuration Mode Field Descriptions

User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Login Code Default = Blank. Range = Up to 31 digits.
The code that has to be entered, as part of a log in sequence, to allow a user to make use
of an extension as if it was their own phone. This entry must be at least 4 digits for DS
port users. Login codes of up to 15 digits are supported with Extn Login buttons. Login
codes of up to 31 digits are supported with Extn Login short codes. Centralized users use
the Login Code for SIP registration on Session Manager.
• For IP phone users, the login code should be limited to 13 digits. The user's login code
is used by IP phones during registration with the system.
• This log in code can be used for hot desking as well as logging back onto your phone
after it has been used by a hot desking user. Hot desking is not supported for
centralized users.
• Users can only log out if they have a Login Code set. Users can log out without having
a Login Code set if they are currently logged in at an extension whose Base Extension
Number (Extension | Extn) no longer matches their own Extension (User | User).
• Supports the short code feature Change Login Code.
• If the user has a login code set, it is used by the Outgoing Call Bar Off short code
feature.
• If the user has a login code set, access to a range of programmable button features will
require entry of the login code. For example access Self Admin and System Phone
features.
Login Idle Period Default = Blank (Off). Range = 0 (Off) to 99999.
(secs)
If the telephone is not used for this period; the user currently logged in is automatically
logged out. This option should be used only in conjunction with Force Login (see below).
Monitor Group Default = <None>
Sets the hunt group whose members the user can monitor if silent monitoring is setup.
See the Call Listen short code.
Privacy Override Default = <None>
Group
The drop-down menu lists the local and network advertised hunt groups. If selected, calls
to this user cannot be seen or picked up by other users unless they are a member of the
selected group.
Coverage Group Default = <None>.
If a group is selected, then in scenarios where an external call would normally have gone
to voicemail, it instead continues ringing and also starts alerting the members of the
coverage group. For further details refer to Coverage Groups.
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 User

Field Description
Status on No Default = Logged On.
Answer
Hunt groups can change the status of call center agents (users with a log in code and set
to forced log in) who do not answer a hunt group call presented to them before it is
automatically presented to the next agent. Use of this is controlled by the Agent's Status
on No Answer Applies To setting of the hunt group. This option is not used for calls
ringing the agent because the agent is in another group's overflow group. The options are:
• Logged On: If this option is selected, the user's status is not changed.
• Busy Wrap-Up: If this option is selected the user's membership status of the hunt group
triggering the action is changed to disabled. The user can still make and receive calls
and will still continue to receive calls from other hunt groups to which they belong.
• Busy Not Available: If this option is selected the user's status is changed to do not
disturb. This is the equivalent of DND and will affect all calls to the user.
• Logged Off: If this option is selected the users status is changed to logged out. In that
state they cannot make calls or receive calls. Hunt group calls go to the next available
agent and personal calls treat the user as being busy.
Reset Longest Default = All Calls.
Idle Time
This setting is used in conjunction with hunt groups set to Longest Waiting (also known as
Idle and Longest Waiting). It defines what type of calls reset the idle time of users who are
members of these hunt groups. Options are All Calls and External Incoming.
ICR Agent Role Note:
This field is available only if you first configure the user as an Integrated Contact
Reporter (ICR) user using the ICR Agent field, which is provided near the end.
Default = Agent.
Select Supervisor to make the user a supervisor. Selecting Supervisor displays the
Enable Huntgroup Monitoring area and lists all the hunt groups available for the
supervisor to monitor. The hunt groups are listed only if they were already configured.
Select the hunt groups for supervisor to monitor.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Force Login Default = Off
If checked, the user must log in using their Login Code to use any extension including an
extension to which they are the default associated user (Base Extension). For example, if
Force Login is ticked for user A and user B has logged onto A's phone, when B logs off
user A is not automatically associated with their normal phone and instead must log back
on. If Force Login was not ticked, A would be automatically logged back in.
Force Account Default = Off
Code
If checked, the user must enter a valid account code to make an external call.
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Configuration Mode Field Descriptions

Field Description
Force Default = Off.
Authorization
If checked, the user must enter a valid authorization code to make an external call. That
Code
authorization code must be one associated with the user or the user rights to which the
user belongs.
Incoming Call Bar Default = Off
When enabled, this setting stops a user from receiving any external calls. On the calling
phone, the call is rejected.
Outgoing Call Bar Default = Off
When enabled, this setting stops a user from making any external calls except those that
use dial emergency features. On many Avaya display phones, this causes a B to be
displayed. The following features can be used with outgoing call bar: Outgoing Call Bar
On, Outgoing Call Bar Off and Change Login Code.
Inhibit Off-Switch Default = Off.
Forward/Transfers
When enabled, this setting stops the user from transferring or forwarding calls externally.
This does not stop another user transferring the restricted users calls off-switch on their
behalf. Note that a number of other controls may inhibit the transfer operation.
Can Intrude Default = Off
Check this option if the user can join or interrupt other user's calls using call intrusion
methods other than conferencing.
Cannot be Default = On
Intruded
If checked, this user's calls cannot be interrupted or acquired by other internal users using
call intrusion. For users with Cannot Be Intruded off, private call can be used to indicate
whether a call can be intrude or not.
Can Trace Calls Default = Off. This settings controls whether the user is able to make used of ISDN MCID
controls.
ICR Agent Default = Off.
Enable to make the user an ICR user. If enabled, the ICR Agent Role field becomes
available and the After Call Work related fields are activated.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Automatic After Default = Off.
Call Work
If enabled, the agent goes into After Call Work (ACW) at the end of an ICR and non-ICR
hunt group call to indicate that they are busy with post-call processing activity. During the
ACW state, they are not sent any hunt group calls.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
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 User

Field Description
Can Control After Default = Off.
Call Work
If enabled, the agent can extend the currently active After Call Work time indefinitely.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
After Call Work Default = The value in this field is populated from the Default After Call Work Time field
Time (Sec) located at System | Contact Center.
The time after a call when an agent is busy and unable to deal with hunt group calls.
Change the value if you want to specify ACW time for this user to be different from the
system default.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Can Accept Default = Off [Brazil Only]
Collect Calls
Determines whether the user is able to receive and accept collect calls.
Deny Auto Default = Off.
Intercom Calls
When enabled, any automatic intercom calls to the user's extension are automatically
turned into normal calls.
Enable Hunt Default = Blank
group Monitoring
All the available hunt groups for Integrated Contact Reporter are listed under Hunt Group
Name. Select the check box against the hunt group to enable it for monitoring by the
supervisor. Select the Hunt Group Name check box to enable all the hunt groups for
monitoring by the supervisor. The field is activated if you assign the user with Supervisor
role using the ICR Agent Role field.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.

Related links
Telephony on page 421

Multi-line Options
Navigation: User | Telephony | Multi-line Options
Additional configuration information
• For additional configuration information, see Appearance Button Operation on page 1058.
• For the Reserve Last CA setting, 1400, 1600, 9500 and 9600 Series telephone users can
put a call on hold pending transfer if they already have held calls even if they have no free
call appearance button available. For additional information, see Context Sensitive
Transfer on page 720.

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Configuration Mode Field Descriptions

Configuration settings
Multi-line options are applied to a user's phone when the user is using an Avaya phones which
supports appearance buttons (call appearance, line appearance, bridged and call coverage).
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Individual Default = 10 seconds, Range 1 to 99999 seconds.
Coverage Time
(secs) This function sets how long the phone will ring at your extension before also alerting at any
call coverage users. This time setting should not be equal to or greater than the No
Answer Time applicable for the user.
Ring Delay Default = Blank (Use system setting). Range = 0 (use system setting) to 98 seconds.
This setting is used when any of the user's programmed appearance buttons is set to
Delayed ringing. Calls received on that button will initially only alert visually. Audible
alerting will only occur after the ring delay has expired.
Coverage Ring Default = Ring.
This field selects the type of ringing that should be used for calls alerting on any the user's
call coverage and bridged appearance buttons. Ring selects normal ringing. Abbreviated
Ring selects a single non-repeated ring. No Ring disables audible ringing. Note that each
button's own ring settings (Immediate, Delayed Ring or No Ring) are still applied.
The ring used for a call alerting on a call coverage or bridged appearance button will vary
according to whether the user is currently connected to a call or not.
• If not currently on a call, the Coverage Ring setting is used.
• If currently on a call, the quieter of the Coverage Ring and Attention Ring settings is
used.
Attention Ring Coverage Ring Setting
Setting
Ring Abbreviated Off
Ring Ring Abbreviated Off
Abbreviated Abbreviated Abbreviated Off
Attention Ring Default = Abbreviated Ring. This field selects the type of ringing that should be used for
calls alerting on appearance buttons when the user already has a connected call on one of
their appearance buttons. Ring selects normal ringing. Abbreviated Ring selects a single
ring. Note that each button's own ring settings (Immediate, Delayed Ring or No Ring) are
still applied.
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 User

Field Description
Ringing Line Default = On.
Preference
For users with multiple appearance buttons. When the user is free and has several calls
alerting, ringing line preference assigns currently selected button status to the appearance
button of the longest waiting call. Ringing line preference overrides idle line preference.
Idle Line Default = On. For users with multiple appearance buttons. When the user is free and has
Preference no alerting calls, idle line preference assigns the currently selected button status to the first
available appearance button.
Delayed Ring Default = Off.
Preference
This setting is used in conjunction with appearance buttons set to delayed or no ring. It
sets whether ringing line preference should use or ignore the delayed ring settings applied
to the user's appearance buttons.
When on, ringing line preference is only applied to alerting buttons on which the ring delay
has expired.
When off, ringing line preference can be applied to an alerting button even if it has delayed
ring applied.
Answer Pre- Default = Off.
Select
Normally when a user has multiple alerting calls, only the details and functions for the call
on currently selected button are shown. Pressing any of the alerting buttons will answer the
call on that button, going off-hook will answer the currently selected button. Enabling
Answer Pre-Select allows the user to press any alerting button to make it the current
selected button and displaying its call details without answering that call until the user
either presses that button again or goes off-hook. Note that when both Answer Pre-Select
and Ringing Line Preference are enabled, once current selected status is assigned to a
button through ringing line preference it is not automatically moved to any other button.
Reserve Last CA Default = Off.
Used for users with multiple call appearance buttons. When selected, this option stops the
user's last call appearance button from being used to receive incoming calls. This ensures
that the user always has a call appearance button available to make an outgoing call and
to initiate actions such as transfers and conferences.
1400, 1600, 9500 and 9600 Series telephone users can put a call on hold pending transfer
if they already have held calls even if they have no free call appearance button available.

Related links
Telephony on page 421

Call Log
Navigation: User | Telephony | Call Log
The system can store a centralized call log for users. Each users' centralized call log can contain
up to 30 call records for user calls. When this limit is reached, each new call records replaces the
oldest previous record.
On Avaya phones with a fixed Call Log or History button (1400, 1600, 9500 and 9600 Series),
that button can be used to display the user's centralized call log. The centralized call log is also

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Configuration Mode Field Descriptions

used for M-Series and T-Series phone. The user can use the call log to make calls or to store as a
personal speed dial. They can also edit the call log to remove records. The same call log is also
used if the user logs into one-X Portal.
The centralized call log moves with the user if they log on and off from different phones. This
includes if they hot desk within a network.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Centralized Call Default = System Default (On)
Log
This setting allows the use of centralized call logging to be enabled or disabled on a per
user basis. The default is to match the system setting System | Telephony | Call Log |
Default Centralized Call Log On.
The other options are On or Off for the individual user. If set to Off, the user receives the
message “Call Log Disabled” when the Call Log button is pressed.
Delete records Default = 00:00 (Never).
after
(hours:minutes) If a time period is set, records in the user's call log are automatically deleted after this
period.
Groups Default = System Default (On).
This section contains a list of hunt groups on the system. If the system setting System |
Telephony | Call Log | Log Missed Huntgroup Calls has been enabled, then missed
calls for those groups selected are shown as part of the users call log. The missed calls
are any missed calls for the hunt group, not just group calls presented to the user and not
answered by them.

Related links
Telephony on page 421

TUI
Navigation: User | Telephony | TUI
These settings can be used to control access to selected menu options on 1400, 1600, 9500 and
9600 Series telephones.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Features Menu Controls
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 User

Field Description
User Setting Default = Same as System
When set to Same as System, matches the system-wide settings of the System |
Telephony | TUI menu options. When set to Custom, uses the Features Menu
settings below.
Features Menu Default = On
When set to off, TUI feature menus are not available. When set to on, you can select
to turn individual feature menus off or on. The following feature menus are listed:
• Basic Call Functions: If selected, the user can access menu options for call
pickup, park, unpark and transfer to mobile functions.
• Advanced Call Functions: If selected, the user can access the menu options for do
not disturb, account code, withhold number and internal auto-answer functions.
Note, the Account Code menu is only shown if the system has been configured
with accounts codes.
• Forwarding: If selected, the user the phone's menus for forwarding and follow me
functions.
• Hot Desk Functions: If selected, the user can access the menu options for logging
in and out.
• Passcode Change: If selected, the user can change their login code (security
credentials) through the phone menus..
• Phone Lock: If selected, the user can access the menu options for locking the
phone and for setting it to automatically lock.
• Self Administration: If selected, the user can access the phone’s Self-
Administration menu options.
• Voicemail Controls: If set, the user can access the Visual Voice option through the
phone's Features menu.

Related links
Telephony on page 421

User | Forwarding
Navigation: User | Forwarding
Additional configuration information
For additional configuration information, see DND, Follow Me, and Forwarding on page 674.
Configuration settings
Use this page to check and adjust a user's call forwarding and follow me settings.
Follow Me is intended for use when the user is present to answer calls but for some reason is
working at another extension. For example; temporarily sitting at a colleague's desk or in another
office or meeting room. As a user, you would use Follow Me instead of Hot-Desking if you don't

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Configuration Mode Field Descriptions

have a log in code or you don't want to interrupt you colleague also receiving their own calls.
Multiple users can use follow me to the same phone.
Forwarding is intended for use when, for some reason, the user is unable to answer a call. They
may be busy on other calls, unavailable or simply don't answer. Calls may be forwarded to internal
or, subject to the user's call barring controls, external numbers.
To bar a user from forwarding calls to an external number, select the setting User | Telephony |
Supervisor Settings | Inhibit Off-Switch Forward/Transfers.
To bar all users from forwarding calls to external numbers, select the setting System | Telephony
| Telephony | Inhibit Off-Switch Forward/Transfers.
Note that analog lines doe not provide call progress signalling. Therefore calls forwarded off-
switch via an analog line are treated as answered and are not recalled.
Once a call has been forwarded to an internal target, it will ignore the target’s Forward No
Answer or Forward on Busy settings but may its Forward Unconditional settings unless they
create a loop.
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Block Default = Off.
Forwarding
When enabled, call forwarding is blocked for this user.
The following actions are blocked: Follow me, Forward unconditional, Forward on
busy, Forward on no answer and Hot Desking
The following actions are not blocked: Do not disturb, Voicemail and Twinning
Follow Me Default = Blank. Range = Internal extension number.
Number
Redirects the user's calls to the internal extension number entered. If the redirected
call receives busy or is not answered, it follows the user's forwarding and or voicemail
settings as if it had been presented to their normal extension. When a user has follow
me in use, their normal extension will give alternate dialtone when off hook. Using
Follow Me overrides Forward Unconditional.
Calls targeting longest waiting type hunt groups ignore Follow Me.
Calls triggered by actions at the user's original extension, for example voicemail
ringback, ignore Follow Me.
Park, hold and transfer return calls will go to the extension at which the user initiated
the park, hold or transfer action.
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 User

Field Description
Forward Default = Off
Unconditional
This option, when checked and a Forward Number also set, forwards all external
calls immediately. Additional options allow this forwarding to also be applied to internal
calls and to hunt group calls if required. When a user has forward unconditional in
use, their normal extension will give alternate dialtone when off hook. If the destination
is an internal user on the same system, they are able to transfer calls back to the user,
overriding the Forward Unconditional.
After being forwarded for the user’s no answer time, if still unanswered, the system
can apply additional options. It does this if the user has forward on no answer set for
the call type or if the user has voicemail enabled.
• If the user has forward on no answer set for the call type, the call is recalled and
then forwarded to the forward on no answer destination.
• If the user has voicemail enabled, the call is redirected to voicemail.
• If the user has both options set, the call is recalled and then forwarded to the
forward on no answer destination for their no answer time and then if still
unanswered, redirected to voicemail.
• If the user has neither option set, the call remains redirected by the forward
unconditional settings.
Note that for calls redirected via external trunks, detecting if the call is still unanswered
requires call progress indication. For example, analog lines do not provide call
progress signalling and therefore calls forwarded via an analog lines are treated as
answered and not recalled.
To Voicemail Default = Off.
If selected and forward unconditional is enabled, calls are forwarded to the user's
voicemail mailbox. The Forward Number and Forward Hunt Group Calls settings
are not used. This option is not available if the system's Voicemail Type is set to
None. 1400, 1600, 9500 and 9600 Series phone users can select this setting through
the phone menu. Note that if the user disables forward unconditional the To
Voicemail setting is cleared.
Forward Number Default = Blank. Range = Internal or External number. Up to 33 characters.
This option sets the destination number to which calls are forwarded when Forward
Unconditional is checked. The number can be an internal or external number. This
option is also used for Forward on Busy and Forward on No Answer if no separate
Forward Number is set for those features. If a user forwards a call to a hunt group of
which they are a member, the group call is not presented to them but is presented to
other members of the hunt group.
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Configuration Mode Field Descriptions

Field Description
Forward Hunt Default = Off
Group Calls
Hunt group calls (internal and external) are not normally presented to a user who has
forward unconditional active. Instead they are presented to the next available member
of the hunt group. This option, when checked, sets that hunt group calls (internal and
external) are also forwarded when forward unconditional is active. The group's Ring
Type must be Sequential or Rotary, not Collective or Longest Waiting. The call is
forwarded for the period defined by the hunt group's No Answer Time after which it
returns to the hunt group if unanswered. Note also that hunt group calls cannot be
forwarded to another hunt group.
Forward Internal Default = On.
Calls
This option, when checked, sets that internal calls should be also be forwarded
immediately when forward unconditional is active.
Forward On Default = Off
Busy
When checked and a forward number is set, external calls are forwarded when the
user's extension is busy. The number used is either the Forward Number set for
Forward Unconditional or if set, the separate Forward Number set under Forward
On Busy. Having Forward Unconditional active overrides Forward on Busy.
If the user has Busy on Held selected, if forward on busy is active it is applied when
the user is free to receive calls but already has a call on hold.
If the user's phone has multiple call appearance buttons, the system will not treat
them as busy until all the call appearance buttons are in use unless the last
appearance button has been reserved for outgoing calls only.
Forward On No Default = Off When checked and a forward number is set, calls are forwarded when
Answer the user does not answer within their set No Answer Time (User | Telephony | Call
Settings).
Forward Number Default = Blank. Range = Internal or External number. Up to 33 characters.
If set, this number is used as the destination for Forward On Busy and Forward On
No Answer when on. If not set, the Forward Number set for Forward Unconditional
is used. If a user forwards a call to a hunt group of which they are a member, the
group call is not presented to them but is presented to other members of the hunt
group.
Forward Internal Default = On. When checked, this option sets that internal calls should be also be
Calls forwarded when forward on no answer or forward on busy is active.

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Dial In
Navigation: User | Dial In
Use this dialogue box to enable dial in access for a remote user. An Incoming Call Route and RAS
service must also be configured.

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 User

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Dial In On Default = Off
When enabled, dial in access into the system is available via this user.
Dial In Time Default = <None>
Profile
Select the Time Profile applicable to this User account. A Time Profile can be used to set
time restrictions on dial in access via this User account. Dial In is allowed during the times
set in the Time Profile form. If left blank, then there are no restrictions.
Dial In Firewall Default = <None>
Profile
Select the Firewall Profile to restrict access to the system via this User account. If blank,
there are no Dial In restrictions.

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Voice Recording
Navigation: User | Voice Recording
Used to activate the automatic recording of user's external calls. The recording of internal calls is
also supported.
Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
Note the following:
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Calls parked or held pause recording until the unparked or taken off hold (does not apply to
SIP terminals).
• Recording is stopped if:
- User recording stops if the call is transferred to another user.
- User account code recording stops if the call is transferred to another user.
- Hunt group recording stops if the call is transferred to another user who is not a member of
the hunt group.
- Incoming call route recording continues for the duration of the call on the system.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Auto Recording
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Configuration Mode Field Descriptions

Field Description
Inbound Default = None.
Select whether automatic recording of incoming calls is enabled. The field to the right
sets whether just external, just internal, or both external and internal calls are included.
The options are:
• None: Do not automatically record calls.
• On: Record the call if possible. If not possible to record, allow the call to continue.
• Mandatory: Record the call if possible. If not possible to record, block the call and
return busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Outbound Default = None.
Select whether automatic recording of out going calls is enabled. The field to the right
sets whether just external, just internal, or both external and internal calls are included.
The options are:
• None: Do not automatically record calls.
• On: Record the call if possible. If not possible to record, allow the call to continue.
• Mandatory: Record the call if possible. If not possible to record, block the call and
return busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Destination Default = None.
Sets the destination for automatically triggered recordings. The options are:
• Voice Recording Library: This options set the destination for the recording to be a
VRL folder on the voicemail server. The ContactStore application polls that folder
and collects waiting recordings which it then places in its own archive. Recording is
still done by Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to Voice Recording
Library above but instructs the voicemail server to create an authenticated recording.
If the file contents are changed, the file is invalidated though it can still be played.
This option is not currently supported with Linux based systems.
Time Profile Default = None. (Any time).
Used to select a time profile during which automatic call recording of incoming calls is
applied. If no profile is selected, automatic recording of incoming calls is active at all
times.

Manual Recording
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 User

Field Description
Destination Default = None.
Sets the destination for automatically triggered recordings. The options are:
• Voice Recording Library: This options set the destination for the recording to be a
VRL folder on the voicemail server. The ContactStore application polls that folder
and collects waiting recordings which it then places in its own archive. Recording is
still done by Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to Voice Recording
Library above but instructs the voicemail server to create an authenticated recording.
If the file contents are changed, the file is invalidated though it can still be played.

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Button Programming
Navigation: User | Button Programming
Additional configuration information
For additional information on programming button actions, see Button Programming Overview on
page 902.
For a description of each button action, see Button Programming Actions on page 914.
Used to assign functions to the programmable keys provided on many Avaya telephones. For full
details of button programming refer to the section Button Programming.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Button No. The number of the DSS key against which the function is being set. To set a function
against a button double-click it or select it and then click Edit.
Label This is a text label for display on the phone. If no label is entered, the default label for the
selected action is used.
Action Defines the action taken by the menu item.
Action Data This is a parameter used by the selected action. The options here will vary according to
the selected button action.
Display All The number of button displayed is based on the phone associated with the user when the
configuration was loaded. This can be overridden by selecting Display All Buttons. This
may be necessary for users who switch between different phones using hot desking or
have an expansion unit attached to their phone.

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Configuration Mode Field Descriptions

User | Menu Programming

Navigation: User | Menu Programming
These menus control a range of options that are specific to different types of phones. The
functions become accessible when the user logs in on the appropriate type of phone.
Related links
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Huntgroup on page 440
4400/6400 on page 440

Huntgroup
Navigation: User | Menu Programming | Hunt Group
Avaya 1400, 1600, 9500 and 9600 Series phone users can control various settings for selected
hunt groups. These settings are also used for one-X Portal for IP Office.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Can Change Default = Off
Membership
This list shows the hunt groups of which the user is a member. Up to 10 of these
groups can be checked; those group and the users current membership status are
then displayed on the phone. The user can change their membership status through
the phone's menus.
Can Change Default = Off
Service Status
This list shows all the hunt groups on the system. Up to 10 of these groups can be
checked.
Can Change Default = Off.
Night Service
If selected, the user can change the fallback group used when the hunt group is in
Group
Night Service mode.
Can Change Out Default = Off. If selected, the user can change the fallback group used when the hunt
of Service Group group is in Out of Service mode.

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4400/6400
Navigation: User | Menu Programming | 4400/6400
4412, 4424, 4612, 4624, 6408, 6416 and 6424 phones have a Menu key, sometimes marked with
an icon. When Menu is pressed, a number of default functions are displayed. The < and >
keys can be used to scroll through the functions while the keys below the display can be used to
select the required function.

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 User

The default functions can be overwritten by selections made within this tab.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Menu No. The menu position which the function is being set.
Label This is a text label for display on the phone. If no label is entered, the default label for
the selected action is used. Labels can also be changed through the menu on some
phones, refer to the appropriate telephone user guide.
Action Defines the action taken by the menu button.
Action Data This is a parameter used by the selected action. The options here will vary according
to the selected button action.

Related links
User | Menu Programming on page 440

Mobility
Navigation: User | Mobility
Additional configuration information
For additional configuration information regarding the Mobile Call Control setting, see Mobile Call
Control. on page 707
Configuration settings
These settings relate to twinning features where a user has a main or primary extension but also
regularly answer calls at a secondary or twinned phone. These features are intended for a single
user. They are not aimed at two users answering calls presented to a single primary extension.
Twinning allows a user's calls to be presented to both their current extension and to another
number. The system supports two modes of twinning:
Internal Mobile
Twinning Destination Internal extensions only External numbers only.
Supported in All locales. All locales.
License Required The primary phone user must be Yes
a licensed user.

In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours
User Rights settings. The user rights applied can be controlled by a time profile selected as the
user's Working Hours Time Profile setting. The effect of the user rights can be displayed using the
User Rights View control.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Internal Twinning
Select this option to enable internal twinning for a user. Internal Twinning cannot be selected for a user if they
already have Mobility Features selected. Internal twinning is not supported across an SCN or SE network.
Internal twinning is not supported during resilience.
Twinned Handset Default = Blank.
For internal twinning, the drop-down list can be used to select an available user as the
twinned calls destination. The secondary phone:
• must be on the same system
• must not be a simultaneous mode phone. For example, Avaya Communicator (for
Windows, iPad, or Web), or WebRTC web client.
If the list is grayed out, the user is a twinning destination and the primary to which they
are twinned is displayed.
All User | Mobility fields are grayed out for unlicensed users.
Maximum Number Default = 1.
of Calls
If set to one, when either the primary or secondary phone are in use, any additional
incoming call receives busy treatment. If set to two, when either phone is in use, it
receives call waiting indication for any second call. Any further calls above two receive
busy treatment.
Twin Bridge Default = Off.
Appearances
By default only calls alerting on the primary phone's call appearance buttons also alert at
the secondary. When this option is enabled, calls alerting on a bridged appearance button
at the primary can also alert at the secondary.
Twin Coverage Default = Off.
Appearances
By default only calls alerting on the primary phone's call appearance buttons also alert at
the secondary. When this option is enabled, calls alerting on a coverage appearance
button at the primary can also alert at the secondary.
Twin Line Default = Off.
Appearances:
By default only calls alerting on the primary phone's call appearance buttons also alert at
the secondary. When this option is enabled, calls alerting on a line appearance button at
the primary can also alert at the secondary.

Mobility Features
If enabled this option allows any of the mobility features to be enabled for the user.
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 User

Field Description
Mobile Twinning If selected, the user is enable for mobile twinning. The user can control this option
through a Twinning programmable button on their a phone.
For user's setup for one-X Mobile Client, changes to their Mobile Twinning status made
through the system configuration or using a Twinning button are not reflected in the
status of the Extension to Cellular icon on their mobile client. However, changes to the
Extension to Cellular status made from the mobile client are reflected by the Mobile
Twinning field in the system configuration. Therefore, for one-X Mobile Client users, it is
recommended that they control their Mobile Twinning status through the one-X Mobile
Client rather than through a Twinning button.
Fallback Twinning Default = Disabled
When Fallback Twinning is enabled and the user’s primary extensions are unreachable,
IP Office redirects the calls to the Twinned Mobile Number even if Mobile Twinning is
disabled. The Mobile Dial Delay time set up by the user is not considered during Fallback
Twinning.
Twinned Mobile Default = Blank.
Number
This field sets the external destination number for mobile twinned calls. It is subject to
normal short code processing and should include any external dialing prefix if necessary.
For users of Mobile Call Control, the number in this field is used to match the users
setting to the incoming CLI.
Twinning Time Default = <None> (Any time)
Profile
This field allows selection of a time profile during which mobile twinning will be used.
Mobile Dial Delay Default = 2 seconds
This setting controls how long calls should ring at the user's primary extension before also
being routed to ring at the twinning destination number. This setting may be used at the
user's choice, however it may also be a necessary control. For example, if the twinning
number is a mobile device that has been switched off, the mobile service provider may
immediately answer the call with their own voicemail service. This would create a
scenario where the user's primary extension does not ring or ring only briefly.
Mobile Answer Default = 0 (Off). Range = 0 to 99 seconds. This control can be used in situations where
Guard calls sent to the twinned destination are automatically answered by a voicemail service or
automatic message if the twinned device is not available. If a twinned call is answered
before the Mobile Answer Guard expires, the system will drop the call to the twin.
Hunt group calls Default = Off
eligible for mobile
twinning This setting controls whether hunt group calls ringing the user's primary extension should
also be presented to the mobile twinning number.
Forwarded calls Default = Off This setting controls whether calls forwarded to the user's primary
eligible for mobile extension should also be presented to the mobile twinning number.
twinning
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Configuration Mode Field Descriptions

Field Description
Twin When Default = Off.
Logged Out
If enabled, if the user logs off their primary extension, calls to that extension will still alert
at their twinned device rather than going immediately to voicemail or busy.
• When logged out but twinned, Mobile Dial Delay is not applied.
• Hunt group calls (all types) will be twinned if Hunt group calls eligible for mobile
twinning is enabled. When this is the case the user's idle time is reset for each
externally twinned call answered. Note that calls twinned over analog and analog
emulation trunks are automatically treated as answered.
• When the user's Mobile Time Profile, if configured, is not active they will not get
twinning calls. Calls will be treated the same as the user was logged out user with no
twinning.
• Callback calls initiated by the user will mature to the Twinned Mobile Number. It will
also be possible to initiate Automatic Callback to the user with external twinning and
their busy/free state will be tracked for all calls via the system.
• Any Bridged Appearance set to the user will not alert. Coverage appearance buttons for
the user will continue to operate.
• The BLF/user button status shown for a logged out user with Logged Off Mobile
Twinning is as follows:
- If there are any calls alerting or in progress through the system to the twin the user
status is shown as alerting or in-use as appropriate. This includes the user showing
as busy/in-use if they have such a call on hold and they have Busy on Held enabled.
- If the user enables DND through Mobile Call Control or one-X Mobile client their
status will show as DND/busy.
- Calls from the system dialed direct to the users twinned destination rather than
directed by twinning from their primary extension will not change the user's status.
one-X Mobile Default = Off.
Client
one-X Mobile Client is a software application that can be installed on Windows Mobile and
Symbian mobile cell phones. It allows the user to access a number of system features.
Mobile Call Default = Off.
Control
This feature allows a user receiving a call on their twinned device to access system dial
tone and then perform dialing action including making calls and activating short codes.
See Mobile Call Control on page 707.
Mobile Callback Default = Off.
Mobile callback allows the user to call the system and then hang up. The system will then
make a call to the user's CLI and when answered, provide them with dial tone from the
system to make calls. See the Mobile Callback topic in Administering Avaya IP Office™
Platform with Web Manager.

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 User

Group Memberships
Navigation: User | Group Membership
This tab displays the hunt group of which the user has been made a member. The tick boxes
indicate whether the user's membership of each of those groups is currently enabled or disabled.
Related links
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Announcements
Navigation: User | Announcements
Announcements are played to callers waiting to be answered. This includes callers being
presented to hunt group members, ie. ringing, and callers queued for presentation.
• The system supports announcements using Voicemail Pro or Embedded Voicemail.
• If no voicemail channel is available for an announcement, the announcement is not played.
• In conjunction with Voicemail Pro, the system allows a number of voicemail channels to be
reserved for announcements. See System | Voicemail.
• With Voicemail Pro, the announcement can be replaced by the action specified in a Queued
(1st announcement) or Still Queued (2nd announcement) start point call flow. Refer to the
Voicemail Pro Installation and Maintenance documentation for details.
• Calls can be answered during the announcement. If it is a mandatory requirement that
announcements should be heard before a call is answered, then a Voicemail Pro call flow
should be used before the call is presented.
Note:
Call Billing and Logging
A call becomes connected when the first announcement is played to it. That connected
state is signaled to the call provider who may start billing at that point. The call will also
be recorded as answered within the SMDR output once the first announcement is played.
• If a call is rerouted, for example forwarded, the announcement plan of the original user is still
applied until the call is answered. The exception is calls rerouted to a hunt group at which
point the hunt group announcement settings are applied.
• For announcements to be used effectively, either the user's no answer time must be
extended beyond the default 15 seconds or Voicemail On should be deselected.
Recording Announcements
Voicemail Pro:
There is no mechanism within the telephony user interfaces (TUI) to record user announcements.
To provide custom announcements, user queued and still queued start points must be configured
with Voicemail Pro with the required prompts played by a generic action.
Embedded Voicemail:

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Configuration Mode Field Descriptions

Embedded Voicemail does not include any default announcement or method for recording an
announcement. The Record Message short code feature is provided to allow the recording of
announcements. The telephone number field of short codes using this feature requires the
extension number followed by either ".1" for announcement 1 or ".2" for announcement 2. For
example, for extension number 300, the short codes *91N# | Record Message | N".1" and *92N#
| Record Message | N".2" could be used to allow recording of the announcements by dialing
*91300# and *92300#.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Announcements Default = Off.
On
This setting enables or disables announcements.
Wait before 1st Default = 10 seconds. Range = 0 to 255 seconds.
announcement:
This setting sets the time delay from the calls presentation, after which the first
announcement should be played to the caller.
Flag call as Default = Off.
answered
This setting is used by the CCC and CBC applications. By default they do not regarded a
call as answered until it has been answered by a person or by a Voicemail Pro action with
Flag call as answered selected. This setting allows calls to be marked as answered
once the caller has heard the first announcement. This setting is not used by the
Customer Call Reporter application.
Post Default = Music on hold.
announcement
Following the first announcement, you can select whether the caller should hear Music on
tone
Hold, Ringing or Silence until answered or played another announcement.
2nd Default = On.
Announcement
If selected, a second announcement can be played to the caller if they have still not been
answered.
Wait before 2nd Default = 20 seconds. Range = 0 to 255 seconds.
announcement
This setting sets the wait between the 1st and the 2nd announcement.
Repeat last Default = On.
announcement
If selected, the last announcement played to the caller is repeated until they are
answered or hang-up.
Wait before repeat Default = 20 seconds. Range = 0 to 255 seconds.
If Repeat last announcement is selected, this setting sets is applied between each
repeat of the last announcement.

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SIP
Navigation: User | SIP

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 User

This tab is available when either of the following has been added to the configuration:
• an IP Office Line
• a SIP trunk with a SIP URI record containing a field that has been set to Use Internal Data.
Various fields within the URI settings used by SIP trunks can be set to Use Internal Data. When
that is the case, the values from this tab are used inserted into the URI when the user makes or
receives a SIP call. Within a multi-site network, that includes calls which break out using a SIP
trunk on another system within the network.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
SIP Name Default = Blank on Voicemail tab/Extension number on other tabs.
The value from this field is used when the From field of the SIP URI being used for a SIP
call is set to Use Internal Data.
SIP Display Name Default = Blank on Voicemail tab/Name on other tabs.
(Alias)
The value from this field is used when the Display Name field of the SIP URI being used
for a SIP call is set to Use Internal Data.
Contact Default = Blank on Voicemail tab/Extension number on other tabs.
The value from this field is used when the Contact field of the SIP URI being used for a
SIP call is set to Use Internal Data.
Anonymous Default = On on Voicemail tab/Off on other tabs.
If the From field in the SIP URI is set to Use Internal Data, selecting this option inserts
Anonymous into that field rather than the SIP Name set above.

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Personal Directory
Navigation: User | Personal Directory
Each user is able to have up to 250 personal directory records (100 pre-Release 10.0), up to the
overall system limit.
These records are used as follows:
• When using ETR, J129, M-Series, T-Series, 1400, 1600, 9500 or 9600 Series phones, the
user is able to view and call their personal directory numbers.
• When using a J129, 1400, 1600, 9500 or 9600 Series phone, the user is also able to edit and
add personal directory records.
• If the user hot desks to a 1400, 1600, 9500 or 9600 Series phone on another system in a
multi-site network, they can still access their personal directory.
Users are able to view and edit their personal directory through their phone. Directory records are
used for dialing and caller name matching.

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Configuration Mode Field Descriptions

Dialing
Directory Dialing:
Directory numbers are displayed by user applications such as SoftConsole. Directory numbers are
viewable through the Dir function on many Avaya phones (Contacts or History). They allow the
user to select the number to dial by name. The directory will also contain the names and numbers
of users and hunt groups on the system.
The Dir function groups directory records shown to the phone user into the following categories.
Depending on the phone, the user may be able to select the category currently displayed. In some
scenarios, the categories displayed may be limited to those supported for the function being
performed by the user:
• External Directory records from the system configuration. This includes HTTP and LDAP
imported records.
• Groups Groups on the system. If the system is in a multi-site network, it will also include
groups on other systems in the network. For pre-Release 5 systems, this feature requires the
systems to have Advanced Small Community Networking licenses.
• Users or Index Users on the system. If the system is in a multi-site network it will also
include users on other systems in the network. For pre-Release 5 systems, this feature
requires the systems to have Advanced Small Community Networking licenses.
• Personal Available on 1400, 1600, 9500 and 9600 Series phones. These are the user's
personal directory records stored within the system configuration.
Speed Dialing:
On M-Series and T-Series phones, a Speed Dial button or dialing Feature 0 can be used to
access personal directory records with an index number.
• Personal: Dial Feature 0 followed by * and the 2-digit index number in the range 01 to 99.
• System: Dial Feature 0 followed by 3-digit index number in the range 001 to 999.
• The Speed Dial short code feature can also be used to access a directory speed dial using its
index number from any type of phone.
Caller Name Matching
Directory records are also used to associate a name with the dialled number on outgoing calls or
the received CLI on incoming calls. When name matching is being done, a match in the user's
personal directory overrides any match in the system directory. Note that some user applications
also have their own user directory.
SoftConsole applications have their own user directories which are also used by the applications
name matching. Matches in the application directory may lead to the application displaying a
different name from that shown on the phone.
Name matching is not performed when a name is supplied with the incoming call, for example
QSIG trunks. On SIP trunks the use of the name matching or the name supplied by the trunk can
be selected using the setting System | Telephony | Telephony | Default Name Priority. This
setting can also be adjusted on individual SIP lines to override the system setting.
Directory name matching is not supported for DECT handsets. For information on directory
integration, see IP DECT R4 Installation Manual.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 User

Field Description
Index Range = 00 to 99 or None.
This value is used with personal speed dials set and dialed from M and T-Series phones.
The value can be changed but each value can only be applied to one directory record at
any time. Setting the value to None makes the speed dial inaccessible from M and T-
Series phones, however it may still be accessible from the directory functions of other
phones and applications. The Speed Dial short code feature can be used to create short
codes to dial the number stored with a specific index value. Release 10.0 allows users to
have up to 250 personal directory entries. However, only 100 of those can be assigned
index numbers.
Name Range = Up to 31 characters.
Enter the text to be used to identify the number.
Number Range = Up to 31 digits plus * and #. Enter the number, without spaces, to be dialed.
Wildcards are not supported in user personal directory records. Note that if the system
has been configured to use an external dialing prefix, that prefix should be added to
directory numbers.

Related links
User on page 401

Web Self Administration

Navigation: User | Web Self Administration
Use this page to enable self administration for users.
For a description of the configuration fields available to the user, open the help from the Self
Administration user interface or see the document Avaya IP Office™ Platform Web Self
Administration.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Name Description
Self Adminstration Default = Off.
When enabled, users can log in to the Web Self Administration interface. In a web
browser, enter the IP address of the system in the format http://
<ip_address> and select IP Office Self Administration.
Configuration settings are grouped under the following categories.
• User
• Voicemail
• DND
• Forwarding
• Mobility
• Personal Directory
• Button Programming
• Download Applications
For each option, except Download Applications, the following can be selected:
• Visible: If selected, the user can view the matching settings in the Self
Administration user interface.
• Write: If selected, the users can change the matching settings in the Self
Administration interface.
Download Applications can be set to Visble.
Media Manager Default = Off.
Replay Self-
When enabled, users can replay recordings on the Web Self Administration user
Administration
interface. Configuration settings:
• Enable Media Manager Replay: The field to enable replay feature for a user.
• Replay All Recordings: The field to enable replay of all recordings for a user.
• Replay Own Recordings: The field to enable replay of own recordings for a
user.
• Replay Recordings For Group: The field to enable replay the recordings of the
selected groups for a user. Select the groups from the listed groups.
• Replay Recordings For Others: The field to enable replay of the recordings for
other users. List the users in the text box.
• Download Recordings: The field to enable downloading of recordings for
users.

Related links
User on page 401

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 Group

Group
Additional configuration information
This section provides the Group field descriptions. For additional configuration information, see
Group Operation on page 694.
Related links
Configuration Mode Field Descriptions on page 186
Group on page 451
Queuing on page 456
Overflow on page 459
Fallback on page 461
Group | Voicemail on page 463
Voice Recording on page 469
Announcements on page 470
SIP on page 473

Group
Navigation: Group | Group
Additional configuration information
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Configuration settings
The Group settings are used to define the name, extension number and basic operation of the
group. It is also used to select the group members.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Name Range = Up to 15 characters
The name to identify this group. This field is case sensitive and must be unique.
Names should not start with a space. Do not use punctuation characters such as #, ?, /, ^,
> and ,.
Voicemail uses the name to match a group and its mailbox. Changing a group's name will
route its voicemail calls to a new mailbox. Note however that Voicemail Pro will treat
names such as "Sales", "sales" and "SALES" as being the same.
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Configuration Mode Field Descriptions

Field Description
Profile Default = Standard Hunt Group
Defines the group type. The options are:
• Standard Hunt Group: The default group type and the standard method for creating IP
Office user groups.
• ICR Agent Group: The ICR Agent Group is available as a hunt group in the Profile drop-
down menu if Integrated Contact Reporter is selected as a Contact Center Application at
System > Contact Center . All the three ring modes, that is, Sequential, Rotary, and
Longest waiting are supported with the hunt group configuration. Queuing is supported
with all Integrated Contact Reporter hunt groups.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
• XMPP Group: Extensible Messaging and Presence Protocol (XMPP) is a
communications protocol for presence status and Instant Messaging (IM). Select XMPP
to enable presence information and instant messaging within a defined group of XMPP
enabled one-X clients. Two users can see each other's presence and exchange instant
messages only if they are members of the same XMPP group. A user can be a member
of zero or more groups.

Important:
Before adding a user to an XMPP group, the user must be added to the
configuration and the configuration saved. If the user is added to the group before
the directory is synchronized, the user will not be visible in one-X Portal.
• Centralized Group: Select Centralized Group for extensions that are normally handled
by the core feature server (Avaya Aura Communication Manager) and are handled by
the IP Office only when in survival mode due to loss of connection to the Avaya Aura®
Session Manager. Calls arriving to a centralized hunt group number when the Avaya
Aura Session Manager line is in-service are sent by the IP Office> to Avaya Aura
Session Manager and are then processed by the core feature server according to the
core feature server hunt group configuration. Calls arriving to a centralized hunt group
number when the Avaya Aura Session Manager line is out-of-service are processed by
the IP Office and targeted to the hunt group members as configured on the IP Office.
To provide consistent operation when the Avaya Aura Session Manager line is in-service
or out-of-service, the following is recommended:
- The IP Office hunt group should be configured consistently with the hunt group
administration at the core feature server that serves the survivable branch endpoints in
normal mode.
- Members included in the IP Office hunt group should be only those members that are
in the local branch, even if the core feature server hunt group includes additional
members from other branches (that is, centralized users).
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 Group

Field Description
Extension Range = 1 to 15 digits.
This sets the directory number for calls to the hunt group.
• Groups for CBC and CCC should only use up to 4 digit extension numbers.
• Extension numbers in the range 8897 to 9999 are reserved for use by the IP Office Delta
Server.
Exclude From Default = Off
Directory
When on, the user does not appear in the directory list shown by the user applications and
on phones with a directory function.
Ring Mode Default = Sequential
Sets how the system determines which hunt group member to ring first and the next hunt
group member to ring if unanswered. This is used in conjunction with the User List which
list the order of group membership. The options are:
• Collective All available phones in the User List phones ring simultaneously. Although
DECT handsets can be programmed as members of groups and receive calls in the
same manner as any other extension within that group, you must not configure DECT
handsets into collective groups.
• Collective Call Waiting This is a Collective hunt group as above but with hunt group
call waiting also enabled (previous versions of Manager used a separate Call Waiting
On control to select this option for a Collective group). When an additional call to the
hunt group call is waiting to be answered, users in the group who are already on a call
will receive call waiting indication. On phones with call appearance buttons, the call
waiting indication takes the form of an alert on the next available call appearance button.
On other phones, call waiting indication is given by a tone in the speech path (the tone is
locale specific).
The user's own Call Waiting On setting is overridden when they are using a phone with
call appearances. Otherwise the user's Call Waiting On setting is used in conjunction
with the hunt group setting.
• Sequential Each extension is rung in order, one after the other, starting from the first
extension in the list each time.
• Rotary Each extension is rung in order, one after the other. However, the last extension
used is remembered. The next call received rings the next extension in the list.
• Longest Waiting The extension that has been unused for the longest period rings first,
then the extension that has been idle second longest rings, etc. For extensions with
equal idle time, 'sequential' mode is used.
Where hunt group calls are being presented to a twinned extension, the longest waiting
status of the user can be reset by calls answered at either their master or twinned
extension.
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Configuration Mode Field Descriptions

Field Description
No Answer Time Default = System Default. Range = System Default or 6 to 99999 seconds.
(secs)
The number of seconds an extension rings before the call is passed to another extension
in the list. This applies to all telephones in this group and also any Overflow Groups it
uses. For collective hunt groups, the idea of moving to the next member when the No
Answer Time expires does not apply, instead calls will continue ringing unless overflow or
voicemail is applied.
Hold Music Default = No Change.
Source
The system can support multiple music on hold sources; the System Source (either an
internal file or the external source port or tones) plus a number of additional internal
sources (3 on IP500 V2 systems, 31 on Linux systems), see System | Telephony | Tones &
Music. Before reaching a hunt group, the source used is set by the system wide setting or
by the Incoming Call Route that routed the call. If the system has several hold music
sources available, this field allows selection of the source to associate with calls presented
to this hunt group or to leave it unchanged. The new source selection will then apply even
if the call is forwarded or transferred out of the hunt group unless changed again by
another hunt group. If the call is routed to another system in a multi-site network, the
matching source on that system is used if available.
Calls overflowing from a hunt group will use the hold music source setting of the original
hunt group and ignore the setting of the overflow group.
Calls going to night service or out of service fallback group use the hold music source
setting of the original hunt group and then, if different, the setting of the fallback group. The
setting of further fallback groups from the first are ignored.
Ring Tone Default = Blank
Override
If ring tones have been configured in the System | Telephony | Ring Tones tab, they are
available in this list. Setting a ring tone override applies a unique ring tone for the hunt
group. Ring tone override features are only supported on 1400 Series, 9500 Series and
J100 Series (except J129) phones.
Agent's Status on Default = None (No status change).
No-Answer
For call center agents, that is hunt group members with a log in code and set to forced log
Applies To
in, the system can change the agent's status if they do not answer a hunt group call
presented to them before being automatically presented to the next available agent.
• This setting defines what type of hunt group calls should trigger use of the agent's
Status on No Answer setting. The options are None, Any Call and External Inbound
Calls Only.
• The new status is set by the agent's Status on No Answer (User | Telephony | Supervisor
Settings) setting.
• This action is only applied if the call is unanswered at the agent for the hunt group's No
Answer Time or longer. It does not apply if the call is presented and, before the No
Answer Time expires, is answered elsewhere or the caller disconnects.
• This option is not used for calls ringing the agent because the agent is in another group's
overflow group.
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 Group

Field Description
User List This is an ordered list of the users who are members of the hunt group. For Sequential
and Rotary groups it also sets the order in which group members are used for call
presentation.
• Repeated numbers can be used, for example 201, 202, 201, 203, etc. Each extension
will ring for the number of seconds defined by the No Answer Time before moving to the
next extension in the list, dependent on the Hunt Type chosen.
• The check box next to each member indicates the status of their membership. Group
calls are not presented to members who have their membership currently disabled.
However, those users are still able to perform group functions such as group call pickup.
• The order of the users can be changed by dragging the existing records to the required
position.
• To add records select Edit. A new menu is displayed that shows available users on the
left and current group members of the right. The lists can be sorted and filtered.
• Users on remote systems in a multi-site network can also be included. Groups containing
remote members are automatically advertised within the network.
• Before adding a user to an XMPP group, the user must be added to the configuration
and the configuration saved. If the user is added to the group before the directory is
synchronized, the user will not be visible in one-X Portal.

Related links
Group on page 451
User List Select Members on page 455

User List Select Members

Navigation: Group | Group | Edit
To open the Select Memebers window, click Edit.
The hunt group Select Members form is used to add and remove users from the hunt group. For
hunt group's with a Ring Mode of Sequential or Rotary it is also used to set the order of use for
the members of the hunt group.
The filters section at the top of the form can be used to filter the users shown. Note for hunt
groups set as a CCR Agent Group, only users set as CCR Agent are shown.
The controls and data on the form vary depending on the hunt group's Ring Mode setting and on
whether the system is in a multi-site network.
To sort either table, click on the column header that should be used for the sort the table. Sort
changes the order of display only, it does not change the actual order of hunt group membership.
For Sequential and Rotary hunt groups, an Order column is shown. To change the order position
of a hunt group member, select the member and then use the up and down arrow buttons.

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Configuration Mode Field Descriptions

During the actions below, the Shift and Ctrl keys can be used as normal to select multiple users.
Note that the list of members has been sorted, the sort is updated after adding or moving
members.
• Add Before Using the Shift and/or Ctrl keys, select the users you want to add and then on
the right select the existing member that you want to add them before.
• Add After Using the Shift and/or Ctrl keys, select the users you want to add and then on the
left select the existing member after which you want them added.
• Append Add the selected users on the left to the hunt group members on the right as the last
member in the group order.
• Remove Remove the selected users on the right from the list of hunt group members.
• Move the selected member on the right up or down the membership order of the group.
Related links
Group on page 451

Queuing
Navigation: Group | Queuing
Any calls waiting to be answered at a hunt group are regarded as being queued. The Normalise
Queue Length control allows selection of whether features that are triggered by the queue length
should include or exclude ringing calls. Once one call is queued, any further calls are also queued.
When an available hunt group member becomes idle, the first call in the queue is presented. Calls
are added to the queue until the hunt group's Queue Limit, if set, is reached.
• When the queue limit is reached, any further calls are redirected to the hunt group's
voicemail if available.
• If voicemail is not available excess calls receive busy tone. An exception to this are analog
trunk and T1 CAS trunk calls which will remain queued regardless of the queue limit if no
alternate destination is available.
• If an existing queued call is displaced by a higher priority call, the displaced call will remain
queued even if it now exceeds the queue limit.
Hunt group announcements are separate from queuing. Announcements can be used even if
queuing is turned off and are applied to ringing and queued calls. See Hunt Group |
Announcements.
There are several methods of displaying a hunt group queue.
• Group Button: On phones, with programmable buttons, the Group function can be assigned
to monitor a specified group. The button indicates when there are calls ringing within the
group and also when there are calls queued. The button can be used to answer the longest
waiting call.
• SoftConsole: The SoftConsole applications can display queue monitors for up to 7 selected
hunt groups. This requires the hunt group to have queuing enabled. These queues can be
used by the SoftConsole user to answer calls.

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 Group

When a hunt group member becomes available, the first call in the queue is presented to that
member. If several members become available, the first call in the queue is simultaneously
presented to all the free members.
Overflow Calls Calls that overflow are counted in the queue of the original hunt group from which
they overflow and not that of the hunt group to which they overflow. This affects the Queue Limit
and Calls in Queue Threshold.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Queuing On Default = On
This settings allows calls to this hunt group to be queued. This option is automatically
enabled and cannot be disabled for a CCR agent group.
Queue Length Default = No Limit. Range = No Limit, 1 to 99 calls.
This setting can be used to limit the number of calls that can be queued. Calls exceeding
this limit are passed to voicemail if available or otherwise receive busy tone. This value is
affected by Normalize Queue Length setting.
• If voicemail is not available excess calls receive busy tone. An exception to this is
analog trunk and T1 CAS trunk calls which will remain queued regardless of the queue
limit if no alternate destination is available. This is due to the limited call status signalling
supported by those trunks which would otherwise create scenarios where the caller has
received ringing from the local line provider and then suddenly gets busy from the
system, creating the impression that the call was answered and then hung up.
• If priority is being used with incoming call routes, high priority calls are place ahead of
lower priority calls. If this would exceed the queue limit the limit is temporarily increased
by 1.
• If an existing queued call is displaced by a higher priority call, the displaced call will
remain queued even if it now exceeds the queue limit.
Normalize Queue Default = On.
Length
Calls both waiting to ring and ringing are regarded as being queued. This therefore affects
the use of the Queue Limit and Calls in Queue Alarm thresholds. If Normalize Queue
Length is enabled, the number of hunt group members logged in and not on DND is
added to those thresholds.
For example, a customer has two products that it is selling through a call center with 10
available agents; one product with a $10 margin and one with a $100 margin. Separate
hunt groups with the same 10 members are created for each product.
• The $100 product has a Queue Limit of 5 and Normalize Queue Length is on. The
maximum number of $100 calls that can be waiting to be answered will be 15 (10
ringing/connected + 5 waiting to ring).
• The $10 product has a Queue Limit of 5 and Normalize Queue Length is off. The
maximum number of $10 calls that can be waiting to be answered is 5 (5 ringing/
connected).
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Configuration Mode Field Descriptions

Field Description
Queue Type Default = Assign Call On Agent Answer.
When queuing is being used, the call that the agent receives when they answer can be
assigned in one of two ways:
• Assign Call On Agent Answer In this mode the call answered by the hunt group
member will always be the longest waiting call of the highest priority. The same call will
be shown on all ringing phones in the group. At the moment of answering that may not
necessarily be the same call as was shown by the call details at the start of ringing.
• Assign Call on Agent Alert In this mode, once a call has been presented to a hunt
group member, that is the call they will answer if they go off hook. This mode should be
used when calls are being presented to applications which use the call details such as a
fax server, CTI or TAPI.
Calls In Queue The system can be set to send an alert to a analog specified extension when the number
Alarm of calls queued for the hunt group reaches the specified threshold.
Calls In Queue Default = Off. Range = 1 to 99.
Threshold
Alerting is triggered when the number of queued calls reaches this threshold. Alerting will
stop only when the number of queued calls drops back below this threshold. This value is
affected by Normalize Queue Length setting above.
Analog Extension Default = <None>.
to Notify
This should be set to the extension number of a user associated with an analog extension.
The intention is that this analog extension port should be connected to a loud ringer or
other alerting device and so is not used for making or receiving calls. The list will only
shown analog extensions that are not members of any hunt group or the queuing alarm
target for any other hunt group queue. The alert does not follow user settings such as
forwarding, follow me, DND, call coverage, etc or receive ICLID information.

Group Queue Controls

Group Queue Settings
Manager Hunt group queuing is enabled using the Queuing On option on the Hunt Group |
Queuing tab.
Controls The following short code features/button programming actions can be used:

SoftConsole SoftConsole can display up to 7 hunt group queues (an eight queue is reserved for recall
calls). They are configured by clicking and selecting the Queue Mode tab. For each
queue alarm threshold can be set based on number of queued calls and longest queued
call time. Actions can then be selected for when a queue exceeds its alarm threshold;
Automatically Restore SoftConsole, Ask me whether to restore SoftConsole or
Ignore the Alarm.
Within the displayed queues, the number of queued calls is indicated and the time of the
longest queued call is shown. Exceeding an alarm threshold is indicated by the queue
icons changing from white to red. The longest waiting call in a queue can be answered by
clicking on the adjacent button.

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 Group

Related links
Group on page 451

Overflow
Navigation: Group | Overflow
Overflow can be used to expand the list of group members who can be used to answer a call. This
is done by defining an overflow group or groups. The call is still targeted to the original group and
subject to that group's settings, but is now presented to available members in the overflow groups
in addition to its own available members.
Overflow calls still use the settings of the original target group. The only settings of the overflow
group that is used is it's Ring Mode. For example:
• Calls that overflow use the announcement settings of the group from which they are
overflowing.
• Calls that overflow use the Voicemail Answer Time of the original group from which are are
overflowing.
• Calls that are overflowing are included in the overflowing group's Queue Length and Calls
In Queue Threshold. They are not included in those values for the hunt group to which they
overflow.
• The queuing and overflow settings of the overflow groups are not used, ie. calls cannot
cascade through a series of multiple overflows.
A call will overflow in the following scenarios:
• If Queuing is off and all members of the hunt group are busy, a call presented to the group
will overflow immediately, irrespective of the Overflow Time.
• If Queuing is on and all members of the hunt group are busy, a call presented to the group
will queue for up to the Overflow Time before overflowing.
• If Queuing is on but there are no members logged in or enabled, calls can be set to overflow
immediately by setting the Overflow Immediate setting to No Active Members. Otherwise
calls will queue until the Overflow Time expires.
• If no Overflow Time is set, a call will overflow when it has rung each available hunt group
member without being answered.
• Once one call is in overflow mode, any additional calls will also overflow if the Overflow
Mode is set to Group (the default).
An overflow call is presented to available group members as follows:
• Once a call overflows, it is presented to the first available member of the first overflow group
listed. The Ring Mode of the overflow group is used to determine its first available member.
However the No Answer Time of the original targeted group is used to determine how long
the call is presented.
• When the No Answer Time expires, the call is presented to the next available member in the
overflow group. If all available members in the overflow group have been tried, the first
member in the next listed overflow group is tried.

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Configuration Mode Field Descriptions

• When the call has been presented to all available members in the overflow groups, it is
presented back to the first available member in the original target group.
• While the call is being presented to members in an overflow group, the announcement and
voicemail settings of the original targeted group are still applied.
For calls being tracked by the Customer Call Reporter application, overflow calls are recorded
against the original targeted group but using separate statistics; Overflowed Calls, Overflowed
Calls Waiting, Overflowed Answered and Overflowed Lost. For full details refer to the
Customer Call Reporter User Guide.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Overflow Time Default = Blank. Range = Off or 1 to 3600 seconds.
For a group using queuing, the Overflow Time sets how long a call queues before being
presented to available agents in the group's Overflow Group List. Note that if the call is
currently ringing an agent when the timer expires, it will complete ringing for the group's
No Answer Time before overflowing.
Overflow Mode Default = Group.
This option allows selection of whether the overflow of queued calls is determined on an
individual call by call basis or is applied to all calls once any one call overflows. The
options are:
• Group: In this mode, once one call overflows all additional queued calls also overflow.
• Call: In this mode, each individual call will follow the group's overflow settings before it
overflows.
Immediate Default = Off.
Overflow:
For groups which are using queueing, this setting can be used to control whether calls
should overflow immediately when there are no available or active agents. The options
are:
• Off: Do not overflow immediately. Use the Overflow Time setting as normal.
• No Active Agents: Overflow immediately if there are no available or active agents as
defined above, regardless of the Overflow Time setting.
- An active agent is an agent who is either busy on a call or in after call work. An
available agent is one who is logged in and enabled in the hunt group but is otherwise
idle.
- A hunt group is automatically treated as having no available or active agents if:
- The group's extension list is empty.
- The group's extension list contains no enabled users.
- The group's extension list contains no extensions that resolve to a logged in agent (or
mobile twin in the case of a user logged out mobile twinning).
Table continues…

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 Group

Field Description
Overflow Group This list is used to set the group or groups that are used for overflow. Each group is used
List in turn, in order from the top of the list. The call is presented to each overflow group
member once, using the Ring Mode of the overflow group. If the call remains
unanswered, the next overflow group in the list is used. If the call remains unanswered at
the end of the list of overflow groups, it is presented to available members of the original
targeted group again and then to those in its overflow list in a repeating loop. A group can
be included in the overflow list more than once if required and the same agent can be in
multiple groups.

Related links
Group on page 451

Fallback
Navigation: Group | Fallback
Fallback settings can be used to make a hunt group unavailable and to set where the hunt group's
calls should be redirected at such times. Hunt groups can be manually placed In Service, Out of
Service or in Night Service. Additionally using a time profile, a group can be automatically placed
in Night Service when outside the Time Profile settings.
Fallback redirects a hunt group's calls when the hunt group is not available, for example outside
normal working hours. It can be triggered either manually or using an associated time profile.
Group Service States:
A hunt group can be in one of three states; In Service, Out of Service or Night Service. When In
Service, calls are presented as normal. In any other state, calls are redirected as below.
Call Redirection:
The following options are possible when a hunt group is either Out of Service or in Night
Service.
• Destination: When in Out of Service, if an Out of Service Destination has been set, calls
are redirected to that destination. When in Night Service, if a Night Service Destination
has been set, calls are redirected to that destination.
• Voicemail: If no fallback destination has been set but voicemail is enabled for the group,
calls are redirected to voicemail.
• Busy Tone: If no fallback destination has been set and voicemail is not available, busy tone
is returned to calls.
Manually Controlling the Service State:
Manager and or short codes can be used to change the service state of a hunt group. The short
code actions can also be assigned to programmable buttons on phones.
•
The icon is used for a hunt group manually set to Night Service mode.
•
The icon is used for a hunt group manually set to Out of Service mode.

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Configuration Mode Field Descriptions

Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported. You can manually override a time profile.
Time Profile:
A Day Service Time Profile can be associated with the hunt group. A time profile if required, is
set through Time Profile | Time Profile.
When outside the time profile, the hunt group is automatically placed into night service. When
inside the time profile, the hunt group uses manually selected mode.
• When outside the time profile and therefore in night service, manual night service controls
cannot be used to override the night service. However the hunt group can be put into out of
service.
• When a hunt group is in Night Service due to a time profile, this is not indicated within
Manager.
• Time profile operation does not affect hunt groups set to Out of Service.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Day Service Time Default = <None> (No automatic night service)
Profile
This field allows selection of a previously created Time Profile. That profile then specifies
the times at which it should use the manually selected Service Mode settings. Outside the
period defined in the time profile, the hunt group behaves as if set to Night Service mode.
Note that when a hunt group is in Night Service due to it associated time profile, this is not
reflected by the Service Mode on this tab. Note also that the manual controls for changing
a hunt group's service mode cannot be used to take a hunt group out of time profile night
service.
Night Service Default = <None> (Voicemail or Busy Tone)
Destination
This field sets the alternate destination for calls when this hunt group is in Night Service
mode. The destination can be a group, a user, a short code, or an Auto Attendant. Select
a group or user from the drop down list. Manually enter a short code or an Auto Attendant
name.
If left blank, calls are redirected to voicemail if available or otherwise receive busy tone.
Out of Service Default = <None> (Voicemail or Busy Tone)
Fallback Group
This field sets the alternate destination for calls when this hunt group is in Out of Service
mode. The destination can be a group, a user, a short code, or an Auto Attendant. Select
a group or user from the drop down list. Manually enter a short code or an Auto Attendant
name. For Auto Attendant names, use the format AA:Name.
If left blank, calls are redirected to voicemail if available or otherwise receive busy tone.
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 Group

Field Description
Mode Default = In Service
This field is used to manually select the current service mode for the hunt group. The
options are:
• In Service: When selected the hunt group is enabled. This is the default mode.
• Night Service: When selected, calls are redirected using the Night Service Fallback
Group setting. This setting can also be manually controlled using the short code and
button programming features Set Hunt Group Night Service and Clear Hunt Group Night
Service.
• Out of Service: When selected, calls are redirected using the Out of Service Fallback
Group setting. This setting can also be manually controlled using the short code and
button programming features Set Hunt Group Out of Service and Clear Hunt Group Out
of Service.

Hunt Group Fallback Controls

The following short code features and button programming actions can be used.
Feature/Action Short Code Default Button
Set Hunt Group Night Yes *20*N# Yes — Toggles
Service
Clear Hunt Group Night Yes *21*N# Yes
Service
Set Hunt Group Out of No No Yes — Toggles
Service
Clear Hunt Group Out of No No Yes
Service

Note that for a hunt group using a time profile, these controls only are only applied when the hunt
group is within the specified time profile period. When outside its time profile, the hunt group is in
night service mode and cannot be overridden.
Related links
Group on page 451

Group | Voicemail
Navigation: Group | Voicemail
The system supports voicemail for hunt groups in addition to individual user voicemail mailboxes.
If voicemail is available and enabled for a hunt group, it is used in the following scenarios.
• Voicemail Answer Time: A call goes to voicemail when this timeout is reached, regardless
of any announcement, overflow, queuing or other settings. The default timeout is 45 seconds.

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Configuration Mode Field Descriptions

• Unanswered Calls: A call goes to voicemail when it has been presented to all the available
hunt group members without being answered. If overflow is being used, this includes be
presented to all the available overflow group members.
• Night Service: A call goes to voicemail if the hunt group is in night service with no Night
Service Fallback Group set.
• Out of Service: A call goes to voicemail if the hunt group is out of service with no Out of
Service Fallback Group set.
• Queue Limit Reached: If queuing is being used, it overrides use of voicemail prior to expiry
of the Voicemail Answer Time, unless the number of queued callers exceeds the set Queue
Limit. By default there is no set limit.
• Automatic Call Recording: Incoming calls to a hunt group can be automatically recorded
using the settings on the Hunt Group | Voice Recording tab.
When a caller is directed to voicemail to leave a message, the system indicates the target user or
hunt group mailbox.
The mailbox of the originally targeted user or hunt group is used. This applies even if the call has
been forwarded to another destination. It also includes scenarios where a hunt group call
overflows or is in fallback to another group.
Voicemail Pro can be used to customize which mailbox is used separately from the mailbox
indicated by the system.
By default no user is configured to receive message waiting indication when a hunt group
voicemail mailbox contains new messages. Message waiting indication is configured by adding a
H groupname record to a user's SourceNumbers tab (User | Source Numbers).
By default, no mechanism is provided for access to specific hunt group mailboxes. Access needs
to be configured using either a short code, programmable button or source number.
• Intuity Emulation Mailbox Mode:For systems using Intuity emulation mode mailboxes, the
hunt group extension number and voicemail code can be used during normal mailbox
access.
• Avaya Branch Gateway Mailbox Mode or IP Office Mailbox Mode: For this mode of
mailbox access, short codes or a Voicemail Collect button are required to access the mailbox
directly.
The voicemail system (Voicemail Pro only) can be instructed to automatically forward messages to
the individual mailboxes of the hunt group members. The messages are not stored in the hunt
group mailbox.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Group

Field Description
Voicemail On Default = On
When on, the mailbox is used by the system to answer the any calls to the group that
reach the Voicemail Answer Time. Note that selecting off does not disable use of the
group mailbox. Messages can still be forward to the mailbox and recordings can be
placed in it. The mailbox can also still be accessed to collect messages.
When a caller is directed to voicemail to leave a message, the system indicates the target
user or hunt group mailbox.
• The mailbox of the originally targeted user or hunt group is used. This applies even if
the call has been forwarded to another destination. It also includes scenarios where a
hunt group call overflows or is in fallback to another group.
• Voicemail Pro can be used to customize which mailbox is used separately from the
mailbox indicated by the system.
Voicemail Answer Default = 45 seconds. Range = Off, 1 to 99999 seconds.
Time
This setting sets how long a call should be presented to a hunt group, and its overflow
groups if set, before going to voicemail. When exceeded the call goes to voicemail (if
available) regardless of any announcements, overflow, queuing or any other actions. If
set to Off, voicemail is used when all available members of the hunt group have been
alerted for the no answer time.
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Configuration Mode Field Descriptions

Field Description
Voicemail Code Default = Blank. Range = 0 (no code) to 15 digits.
A code used by the voicemail server to validate access to this mailbox. If remote access
is attempted to a mailbox that has no voicemail code set, the prompt "Remote access is
not configured on this mailbox" is played.
The mailbox access code can be set through IP Office Manager or through the mailbox
telephone user interface (TUI). The minimum password length is:
• Voicemail Pro (Manager) - 0
• Voicemail Pro (Intuity TUI) - 2
• Embedded Voicemail (Manager) - 0
• Embedded Voicemail (Intuity TUI) - 0
Codes set through the Voicemail Pro telephone user interface are restricted to valid
sequences. For example, attempting to enter a code that matches the mailbox extension,
repeat the same number (1111) or a sequence of numbers (1234) are not allowed. If
these types of code are required they can be entered through Manager.
Manager does not enforce any password requirements for the code if one is set through
Manager.
• Embedded Voicemail For Embedded Voicemail running in IP Office mailbox mode, the
voicemail code is used if set.
• IP Office mode The voicemail code is required when accessing the mailbox from a
location that is not set as a trusted number in the user's Source Numbers list.
• Intuity Emulation mode By default the voicemail code is required for all mailbox
access. The first time the mailbox is accessed the user will be prompted to change the
password. Also if the voicemail code setting is left blank, the caller will be prompted to
set a code when they next access the mailbox. The requirement to enter the voicemail
code can be removed by adding a customized user or default collect call flow, refer to
the Voicemail Pro manuals for full details.
• Trusted Source Access The voicemail code is required when accessing the mailbox
from a location that is not set as a trusted number in the user's Source Numbers list.
• Call Flow Password Request Voicemail Pro call flows containing an action where the
action's PIN code set to $ will prompt the user for their voicemail code.
Voicemail Help Default = Off
This option controls whether users retrieving messages are automatically given an
additional prompt "For help at any time press 8." If switched off, users can still press 8 for
help. For voicemail systems running in Intuity emulation mode, this option has no effect.
On those systems the default access greeting always includes the prompt "For help at
any time, press *4" (*H in the US locale).
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 Group

Field Description
Broadcast Default = Off. (Voicemail Pro only).
When enabled, if a voicemail message is left for the hunt group, copies of the message
are forwarded to the mailboxes of the individual group members. The original message in
the hunt group mailbox is deleted unless it occurred as the result of call recording. This
feature is not applied to recordings created by Voice Question actions.
UMS Web Default = Off.
Services
This option is used with Voicemail Pro. If enabled, the hunt group mailbox can be
accessed using either an IMAP email client or a web browser. Note that the mailbox must
have a voicemail code set in order to use either of the UMS interfaces. UMS Web
Service licenses are required for the number of groups configured.
In the License section, double-clicking on the UMS Web Services license display a menu
that allows you to add and remove users and groups from the list of those enabled for
UMS Web Services without having to open the settings of each individual user or group.
Voicemail Email: Default = Blank (No voicemail email features)
This field is used to set the user or group email address used by the voicemail server for
voicemail email operation. When an address is entered, the additional Voicemail Email
control below are selectable to configure the type of voicemail email service that should
be provided.
Use of voicemail email requires the Voicemail Pro server to have been configured to use
either a local MAPI email client or an SMTP email server account. For Embedded
Voicemail, voicemail email is supportedand uses the system's SMTP settings.
The use of voicemail email for the sending (automatic or manual) of email messages with
wav files attached should be considered with care. A one-minute message creates a
1MB .wav file. Many email systems impose limits on emails and email attachment sizes.
For example the default limit on an Exchange server is 5MB.
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Configuration Mode Field Descriptions

Field Description
Voicemail Email Default = Off
If an email address is entered for the user or group, the following options become
selectable. These control the mode of automatic voicemail email operation provided by
the voicemail server whenever the voicemail mailbox receives a new voicemail message.
Users can change their voicemail email mode using visual voice. If the voicemail server is
set to IP Office mode, user can also change their voicemail email mode through the
telephone prompts. The ability to change the voicemail email mode can also be provided
by Voicemail Pro in a call flow using a Play Configuration Menu action or a Generic
action.
If the voicemail server is set to IP Office mode, users can manually forward a message to
email.
The options are:
• Off If off, none of the options below are used for automatic voicemail email. Users can
also select this mode by dialing *03 from their extension.
• Copy If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a copy of the message is attached to an email and sent to the email
address. There is no mailbox synchronization between the email and voicemail
mailboxes. For example reading and deletion of the email message does not affect the
message in the voicemail mailbox or the message waiting indication provided for that
new message.
• Forward If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, that message is attached to an email and sent to the email address.
No copy of the voicemail message is retained in the voicemail mailbox and their is no
message waiting indication. As with Copy, there is no mailbox synchronization between
the email and voicemail mailboxes. Users can also select this mode by dialing *01 from
their extension.
Note that until email forwarding is completed, the message is present in the voicemail
server mailbox and so may trigger features such as message waiting indication.
• UMS Exchange 2007 With Voicemail Pro, the system supports voicemail email to an
Exchange 2007 server email account. For users and groups also enabled for UMS Web
Services this significantly changes their mailbox operation. The Exchange Server inbox
is used as their voicemail message store and features such as message waiting
indication are set by new messages in that location rather than the voicemail mailbox
on the voicemail server. Telephone access to voicemail messages, including Visual
Voice access, is redirected to the Exchange 2007 mailbox.
• Alert If this mode is selected, each time a new voicemail message is received in the
voicemail mailbox, a simple email message is sent to the email address. This is an
email message announcing details of the voicemail message but with no copy of the
voicemail message attached. Users can also select this mode by dialing *02 from their
extension.

Related links
Group on page 451

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 Group

Voice Recording
Navigation: Group | Voice Recording
This tab is used to configure automatic recording of external calls handled by hunt group
members. The recording of internal calls as well is also supported.
Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
Note the following:
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Calls parked or held pause recording until the unparked or taken off hold (does not apply to
SIP terminals).
• Recording is stopped if:
- User recording stops if the call is transferred to another user.
- User account code recording stops if the call is transferred to another user.
- Hunt group recording stops if the call is transferred to another user who is not a member of
the hunt group.
- Incoming call route recording continues for the duration of the call on the system.
A destination mailbox other than the hunt group's own mailbox can be specified as the destination
for recordings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Record Inbound Default = None
Select whether automatic recording of incoming calls is enabled. The options are:
• None: Do not automatically record calls.
• On: Record the call if possible. If not possible to record, allow the call to continue.
• Mandatory: Record the call if possible. If not possible to record, block the call and
return busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Record Time Default = <None> (Any time)
Profile
Used to select a time profile during which automatic call recording of incoming calls is
applied. If no profile is selected, automatic recording of incoming calls is active at all times.
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Configuration Mode Field Descriptions

Field Description
Recording (Auto) Default = Mailbox
Sets the destination for automatically triggered recordings. The options are:
• Mailbox This option sets the destination for the recording to be a selected user or hunt
group mailbox. The adjacent drop down list is used to select the mailbox.
• Voice Recording Library: This options set the destination for the recording to be a VRL
folder on the voicemail server. The ContactStore application polls that folder and collects
waiting recordings which it then places in its own archive. Recording is still done by the
Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to Voice Recording
Library above but instructs the voicemail server to create an authenticated recording. If
the file contents are changed, the file is invalidated though it can still be played. This
option is currently not supported with Linux based servers.
Auto Record Calls Default = External.
This setting allows selection of whether External or External & Internal calls are subject
to automatic call recording.

Related links
Group on page 451

Announcements
Navigation: Group | Announcements
Announcements are played to callers waiting to be answered. This includes callers being
presented to hunt group members, ie. ringing, and callers queued for presentation.
• The system supports announcements using Voicemail Pro or Embedded Voicemail.
• If no voicemail channel is available for an announcement, the announcement is not played.
• In conjunction with Voicemail Pro, the system allows a number of voicemail channels to be
reserved for announcements. See System | Voicemail.
• With Voicemail Pro, the announcement can be replaced by the action specified in a Queued
(1st announcement) or Still Queued (2nd announcement) start point call flow. Refer to the
Voicemail Pro Installation and Maintenance documentation for details.
• Calls can be answered during the announcement. If it is a mandatory requirement that
announcements should be heard before a call is answered, then a Voicemail Pro call flow
should be used before the call is presented.
Note:
Call Billing and Logging
Acall becomes connected when the first announcement is played to it. That connected
state is signaled to the call provider who may start billing at that point. The call will also
be recorded as answered within the SMDR output once the first announcement is played.

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 Group

• If a call is rerouted to a hunt group's Night Service Group or Out of Service Fallback Group,
the announcements of the new group are applied.
• If a call overflows, the announcements of the original group are still applied, not those of the
overflow group.
• For announcements to be used effectively, the hunt group's Voicemail Answer Time must
be extended or Voicemail On must be unselected.
Recording the Group Announcement
Voicemail Pro provides a default announcement "I'm afraid all the operators are busy but please
hold and you will be transferred when somebody becomes available". This default is used for
announcement 1 and announcement 2 if no specific hunt group announcement has been
recorded. Embedded Voicemail does not provide any default announcement. Voicemail Lite also
provides the default announcements.
The maximum length for announcements is 10 minutes. New announcements can be recorded
using the following methods.
Voicemail Lite: Access the hunt group mailbox and press 3. Then press either 3 to record the 1st
announcement for the hunt group or 4 to record the 2nd announcement for the hunt group.
Voicemail Pro : The method of recording announcements depends on the mailbox mode being
used by the voicemail server.
• IP Office Mailbox Mode: Access the hunt group mailbox and press 3. Then press either 3 to
record the 1st announcement for the hunt group or 4 to record the 2nd announcement for the
hunt group.
• Intuity Emulation Mailbox Mode: There is no mechanism within the Intuity telephony user
interface (TUI) to record hunt group announcements. To provide custom announcements,
hunt group queued and still queued start points must be configured with Voicemail Pro with
the required prompts played by a generic action.
Embedded Voicemail: Embedded Voicemail does not include any default announcement or
method for recording announcements. The Record Message short code feature is provided to
allow the recording of announcements. The telephone number field of short codes using this
feature requires the extension number followed by either ".1" for announcement 1 or ".2" for
announcement 2. For example, for extension number 300, the short codes *91N# | Record
Message | N".1" and *92N# | Record Message | N".2" could be used to allow recording of the
announcements by dialing *91300# and *92300#.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Announcements Default = Off.
On
This setting enables or disables announcements.
Wait before 1st Default = 10 seconds. Range = 0 to 255 seconds.
announcement:
This setting sets the time delay from the calls presentation, after which the first
announcement should be played to the caller. If Synchronize Calls is selected, the actual
wait may differ, see below.
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Configuration Mode Field Descriptions

Field Description
Flag call as Default = Off.
answered
This setting is used by the CCC and CBC applications. By default they do not regarded a
call as answered until it has been answered by a person or by a Voicemail Pro action with
Flag call as answered selected. This setting allows calls to be marked as answered
once the caller has heard the first announcement. This setting is not used by the
Customer Call Reporter application.
Post Default = Music on hold.
announcement
Following the first announcement, you can select whether the caller should hear Music on
tone
Hold, Ringing or Silence until answered or played another announcement.
2nd Default = On.
Announcement
If selected, a second announcement can be played to the caller if they have still not been
answered.
Wait before 2nd Default = 20 seconds. Range = 0 to 255 seconds.
announcement
This setting sets the wait between the 1st and the 2nd announcement. If Synchronize
Calls is selected, the actual wait may differ, see below.
Repeat last Default = On.
announcement
If selected, the last announcement played to the caller is repeated until they are answered
or hang-up.
Wait before repeat Default = 20 seconds. Range = 0 to 255 seconds.
If Repeat last announcement is selected, this setting sets is applied between each
repeat of the last announcement. If Synchronize Calls is selected, this value is grayed
out and set to match the Wait before 2nd announcement setting.
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 Group

Field Description
Synchronize calls Default = Off
This option can be used to restrict how many voicemail channels are required to provide
the announcements.
When Synchronize calls is off, announcement are played individually for each call. This
requires a separate voicemail channel each time an announcement is played to each
caller. While this ensures accurate following of the wait settings selected, it does not make
efficient use of voicemail channels.
When Synchronize calls is on, if a required announcement is already being played to
another caller, further callers wait until the announcement been completed and can be
restarted. In addition, when a caller has waited for the set wait period and the
announcement is started, any other callers waiting for the same announcement hear it
even if they have not waited for the wait period. Using this setting, the maximum number
of voicemail channels ever needed is 1 or 2 depending on the number of selected
announcements.

Note:
Interaction with Voicemail Pro Queued and Still Queued Start Points If either
custom Queued or Still Queued start point call flows are being used for the
announcements, when Synchronize Calls is enabled those call flows will support
the playing of prompts only. Voicemail Pro actions such as Speak ETA, Speak
Position, Menu, Leave Mail, Transfer and Assisted Transfer, etc. are not
supported.

Related links
Group on page 451

SIP
Navigation: Group | SIP
Each hunt group can be configured with its own SIP URI information. For calls received on a SIP
line where any of the line's SIP URI fields are set to Use Internal Data, if the call is presented to
the hunt group that data is taken from these settings.
This form is hidden if there are no system multi-site network lines in the configuration or no SIP
lines with a URI set to Use Internal Data.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
SIP Name: Default = Blank on Voicemail tab/Extension number on other tabs.
The value from this field is used when the From field of the SIP URI being used for a SIP
call is set to Use Internal Data.
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Configuration Mode Field Descriptions

Field Description
SIP Display Name Default = Blank on Voicemail tab/Name on other tabs.
(Alias)
The value from this field is used when the Display Name field of the SIP URI being used
for a SIP call is set to Use Internal Data.
Contact Default = Blank on Voicemail tab/Extension number on other tabs. The value from this
field is used when the Contact field of the SIP URI being used for a SIP call is set to Use
Internal Data.
Anonymous Default = On on Voicemail tab/Off on other tabs. If the From field in the SIP URI is set to
Use Internal Data, selecting this option inserts Anonymous into that field rather than the
SIP Name set above.

Related links
Group on page 451

Short Code
Navigation: Short Code | Short Code
Additional configuration information
This section provides the Short Code field descriptions. For additional configuration information,
see Short Code Features on page 809.
Configuration settings
These settings are used to create System Short Codes. System short codes can be dialed by all
system users. However the system short code is ignored if the user dialing matches a user or user
rights short code.
Warning:
User dialing of emergency numbers must not be blocked. If short codes are edited, the users
ability to dial emergency numbers must be tested and maintained.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Code The dialing digits used to trigger the short code. Maximum length 31 characters.
Feature Select the action to be performed by the short code.
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 Service

Field Description
Telephone The number dialed by the short code or parameters for the short code feature. This field
Number can contain numbers and characters. For example, it can contain Voicemail Pro start point
names, user names, hunt group names and telephone numbers (including those with
special characters). Maximum length 31 characters.
The majority of North-American telephony services use 'en-bloc' dialing, ie. they expect to
receive all the routing digits for a call as a single simultaneous set of digits. Therefore the
use of a ; is recommended at the end of all dialing short codes that use an N. This is also
recommended for all dialing where secondary dial tone short codes are being used.
Line Group ID Default = 0.
For short codes that result in the dialing of a number, that is short codes with a Dial
feature, this field is used to enter the initially routing destination of the call. The drop down
can be used to select the following from the displayed list:
• Outgoing Group ID: The Outgoing Group ID's current setup within the system
configuration are listed. If an Outgoing Group ID is selected, the call will be routed to
the first available line or channel within that group.
• ARS: The ARS records currently configured in the system are listed. If an ARS record is
selected, the call will be routed by the setting within that ARS record. Refer to ARS
Overview.
Locale Default = Blank.
For short codes that route calls to voicemail, this field can be used to set the prompts
locale that should be used if available on the voicemail server.
Force Account Default = Off.
Code
For short codes that result in the dialing of a number, this field trigger the user being
prompted to enter a valid account code before the call is allowed to continue.
Force Default = Off.
Authorization
This option is only shown on systems where authorization codes have been enabled. If
Code
selected, then for short codes that result in the dialing of a number, the user is required to
enter a valid authorization code in order to continue the call.

Related links
Configuration Mode Field Descriptions on page 186

Service
Normal, WAN, or Internet Services
Services are used to configure the settings required when a user or device on the LAN needs to
connect to a off-switch data service such as the Internet or another network. Services can be used
when making data connections via trunk or WAN interfaces.

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Configuration Mode Field Descriptions

Once a service is created, it can be used as the destination for an IP Route record. One service
can also be set as the Default Service. That service will then be used for any data traffic received
by the system for which no IP Route is specified.
The system supports the following types of service:
• Normal Service This type of service should be selected when for example, connecting to an
ISP.
• WAN Service This type of service is used when creating a WAN link. A User and RAS
Service will also be created with the same name. These three records are automatically
linked and each open the same form. Note however, that this type of Service cannot be used
if the Encrypted Password option is checked. In this case the RAS Service name must match
the Account Name. Therefore either create each record manually or create an Intranet
Service.
• Intranet Service This type of service can be selected to automatically create a User with the
same name at the same time. These two records are linked and will each open the same
form. The User's password is entered in the Incoming Password field at the bottom on the
Service tab. An Intranet Services shares the same configuration tabs as those available to
the WAN Service.
SSL VPN Services
For full details on how to configure and administer SSL VPN services, refer to Deploying Avaya IP
Office™ Platform SSL VPN Services.
Related links
Configuration Mode Field Descriptions on page 186
Service on page 476
Bandwidth on page 477
IP on page 479
Autoconnect on page 481
Quota on page 481
PPP on page 482
Fallback on page 484
Dial In on page 485
SSL VPN Service on page 485

Service
Navigation: Service | Service
Additional configuration information
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Configuration settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Service

Field Description
Service Name The name of the service. It is recommended that only alphanumeric characters be used.
Account Name The user name that is used to authenticate the connection. This is provided by the ISP or
remote system.
Password Default = Blank
Enter the password that is used to authenticate the connection. This is provided by the
ISP or remote system.
Telephone Number Default = Blank
If the connection is to be made via ISDN enter the telephone number to be dialed. This is
provided by the ISP or remote system.
Firewall Profile Default = Internet01 if present, otherwise <None>
From the list box select the Firewall Profile that is used to allow/disallow protocols
through this Service.
Encrypted Default = Off When enabled the password is authenticated via CHAP (this must also be
Password supported at the remote end). If disabled, PAP is used as the authentication method.
Default Route Default = Off
When enabled this Service is the default route for data packets unless a blank IP Route
has been defined in the system IP Routes. A green arrow appears to the left of the
Service in the Configuration Tree. Only one Service can be the default route. If disabled,
a route must be created under IP Route.
Incoming Default = Blank Shown on WAN and Intranet services. Enter the password that will be
Password used to authenticate the connection from the remote Control Unit. (If this field has
appeared because you have created a Service and User of the same name, this is the
password you entered in the User's Password field).

Related links
Service on page 475

Bandwidth
Navigation: Service | Bandwidth
These options give the ability to make ISDN calls between sites only when there is data to be sent
or sufficient data to warrant an additional call. The calls are made automatically without the users
being aware of when calls begin or end. Using ISDN it is possible to establish a data call and be
passing data in less that a second.
Note:
The system will check Minimum Call Time first, then Idle Period, then the Active Idle
Period.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Minimum No of Default = 1. Range = 1 to 30.
Channels
Defines the number of channels used to connect for an outgoing connection. The
initial channel must be established and stable, before further calls are made.
Maximum No of Default = 1. Range = 1 to 30.
Channels
Defines the maximum number of channels to can be used. This field should contain a
value equal to or greater than the Minimum Channels field.
Extra BW Default = 50%. Range = 0 to 100%.
Threshold
Defines the utilization threshold at which extra channels are connected. The value
entered is a %. The % utilization is calculated over the total number of channels in
use at any time, which may be one, two etc.
For example, if Minimum Channels set to 1, Maximum Channels set to 2 and Extra
Bandwidth set to 50 - once 50% of first channel has been used the second channel
is connected.
Reduce BW Default = 10%. Range = 0 to 100%.
Threshold
Defines the utilization threshold at which additional channels are disconnected. The
value entered is a %. Additional calls are only dropped when the % utilization,
calculated over the total number of channels in use, falls below the % value set for a
time period defined by the Service-Idle Time. The last call (calls - if Minimum Calls is
greater than 1) to the Service is only dropped if the % utilization falls to 0, for a time
period defined by the Service-Idle Time. Only used when 2 or more channels are set
above.
For example, if Minimum Channels set to 1, Maximum Channels set to 2 and Reduce
Bandwidth is set to 10 - once the usage of the 2 channels drops to 10% the number
of channels used is 1.
Callback Default = Blank
Telephone
The number that is given to the remote service, via BAP, which the remote Control
Number
Unit then dials to allow the bandwidth to be increased. Incoming Call routing and RAS
Services must be appropriately configured.
Idle Period Default = 10 seconds. Range = 0 to 999999 seconds.
(secs)
The time period, in seconds, required to expire after the line has gone idle. At this
point the call is considered inactive and is completely closed.
For example, the 'Idle Period' is set to X seconds. X seconds before the 'Active Idle
Period' timeouts the Control Unit checks the packets being transmitted/received, if
there is nothing then at the end of the 'Active Idle Period' the session is closed & the
line is dropped. If there are some packets being transmitted or received then the line
stays up. After the 'Active Idle Period' has timed out the system performs the same
check every X seconds, until there are no packets being transferred and the session
is closed and the line dropped.
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 Service

Field Description
Active Idle Default = 180 seconds. Range = 0 to 999999 seconds.
Period (secs):
Sets the time period during which time the line has gone idle but there are still active
sessions in progress (for example an FTP is in process, but not actually passing data
at the moment). Only after this timeout will call be dropped.
For example, you are downloading a file from your PC and for some reason the other
end has stopped responding, (the remote site may have a problem etc.) the line is
idle, not down, no data is being transmitted/ received but the file download session is
still active. After the set time period of being in this state the line will drop and the
sessions close. You may receive a remote server timeout error on your PC in the
Browser/FTP client you were using.
Minimum Call Default = 60 seconds. Range = 0 to 999999 seconds.
Time (secs):
Sets the minimum time that a call is held up after initial connection. This is useful if
you pay a minimum call charge every time a call is made, no matter the actual length
of the call. The minimum call time should be set to match that provided by the line
provider.
Extra Bandwidth Default = Incoming Outgoing
Mode
Defines the mode of operation used to increases bandwidth to the initial call to the
remote Service. The options are:
• Outgoing Only Bandwidth is added by making outgoing calls.
• Incoming Only Bandwidth is added by the remote service calling back on the
BACP number (assuming that BACP is successfully negotiated).
• Outgoing Incoming Uses both methods but bandwidth is first added using
outgoing calls.
• Incoming Outgoing Uses both methods but bandwidth is first added using
incoming BACP calls.

Related links
Service on page 475

IP
Navigation: Service | IP
The fields in this tab are used to configure network addressing for the services you are running.
Depending on how your network is configured, the use of Network Address Translation (NAT) may
be required.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
IP Address Default = 0.0.0.0 (address assigned by ISP)
An address should only be entered here if a specific IP address and mask have been
provided by the Service Provider. Note that if the address is in a different domain from the
system then NAT is automatically enabled
IP Mask Default = 0.0.0.0 (use NAT)
Enter the IP Mask associated with the IP Address if an address is entered.
Primary Transfer Default = 0.0.0.0 (No transfer)
IP Address
This address acts as a primary address for incoming IP traffic. All incoming IP packets
without a session are translated to this address. This would normally be set to the local
mail or web server address.
For control units supporting a LAN1 and LAN2, the primary transfer address for each LAN
can be set through the System | LAN1 and System | LAN2 tabs.
RIP Mode Default = None
Routing Information Protocol (RIP) is a method by which network routers can exchange
information about device locations and routes. RIP can be used within small networks to
allow dynamic route configuration as opposed to static configuration using. The options
are:
• None The LAN does not listen to or send RIP messages.
• Listen Only (Passive) Listen to RIP-1 and RIP-2 messages in order to learn RIP routes
on the network.
• RIP1 Listen to RIP-1 and RIP-2 messages and send RIP-1 responses as a sub-network
broadcast.
• RIP2 Broadcast (RIP1 Compatibility) Listen to RIP-1 and RIP-2 messages and send
RIP-2 responses as a sub-network broadcast.
• RIP2 Multicast Listen to RIP-1 and RIP-2 messages and send RIP-2 responses to the
RIP-2 multicast address.
Request DNS Default = Off.
When selected, DNS information is obtained from the service provider. To use this, the
DNS Server addresses set in the system configuration (System | DNS) should be blank.
The PC making the DNS request should have the system set as its DNS Server. For
DHCP clients the system will provide its own address as the DNS server.
Forward Multicast Default = On.
Messages
By default this option is on. Multicasting allows WAN bandwidth to be maximized through
the reduction of traffic that needs to be passed between sites.

Related links
Service on page 475

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 Service

Autoconnect
Navigation: Service | Autoconnect
These settings enable you to set up automatic connections to the specified Service.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Auto Connect Default = 0 (disabled). Range = 0 to 99999 minutes.
Interval (mins):
This field defines how often this Service will automatically be called ("polled"). For
example setting 60 means the system will call this Service every hour in the absence of
any normally generated call (this timer is reset for every call; therefore if the service is
already connected, then no additional calls are made). This is ideal for SMTP Mail polling
from Internet Service Providers.
Auto Connect Default = <None>
Time Profile
Allows the selection of any configured Time Profiles. The selected profile controls the time
period during which automatic connections to the service are made. It does NOT mean
that connection to that service is barred outside of these hours. For example, if a time
profile called "Working Hours" is selected, where the profile is defined to be 9:00AM to
6:00PM Monday to Friday, then automatic connection to the service will not be made
unless its within the defined profile. If there is an existing connection to the service at
9:00AM, then the connection will continue. If there is no connection, then an automatic
connection will be made at 9:00AM.

Related links
Service on page 475

Quota
Navigation: Service | Quota
Quotas are associated with outgoing calls, they place a time limit on calls to a particular IP
Service. This avoids excessive call charges when perhaps something changes on your network
and call frequency increases unintentionally.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Quota Time Default = 240 minutes. Range = 0 to 99999 minutes.
(mins)
Defines the number of minutes used in the quota. When the quota time is used up no
further data can be passed to this service. This feature is useful to stop things like an
internet game keeping a call to your ISP open for a long period.

Warning:
Setting a value here without selecting a Quota period below will stop all further calls
after the Quota Time has expired.
Quota: Default = Daily. Range = None, Daily, Weekly or Monthly
Sets the period during which the quota is applied. For example, if the Quota Time is 60
minutes and the Quota is set to Daily, then the maximum total connect time during any
day is 60 minutes. Any time beyond this will cause the system to close the service and
prevent any further calls to this service. To disable quotas select None and set a Quota
Time of zero.

Note:
The ClearQuota feature can be used to create short codes to refresh the quota time.

Related links
Service on page 475

PPP
Navigation: Service | PPP
These settings enable you to configure Point to Point Protocol (PPP) in relation to this particular
service. PPP is a protocol for communication between two computers using a Serial interface.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Chap Challenge Default = 0 (disabled). Range = 0 to 99999 seconds. The period between CHAP
Interval (secs) challenges. Blank or 0 disables repeated challenges. Some software such as Windows 95
DUN does not support repeated CHAP challenges.
Bi-Directional Default =Off.
Chap
Header Default = None selected
Compression
Enables the negotiation and use of IP Header Compression. Supported modes are IPHC
and VJ. IPHC should be used on WAN links.
Table continues…

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 Service

Field Description
PPP Compression Default = MPPC
Mode
Enables the negotiate and use of compression. Do not use on VoIP WAN links. The
options are:
• Disable Do not use or attempt to use compression.
• StacLZS Attempt to use STAC compression (Mode 3, sequence check mode).
• MPPC Attempt to use MPPC compression. Useful for NT Servers.
PPP Callback Default = Disabled.
Mode
The options are:
• Disable Callback is not enabled
• LCP (Link Control Protocol) After authentication the incoming call is dropped and an
outgoing call to the number configured in the Service is made to re-establish the link.
• Callback CP (Microsoft's Callback Control Protocol) After acceptance from both ends
the incoming call is dropped and an outgoing call to the number configured in the
Service is made to re-establish the link.
• Extended CBCP (Extended Callback Control Protocol) Similar to Callback CP except
the Microsoft application at the remote end prompts for a telephone number. An
outgoing call is then made to that number to re-establish the link.
PPP Access Mode Default = Digital64
Sets the protocol, line speed and connection request type used when making outgoing
calls. Incoming calls are automatically handled (see RAS services). The options are:
• Digital64 Protocol set to Sync PPP, rate 64000 bps, call presented to local exchange as
a "Data Call".
• Digital56 As above but rate 56000 bps.
• Voice56 As above but call is presented to local exchange as a "Voice Call".
• V120 Protocol set to Async PPP, rate V.120, call presented to local exchange as a "Data
Call". This mode runs at up to 64K per channel but has a higher Protocol overhead than
pure 64K operation. Used for some bulletin board systems as it allows the destination
end to run at a different asynchronous speed to the calling end.
• V110 Protocol is set to Async PPP, rate V.110. This runs at 9600 bps, call is presented
to local exchange as a "Data Call". It is ideal for some bulletin boards.
• Modem Allows Asynchronous PPP to run over an auto-adapting Modem to a service
provider (requires a Modem2 card in the main unit)
Data Pkt. Size Default = 0. Range = 0 to 2048.
Sets the size limit for the Maximum Transmissible Unit.
BACP Default = Off.
Enables the negotiation and use of BACP/BCP protocols. These are used to control the
addition of B channels to increase bandwidth.
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Configuration Mode Field Descriptions

Field Description
Incoming traffic Default = On.
does not keep link
When enabled, the link is not kept up for incoming traffic only.
up
Multilink/QoS Default = Off.
Enables the negotiation and use of Multilink protocol (MPPC) on links into this Service.
Multilink must be enabled if there is more than one channel that is allowed to be Bundled/
Multilinked to this RAS Service.

Related links
Service on page 475

Fallback
Navigation: Service | Fallback
These settings allow you to set up a fallback for the Service. For example, you may wish to
connect to your ISP during working hours and at other times take advantage of varying call
charges from an alternative carrier. You could therefore set up one Service to connect during peak
times and another to act as fallback during the cheaper period.
You need to create an additional Service to be used during the cheaper period and select this
service from the Fallback Service list box (open the Service form and select the Fallback tab).
If the original Service is to be used during specific hours and the Fallback Service to be used
outside of these hours, a Time Profile can be created. Select this Time Profile from the Time
Profile list box. At the set time the original Service goes into Fallback and the Fallback Service is
used.
A Service can also be put into Fallback manually using short codes, for example:
Put the service "Internet" into fallback:
• Short Code: *85
• Telephone Number: "Internet"
• Line Group ID: 0
• Feature: SetHuntGroupNightService
Take the service "Internet" out of fallback:
• Short Code: *86
• Telephone Number: "Internet"
• Line Group ID: 0
• Feature: ClearHuntGroupNightService
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Service

Field Description
In Fallback Default = Off.
This option indicates whether the Service is in Fallback or not. A service can be set into
fallback using this setting. Alternatively a service can be set into fallback using a time
profile or short codes.
Time profile Default = <None> (No automatic fallback)
Select the time profile you wish to use for the service. The time profile should be set up for
the hours that you wish this service to be operational, out of these hours the Fallback
Service is used.
Fallback Service Default = <None>
Select the service that is used when this service is in fallback.

Related links
Service on page 475

Dial In
Navigation: Service | Dial In
Only available for WAN and Intranet Services. This tab is used to define a WAN connection.
To define a WAN connection, click Add and enter WAN if the service is being routed via a WAN port
on a WAN3 expansion module.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Related links
Service on page 475

SSL VPN Service

The SSL VPN service provides secure tunneling between the Avaya IP Office hardware installed
at a customer site and a remote Avaya VPN Gateway (AVG). This secure tunnel allows support
personnel to offer remote management services to customers, such as fault management,
monitoring, and administration.
For full details on how to configure and administer SSL VPN services, refer to Deploying Avaya IP
Office™ Platform SSL VPN Services.
Related links
Service on page 475
Service on page 486
Session on page 486
NAPT on page 487
Fallback on page 488

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Configuration Mode Field Descriptions

Service
Navigation: SSL VPN Service | Service
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Service Name Enter a name for the SSL VPN service.
Account Name Enter the SSL VPN service account name. This account name is used for authenticating
the SSL VPN service when connecting with the Avaya VPN Gateway (AVG).
Account Password Enter the password for the SSL VPN service account.
Confirm Password Confirm the password for the SSL VPN service account.
Server Address Enter the address of the VPN gateway. The address can be a fully qualified domain
name or an IPv4 address
Server Type Default = AVG. This field is fixed to AVG (Avaya VPN Gateway).
Server Port Default = 443. Select a port number.
Number

Related links
SSL VPN Service on page 485

Session
Navigation: SSL VPN Service | Session
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Session Mode Default = Always On.
This setting is greyed out and cannot be adjusted.
Preferred Data Default = UDP.
Transport Protocol
This is the protocol used by the SSL VPN service for data transport. Only TCP is
supported. If you select UDP as the protocol when you configure the connection, UDP
displays in this field but the SSL VPN service falls back to TCP.
Heartbeat Interval Default = 30 seconds. Range = 1 to 600 seconds.
Enter the length of the interval between heartbeat messages, in seconds. The default
value is 30 seconds.
Heartbeat Retries Default = 4. Range = 1 to 10.
Enter the number of unacknowledged heartbeat messages that IP Office sends to AVG
before determining that AVG is not responsive. When this number of consecutive
heartbeat messages is reached and AVG has not acknowledged them, IP Office ends
the connection.
Table continues…

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 Service

Field Description
Keepalive Interval Default = 10 seconds. Range = 0 (Disabled) to 600 seconds.
Not used for TCP connections. Keepalive messages are sent over the UDP data
transport channel to prevent sessions in network routers from timing out.
Reconnection Default = 60 seconds. Range = 1 to 600 seconds.
Interval on Failure
The interval the system waits attempting to re-establish a connection with the AVG. The
interval begins when the SSL VPN tunnel is in-service and makes an unsuccessful
attempt to connect with the AVG, or when the connection with the AVG is lost. The
default is 60 seconds.

Related links
SSL VPN Service on page 485

NAPT
Navigation: SSL VPN Service | NAPT
The Network Address Port Translation (NAPT) rules are part of SSL VPN configuration. NAPT
rules allow a support service provider to remotely access LAN devices located on a private IP
Office network. You can configure each SSL VPN service instance with a unique set of NAPT
rules. You can configure up to 64 rules.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
The SSL VPN restarts after a setting change.
Field Description
Application Default = Blank
Defines the communication application used to connect to the LAN device through the SSL
VPN tunnel. When you select an application, the Protocol and Port Number fields are
filled with the default values. The drop-down Application selector options and the
associated default values are:
External and Internal Port
Application Protocol Number
Custom TCP 0
VMPro TCP 50791
OneXPortal TCP 8080
SSH TCP 22
TELNET TCP 23
RDP TCP 3389
WebControl TCP 7070
Protocol Default = TCP
The protocol used by the application. The options are TCP and UDP.
Table continues…

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Configuration Mode Field Descriptions

Field Description
External Port Default = the default port number for the application. Range = 0 to 65535
Number
Defines the port number used by the application to connect from the external network to
the LAN device in the customer private network.
Internal IP Default = Blank.
address
The IP address of the LAN device in the customer network.
Internal Port Default = the default port number for the application. Range = 0 to 65535
Number
Defines the port number used by the application to connect to the LAN device in the
customer private network.

Related links
SSL VPN Service on page 485

Fallback
Navigation: SSL VPN Service | Fallback
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
In Fallback Default = Off.
This setting is used to indicate whether the SSL VPN service is in use or not.
• To configure the service without establishing an SSL VPN connection, or to disable an
SSL VPN connection, select this option.
• To enable the service and establish an SSL VPN connection, de-select this option.
• The Set Hunt Group Night Service and Clear Hunt Group Night Service short code
and button features can be used to switch an SSL VPN service off or on respectively.
The service is indicated by setting the service name as the telephone number or action
data. Do not use quotation marks.

Related links
SSL VPN Service on page 485

RAS
Navigation: RAS | RAS
A Remote Access Server (RAS) is a piece of computer hardware which sits on a corporate LAN
and into which employees dial on the public switched telephone network to get access to their
email and to software and data on the corporate LAN.
This form is used to create a RAS service that the system offers Dial In users. A RAS service is
needed when configuring modem dial in access, digital (ISDN) dial in access and a WAN link.

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 RAS

Some systems may only require one RAS service since the incoming call type can be
automatically sensed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Name A textual name for this service. If Encrypted Password below is used, this name must
match the Account Name entered in the Service form.
Extension Enter an extension number if this service is to be accessed internally.
COM Port For future use.
TA Enable Default = Off
Select to enable or disable - if enabled RAS will pass the call onto a TA port for external
handling.
Encrypted Default = Off
Password
This option is used to define whether Dial In users are asked to use PAP or CHAP during
their initial log in to the RAS Service. If the Encrypted Password box is checked then Dial
In users are sent a CHAP challenge, if the box is unchecked PAP is used as the Dial In
Authorization method.

Related links
Configuration Mode Field Descriptions on page 186
PPP on page 489

PPP
Navigation: RAS | PPP
PPP (Point-to-Point Protocol) is a Protocol for communication between two computers using a
Serial interface, typically a personal computer connected by phone line to a server.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
CHAP Challenge Default = 0 (disabled). Range = 0 to 99999 seconds.
Interval (secs)
The period between successive CHAP challenges. Blank or 0 disables repeated
challenges. Some software, for example Windows 95 DUN, does not support repeated
CHAP challenges.
Header Default = Off
Compression
Enables the negotiation and use of IP Header Compression as per RFC2507, RFC2508
and RFC2509.
Table continues…

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Configuration Mode Field Descriptions

Field Description
PPP Compression Default = MPPC This option is used to negotiate compression (or not) using CCP. If set to
Mode MPPC or StacLZS the system will try to negotiate this mode with the remote Control Unit.
If set to Disable CCP is not negotiated. The options are:
• Disable Do not use or attempt to use compression.
• StacLZS Attempt to use and negotiate STAC compression (the standard, Mode 3)
• MPPC Attempt to use and negotiate MPPC (Microsoft) compression. Useful for dialing
into NT Servers.
PPP Callback Default = Disable
Mode
The options are:
• Disable: Callback is not enabled
• LCP: (Link Control Protocol) After authentication the incoming call is dropped and an
outgoing call to the number configured in the Service will be made to reestablish the
link.
• Callback CP: (Microsoft's Callback Control Protocol) After acceptance from both ends
the incoming call is dropped and an outgoing call to the number configured in the
Service is made to reestablish the link.
• Extended CBCP: (Extended Callback Control Protocol) Similar to Callback CP however
the Microsoft application at the remote end will prompt for a telephone number. An
outgoing call will then be made to that number to reestablish the link.
Data Pkt. Size Default = 0. Range = 0 to 2048.
This is the number of data bytes contained in a Data Packet.
BACP Default = Off
Allows negotiation of the BACP/BCP protocols. These are used to control the addition of
additional B channels to simultaneously improve data throughput.
Multilink Default = Off
When enabled the system attempts to negotiate the use of the Multilink protocol (MPPC)
on the link(s) into this Service. Multilink must be enabled if the more than one channel is
allowed to be Bundled/Multilinked to this RAS Service.

Related links
RAS on page 488

Incoming Call Route

Incoming call routes are used to determine the destination of voice and data calls received by the
system. On systems where a large number incoming call routes need to be setup for DID
numbers, the MSN/DID Configuration tool can be used.

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 Incoming Call Route

Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Determining which incoming call route is used is based on the call matching a number of possible
criteria. In order of highest priority first, the criteria, which if set must be matched by the call in
order for the call to use that route are:
1. The Bearer Capability indicated, if any, with the call. For example whether the call is a
voice, data or video call.
2. The Incoming Group ID of the trunk or trunk channel on which the call was received.
3. The Incoming Number received with the call.
4. The Incoming Sub Address received with the call.
5. The Incoming CLI of the caller.
Multiple Matches
If there is a match between more than one incoming call route record, the one added to the
configuration first is used.
Incoming Call Route Destinations
Each incoming route can include a fallback destination for when the primary destination is busy. It
can also include a time profile which control when the primary destination is used. Outside the
time profile calls are redirected to a night service destination. Multiple time profiles can be
associated with an incoming call route. Each time profile used has its own destination and fallback
destination specified.
Incoming Call Routing Examples
Example 1
For this example, the customer has subscribes to receive two 2-digit DID numbers. They want
calls on one routed to a Sales hunt group and calls on the other to a Services hunt group. Other
calls should use the normal default route to hunt group Main. The following incoming call routes
were added to the configuration to achieve this:
Line Group Incoming Number Destination
0 77 Sales
0 88 Services
0 blank Main

Note that the incoming numbers could have been entered as the full dialed number, for example
7325551177 and 7325551188 respectively. The result would still remain the same as incoming
number matching is done from right-to-left.
Line Group Incoming Number Destination
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Configuration Mode Field Descriptions

0 7325551177 Sales
0 7325551188 Services
0 blank Main

Example 2
In the example below the incoming number digits 77 are received. The incoming call route records
677 and 77 have the same number of matching digit place and no non-matching places so both a
potential matches. In this scenario the system will use the incoming call route with the Incoming
Number specified for matching.
Line Group Incoming Number Destination
0 677 Support
0 77 Sales
0 7 Services
0 blank Main

Example 3
In the following example, the 677 record is used as the match for 77 as it has more matching digits
than the 7 record and no non-matching digits.
Line Group Incoming Number Destination
0 677 Support
0 7 Services
0 blank Main

Example 4
In this example the digits 777 are received. The 677 record had a non-matching digit, so it is not a
match. The 7 record is used as it has one matching digit and no non-matching digits.
Line Group Incoming Number Destination
0 677 Support
0 7 Services
0 blank Main

Example 5
In this example the digits 77 are received. Both the additional incoming call routes are potential
matches. In this case the route with the shorter Incoming Number specified for matching is used
and the call is routed to Services.
Line Group Incoming Number Destination
0 98XXX Support
0 8XXX Services
0 blank Main

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 Incoming Call Route

Example 6
In this example two incoming call routes have been added, one for incoming number 6XXX and
one for incoming number 8XXX. In this case, any three digit incoming numbers will potential match
both routes. When this occurs, potential match that was added to the system configuration first is
used. If 4 or more digits were received then an exact matching or non-matching would occur.
Line Group Incoming Number Destination
0 6XXX Support
0 8XXX Services
0 blank Main

Related links
Configuration Mode Field Descriptions on page 186
Standard on page 493
Voice Recording on page 497
Destinations on page 498

Standard
Navigation: Incoming Call Route | Standard
Additional configuration information
For additional information on the Tag setting, see Call Tagging on page 664.
Incoming call routes are used to match call received with destinations. Routes can be based on
the incoming line group, the type of call, incoming digits or the caller's ICLID. If a range of
MSN/DID numbers has been issued, this form can be populated using the MSN Configuration tool.
In Manager, see Tools > MSN Configuration.
Default Blank Call Routes
By default the configuration contains two incoming calls routes; one set for Any Voice calls
(including analog modem) and one for Any Data calls. While the destination of these default
routes can be changed, it is strongly recommended that the default routes are not deleted.
• Deleting the default call routes, may cause busy tone to be returned to any incoming external
call that does not match any incoming call route.
• Setting any route to a blank destination field, may cause the incoming number to be checked
against system short codes for a match. This may lead to the call being rerouted off-switch.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can
be used to modify the incoming digits.
If there is no matching incoming call route for a call, matching is attempted against system short
codes and finally against voicemail nodes before the call is dropped.

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Configuration Mode Field Descriptions

SIP Calls
For SIP calls, the following fields are used for call matching:
• Line Group ID This field is matched against the Incoming Group settings of the SIP URI
(Line | SIP URI). This must be an exact match.
• Incoming Number This field can be used to match the called details (TO) in the SIP header
of incoming calls. It can contain a number, SIP URI or Tel URI. For SIP URI's the domain part
of the URI is removed before matching by incoming call routing occurs. For example, for the
SIP URI [email protected] , only the user part of the URI, ie. mysip, is used for matching.
The Call Routing Method setting of the SIP line can be used to select whether the value used for
incoming number matching is taken from the To Header or the Request URI information provided
with incoming calls on that line.
Incoming CLI This field can be used to match the calling details (FROM) in the SDP header of
incoming SIP calls. It can contain a number, SIP URI, Tel URI or IP address received with SIP
calls. For all types of incoming CLI except IP addresses a partial record can be used to achieve
the match, records being read from left to right. For IP addresses only full record matching is
supported.
Configuration Settings
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Incoming Call Matching Fields:
The following fields are used to determine if the Incoming Call Route is a potential match for the
incoming call. By default the fields are used for matching in the order shown starting with Bearer
Capability.
Field Description
Bearer Capability Default = Any Voice
The type of call selected from the list of standard bearer capabilities. The options are:
• Any
• Any Voice
• Any Data
• Speech
• Audio 3K1
• Data 56K
• Data 64K
• Data V110
• Video
Table continues…

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 Incoming Call Route

Field Description
Line Group ID Default = 0. Range = 0 to 99999.
Matches against the Incoming Line Group to which the trunk receiving the call belongs.
For Server Edition systems, the default value 0 is not allowed. You must change the
default value and enter the unique Line Group ID for the line.
Incoming Default = Blank (Match any unspecified)
Number
Matches to the digits presented by the line provider. A blank record matches all calls that
do not match other records. By default this is a right-to-left matching. The options are:
• * = Incoming CLI Matching Takes Precedence
• – = Left-to-Right Exact Length Matching Using a - in front of the number causes a left-
to-right match. When left-to-right matching is used, the number match must be the same
length. For example -96XXX will match a DID of 96000 but not 9600 or 960000.
• X = Single Digit Wildcard Use X's to enter a single digit wild card character. For
example 91XXXXXXXX will only match DID numbers of at least 10 digits and starting
with 91, -91XXXXXXXX would only match numbers of exactly 10 digits starting with 91.
Other wildcard such as N, n and ? cannot be used.
Where the incoming number potentially matches two incoming call routes with X
wildcards and the number of incoming number digits is shorter than the number of
wildcards, the one with the shorter overall Incoming Number specified for matching is
used.
• i = ISDN Calling Party Number 'National' The i character does not affect the incoming
number matching. It is used for Outgoing Caller ID Matching, see notes below.
Incoming Sub Default = Blank (Match all)
Address
Matches any sub address component sent with the incoming call. If this field is left blank, it
matches all calls.
Incoming CLI Default = Blank (Match all) Enter a number to match the caller's ICLID provided with the
call. This field is matched left-to-right. The number options are:
• Full telephone number.
• Partial telephone number, for example just the area code.
• ! : Matches calls where the ICLID was withheld.
• ? : for number unavailable.
• Blank for all.

Call Setting Fields:

For calls routed using this Incoming Call Route, the settings of the following fields are applied to
the call regardless of the destination.

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Configuration Mode Field Descriptions

Field Description
Locale Default = Blank (Use system setting)
This option specifies the language prompts, if available, that voicemail should use for the call
if it is directed to voicemail.
Priority Default = 1-Low. Range = 1-Low to 3-High.
This setting allows incoming calls to be assigned a priority. Other calls such as internal calls
are assigned priority 1-Low
In situations where calls are queued, high priority calls are placed before calls of a lower
priority. This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where Voicemail Pro is
being used to provided queue ETA and queue position messages to callers since those
values will no longer be accurate when a higher priority call is placed into the queue. Note
also that Voicemail Pro will not allow a value already announced to an existing caller to
increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt group's
Queue Length Limit, the limit is temporarily raised by 1. This means that calls already
queued are not rerouted by the addition of a higher priority call into the queue.
A timer can be used to increase the priority of queued calls, see the setting System |
Telephony | Telephony | Call Priority Promotion Time.
The current priority of a call can be changed through the use of the p short code character in
a short code used to transfer the call.
Tag Default = Blank (No tag).
Allows a text tag to be associated with calls routed by this incoming call route. This tag is
displayed with the call within applications and on phone displays.
Hold Music Default = System source.
Source
The system can support several music on hold sources. See System | Telephony | Tones
and Music.
If the system has several hold music sources available, this field allows selection of the
source to associate with calls routed by this incoming call route. The new source selection
will then apply even if the call is forwarded or transferred away from the Incoming Call Route
destination. If the call is routed to another system in a multi-site network, the matching
source on that system is used if available. The hold music source associated with a call can
also be changed by a hunt group's Hold Music Source setting.
Ring Tone Default = Blank
Override
If ring tones have been configured in System | Telephony | Ring Tones, they are available
in this list. Setting a ring tone override applies a unique ring tone for the incoming call route.

Outgoing Caller ID Matching

In cases where a particular Incoming Number is routed to a specific individual user, the system will
attempt to use that Incoming Number as the user's caller ID when they make outgoing calls if no
other number is specified. This requires that the Incoming Number is a full number suitable for
user as outgoing caller ID and acceptable to the line provider.

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 Incoming Call Route

When this is the case, the character i can also be added to the Incoming Number field. This
character does not affect the incoming call routing. However when the same Incoming Number is
used for an outgoing caller ID, the calling party number plan is set to ISDN and the type is set to
National. This option may be required by some network providers.
For internal calls being forwarded or twinned, if multiple incoming call route entries match the
extension number used as caller ID, the first entry created is used. This entry should start with a
“-” character (meaning fixed length) and provide the full national number. These entries do not
support wildcards. If additional entries are required for incoming call routing, they should be
created after the entry required for reverse lookup.
Related links
Incoming Call Route on page 490

Voice Recording
Navigation: Incoming Call Route | Voice Recording
These settings are used to activate the automatic recording of incoming calls that match the
incoming call route.
Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.
Note the following:
• Calls to and from IP devices, including those using Direct media, can be recorded.
• Calls parked or held pause recording until the unparked or taken off hold (does not apply to
SIP terminals).
• Recording is stopped if:
- User recording stops if the call is transferred to another user.
- User account code recording stops if the call is transferred to another user.
- Hunt group recording stops if the call is transferred to another user who is not a member of
the hunt group.
- Incoming call route recording continues for the duration of the call on the system.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Record Inbound Default = None
Select whether automatic recording of incoming calls is enabled. The options are:
• None: Do not automatically record calls.
• On: Record the call if possible. If not possible to record, allow the call to continue.
• Mandatory: Record the call if possible. If not possible to record, block the call and return
busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Record Time Default = <None> (Any time)
Profile
Used to select a time profile during which automatic call recording of incoming calls is
applied. If no profile is selected, automatic recording of incoming calls is active at all times.
Recording Default = Mailbox
(Auto)
Sets the destination for automatically triggered recordings. The options are:
• Mailbox This option sets the destination for the recording to be a selected user or hunt
group mailbox. The adjacent drop down list is used to select the mailbox.
• Voice Recording Library: This options set the destination for the recording to be a VRL
folder on the voicemail server. The ContactStore application polls that folder and collects
waiting recordings which it then places in its own archive. Recording is still done by the
Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to Voice Recording
Library above but instructs the voicemail server to create an authenticated recording. If the
file contents are changed, the file is invalidated though it can still be played. This option is
not currently supported with Linux based systems.

Related links
Incoming Call Route on page 490

Destinations
Navigation: Incoming Call Route | Destinations
The system allows multiple time profiles to be associated with an incoming call route. For each
time profile, a separate Destination and Fallback Extension can be specified.
When multiple records are added, they are resolved from the bottom up. The record used will be
the first one, working from the bottom of the list upwards, that is currently 'true', ie. the current day
and time or date and time match those specified by the Time Profile. If no match occurs the
Default Value options are used.
Once a match is found, the system does not use any other destination set even if the intended
Destination and Fallback Extension destinations are busy or not available.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 Incoming Call Route

For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Time Profile This column is used to specify the time profiles used by the incoming call routes. It displays a
drop-down list of existing time profiles from which a selection can be made. To remove an
existing entry, select it by clicking on the button on the left of the row, then right-click on the
row and select Delete.
The Default Value entry is fixed and is used if no match to a time profile below occurs.
Destination Default = Blank
Either enter the destination manually or select the destination for the call from the drop-down
list. The dr box which contains all available extensions, users, groups, RAS services and
voicemail. System short codes and dialing numbers can be entered manually. Once the
incoming call is matched the call is passed to that destination.
The following options appear in the drop-down list:
• Voicemail allows remote mailbox access with voicemail. Callers are asked to enter the
extension ID of the mailbox required and then the mailbox access code.
• Local user names.
• Local hunt groups names.
• AA: Name directs calls to an Embedded Voicemail auto-attendant services.
In addition to short codes, extension and external numbers, the following options can be also
be entered manually:
• VM:Name Directs calls to the matching start point in Voicemail Pro.
• A . matches the Incoming Number field. This can be used even when X wildcards are being
used in the Incoming Number field.
• A # matches all X wildcards in the Incoming Number field. For example, if the Incoming
Number was -91XXXXXXXXXXX, the Destination of # would match XXXXXXXXXXX.
• Text and number strings entered here are passed through to system short codes, for
example to direct calls into a conference. Note that not all short code features are
supported.
• If necessary, quote marks can be used to stop characters in the destination string being
interpreted as special characters.
Fallback Default = Blank (No fallback)
Extension
Defines an alternate destination which should be used when the current destination, set in
the Destination field cannot be obtained. For example if the primary destination is a hunt
group returning busy and without queuing or voicemail.

Related links
Incoming Call Route on page 490

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Configuration Mode Field Descriptions

WAN Port
These settings are used to configure the operation of system WAN ports and services.
WAN services can be run over a T1 PRI trunk connection. This requires creation of a virtual WAN
port. For full details refer to Using a Dedicated T1/PRI ISP Link.
Related links
Configuration Mode Field Descriptions on page 186
WAN Port on page 500
Frame Relay on page 501
DLCIs on page 501
Advanced on page 503

WAN Port
Navigation: WAN Port | WAN Port
Use these settings to configure a WAN port.
On IP500 V2 systems, these settings configure the leased line connected to the WAN port on the
Control Unit. Normally this connection is automatically detected by the control unit. If a WAN Port
is not displayed, connect the WAN cable, reboot the Control Unit and receive the configuration.
The WAN Port configuration form is now be added.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Name The physical ID of the Extension port,. This parameter is not configurable; it is allocated
by the system.
Speed The operational speed of this port. For example for a 128K connection, enter 128000.
This should be set to the actual speed of the leased line as this value is used in the
calculation of bandwidth utilization. If set incorrectly, additional calls may be made to
increase Bandwidth erroneously.
Mode Default = SyncPPP
Select the protocol required. The options are:
• SyncPPP For a data link.
• SyncFrameRelay For a link supporting Frame Relay.
RAS Name If the Mode is SyncPPP, selects the RAS service to associate with the port. If the Mode
is SyncFrameRelay, the RAS Name is set through the DLCIs tab.

Related links
WAN Port on page 500

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 WAN Port

Frame Relay
Navigation: WAN Port | Frame Relay
These settings are for Frame Relay configuration.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Frame This must match the management type expected by the network provider. Selecting
Management Type AutoLearn allows the system to automatically determine the management type based on
the first few management frames received. If a fixed option is required the following
options are supported:
• Q933 AnnexA 0393
• Ansi AnnexD
• FRFLMI
• None
Frame Learn Mode This parameter allows the DLCIs that exist on the given WAN port to be provisioned in a
number of different ways.
• None No automatic learning of DLCIs. DLCIs must be entered and configured
manually.
• Mgmt Use LMI to learn what DLCIs are available on this WAN.
• Network Listen for DLCIs arriving at the network. This presumes that a network
provider will only send DLCIs that are configured for this particular WAN port.
• NetworkMgmt Do both management and network listening to perform DLCI learning
and creation.
Max Frame Length Maximum frame size that is allowed to traverse the frame relay network.
Fragmentation The options are:
Method
• RFC1490
• RFC1490+FRF12

Related links
WAN Port on page 500

DLCIs
Navigation: WAN Port | DLCIs
DLCIs are created for Frame Relay connections. These settings are not mergeable. Changes to
these settings will require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Frame Link Type Default = PPP
Data transfer encapsulation method. Set to the same value at both ends of the PVC
(Permanent Virtual Channel). The options are:
• None
• PPP Using PPP offers features such as out of sequence traffic reception, compression
and link level connection management.
• RFC 1490 RFC 1490 encapsulation offers performance and ease of configuration and
more inter-working with third party CPE.
• RFC1490 + FRF12 Alternate encapsulation to PPP for VoIP over Frame Relay. When
selected all parameters on the Service | PPP tab being used are overridden.
DLCI Default = 100 This is the Data Link Connection Identifier, a unique number assigned to a
PVC end point that has local significance only. Identifies a particular PVC endpoint
within a user's physical access channel in a frame relay.
RAS Name Select the RAS Service you wish to use.
Tc Default = 10
This is the Time Constant in milliseconds. This is used for measurement of data traffic
rates. The Tc used by the system can be shorter than that used by the network provider.
CIR (Committed Information Rate) Default = 64000 bps This is the Committed Information
Rate setting. It is the maximum data rate that the WAN network provider has agreed to
transfer. The committed burst size (Bc) can be calculated from the set Tc and CIR as Bc
= CIR x Tc. For links carrying VoIP traffic, the Bc should be sufficient to carry a full VoIP
packet including all its required headers. See the example below.
EIR (Excess Information Rate) Default = 0 bps This is the maximum amount of data in
excess of the CIR that a frame relay network may attempt to transfer during the given
time interval. This traffic is normally marked as De (discard eligible). Delivery of De
packets depends on the network provider and is not guaranteed and therefore they are
not suitable for UDP and VoIP traffic. The excess burst size (Be) can be calculated as
Be = EIR x Tc.

Example: Adjusting the Tc Setting

G.729 VoIP creates a 20 byte packet every 20ms. Adding typical WAN PPP headers results in a
33 byte packet every 20ms.
For a Committed Information Rate (CIR) of 14Kbps, with the Time Constant (Tc) set to 10ms; we
can calculate the Committed Burst size:
Bc = CIR x Tc = 14,000 x 0.01 = 140 bits = 17.5 bytes.
Using 10ms as the Tc, a full G.729 VoIP packet (33 bytes) cannot be sent without exceeding the
Bc. The most likely result is lost packets and jitter.
If the Tc is increased to 20ms:
Bc = CIR x Tc = 14,000 x 0.02 = 280 bits = 35 bytes.
The Bc is now sufficient to carry a full G.729 VoIP packet.

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 WAN Port

Notes:
1. Backup over Frame Relay is not supported when the Frame Link Type is set to RFC1490.
2. When multiple DLCIs are configured, the WAN link LED is switched off if any of those
DLCIs is made inactive, regardless of the state of the other DLCIs. Note also that the WAN
link LED is switched on following a reboot even if one of the DLCIs is inactive. Therefore
when multiple DLCIs are used, the WAN link LED cannot be used to determine the current
state of all DLCIs.
3. When the Frame Link Type is set to RFC1490, the WAN link LED is switched on when the
WAN cable is attached regardless other whether being connected to a frame relay
network.
Related links
WAN Port on page 500

Advanced
Navigation: WAN Port | Advanced
These settings are used for Frame Relay connections.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Address Length The address length used by the frame relay network. The network provider will indicate if
lengths other than two bytes are to be used.
N391 Full Status Polling Counter
Polling cycles count used by the CPE and the network provider equipment when
bidirectional procedures are in operation. This is a count of the number of link integrity
verification polls (T391) that are performed (that is Status Inquiry messages) prior to a Full
Status Inquiry message being issued.
N392 Error Threshold Counter
Error counter used by both the CPE and network provider equipment. This value is
incremented for every LMI error that occurs on the given WAN interface. The DLCIs
attached to the given WAN interface are disabled if the number of LMI errors exceeds this
value when N393 events have occurred. If the given WAN interface is in an error condition
then that error condition is cleared when N392 consecutive clear events occur.
N393 Monitored Events Counter
Events counter measure used by both the CPE and network provider equipment. This
counter is used to count the total number of management events that have occurred in
order to measure error thresholds and clearing thresholds.
T391 Link Integrity Verification Polling Timer
The link integrity verification polling timer normally applies to the user equipment and to
the network equipment when bidirectional procedures are in operation. It is the time
between transmissions of Status Inquiry messages.
Table continues…

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Configuration Mode Field Descriptions

Field Description
T392 Polling Verification Timer The polling verification timer only applies to the user equipment
when bidirectional procedures are in operation. It is the timeout value within which to
receive a Status Inquiry message from the network in response to transmitting a Status
message. If the timeout lapses an error is recorded (N392 incremented).

Related links
WAN Port on page 500

Directory Entry
Navigation: Directory | Directory Entry
Additional configuration information
For additional configuration information, see Centralized System Directory on page 589.
Configuration settings
Use these settings to create directory records that are stored in the system's configuration.
Directory records can also be manually imported from a CSV file. The system can also use
Directory Services to automatically import directory records from an LDAP server at regular
intervals.
A system can also automatically import directory records from another system. Automatically
imported records are used as part of the system directory but are not part of the editable
configuration. Automatically imported records cannot override manually entered records.
For a Server Edition network, these settings can only be configured at the network level and they
are stored in the configuration of the Primary Server. All other systems in the network are
configured to share the directory settings of the Primary Server through their Manager settings at
System | Directory Services | HTTP.
Directory Special Characters
The following characters are supported in directory records. They are supported in both system
configuration records and in imported records.
• ? = Any Digit Directory records containing a ? are only used for name matching against the
dialed or received digits on outgoing or incoming calls. They are excluded from the dialable
directory. In the following example, any calls where the dialed or received number that starts
9732555 will have the display name Homdel associated with them.
- Name: Holmdel
- Number: 9732555?
• ( ) = Optional Digits Brackets can be used to enclose an optional portion of a number,
typically the area code. Only one pair of brackets are supported in a number. Records
containing digits inside ( ) brackets are only used for user dialling. The full string is dialed with
the ( ) brackets removed.

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 Time Profile

• - Characters Directory records can also contain - characters. Records containing -

characters are only used for user dialing. The full string is dialed with the - character
removed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Index Range = 000 to 999 or None.
This value is used with system speed dials dialed from M and T-Series phones. The value
can be changed but each value can only be applied to one directory record at any time.
Setting the value to None makes the speed dial inaccessible from M and T-Series phones,
however it may still be accessible from the directory functions of other phone types and
applications. The Speed Dial short code feature can be used to create short codes to dial
the number stored with a specific index value.
Name Enter the text, to be used to identify the number. Names should not begin with numbers.
Number Enter the number to be matched with the above name. The number is processed against
the applicable user and system short codes. Note that if the system has been configured
to use an external dialing prefix, that prefix should be added to directory numbers.

Related links
Configuration Mode Field Descriptions on page 186

Time Profile
Navigation: Time Profile | Time Profile
Additional configuration information
This section provides the Time Profiles field descriptions.
For additional configuration information, see:
• Configuring Time Profiles on page 582
• The button action Time Profile on page 1042
Configuration settings
For a time profile with multiple records, for example a week pattern and some calendar records,
the profile is valid when any entry is valid. For Server Edition, this type of configuration record can
be saved as a template and new records created from a template.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.

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Configuration Mode Field Descriptions

Field Description
Name Range = Up to 15 characters
This name is used to select the time profile from within other tabs.
Manual Override Default = Off.
You can manually override a time profile. The override settings allow you to mix timed and
manual settings. The options are:
• Active Until Next Timed Inactive: Use for time profiles with multiple intervals. Select to
make the current timed interval active until the next inactive interval.
• Inactive Until Next Timed Active: Use for time profiles with multiple intervals. Select to
make the current active timed interval inactive until the next active interval.
• Latch Active: Set the time profile to active. Timed inactive periods are overridden and
remain active. The setting is retained over a reboot.
• Latch Inactive: Set the time profile to inactive. Timed active periods are overridden and
remain active. The setting is retained over a reboot.
Time Entry List
This list shows the current periods during which the time profile is active. Clicking on an existing entry will
display the existing settings and allows them to be edited if required. To remove an entry, selecting it and then
click on Remove or right-click and select Delete.
Recurrence When a new time entry is required, click Add Recurring and then enter the settings for
Pattern (Weekly the entry using the fields displayed. Alternately right-click and select Add Recurring Time
Time Pattern) Entry. This type of entry specific a time period and the days on which it occurs, for
example 9:00 - 12:00, Monday to Friday. A time entry cannot span over two days. For
example you cannot have a time profile starting at 18:00 and ending 8:00. If this time
period is required two Time Entries should be created - one starting at 18:00 and ending
11:59, the other starting at 00:00 and ending 8:00.
• Start Time The time at which the time period starts.
• End Time The time at which the time period ends. Note that the endtime is at the end of
the minute, for example 11:00 is interpreted as 11:00:59, not 11:00:00.
• Days of Week The days of the week to which the time period applies.
Recurrence When a new calendar date entry is required, click Add Date and then enter the settings
Pattern (Calendar required. Alternately right-click and select Add Calendar Time Entry. Calendar records
Date) can be set for up to the end of the next calendar year.
• Start Time The time at which the time period starts.
• End Time The time at which the time period ends.
• Year Select either the current year or the next calendar year.
• Date To select or de-select a particular day, double-click on the date. Selected days are
shown with a dark gray background. Click and drag the cursor to select or de-select a
range of days.

Related links
Configuration Mode Field Descriptions on page 186

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 Firewall Profile

Firewall Profile
The system can act as a firewall, allowing only specific types of data traffic to start a session
across the firewall and controlling in which direction such sessions can be started.
The system supports Static NAT address translation by a firewall profiles. If the Firewall Profile
contains any Static NAT records, all packets received by the firewall must match one of those
static NAT records to not be blocked.
If Network Address Translation (NAT) is used with the firewall (which it typically is), then you must
also configure the setting Service | IP | Primary Trans. IP Address if you wish sessions to be
started into your site (typically for SMTP) from the Internet.
On Server Edition Linux systems, to ensure that the firewall starts after a reboot, you must enable
the Activate setting in the Web Control menus. See Using the Server Edition Web Control Menus.
System firewall profiles can be applied in the following areas of operation.
System:
A firewall profile can be selected to be applied to traffic between LAN1 and LAN2.
User:
Users can be used as the destination of incoming RAS calls. For those users a firewall profile can
be selected on the user's Dial In tab.
Service:
Services are used as the destination for IP routes connection to off-switch data services such as
the internet. A Firewall Profile can be selected for use with a service.
Related links
Configuration Mode Field Descriptions on page 186
Firewall | Standard on page 507
Firewall | Custom on page 509
Static NAT on page 511

Firewall | Standard
Navigation: Firewall Profile | Standard
Additional configuration information
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Configuration settings
By default, any protocol not listed in the standard firewall list is dropped unless a custom firewall
entry is configured for that protocol.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Name Range = Up to 15 characters. Enter the name to identify this profile.
Protocol Control For each of the listed protocols, the options Drop, In (Incoming traffic can start a session),
Out (Outgoing traffic can start a session) and Both Directions can be selected. Once a
session is started, return traffic for that session is also able to cross the firewall.
Protocol Default Description
TELNET Out Remote terminal log in.
FTP Out File Transfer Protocol.
SMTP Out Simple Mail Transfer
Protocol.
TIME Out Time update protocol.
DNS Out Domain Name System.
GOPHER Drop Internet menu system.
FINGER Drop Remote user information
protocol.
RSVP Drop Resource Reservation
Protocol.
HTTP/S Bothway Hypertext Transfer Protocol.
POP3 Out Post Office Protocol.
NNTP Out Network News Transfer
Protocol.
SNMP Drop Simple Network
Management Protocol.
IRC Out Internet Relay Chat.
PPTP Drop Point to Point Tunneling
Protocol.
IGMP Drop Internet Group Membership
Protocol.
Service Control For each of the listed services, the options Drop, In, Out and Both Directions can be
selected. Once a session is started, return traffic for that session is also able to cross the
firewall.
Protocol Default Description
SSI In System Status Application
access.
SEC Drop TCP security settings
access.
CFG Drop TCP configuration settings
access.
TSPI In TSPI service access.
WS Drop IP Office web management
services.

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 Firewall Profile

Related links
Firewall Profile on page 507

Firewall | Custom
Navigation: Firewall Profile | Custom
The tab lists custom firewall settings added to the firewall profile. The Add, Edit and Remove
controls can be used to amend the settings in the list.
Example Custom Firewall Records
Dropping NetBIOS searches on an ISPs DNS:
We suggest that the following filter is always added to the firewall facing the Internet to avoid
costly but otherwise typically pointless requests from Windows machines making DNS searches
on the DNS server at your ISP.
Direction: Drop
IP Protocol: 6 (TCP)
Match Offset: 20
Match Length: 4
Match Data: 00890035
Match Mask: FFFFFFFF
Browsing Non-Standard Port Numbers:
The radio button for HTTP permits ports 80 and 443 through the firewall. Some hosts use non-
standard ports for HTTP traffic, for example 8080, 8000, 8001, 8002, etc. You can add individual
filters for these ports as you find them.
You wish to access a web page but you cannot because it uses TCP port 8000 instead of the
more usual port 80, use the entry below.
Direction: Out
IP Protocol: 6 (TCP)
Match Offset: 22
Match Length: 2
Match Data: 1F40
Match Mask: FFFF
A more general additional entry given below allows all TCP ports out.
Direction: Out
IP Protocol: 6 (TCP)
Match Offset: 0
Match Length: 0

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Configuration Mode Field Descriptions

Match Data: 00000000000000000000000000000000

Match Mask: 00000000000000000000000000000000
Routing All Internet Traffic through a WinProxy:
If you wish to put WinProxy in front of all Internet traffic via the Control Unit. The following firewall
allows only the WinProxy server to contact the Internet : -
1. Create a new Firewall profile and select Drop for all protocols
2. Under Custom create a new Firewall Entry
3. In Notes enter the name of the server allowed. Then use the default settings except in
Local IP Address enter the IP address of the WinProxy Server, in Local IP Mask enter
255.255.255.255 and in Direction select Both Directions.
Stopping PINGs:
You wish to stop pings - this is ICMP Filtering. Using the data below can create a firewall filter that
performs the following; Trap Pings; Trap Ping Replies; Trap Both.
Trap Pings: Protocol = 1, offset = 20, data = 08, mask = FF
Trap Ping Replies: Protocol = 1, offset = 20, data = 00, mask = FF
Trap Both: Protocol = 1, offset = 20, data = 00, mask = F7, Traps Both.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Notes For information only. Enter text to remind you of the purpose of the custom firewall record.
Remote IP The IP address of the system at the far end of the link. Blank allows all IP addresses.
Address
Remote IP Mask The mask to use when checking the Remote IP Address. When left blank no mask is set,
equivalent to 255.255.255.255 - allow all.
Local IP Address The address of devices local to this network (pre-translated). Blank allows all IP addresses.
Local IP Mask The mask to use when checking the Local IP Address. When left blank no mask is set,
equivalent to 255.255.255.255 - allow all.
IP Protocol The value entered here corresponds to the IP Protocol which is to be processed by this
Firewall profile: 1 for ICMP, 6 for TCP, 17 for UDP or 47 for GRE. This information can be
obtained from the "pcol" parameter in a Monitor trace.
Match Offset The offset into the packet (0 = first byte of IP packet) where checking commences for either
a specific port number, a range of port numbers, or data.
Match Length The number of bytes to check in the packet, from the Match Offset point, that are checked
against the Match Data and Match Mask settings.
Match Data The values the data must equal once masked with the Match Mask. This information can
be obtained from "TCP Dst" parameter in a Monitor trace (the firewall uses hex so a port
number of 80 is 50 in hex)
Match Mask This is the byte pattern, which is logically ANDed with the data in the packet from the offset
point. The result of this process is then compared against the contents of the "Match Data"
field.
Table continues…

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 IP Route

Field Description
Direction The direction that data may take if matching this filter.
Drop All matching traffic is dropped.
In Incoming traffic can start a session.
Out Outgoing traffic can start a session.
Both Directions Both incoming and outgoing traffic can start sessions.

Related links
Firewall Profile on page 507

Static NAT
Navigation: Firewall Profile | Static NAT
The Static NAT table allows the firewall to perform address translation between selected internal
and external IP addresses. Up to 64 internal and external IP address pairs can be added to the
Static NAT section of a Firewall Profile.
This feature is intended for incoming maintenance access using applications such as PC-
Anywhere, Manager and the Voicemail Pro Client. The address translation is used for destinations
such a Voicemail Pro server or the system's own LAN1 address.
• If there are any records in the Static NAT settings of a Firewall Profile, each packet
attempting to pass through the firewall must match one of the static NAT pairs or else the
packet will be dropped.
• The destination address of incoming packets is checked for a matching External IP
Address. If a match is found, the target destination address is changed to the corresponding
Internal IP Address.
• The source address of outgoing packets is checked for a matching Internal IP Address. If a
match is found, the source address is changed to the corresponding External IP Address.
• Even when a static NAT address match occurs, the other settings on the Firewall Profile
Standard and Custom tabs are still applied and may block the packet.
Related links
Firewall Profile on page 507

IP Route
Additional configuration information
This section provides the IP Route field descriptions. For additional configuration information, see
Configuring IP Routes on page 642.

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Configuration Mode Field Descriptions

Related links
Configuration Mode Field Descriptions on page 186
IP Route | IP Route on page 512
RIP Dynamic Routing on page 513

IP Route | IP Route
Navigation: IP Route | IP Route
Additional configuration information
For additional configuration information, see Configuring IP Routes on page 642.

Configuration settings
These settings are used to setup static IP routes from the system. These are in addition to RIP if
RIP is enabled on LAN1 and or LAN2. Up to 100 routes are supported.
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
Warning:
The process of 'on-boarding' (refer to the IP Office SSL VPN Solutions Guide) may
automatically add a static route to an SSL VPN service in the system configuration when the
on-boarding file is uploaded to the system. Care should be taken not to delete or amend such
a route except when advised to by Avaya.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
IP Address The IP address to match for ongoing routing. Any packets meeting the IP Address and IP
Mask settings are routed to the entry configured in the Destination field. When left blank
then an IP Address of 255.255.255.255 (all) is used.
IP Mask The subnet mask used to mask the IP Address for ongoing route matching. If blank, the
mask used is 255.255.255.255 (all).
A 0.0.0.0 entry in the IP Address and IP Mask fields routes all packets for which there is
no other specific IP Route available. The Default Route option with Services can be used
to do this if a blank IP route is not added.
Gateway IP Default = Blank The address of the gateway where packets for the above address are to
Address be sent. If this field is set to 0.0.0.0 or is left blank then all packets are just sent down to
the Destination specified, not to a specific IP Address. This is normally only used to
forward packets to another Router on the local LAN.
Destination Allows selection of LAN1, LAN2 and any configured Service, Logical LAN or Tunnel
(L2TP only).
Metric: Default = 0
The number of "hops" this route counts as.
Table continues…

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 IP Route

Field Description
Proxy ARP Default = Off
This allows the system to respond on behalf of this IP address when receiving an ARP
request.

Related links
IP Route on page 511

RIP Dynamic Routing

Navigation: IP Route | RIP Dynamic Routing
Routing Information Protocol (RIP) is a protocol which allows routers within a network to exchange
routes of which they are aware approximately every 30 seconds. Through this process, each
router adds devices and routes in the network to its routing table.
Each router to router link is called a 'hop' and routes of up to 15 hops are created in the routing
tables. When more than one route to a destination exists, the route with the lowest metric (number
of hops) is added to the routing table.
When an existing route becomes unavailable, after 5 minutes it is marked as requiring 'infinite' (16
hops). It is then advertised as such to other routers for the next few updates before being removed
from the routing table. The system also uses 'split horizon' and 'poison reverse'.
RIP is a simple method for automatic route sharing and updating within small homogeneous
networks. It allows alternate routes to be advertised when an existing route fails. Within a large
network the exchange of routing information every 30 seconds can create excessive traffic. In
addition the routing table held by each system is limited to 100 routes (including static and internal
routes).
It can be enabled on LAN1, LAN2 and individual services. The normal default is for RIP to be
disabled.
• Listen Only (Passive): The system listens to RIP1 and RIP2 messages and uses these to
update its routing table. However the system does not respond.
• RIP1: The system listens to RIP1 and RIP2 messages. It advertises its own routes in a RIP1
sub-network broadcast.
• RIP2 Broadcast (RIP1 Compatibility): The system listens to RIP1 and RIP2 messages. It
advertises its own routes in a RIP2 sub-network broadcast. This method is compatible with
RIP1 routers.
• RIP2 Multicast: The system listens to RIP1 and RIP2 messages. It advertises its own routes
to the RIP2 multicast address (249.0.0.0). This method is not compatible with RIP1 routers.
Broadcast and multicast routes (those with addresses such as 255.255.255.255 and 224.0.0.0)
are not included in RIP broadcasts. Static routes (those in the IP Route table) take precedence
over a RIP route when the two routes have the same metric.
Related links
IP Route on page 511

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Configuration Mode Field Descriptions

Account Code
Additional configuration information
This section provides the Account Code field descriptions. For additional configuration
information, see Configuring Account Codes on page 672.
Account codes are commonly used to control cost allocation and out-going call restriction. The
account code used on a call is included in the call information output by the system's call log.
Incoming calls can also trigger account codes automatically by matching the Caller ID stored with
the account code.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Once a call has been completed using an account code, the account code information is removed
from the user's call information. This means that re-dial functions will not re-enter the account
code. The maximum recommended number of accounts codes is 1000.
Related links
Configuration Mode Field Descriptions on page 186

Account Code
Navigation: Account Code | Account Code
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Descriptions
Account Code Enter the account code required. It can also include wildcards; ? matches a single digit
and * matches any digits.
Caller ID A caller ID can be entered and used to automatically assign an account code to calls
made to or received from caller ID.

Voice Recording
Navigation: Account Code | Voice Recording
These settings are used to activate the automatic recording of external calls when the account
code is entered at the start of the call.
Call recording requires Voicemail Pro to be installed and running. Call recording also requires
available conference resources similar to a 3-way conference.

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 Account Code

Note the following:

• Calls to and from IP devices, including those using Direct media, can be recorded.
• Calls parked or held pause recording until the unparked or taken off hold (does not apply to
SIP terminals).
• Recording is stopped if:
- User recording stops if the call is transferred to another user.
- User account code recording stops if the call is transferred to another user.
- Hunt group recording stops if the call is transferred to another user who is not a member of
the hunt group.
- Incoming call route recording continues for the duration of the call on the system.
The destination mailbox for the recording can be specified.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Record Outbound Default = None
Select whether automatic recording of outgoing calls is enabled. The Auto Record Calls
option sets whether just external calls or external and internal calls are included. The
options are:
• None: Do not automatically record calls.
• On: Record the call if possible. If not possible to record, allow the call to continue.
• Mandatory: Record the call if possible. If not possible to record, block the call and return
busy tone.
• Percentages of calls: Record a selected percentages of the calls.
Record Time Default = <None> (Any time)
Profile
Used to select a time profile during which automatic call recording of outgoing calls is
applied. If no profile is selected, automatic recording of outgoing calls is active at all times.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Recording (Auto) Default = Mailbox
Sets the destination for automatically triggered recordings. The options are:
• Mailbox This option sets the destination for the recording to be a selected user or hunt
group mailbox. The adjacent drop down list is used to select the mailbox.
• Voice Recording Library: This options set the destination for the recording to be a VRL
folder on the voicemail server. The ContactStore application polls that folder and collects
waiting recordings which it then places in its own archive. Recording is still done by the
Voicemail Pro.
• Voice Recording Library Authenticated: This option is similar to Voice Recording
Library above but instructs the voicemail server to create an authenticated recording. If
the file contents are changed, the file is invalidated though it can still be played. This
option is not currently supported with Linux based systems.

License
Additional configuration information
This section provides the Licenses field descriptions.
For additional configuration information on licensing, see the following.
• Applying Licenses on page 552.
• Converting from Nodal Licensing to Centralized Licensing on page 564
• Migrating ADI Licenses to PLDS on page 565
• “Licenses” in Avaya IP Office™ Platform Solution Description.
Related links
Configuration Mode Field Descriptions on page 186
License on page 516
Remote Server on page 518

License
Navigation: License | License
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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 License

Name Description
License Mode Identifies the status of the system licenses. The two license configuration types are nodal
and WebLM. Nodal licenses are licenses that are present on the system. WebLM licenses
means licenses obtained from the WebLM server.

The possible states are:

• Normal Mode Normal nodal licensing mode. In this mode, WebLM is not configured and
only nodal licensing is allowed.
• Server Error This mode occurs when transitioning to WebLM licensing. WebLM has
been configured but the server is not available.
• Configuration Error This mode occurs when transitioning to WebLM licensing. WebLM
has been configured and the server is available, but there are not enough licenses
available to license all of the configured features. Only nodal licenses are valid on
Standard mode IP500 V2 systems.
• WebLM Normal Mode The system is fully licensed. WebLM has been configured and
there are enough licenses available to license all of the configured features.
• WebLM Error Mode Action is required to correct the License Mode. Refer to the License
Status column and the Error List section at the bottom of the screen to determine why the
system is in License Error Mode. A 30-day grace period provides access to the
capacities and features of the installed license when the system is in License Error Mode.
• WebLM Restricted Mode When the system is in License Error Mode, if the problem is
not resolved with the 30-day grace period, the system will enter License Restricted Mode.
When in this mode, configuration changes are blocked, except for fixing the licensing
errors. If a feature license cannot be acquired from the WebLM server, the feature will not
function.
Type Mode WebLM Configured Virtual License and
Grace Period (30
days)
Nodal Normal No No
WebLM Server Error Yes No
WebLM Configuration Error Yes No
WebLM Normal Yes No
WebLM Error Yes Yes
WebLM Restricted Yes No
Licensed Version Indicates the software version the system is currently licensed for.
PLDS Host ID The ID used when generating PLDS nodal license files.
Not used with WebLM licensing. WebLM licensing uses the host ID of the WebLM server.
PLDS File Status If a PLDS nodal license file is loaded, this field indicates if the file is valid or not.
Select Licensing Indicates that the system has a valid Select license.
Feature Identifies the licenses installed on the system.
Table continues…

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Configuration Mode Field Descriptions

Name Description
Key This is the license key string supplied. It is a unique value based on the feature being
licensed and the either the system's Dongle Serial Number or System Identification
depending on the type of system.
Not applicable when using PLDS or WebLM licensing. This field is not displayed if there are
no ADI licenses.
Instance For information only. Some licenses enable a number of port, channels or users. When that
is the case, the number of such is indicated here. Multiple licenses for the same feature are
usually cumulative.
Status For information only. This field indicates the current validation status of the license key.
• Unknown This status is shown for licenses that have just been added to the
configuration shown in Manager. Once the configuration has been sent back to the
system and then reloaded, the status will change to one of those below.
• Valid: The license is valid.
• Invalid: The license was not recognized. It did not match the PLDS host ID.
• Dormant: The license is valid but is conditional on some other pre-requisite licenses.
• Obsolete: The license is valid but is one no longer used by the level of software running
on the system.
Expiry Date For information only. Trial licenses can be set to expire within a set period from their issue.
The expiry date is shown here.
Source The source of the license file. The options are:
• ADI Nodal: ADI licenses added locally to the system. This may appear on upgraded
systems.
• PLDS Nodal: PLDS licenses added locally to the system.
• WebLM: Licenses obtained from the WebLM server.
• Virtual: Licenses created by the system. This may appear on upgraded systems.
• Virtual Grace: Licenses created by the system while in WebLM error mode.

Additional Configuration Information

Click PLDS License > Send To IP Office > OK to open the Select PLDS License File dialog
from where you can upload a PLDS license to IP Office. You can browse to a location on your
system and select a file to upload.
Select an existing license and click PLDS License > Delete From IP Office > OK to delete the
selected license.
Related links
License on page 516

Remote Server
Navigation: License | Remote Server

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 License

This tab is used for:

• IP500 V2 systems in a Enterprise Branch deployments which are using WebLM licensing
• Server Edition systems to specify which method of centralized licensing is used.
The Reserved Licenses setting is mergeable. The remaining settings are not mergeable.
Changes to these settings requires a reboot of the system.
The following field two fields control which source the system uses for its licenses. The field shown
depends on the type of system:
Field Description
Licence Source Default = WebLM.
This field is available on Server Edition systems. All systems in the network must use the
same source for licensing. The options are:
• WebLM: Licenses are obtained from the WebLM service. The PLDS license file is
uploaded to the WebLM service. All servers in the network make license reservation
requests to the WebLM service. On Server Edition systems, a Deploy button appears
when you select WebLM as License Source. Click the Deploy button to browse and
select a license file to deploy.
• Local / Primary Server: The PLDS license file is uploaded to the IP Office service, not
WebLM. Depending on the particular license, some are obtained by reservation
requests to the primary server, others are obtained from the server’s own license file.
Enable Remote Default = Off.
Server
This field is available on non-Server Edition IP500 V2 systems. The options are:
• If disabled, the system is licensed locally by uploading a license file to the system.
• If enabled, the system uses licenses requested from a remote WebLM server. This
option is only supported for systems in a branch enterprise supported via Avaya System
Manager.

The additional fields displayed depend on the license source selection above:
Local/Primary Server Licensed Server Settings
Field Description
License Server IP Default = 127.0.0.1 on Primary. On Secondary and expansion systems, the default is the
Address Primary IP address.
This field is available when the Licence Source is set to Local Primary Server. This
field contains the IP address of the Server Edition Primary server.

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Configuration Mode Field Descriptions

WebLM Licensed Primary Server Settings

Field Description
Domain Name Default = Blank for IP500 V2 systems and for Server Edition Primary hosted
(URL) deployments. For Server Edition, the IP address of the Primary Server.
• For Enterprise Branch deployments, the domain name or IP address of the WebLM
server or the domain name of System Manager if the system is under System Manager
control.
• For Server Edition deployments, the domain name or IP address of the Primary Server.
• For Server Edition hosted deployments, the domain name of the WebLM server.

The format can be the FQDN or the IP address prefixed with https://.
Path Default = WebLM/LicenseServer.
The path on the web server of the WebLM resource.
Port Number Default = 52233.
The port number of the WebLM server.
WebLM Client ID An ID based on MAC address of the system. This is a read only field used by the WebLM
server to identify the system.
WebLM Node ID An ID based on MAC address and hostname of the system. This is a read only field used
by the WebLM server to identify the system.

WebLM Licensed Server (non-Primary) Settings

Field Description
Enable proxy via Default = On.
Primary IP Office
Available on Server Edition Secondary and Expansion systems.
line
• Enables retrieval of licenses from the WebLM server through the IP Office Line
connection to the Server Edition Primary server.
• If the check box is cleared, the WebLM request is done directly to the WebLM server.
Note that this field is not available if the node is not configured as a WebSocket client to
the Server Edition Primary server.
Primary IP Default = The IP address of the Server Edition Primary server.
Address
Available on Server Edition Secondary and Expansion systems when Enable proxy via
Primary IP Office line is enabled
WebLM Client ID An ID based on MAC address of the system. This is a read only field used by the WebLM
server to identify the system.
WebLM Node ID An ID based on MAC address and hostname of the system. This is a read only field used
by the WebLM server to identify the system.

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 License

Reserved Licenses
These fields are used to reserve licenses from the license server, WebLM or, if using nodal
licensing, the Primary server. There are two types of reservation field; manual and automatic.
• Manual fields can be used to set the number of licenses that the server should request from
those available on the primary/WebLM server.
• Automatic fields are set to match other aspects of the server configuration, for example the
number of configured power users. Note that these values may not change until after the
configuration is saved and then reloaded.

WebLM Reserved Primary Server Secondary Expansion Expansion

Licenses — Manual Server (Linux) (IP500 V2)
SIP Trunk Sessions Yes Yes Yes Yes
SM Trunk Sessions Yes Yes Yes Yes
Voicemail Pro Ports Yes Yes - -
VMPro Recordings Yes Yes - -
Administrators
VMPro TTS Professional Yes Yes - -
Wave Users - - - Yes
CTI Link Pro Yes Yes Yes Yes
UMS Web Services Yes Yes Yes Yes
MAC Softphones Yes Yes Yes Yes
Avaya Contact Center Yes Yes - -
Select
Third Party Recorder Yes Yes - -
VM Media Manager Yes Yes Yes -
Customer Service Yes Yes Yes Yes
Supervisor
Customer Service Agent Yes Yes Yes Yes

Nodal Reserved Primary Server Secondary Expansion Expansion

Licenses — Manual Server (Linux) (IP500 V2)
SIP Trunk Sessions Yes Yes Yes Yes

WebLM/Nodal Primary Server Secondary Expansion Expansion

Reserved Licenses — Server (Linux) (IP500 V2)
Automatic
Server Edition Yes Yes Yes Yes
Avaya IP Endpoints Yes Yes Yes Yes
3rd Party IP Endpoints Yes Yes Yes Yes
Receptionist Yes Yes Yes Yes
Table continues…

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Configuration Mode Field Descriptions

Office Worker Yes Yes Yes Yes

Power User Yes Yes Yes Yes
Avaya Softphone Yes Yes Yes Yes
Web Collaboration Yes Yes Yes Yes
Universal PRI Additional - - - Yes
Channels
IPSec Tunneling - - - Yes

Related links
License on page 516

Tunnel

Tunneling allows additional security to be applied to IP data traffic. This is useful when sites
across an unsecure network such as the public internet. The system supports two methods of
tunneling, L2TP and IPSec. Once a tunnel is created, it can be used as the destination for
selected IP traffic in the IP Route table.
The use of tunnels is not supported by Linux based systems. On other systems, two types of
tunneling are supported.
L2TP:
Layer 2 Tunneling Protocol PPP (Point to Point Protocol) authentication normally takes place
between directly connected routing devices. For example when connecting to the internet,
authentication is between the customer router and the internet service provider's equipment. L2TP
allows additional authentication to be performed between the routers at each end of the
connection regardless of any intermediate network routers. The use of L2TP does not require a
license.
IPSec:
IPSec allows data between two locations to be secured using various methods of sender
authentication and or data encryption. The use of IPSec requires entry of an IPSec Tunneling
license into the system at each end.
Related links
Configuration Mode Field Descriptions on page 186
L2TP Tunnel on page 523
IP Security Tunnel on page 525

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 Tunnel

L2TP Tunnel
Related links
Tunnel on page 522
L2PT Tunnel on page 523
L2TP on page 524
L2TP PPP on page 524

L2PT Tunnel
Navigation: Tunnel | Tunnel (L2TP)
Additional configuration information
This type of configuration record can be saved as a template and new records created from a
template. See Working with Templates on page 585.
Configuration settings
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Name Default = Blank.
A unique name for the tunnel. Once the tunnel is created, the name can be selected as a
destination in the IP Route table.
Local Configuration
The account name and password is used to set the PPP authentication parameters.
Local Account The local user name used in outgoing authentication.
Name
Local Account The local user password. Used during authentication.
Password/
Confirm
Password
Local IP Address The source IP address to use when originating an L2TP tunnel. By default (un-configured),
the system uses the IP address of the interface on which the tunnel is to be established as
the source address of tunnel.

Remote Configuration
The account name and password is used to set the PPP authentication parameters.
Remote Account The remote user name that is expected for the authentication of the peer.
Name
Remote Account The password for the remote user. Used during authentication.
Password/
Confirm
Password
Table continues…

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Configuration Mode Field Descriptions

Field Description
Remote IP The IP address of the remote L2TP peer or the local VPN line IP address or the WAN IP
Address address.
Minimum Call Default = 60 minutes. Range = 1 to 999.
Time (Mins)
The minimum time that the tunnel will remain active.
Forward Default = On
Multicast
Allow the tunnel to carry multicast messages when enabled.
Messages
Encrypted Default = Off
Password
When enabled, the CHAP protocol is used to authenticate the incoming peer.

Related links
L2TP Tunnel on page 523

L2TP
Navigation: Tunnel | L2TP
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Shared Secret/ User setting used for authentication. Must be matched at both ends of the tunnel. This
Confirm password is separate from the PPP authentication parameters defined on the L2TP|Tunnel
Password tab.
Total Control Default = 0. Range = 0 to 65535.
Retransmission
Time delay before retransmission.
Interval
Receive Window Default = 4. Range = 0 to 65535.
Size
The number of unacknowledged packets allowed.
Sequence Default = On
numbers on Data
When on, adds sequence numbers to L2TP packets.
Channel
Add checksum Default = On.
on UDP packets
When on, uses checksums to verify L2TP packets.
Use Hiding Default = Off
When on, encrypts the tunnel's control channel.

Related links
L2TP Tunnel on page 523

L2TP PPP
Navigation: Tunnel | PPP (L2TP)
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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 Tunnel

Field Description
CHAP Challenge Default = 0 (Disabled). Range = 0 to 99999 seconds.
Interval (secs)
Sets the period between CHAP challenges. Blank or 0 disables repeated challenges.
Some software (such as Windows 95 DUN) does not support repeated challenges.
Header Default = None
Compression
Select header compression. Options are: IPHC and/or VJ.
PPP Compression Default = MPPC
Mode
Select the compression mode for the tunnel connection. Options are: Disable, StacLZS or
MPPC.
Multilink/QoS Default = Off
Enable the use of Multilink protocol (MPPC) on the link.
Incoming traffic Default = On
does not keep link
When enabled, the link is not kept up when the only traffic is incoming traffic.
up
LCP Echo Default = 6. Range = 0 to 99999 milliseconds.
Timeout (msecs)
When a PPP link is established, it is normal for each end to send echo packets to verify
that the link is still connected. This field defines the time between LCP echo packets. Four
missed responses in a row will cause the link to terminate.

Related links
L2TP Tunnel on page 523

IP Security Tunnel
Related links
Tunnel on page 522
IPSec Main on page 525
Tunnel | IKE Policies (IPSec) on page 526
IPSec Policies on page 527

IPSec Main
Navigation: Tunnel | Main (IPSec)
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
These settings are not mergeable. Changes to these settings will require a reboot of the system.

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Configuration Mode Field Descriptions

Field Descripition
Name Default = Blank.
A unique name for the tunnel. Once the tunnel is created, the name can be selected as a
destination for traffic in the IP Route table.
Local Configuration
The IP Address and IP Mask are used in conjunction with each other to configure and set the conditions for this
Security Association (SA) with regard to inbound and outbound IP packets.
IP Address The IP address or sub-net for the start of the tunnel.
IP Mask The IP mask for the above address.
Tunnel Endpoint The local IP address to be used to establish the SA to the remote peer. If left un-
IP Address configured, the system will use the IP address of the local interface on which the tunnel is
to be configured.

Remote Configuration
The IP Address and IP Mask are used in conjunction with each other to configure and set the conditions for this
Security Association (SA) with regard to inbound and outbound IP packets.
IP Address The IP address or sub-net for the end of the tunnel.
IP Mask The IP mask for the above address.
Tunnel Endpoint The IP address of the peer to which a SA must be established before the specified local
IP Address and remote addresses can be forwarded.

Related links
IP Security Tunnel on page 525

Tunnel | IKE Policies (IPSec)

Navigation: Tunnel | IKE Policies (IPSec)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Descripition
Shared Secret/ The password used for authentication. This must be matched at both ends of the tunnel.
Confirm
Password
Exchange Type Default = ID Prot
Aggressive provides faster security setup but does not hide the ID's of the communicating
devices. ID Prot is slower but hides the ID's of the communicating devices.
Encryption Default = DES CBC
Select the encryption method used by the tunnel. The options are:
• DES CBC
• 3DES
• Any
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 Tunnel

Field Descripition
Authentication Default = MD5
The method of password authentication. Options are:
• MD5
• SHA
• Any
DH Group Default = Group 1
Life Type Default = KBytes
Sets whether Life (below) is measured in seconds or kilobytes.
Life Range = 0 to 99999999.
Determines the period of time or the number of bytes after which the SA key is refreshed
or re-calculated.

Related links
IP Security Tunnel on page 525

IPSec Policies
Navigation: Tunnel | IKE Policies (IPSec)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field Description
Protocol Default = ESP
The options are:
• ESP (Encapsulated Security Payload)
• AH (Authentication Header, no encryption)
Encryption Default = DES
Select the encryption method used by the tunnel. The options are:
• DES CBC
• 3DES
• Any
Authentication Default = HMAC MD5
The method of password authentication. Options are:
• HMAC MD5
• HMAC SHA
• Any
Table continues…

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Configuration Mode Field Descriptions

Field Description
Life Type Default = KBytes
Sets whether Life (below) is measured in seconds or kilobytes.
Life Determines the period of time or the number of bytes after which the SA key is refreshed
or re-calculated.

Related links
IP Security Tunnel on page 525

Auto Attendant
These settings are used for embedded voicemail provided by the IP Office control unit. This is
setup by adding an Avaya Embedded Voicemail memory card to the control unit and then
selecting Embedded Voicemail as the Voicemail Type.
This tab and its settings are hidden unless the system has been configured to use Embedded
Voicemail on the System | Voicemail tab.
For full details on configuration and operation of Embedded Voicemail auto-attendants refer to the
IP Office Embedded Voicemail Installation Manual.
Up to 40 auto-attendant services can be configured.
Embedded voicemail services include auto-attendant, callers accessing mailboxes to leave or
collect messages and announcements to callers waiting to be answered.
The IP500 V2 supports 2 simultaneous Embedded Voicemail calls by default but can be licensed
for up to 6. The licensed limit applies to total number of callers leaving messages, collecting
messages and or using an auto attendant.
In addition to basic mailbox functionality, Embedded Voicemail can also provide auto-attendant
operation. Each auto attendant can use existing time profiles to select the greeting given to callers
and then provide follow on actions relating to the key presses 0 to 9, * and #.
Time Profiles:
Each auto attendant can use up to three existing time profiles, on each for Morning, Afternoon and
Evening. These are used to decide which greeting is played to callers. They do not change the
actions selectable by callers within the auto attendant. If the time profiles overlap or create gaps,
then the order of precedence used is morning, afternoon, evening.
Greetings:
Four different greetings are used for each auto attendant. One for each time profile period. This is
then always followed by the greeting for the auto-attendant actions. By default a number of system
short codes are automatically created to allow the recording of these greetings from a system
extension. See below.
Actions:

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 Auto Attendant

Separate actions can be defined for the DTMF keys 0 to 9, * and #. Actions include transfer to a
specified destination, transfer to another auto-attendant transfer to a user extension specified by
the caller (dial by number) and replaying the greetings.
• The Fax action can be used to reroute fax calls when fax tone is detected by the auto-
attendant.
• The Dial by Name action can be used to let callers specify the transfer destination.
Short Codes:
Adding an auto attendant automatically adds a number of system short codes. These use the
Auto Attendant short code feature. These short codes are used to provide dialing access to
record the auto attendant greetings.
Four system short codes (*81XX, *82XX, *83XX and *84XX) are automatically added for use with
all auto attendants, for the morning, afternoon, evening and menu options greetings respectively.
These use a telephone number of the form "AA:" N" . Y " where N is the replaced with the auto
attendant number dialed and Y is 1, 2, 3 or 4 for the morning, afternoon, evening or menu option
greeting.
• An additional short code of the form (for example) *80XX/Auto Attendant/"AA:"N can be
added manual if internal dialed access to auto attendants is required.
• To add a short code to access a specific auto attendant, the name method should be used.
• For IP Office deployed in a Enterprise Branch environment, the short codes *800XX,
*801XX…*809XX, *850XX, and *851XX are automatically created for recording a Page
prompt.
Routing Calls to the Auto Attendant:
The telephone number format AA:Name can be used to route callers to an auto attendant. It can
be used in the destination field of incoming call routes and telephone number field of short codes
set to the Auto Attend feature.
Related links
Configuration Mode Field Descriptions on page 186
Auto Attendant on page 529
Actions on page 531

Auto Attendant
Navigation: Auto Attendant | Auto Attendant
These settings are used to define the name of the auto attendant service and the time profiles that
should control which auto attendant greetings are played.
These settings are mergeable. Changes to these settings do not require a reboot of the system.

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Configuration Mode Field Descriptions

Field Description
Name Range = Up to 12 characters
This field sets the name for the auto-attendant service. External calls can be routed to the
auto attendant by entering AA:Name in the destination field of an Incoming Call Route.
Maximum Default = 8 seconds; Range = 1 to 20 seconds.
Inactivity
This field sets how long after playing the prompts the Auto Attendant should wait for a
valid key press. If exceeded, the caller is either transferred to the Fallback Extension set
within the Incoming Call Route used for their call or else the caller is disconnected.
Enable Local Default = On.
Recording
When off, use of short codes to record auto-attendant prompts is blocked. The short
codes can still be used to playback the greetings.
Direct Dial-By- Default = Off.
Number
This setting affects the operation of any key presses in the auto attendant menu set to use
the Dial By Number action.
If selected, the key press for the action is included in any following digits dialed by the
caller for system extension matching. For example, if 2 is set in the actions to Dial by
Number, a caller can dial 201 for extension 201.
If not selected, the key press for the action is not included in any following digits dialed by
the caller for system extension matching. For example, if 2 is set in the actions to Dial by
Number, a caller must dial 2 and then 201 for extension 201.
Dial by Name Default = First Name/Last Name.
Match Order
Determines the name order used for the Embedded Voicemail Dial by Name function. The
options are:
• First then Last
• Last then First
AA Number This number is assigned by the system and cannot be changed. It is used in conjunction
with short codes to access the auto attendant service or to record auto attendant
greetings.
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 Auto Attendant

Field Description
Morning/ Each auto-attendant can consist of three distinct time periods, defined by associated time
Afternoon/ profiles. A greeting can be recorded for each period. The appropriate greeting is played to
Evening/Menu callers and followed by the Menu Options greeting which should list the available actions.
Options The options are:
• Time Profile The time profile that defines each period of auto-attendant operation.
When there are overlaps or gaps between time profiles, precedence is given in the order
morning, afternoon and then evening.
• Short code These fields indicate the system short codes automatically created to allow
recording of the time profile greetings and the menu options prompt.
• Recording Name: Default = Blank. Range = Up to 31 characters. This field appears
next to the short code used for manually recording auto-attendant prompts. It is only
used is using pre-recorded wav files as greeting rather than manually recording
greetings using the indicated short codes. If used, note that the field is case sensitive
and uses the name embedded within the wav file file header rather than the actual file
name.
This field can be used with all systems supporting Embedded Voicemail. The utility for
converting .wav files to the correct format is provided with Manager and can be
launched via File | Advanced | LVM Greeting Utility. Files then need to be manually
transferred to the Embedded Voicemail memory card. For full details refer to the IP
Office Embedded Voicemail Installation manual.

Related links
Auto Attendant on page 528

Actions
Navigation: Auto Attendant | Actions
This tab defines the actions available to callers dependant on which DTMF key they press. To
change an action, select the appropriate row and click Edit. When the key is configured as
required click OK.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Key The standard telephone dial pad keys, 0 to 9 plus * and #.
The option Fax can be used for a transfer to the required fax destination and will then be
triggered by fax tone detection. If left as Not Defined, fax calls will follow the incoming call
routes fallback settings once the auto-attendant Maximum Inactivity Time set on the Auto
Attendant | Auto Attendant tab is reached.
Action
The following actions can be assigned to each key.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Centrex Transfer Used to transfer the incoming call to an external telephone number defined in the
Transfer Number field.
This option is only supported with Embedded Voicemail.
Dial by Name Callers are asked to dial the name of the user they require and then press #. The recorded
name prompts of matching users are then played back for the caller to make a selection.
The name order used is set by the Dial by Name Match Order setting on the Auto
Attendant tab. Note the name used is the user's Full Name if set, otherwise their User
Name is used. Users without a recorded name prompt or set to Exclude From Directory
are not included. For Embedded Voicemail in IP Office mode, users can record their name
by accessing their mailbox and dialing *05. For Embedded Voicemail in Intuity mode,
users are prompted to record their name when they access their mailbox.
Dial By Number This option allows callers with DTMF phones to dial the extension number of the user they
require. No destination is set for this option. The prompt for using this option should be
included in the auto attendant Menu Options greeting. A uniform length of extension
number is required for all users and hunt group numbers. The operation of this action is
affected by the auto attendant's Direct Dial-by-Number setting.
Normal Transfer Can be used with or without a Destination set. When the Destination is not set, this
action behaves as a Dial By Number action. With the Destination is set, this action waits
for a connection before transferring the call. Callers can hear Music on Hold.
Announcements are not heard.
Not Defined The corresponding key takes no action.
Park & Page The Park & Page feature is supported when the system Voicemail Type is designated as
Embedded Voicemail or Voicemail Pro. Park & Page is also supported on systems
where Modular Messaging over SIP is configured as the central voicemail system and
the local Embedded Voicemail provides auto attendant operation. The Park & Page
feature is an option in user mailboxes where a key is configured with the Park & Page
feature. When an incoming call is answered by the voicemail system and the caller dials
the DTMF digit for which Park & Page is configured, the caller hears the Park & Page
prompt. IP Office parks the call and sends a page to the designated extension or hunt
group. When Park & Page is selected in the Action drop-down box, the following fields
appear:
• Park Slot Prefix – the desired Park Slot prefix number. Maximum is 8 digits. A 0-9 will
be added to this prefix to form a complete Park Slot.
• Retry count – number of page retries; the range is 0 to 5.
• Retry timeout – provided in the format M:SS (minute:seconds). The range can be set in
15-second increments. The minimum setting is 15 seconds and the maximum setting is
5 minutes. The default setting is 15 seconds.
• Page prompt – short code to record the page prompt or upload the recorded prompt.
(Prompt can be uploaded to the SD card in the same way the AA prompts are).
Replay Menu Replay the auto-attendant greetings again.
Greeting
Transfer Transfer the call to the selected destination. This is an unsupervised transfer, if the caller
is not answered they will be handled as per a direct call to that number.
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 Authorization Codes

Field Description
Transfer to This action can be used to transfer calls to another existing auto attendant.
Attendant

Destination Sets the destination for the action.

Destination can be a user, a hunt group or a short code.
If the destination field is left blank, callers can dial the user extension number that they
require. Note however that no prompt is provided for this option so it should be included in
the auto attendant Menu Options greeting.

Related links
Auto Attendant on page 528

Authorization Codes
Navigation: Authorization Codes
Note:
In release 9.1, authorization codes can no longer be associated with User Rights. If an
authorization code was configured in relationship with User Rights in an earlier release
configuration, this authorization code will be lost during upgrade. The administrator must re-
configure the authorization code, after upgrade. The authorization code must be associated
with a user.
Authorization codes are enabled by default.
Each authorization code is associated with a particular user. The user can then dial numbers
which are set to trigger forced authorization code entry. Once a code is entered, the short code
settings of the user with which the code is associated are used to completed the call.
This can be used to allow authorized users to make otherwise restricted calls from any extension
without first having to log in to that extension and then log out after the call. Valid/invalid
authorization code entry can be recorded in the SMDR output.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Authorization Code Range = Up to 12 digits.
The digits used for the authorization code. Each code must be unique. Wildcards are
not usable with authorization codes.
User This field is used to select a user with which the authorization code is associated. The
authorization code can then be used to authorize calls made by that user.

Related links
Configuration Mode Field Descriptions on page 186

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Configuration Mode Field Descriptions

User Rights
Additional configuration information
This section provides the User Rights field descriptions. For additional configuration information,
see Configuring User Rights on page 653.
User Rights act as templates for selected user settings. The settings of a user rights template are
applied to all users associated with that template. The use of a template can also be controlled by
a time profile to set when the template is used for a particular user.
Related links
Configuration Mode Field Descriptions on page 186

User
Navigation: User Rights | User
Used to set and lock various user settings.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Name The name for the user rights . This must be set in order to allow the user rights to be
selected within the User Rights drop down list on the User | User tab of individual users.
Application Default = Off.
Servers Group
Set to On if the IP Office system is deployed in an IP Office Contact Center solution or an
Avaya Contact Center Select solution.
Only one user rights record can be configured to be the Application Servers Group. If it is
set on any one group then the control is disabled on all other groups.
Locale Default = Blank
Sets and locks the language used for voicemail prompts to the user, assuming the
language is available on the voicemail server. On a digital extension it also controls the
display language used for messages from the system to the phone. See Avaya IP Office™
Platform Locale Settings.
Priority Default = 5, Range 1 (Lowest) to 5 (Highest)
Sets and locks the user's priority setting for least cost routing.
Do Not Disturb Default = Off Sets and locks the user's DND status setting.

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 User Rights

Short Codes
Navigation: User Rights | Short Codes
Used to set and lock the user's short code set. The tab operates in the same way as the User |
Short Codes tab. User and User Rights short codes are only applied to numbers dialed by that
user. For example they are not applied to calls forwarded via the user.
Warning:
User dialing of emergency numbers must not be blocked. If short codes are edited, the users
ability to dial emergency numbers must be tested and maintained.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.

Button Programming
Navigation: User Rights | Button Programming
This tab is used to set and lock the user's programmable button set. When locked, the user cannot
use Admin or Admin1 buttons on their phone to override any button set by their user rights.
Buttons not set through the user rights can be set through the user's own settings. When Apply
user rights value is selected, the tab operates in the same manner as the User | Button
Programming tab.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Adding Blank Buttons
There are scenarios where users are able to program their own buttons but you may want to force
certain buttons to be blank. This can be done through the user's associated User Rights as
follows:
1. Assign the action Emulation | Inspect to the button. This action has no specific function.
Enter some spaces as the button label.
2. When pressed by the user, this button will not perform any action. However it cannot be
overridden by the user.

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Configuration Mode Field Descriptions

Telephony
Navigation: User Rights | Telephony
Allows various user telephony settings to be set and locked. These match settings found on the
User | Telephony tab.

Call Settings
Navigation: User Rights | Telephony | Call Settings
Additional configuration information
For additional information on ring tones, see Ring Tones on page 604.
Configuration settings
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
No Answer Time Default = Blank (Use system setting). Range = 6 to 99999 seconds.
Sets how long a call rings the user before following forwarded on no answer if set or going
to voicemail. Leave blank to use the system default setting.
Transfer return Default = Blank (Off), Range 1 to 99999 seconds.
Time (secs)
Sets the delay after which any call transferred by the user, which remains unanswered,
should return to the user if possible.
Wrap up Time Default = 2 seconds, Range 0 to 99999 seconds.
(secs)
Specifies the amount of time after ending one call before another call can ring. You may
wish to increase this in a "call center" environment where users may need time to log call
details before taking the next call. It is recommended that this option is not set to less than
the default of 2 seconds. 0 is used for immediate ringing.
Call waiting on/ Default = Off
Enable call
For users on phones without appearance buttons, if the user is on a call and a second call
waiting
arrives for them, an audio tone can be given in the speech path to indicate a waiting call
(the call waiting tone varies according to locale). The waiting caller hears ringing rather
than receiving busy. There can only be one waiting call, any further calls receive normal
busy treatment. If the call waiting is not answered within the no answer time, it follows
forward on no answer or goes to voicemail as appropriate. User call waiting is not used for
users on phones with multiple call appearance buttons.
Busy on held/ Default = Off
Enable busy on
If on, when the user has a call on hold, new calls receive busy tone (ringing for incoming
Held
analog call) or are diverted to voicemail if enabled, rather than ringing the user. Note this
overrides call waiting when the user has a call on hold. Not supported (should be set to off)
for users with call appearance buttons.

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 User Rights

Supervisor Settings
Navigation: User Rights | Telephony | Supervisor Settings
Additional configuration information
Off-Switch Transfer Restriction
Call Barring
Configuration settings
These settings relate to user features normally only adjusted by the user's supervisor.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Can Intrude Default = Off
Check this option if the User can interrupt other user's calls. This setting and the setting
below are used to control the use of the following short code and button features:
• Call Intrude
• Call Listen
• Call Steal
• Dial Inclusion
Cannot be Default = On
Intruded
If checked, this user's calls cannot be interrupted or acquired. In addition to the features
listed above, this setting also affects whether other users can use their appearance
buttons to bridge into a call to which this user has been the longest present user.
Deny Auto Default = Off.
Intercom Calls
When enabled, any automatic intercom calls to the user's extension are automatically
turned into normal calls.
Force Login Default = Off
If checked, the user must log in using their Login Code to use an extension. For example,
if Force Login is ticked for User A and user B has logged into A's phone, after B logs off A
must log back. If Force Login was not ticked, A would be automatically logged back in.
Force Account Default = Off
Code
If checked, the user must enter a valid account code to make an external call.
Inhibit Off-Switch : Default = Off
Forward/Transfer
When enabled, this setting stops the user from transferring or forwarding calls externally.
Note that all user can be barred from forwarding or transferring calls externally by the
System | Telephony | Telephony | Inhibit Off-Switch Forward/Transfers setting.
Table continues…

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Configuration Mode Field Descriptions

Field Description
Outgoing Call Default = Off
Bar
When set, bars the user from making external calls.
Coverage Group Default = <None>
If a group is selected, the system will not use voicemail to answer the users unanswered
calls. Instead the call will continue ringing until either answered or the caller disconnects.
For external calls, after the users no answer time, the call is also presented to the users
who are members of the selected Coverage Group. For further details refer to Coverage
Groups.
ICR Agent Applicable for Integrated Contact Reporter
Default = Off
Enable to configure user right members as ICR agents. Any user configured to use the
user right becomes an ICR agent.
If enabled, it also activates the After Call Work related fields.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Automatic After Applicable for Integrated Contact Reporter
Call Work
Default = Off
If enabled, all ICR agents of the user right go into After Call Work (ACW) at the end of an
ICR and non-ICR hunt group call to indicate that they are busy with post-call processing
activity. During the ACW state, they are not sent any hunt group calls.
For more information about configuring ACW, see Administering Avaya IP Office™
Platform Integrated Contact Reporter.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
Can Control after Applicable for Integrated Contact Reporter
Call Work
Default = Off
If enabled, the ICR agents in the user right can extend the currently active After Call Work
time indefinitely.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.
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 User Rights

Field Description
After Call Work Applicable for Integrated Contact Reporter
Time
Default = The value in this field is populated from the Default After Call Work Time field
located at System | Contact Center.
The time after a call when an agent is busy and unable to deal with hunt group calls.
Change the value if you want to specify ACW time for this all ICR agents in the user right
to be different from the system default.

Note:
Integrated Contact Reporter is not supported in IP Office Release 11.0.

Multi-line Options
Navigation: User Rights | Telephony | Multi-line Options
Additional configuration information
For additional configuration information, see Appearance Button Operation on page 1058.
Configuration settings
Multi-line options are applied to a user's phone when the user is using an Avaya phones which
supports appearance buttons (call appearance, line appearance, bridged and call coverage).
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Individual Default = 10 seconds, Range 1 to 99999 seconds.
Coverage Time
This function sets how long the phone will ring at your extension before also alerting
(secs)
at any call coverage users. This time setting should not be equal to or greater than the
No Answer Time.

Call Log
Navigation: User Rights | Telephony | Call Log
The system can store a centralized call log for users. Each users' centralized call log can contain
up to 30 call records for user calls. When this limit is reached, each new call records replaces the
oldest previous record.
On Avaya phones with a fixed Call Log or History button (1400, 1600, 9500 and 9600 Series),
that button can be used to display the user's centralized call log. The centralized call log is also
used for M-Series and T-Series phone. The user can use the call log to make calls or to store as a
personal speed dial. They can also edit the call log to remove records. The same call log is also
used if the user logs into one-X Portal for IP Office.
The centralized call log moves with the user if they log on and off from different phones. This
includes if they hot desk within a network.

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Configuration Mode Field Descriptions

These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Centralized Call Default = System Default (On)
Log
This setting allows the use of centralized call logging to be enabled or disabled on a per
user basis. The default is to match the system setting System | Telephony | Call Log |
Default Centralized Call Log On.
The other options are On or Off for the individual user. If off is selected, the call log shown
on the users phone is the local call log stored by the phone.
Delete records Default = 00:00 (Never).
after
(hours:minutes) If a time period is set, records in the user's call log are automatically deleted after this
period.
Groups Default = System Default (On).
This section contains a list of hunt groups on the system. If the system setting System |
Telephony | Call Log | Log Missed Huntgroup Calls has been enabled, then missed
calls for those groups selected are shown as part of the users call log. The missed calls
are any missed calls for the hunt group, not just group calls presented to the user and not
answered by them.

User Rights Membership

Navigation: User Rights | User Rights Membership
The tabs display the users associated with the user rights and allows these to be changed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Members of this This tab indicates those users associated with the user rights. If the user has an
User Rights associated Working hours time profile, their association to the user rights applies only
during the periods defined by the time profile. If the user does not have an associated
Working hours time profile, they are associated with the user rights at all times.
Members when This tab indicates those users associated with the user rights outside the time periods
out of service defined by their Working hours time profile. The Members when out of service tab is not
populated unless there are time profiles available within the configuration.

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 User Rights

Voicemail
Navigation: User Rights | Voicemail
Display the users associated with the user rights and allows these to be changed.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Voicemail On Default = On
When on, the mailbox is used by the system to answer the user's unanswered calls or
calls when the user's extension returns busy. Note that selecting off does not disable use
of the user's mailbox. Messages can still be forward to their mailbox and recordings can
be placed in it. The mailbox can also still be accessed to collect messages.
Voicemail Default = Off
Ringback
When enabled and a new message has been received, the voicemail server calls the
user's extension to attempt to deliver the message each time the telephone is put down.
Voicemail will not ring the extension more than once every 30 seconds.
DTMF Breakout
When a caller is directed to voicemail to leave a message, they can be given the option to be transferred to a
different extension. The greeting message needs to be recorded telling the caller the options available. The
extension numbers that they can be transferred to are entered in the fields below. These system default values
can be set for these numbers and are used unless a different number is set within these user settings.
The Park & Page feature is supported when the system voicemail type is configured as Embedded Voicemail
or Voicemail Pro. Park & Page is also supported on systems where Avaya Aura Messaging, Modular
Messaging over SIP, or CallPilot (for Enterprise Branch with CS 1000 deployments) is configured as the central
voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto attendant operation. The
Park & Page feature allows a call to be parked while a page is made to a hunt group or extension. This feature
can be configured for Breakout DTMF 0, Breakout DTMF 2, or Breakout DTMF 3.
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Configuration Mode Field Descriptions

Field Description
Reception/ The number to which a caller is transferred if they press 0while listening to the mailbox
Breakout (DTMF greeting rather than leaving a message (*0 on Embedded Voicemail in IP Office mode).
0)
For voicemail systems set to Intuity emulation mode, the mailbox owner can also access
this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave Mail
action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success results
connections depending on whether the caller pressed 0 before or after the record tone.
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0) setting.
When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
• Paging Number – displays a list of hunt groups and users (extensions). Select a hunt
group or extension to configure this option.
• Retries – the range is 0 to 5. The default setting is 0.
• Retry Timeout – provided in the format M:SS (minute:seconds). The range can be set
in 15-second increments. The minimum setting is 15 seconds and the maximum setting
is 5 minutes. The default setting is 15 seconds
Breakout (DTMF The number to which a caller is transferred if they press 2while listening to the mailbox
2) greeting rather than leaving a message (*2 on Embedded Voicemail in IP Office mode)
Breakout (DTMF The number to which a caller is transferred if they press 3while listening to the mailbox
3) greeting rather than leaving a message (*3 on Embedded Voicemail in IP Office mode).

Forwarding
Navigation: User Rights | Forwarding
Additional configuration information
For additional configuration information, see DND, Follow Me, and Forwarding on page 674.
Configuration settings
Display the users associated with the user rights and allows these to be changed.
These settings are mergeable.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Field Description
Block Forwarding
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 ARS

Field Description
Enable Block Default = Off.
Forwarding
When enabled, call forwarding is blocked.
The following actions are blocked:
• Follow me
• Forward unconditional
• Forward on busy
• Forward on no answer
• Call Coverage
• Hot Desking
The following actions are not blocked:
• Do not disturb
• Voicemail
• Twinning

ARS
ARS (Alternate Route Selection) replaces LCR (Least Cost Routing) used by previous releases of
IP Office. It also replaces the need to keep outgoing call routing short codes in the system short
codes.
Related links
Configuration Mode Field Descriptions on page 186

ARS
Navigation: ARS | ARS
Additional configuration information
This section contains the configuration settings for Alternate Route Selection. For additional
configuration information, see Configuring ARS on page 630
Configuration settings
Each ARS form contains short codes which are used to match the result of the short code that
triggered use of the ARS form, ie. the Telephone Number resulting from the short code is used
rather than the original number dialed by the user.
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.

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Configuration Mode Field Descriptions

These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
ARS Route ID The default value is automatically assigned. Range = 0 to 99999.
For most deployments, do not edit this field.
For those conditions where it is necessary to edit this field, the value must be unique
within ARS and within the line Outbound Group IDs.
Route Name Default = Blank. Range = Up to 15 characters.
The name is used for reference and is displayed in other areas when selecting which
ARS to use.
Dial Delay Time Default = System. Range = 1 to 30 seconds.
This settings defines how long ARS should wait for further dialing digits before assuming
that dialing is complete and looking for a short code match against the ARS form short
codes. When set to System, the system setting System | Telephony | Telephony | Dial
Delay Time is used.
Secondary Dial Defaults = Off.
Tone
When on, this setting instructs the system to play secondary dial tone to the user. The
tone used is set by the field below.
The tone used is set as either System Tone (normal dial tone) or Network Tone
(secondary dial tone). Both tone types are generated by the system in accordance with
the system specific locale setting. Note that in some locales normal dial tone and
secondary dial tone are the same.
When Secondary Dial Tone is selected, the ARS form will return tone until it receives
digits with which it can begin short code matching. Those digits can be the result of user
dialing or digits passed by the short code which invoked the ARS form. For example with
the following system short codes:
In this example, the 9 is stripped from the dialed number and is not part of the telephone
number passed to the ARS form. So in this case secondary dial tone is given until the
user dials another digit or dialing times out.
• Code: 9N
• Telephone Number: N
• Line Group ID: 50 Main
In this example, the dialed 9 is included in the telephone number passed to the ARS form.
This will inhibit the use of secondary dial tone even if secondary dial tone is selected on
the ARS form.
• Code: 9N
• Telephone Number: 9N
• Line Group ID: 50 Main
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 ARS

Field Description
Check User Call Default = Off
Barring
If enabled, the dialing user's Outgoing Call Bar setting and any user short codes set to
the function Barred are checked to see whether they are appropriate and should be used
to bar the call.
Description Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
In Service: Default = On
This field is used to indicate whether the ARS form is in or out of service. When out of
service, calls are rerouted to the ARS form selected in the Out of Service Route field.
Short codes can be used to take an ARS form in and out of service. This is done using
the short code features Disable ARS Form and Enable ARS Form and entering the ARS
Route ID as the short code Telephone Number value.
Out of Service Default = None.
Route
This is the alternate ARS form used to route calls when this ARS form is not in service.
Time Profile Default = None.
Use of a ARS form can be controlled by an associate time profile. Outside the hours
defined within the time profile, calls are rerouted to an alternate ARS form specified in the
Out of Hours Route drop-down. Note that the Time Profile field cannot be set until an Out
of Hours Route is selected.
Out of Hours Default = None.
Route
This is the alternate ARS form used to route calls outside the hours defined within the
Time Profile selected above.
Short Codes Short codes within the ARS form are matched against the "Telephone Number" output by
the short code that routed the call to ARS. The system then looks for another match using
the short codes with the ARS form.
Only short codes using the following features are supported within ARS: Dial, Dial
Emergency, Dial Speech, Dial 56K, Dial64K, Dial3K1, DialVideo, DialV110, DialV120
and Busy.
Multiple short codes with the same Code field can be entered so long as they have
differing Telephone Number and or Line Group ID settings. In this case when a match
occurs the system will use the first match that points to a route which is available.
Alternate Route Default = 3. Range = 1 (low) to 5 (high).
Priority
If the routes specified by this form are not available and an Alternate Route has been
specified, that route will be used if the users priority is equal to or higher than the value
set here. User priority is set through the User | User form and by default is 5. If the users
priority is lower than this value, the Alternate Route Wait Time is applied. This field is
grayed out and not used if an ARS form has not been selected in the Alternate Route
field.
If the caller's dialing matches a short code set to the Barred function, the call remains at
that short code and is not escalated in any way.
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Configuration Mode Field Descriptions

Field Description
Alternate Route Default = 30 seconds. Range = Off, 5 to 60 seconds.
Wait Time
If the routes specified by this form are not available and an Alternate Route has been
specified, users with insufficient priority to use the alternate route immediately must wait
for the period defined by this value. During the wait the user hears camp on tone. If during
that period a route becomes available it is used. This field is grayed out and not used if an
ARS form has not been selected in the Alternate Route field.
Alternate Route Default = None.
This field is used when the route or routes specified by the short codes are not available.
The routes it specifies are checked in addition to those in this ARS form and the first route
to become available is used.

Cause Codes and ARS

ARS routing to digital trunks can be affected by signalling from the trunk.
The following cause codes cause ARS to no longer target the line group (unless it is specified by
an alternate ARS route). The response to cause codes received from the line is as follows.
Code Cause Code
1 Unallocated Number.
2 No route to specific transit network/(5ESS) Calling
party off hold.
3 No route to destination./(5ESS) Calling party
dropped while on hold.
4 Send special information tone/(NI-2) Vacant Code.
5 Misdialed trunk prefix.
8 Preemption/(NI-2) Prefix 0 dialed in error.
9 Preemption, cct reserved/ (NI-2) Prefix 1 dialed in
error.
10 (NI-2) Prefix 1 not dialed.
11 (NI-2) Excessive digits received call proceeding.
22 Number Changed.
28 Invalid Format Number.
29 Facility Rejected.
50 Requested Facility Not Subscribed.
52 Outgoing calls barred.
57 Bearer Capability Not Authorized.
63 Service or Option Unavailable.
65 Bearer Capability Not Implemented.
66 Channel Type Not Implemented.
69 Requested Facility Not Implemented.
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 Location

Code Cause Code

70 Only Restricted Digital Information Bearer
Capability Is Available.
79 Service Or Option Not Implemented.
88 Incompatible.
91 Invalid Transit Network Selection.
95 Invalid Message.
96 Missing Mandatory IE.
97 Message Type Nonexistent Or Not Implemented.
98 Message Not Implemented.
99 Parameter Not Implemented.
100 Invalid IE Contents.
101 Msg Not Compatible.
111 Protocol Error.
127 Interworking Unspecified.

Stop ARS The following cause codes stop ARS targeting completely.
Code Cause Code
17 Busy.
21 Call Rejected.
27 Destination Out of Order.

No Affect All other cause codes do not affect ARS operation.

Location
Navigation: Location | Location
Additional configuration information
This section provides the Location field descriptions. For additional configuration information, see:
• Emergency Call on page 594
• Configuring Call Admission Control on page 600
• PreventingToll Bypass on page 647
• Configuring Location Based Extension Resiliency on page 742
Configuring locations allows you to specify named locations for groups of phones, IP Office
systems, or IP Trunks. The IP Office system must also be assigned a location. Multiple systems in
an SCN or Server Edition group of systems may reside in the same location. In an SCN
environment, locations must be configured at the top level and therefore, all systems must be

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Configuration Mode Field Descriptions

configured with the same settings, except when the emergency ARS needs to be set at the
system level.
Once locations have been defined, extensions can be allocated to them in the extension
configuration. IP phones can be identified by the IP address that they register from. Each location
can have only one subnet defined, but phones outside that subnet can be explicitly assigned that
location.
The Location page allows you to define a physical location and associate a network address with
a physical location. Locations can then be allocated to extensions. Linking a location to an
extension, enables the physical location of a phone to be identified when an emergency call is
made.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description
Location Name Default = Blank.
A meaningful location name, clearly identifying the geographical position of the phone.
Location ID Default = Based on existing configured locations, the next incremental value is assigned.
This field is read only.
Subnet Address Default = Blank.
The IP address associated with this location. The subnet where this IP address resides
must be unique across all configured locations. Overlapping IP address ranges between
locations will cause extensions to use the first match found which may not be the correct
location.
Subnet Mask Default = Blank.
The subnet mask for this IP address.
Emergency ARS Default = None.
The ARS (Alternate Route Selection) that defines how emergency calls from this location
are routed. The drop down list contains all available ARS entries using the format ARS
Route ID: Route Name. For example 50: Main.
Parent Location Default = None.
for CAC
The options are:
• None The default setting.
• Cloud The parent location is an internet address external to the IP Office network.
When set to Cloud, the Call Admission Control (CAC) settings are disabled. Calls to
this location from other configured locations are counted as external, yet no CAC limits
are applied to the location itself.
Call Admission Control
The CAC settings, when not unlimited, restrict the number of calls into and out of the location, The following Call
Admission Control settings can be configured.
Total Maximum Default = Unlimited. Range = 1 - 99, Unlimited.
Calls
Limit of all calls to or from other configured locations and the cloud.
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 Location

Field Description
External Default = Unlimited. Range = 1 - 99, Unlimited.
Maximum Calls
Limit of calls to or from the cloud in this location.
Internal Maximum Default = Unlimited. Range = 1 - 99, Unlimited.
Calls
Limit of calls to or from other configured locations in this location.
Time Settings
The display of location based time is only supported on 1100, 1200, 1600 and 9600 Series (96x0 and 96x1)
phones and D100, E129 and B179 telephones.
Time Zone Default = Same as System
Select a time zone from the list.
Local Time Offset Default is based on the currently selected time zone.
from UTC
Set the time for this location by entering the offset from UTC.
Automatic DST Default is based on the currently selected time zone.
When set to On, the system automatically corrects for daylight saving time (DST) changes
as configured in the Clock Forward/Back Settings below.
Clock Forward/ Default is based on the currently selected time zone.
Back Settings
Click Edit to configure the time and date for DST clock corrections. In the Daylight Time
(Start Date — End Settings window, you can configure the following information:
Date (DST Offset))
• DST Offset: the number of hours to shift for DST.
• Clock Forward/Back: Select Go Forward to set the date when the clock will move
forward. Select Go Backwards to set the date when the clock will move backward.
• Local Time To Go Forward: The time of day to move the clock forward or backward.
• Date for Clock Forward/Back: Set the year, month and day for moving the clock
forwards and backwards.
Once you click OK, the forward and back dates, plus the DST offset, are displayed using
the format (Start Date — End Date (DST Offset)).

Fallback System Default = No override.

The drop down list contains all configured IP Office Lines and the associated IP Office
system. The group of extensions associated with this location can fallback to the alternate
system selected.

Related links
Configuration Mode Field Descriptions on page 186
Address on page 549

Address
Navigation: Location | Address

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Configuration Mode Field Descriptions

Enter address information to define a specific location. The address fields are based on the
standards RFC 4119 and RFC 5139.
If Line | SIP LIne | Advanced | Send Location Info is set to Emergency Calls then the location
defined here is sent as part of the INVITE message when emergency calls are made.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field Description Example
Country The country is identified by the two letter ISO 3166 code. US
Code
A1 National subdivisions (state, region, province, New York
prefecture).
A2 County, parish, gun (JP), district (IN). King’s County
A3 City, township, shi (JP). New York
A4 City division, borough, city district, ward, chou (JP). Manhattan
A5 Neighborhood, block. Morningside Heights
A6 Street. Broadway
RD Primary road or street Broadway
RDSEC Trailing street suffix. SW
RDBR Road branch. Lane 7
RDSUBBR Road sub-branch. Alley 8
PRD Leading street direction. N
POD Trailing street suffix. NE
STS Street suffix. Avenue, Platz, Street
PRM Road pre-modifier. Old
POM Road post-modifier. Extended
HNO House number, numeric part only. 123
HNS House number suffix. A, 1/2
LMK Landmark or vanity address. Low Library
BLD Building (structure). Hope Theatre
LOC Additional location information. Room 543
PLC Place type. Office
FLR Floor. 5
UNIT Unit (apartment, suite). 12a
ROOM Room. 450F
SEAT Seat (desk, cubicle, workstation). WS 181
NAM Name (residence, business, or office occupant). Joe’s Barbershop
ADDCODE Additional Code 13203000003
PCN Postal community name. Leonia
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 Location

Field Description Example

PC Postal code. 10027-0401
POBOX Post office box (P.O. box) U40

Related links
Location on page 547

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Chapter 12: Configure General System
Settings

Applying Licenses
For a description of IP Office licenses and for information on licensing requirements, see Avaya IP
Office Platform™ Solution Description.
Related links
PLDS licensing on page 552
Web License Manager (WebLM) on page 553
Server Edition Centralized Licensing on page 554
Distributing Server Edition Licenses on page 554
Procedures for Applying Licensing on page 559

PLDS licensing
IP Office uses the Avaya Product Licensing and Delivery System (PLDS) to manage licenses.
PLDS is an online, web-based tool for managing license entitlements and electronic delivery of
software and related license files. PLDS provides customers, Avaya Partners, distributors, and
Avaya Associates with easy-to-use tools for managing license entitlements and electronic delivery
of software and related license files. Using PLDS, you can perform operations such as license
activations, license upgrades, license moves, and software downloads. You can access PLDS
from http://plds.avaya.com/.
PLDS license files
Licenses are delivered from PLDS with license files. A PLDS license file is generated for installing
on a specific machine. There are two deployment options:
• PLDS Nodal license files are generated for and installed on particular IP Office nodes.
• PLDS WebLM license files are generated for and installed on a WebLM server that can
license multiple IP Office nodes.
WebLM centralized licensing is supported in IP Office Server Edition and in IP Office Branch
deployments, but not in non-Branch deployments of IP Office Standard mode.

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 Applying Licenses

PLDS host ID
PLDS Nodal license files are machine specific and you must specify the host ID in the PLDS host
ID field on License | License.
IP500 V2 systems: You can find the PLDS host ID in the Licensing tab of IP Office Manager and
Web Manager. The PLDS host ID is made of the two digits “11”, followed by the 10 digit feature
key serial number printed on the IP Office SD card. If the SD card is changed, the PLDS host ID
will also change.
IP Office Linux servers: The PLDS host ID can be found on the server labeling, the server
packaging label, and the system ignition Login screen. The PLDS host ID is derived from the
system ID. If the system ID changes, the PLDS host ID will also change.
WebLM: The WebLM host ID is the Mac address of the WebLM server. In a virtual environment,
the WebLM host ID is a virtual Mac address that starts with the letter “V”. The WebLM host ID
must be used when generating a PLDS license file for the WebLM server in order to implement a
centralized licensing scheme for multiple IP Office systems. The WebLM host ID can be found on
the server labeling, the server packaging label, the system ignition Login screen, and through the
WebLM management interface.
Related links
Applying Licenses on page 552

Web License Manager (WebLM)

The Web License Manager (WebLM) is a web-based application for managing licenses. If you use
the WebLM server running on the IP Office server, then you can use IP Office Web Manager to log
in to the WebLM server by selecting Applications > Web License Manager. WebLM credentials
are managed separately from IP Office system passwords and are not part of single sign on
(SSO).
Note:
• WebLM license management is supported for Server Edition deployments and for
Enterprise Branch deployments using the System Manager WebLM server. It is not
supported for Standard Mode systems.
• When upgrading from a previous release, all systems must be running the same software
level. IP Office Server Edition does not support mixed versioning.
For more information on WebLM, see Administering standalone Avaya WebLM.
To establish communication between IP Office and the WebLM server, you must configure the
remote server profile on License | Remote Server .
Note:
When upgrading from release 9.1, the WebLM server is not started automatically. Perform the
following steps to start the WebLM server.
1. Log in to Web Manager.
2. Select Server Menu > Platform View > System.
3. Under Services, select the WebLM server and click Start.

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Configure General System Settings

Related links
Applying Licenses on page 552

Server Edition Centralized Licensing

Before release 10, Server Edition deployments used nodal licensing. This type of licensing can still
be used in release 10 and higher. However, it is expected that most deployments will prefer to
centralize license management using the Avaya Web License Management (WebLM) server. The
WebLM server is automatically installed on the Server Edition Primary server. For newly installed
systems, centralized licensing is the default configuration.
All systems in the Server Edition solution must use the same License Source.
Nodal licensing
With nodal licensing, license files must be installed on each node in the system. For some
licensed features, the required license can be installed on the Server Edition Primary server and
used by all nodes in the system. However, for other licensed features, the required license must
be installed on the node where the feature is used.
Centralized licensing
As of release 10, you can use the WebLM server running on the Server Edition Primary server to
fully centralize license management. With centralized license management, all licenses are
contained in a single PLDS file uploaded to WebLM. All nodes in the solution obtain their licenses
from WebLM.
The IP Office Secondary server and Expansion systems can be configured to request licenses
directly from the WebLM server, or to use a proxy option. When configured to use the proxy
option, the license requests are sent through the IP Office Primary server, which proxies the
requests to the WebLM server. The Primary server does not allocate licenses, but only acts as a
proxy.
Systems using nodal licensing can be converted to use centralized licensing. Since PLDS license
files are generated using the host ID of the server where they reside, you must regenerate the
license file using the host ID of the WebLM server that will host the license file.
Related links
Applying Licenses on page 552

Distributing Server Edition Licenses

Note:
For a description of IP Office licenses and for information on licensing requirements, see
Avaya IP Office Platform™ Solution Description.
The License | Remote Server page displays the Reserved Licenses allocated to a Server
Edition server.

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 Applying Licenses

Note:
The SIP Trunk Sessions field has replaced the System | Telephony | Telephony | Max SIP
sessions setting.

PLDS File Location

How licenses are allocated depends on the location of the PLDS file. For standalone systems,
SCN deployments, and Server Edition nodal licensing, each node in the system must have a
PLDS file installed.

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PLDS File

Distributed
Primary Server Licenses

Secondary Server Expansion Expansion

Figure 1: PLDS file location for Server Edition Nodal Licensing

For Server Edition centralized licensing, the PLDS file is located on the WebLM server. The
WebLM server can be located on the Primary Server or on a remote server.

PLDS File

Primary Server WebLM

Server

Secondary Server Expansion Expansion

Figure 2: PLDS file location for Server Edition Centralized Licensing

Related links
Applying Licenses on page 552

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Nodal license distribution on page 557

Centralized license distribution on page 558

Nodal license distribution

When the License Source is Local, the Reserved Licenses read-only fields indicate licenses
that are required for the currently configured features.
Nodal licensing for a Server Edition solution is based on a combination of licensing done through
the Server Edition Primary server plus some server-specific licenses. All the user specific and
system specific licenses can be managed from the Server Edition Primary server that also acts as
a licensing server. Licenses are entered into the configuration of the Server Edition Primary server
and are based on the system ID of that server.
Where a license is used to enable features, such as SIP Trunk channels, on other systems, the
Server Edition Primary server only allocates those licenses to other systems after it has met its
own license needs.
When another system loses connection to the Server Edition Primary server, any license
requirements based on those licenses entered in the Server Edition Primary server's configuration
are supported for a grace period of 30 days.
Other server specific licenses are entered into the configuration of the server requiring the feature
and are based on the System ID of that system.
License Primary server Server-specific
Server Edition Yes No
Avaya IP endpoints Yes No
Third-party IP endpoints Yes No
SIP trunk channels Yes No
IP500 universal PRI channels No Yes
Additional voicemail ports [3] Yes No
Web Collaboration No Yes
UMS Web Services [1] No Yes
Office Worker Yes No
Power User Yes No
Office Worker to Power User Yes No
upgrade
Receptionist No Yes
CTI Link Pro No Yes
Messaging TTS Pro [3] Yes No
Voicemail Pro Recording Yes No
Administrator [2] [3]
WAV User No Yes
IPSec tunneling No Yes
Table continues…

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License Primary server Server-specific

Video SoftPhone for Mac No Yes

1. UMS Web Service licenses are for Hunt Groups only.

2. The Voicemail Pro Recording Administrator license refers to Contact Store. Only one
license is required for a Server Edition network.
3. For deployments with dual Voicemail Pro servers, Messaging TTS Pro, Voicemail Pro
Recording Administrator, and Additional voicemail ports licenses must be on the
Secondary Server.
Related links
Distributing Server Edition Licenses on page 554

Centralized license distribution

When the license source is WebLM, the Reserved Licenses read-only fields indicate licenses
that are required for the currently configured features. Editable fields can be used to:
• Request additional licenses from the WebLM server.
• Remove licenses from the IP Office node to apply them elsewhere.
Important:
When reallocating licenses, always reduce the number on the IP Office node where they are
currently applied before applying them on another node. If you exceed the number of licenses
available, you will receive an error message.
Distribution after conversion from Nodal to Centralized licensing
• If the IP Office node needs any of the following licenses, then you must manually configure
the respective Reserved Licenses editable fields. This will allow the IP Office node to
request the licenses from the WebLM server.
- VMPro Recordings Administrators
- VMPro TTS Professional
- CTI Link Pro
Extension Reserved license setting: When the license source is Local, the setting
Extension > VoIP > Reserve License is set to None. Switching the license source to WebLM
changes the setting to Reserve Avaya IP endpoint license. If required, you must manually
change this setting to Reserve 3rd party endpoint license or Both.
License allocation in WebLM
You can use WebLM to view the licenses used by each node in IP Office Server Edition. In the
WebLM navigation pane on the left, click Licensed Products. The Acquired licenses table
displays information about the licenses acquired for each client ID. In IP Office, the WebLM client
ID for each node is displayed on the license Remote Server page.
Related links
Distributing Server Edition Licenses on page 554

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 Applying Licenses

Procedures for Applying Licensing

Applying Licenses

Deployment
Type?

Standalone Enterprise
and SCN Branch

Uploading a PLDS
Configuring the License Server
License
for Enterprise Branch
File to IP Office

Server Edition Server Edition

Nodal Centralized

Configuring a Server Obtaining the Host ID

Edition License Source of the WebLM Server

Uploading a PLDS Installing a License File

License on the WebLM Server
File to IP Office

Configuring a Server
Edition License Source
Configuring Server
Edition Nodal Licensing

Configuring Server
Edition Centralized Licensing

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Related links
Applying Licenses on page 552
Obtaining the Host ID of the WebLM Server on page 560
Installing a License File on the WebLM Server on page 560
Configuring the Server Edition License Source on page 561
Uploading a PLDS License File to IP Office on page 561
Configuring Server Edition Nodal Licensing on page 562
Configuring Server Edition Centralized Licensing on page 562
Configuring the License Server in an Enterprise Branch Deployment on page 564

Obtaining the Host ID of the WebLM Server

The WebLM Host ID is required to generate a PLDS license file for centralized licensing. The
license file is uploaded to the WebLM server.
Procedure
1. In Web Manager, select Applications > Web License Manager.
2. Log in to WebLM.
3. In the navigation pane on the left, click Server Properties.
The Server Properties page displays the Host ID. The host ID is the MAC address of the
Server Edition Primary server.
Record the host ID.
Related links
Procedures for Applying Licensing on page 559

Installing a License File on the WebLM Server

Use Web Manager to log in to the WebLM license server and install a license file.
Before you begin
Obtain the license file from the Avaya Product Licensing and Delivery System (PLDS) Web site at
https://plds.avaya.com.
You must know the user ID and password for the WebLM server. WebLM credentials are managed
separately from IP Office system passwords and are not part of single sign on.
Procedure
1. Log in to Web Manager.
2. Select Applications > Web License Manager.
3. Log in to the WebLM server.
4. In the left navigation pane, click Install license.
5. On the Install license page, click Browse to select the license file.

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6. Click Install to install the license file.

WebLM displays a message upon successful installation of the license file.
If the installation is not successful, for troubleshooting information see Administering Avaya
WebLM, available on the Avaya support site at https://downloads.avaya.com/css/P8/
documents/100157154.
Related links
Procedures for Applying Licensing on page 559

Configuring the Server Edition License Source

For Server Edition deployments, the license source can be centralized or nodal.
• With centralized licensing, the PLDS license file resides on the WebLM server. The WebLM
server is the license source and all nodes in the solution receive licenses from the WebLM
server. The WebLM server can run on a remote machine or on the Primary Server.
• With nodal licensing, a PLDS license file is uploaded to each node.
All systems in the Server Edition solution must use the same license source. The license source is
defined by the configuration setting License | License | License Source. Use this procedure to
set all nodes to use the same license source.
Procedure
1. Log in to Manager.
2. On the Solution view page, on the right hand side, select Set All Nodes License Source.
3. In the Select License Source window, select either
• Local/Primary Server for nodal licensing.
• WebLM for centralized licensing.
All nodes in the solution are set to the same license source.
Related links
Procedures for Applying Licensing on page 559

Uploading a PLDS License File to IP Office

Use this procedure to upload a PLDS license file for nodal license management. Nodal license
management is used for standalone IP500 V2 systems and is an option for Server Edition
systems.
Before you begin
The PLDS license file must be on the local machine where IP Office Manager is running
Procedure
1. In IP Office Manager, select the Primary Server and then select License | License.
2. Click Add.

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3. In the Upload Files window, navigate to the license file.

4. Select the file and click Open.
Related links
Procedures for Applying Licensing on page 559

Configuring Server Edition Nodal Licensing

With nodal licensing, licenses are managed using license files installed on each node in the
system. For information on license distribution, see Distributing Nodal Licenses on page 557.
Procedure
1. In IP Office Manager, select the Primary Server and then select License | Remote Server.
2. In the Licence Source field, select Primary.
Note:
All systems in the Server Edition solution must use the same License Source. In
Manager, on the Solution page, you can select Set All Nodes License Source to
configure the setting for all nodes in the solution.
3. Enter the Server Edition Primary server IP address in the License Server IP Address
field.
4. Under Reserved Licenses, the right hand column indicates which licenses have been
reserved for this system. Use the left hand column to request additional licenses for this
system.
5. Click OK.
Licenses are displayed in the table.
6. Repeat steps 1 to 5 for the Server Edition Secondary server and all Server Edition
Expansion Systems.
Related links
Procedures for Applying Licensing on page 559

Configuring Server Edition Centralized Licensing

With centralized licensing, licenses are managed from a central WebLM server.
Before you begin
You must have a PLDS license file activated with the host ID of the WebLM server
Procedure
1. In IP Office Manager, select the Primary Server and then select License | Remote Server.
2. Ensure Licence Source is set to WebLM.

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Note:
All systems in the Server Edition solution must use the same License Source. In
Manager, on the Solution page, you can select Set All Nodes License Source to
configure the setting for all nodes in the solution.
3. The WebLM server can be located on the Server Edition Primary server or on a separate
server. Enter the domain name or IP address of the WebLM server in the Domain Name
(URL) field.
Note that the domain name URL must use https://.
4. If required, change the path to the WebLM server in the Path field.
5. Under Reserved Licenses, the right hand column indicates which licenses will be
automatically requested from the WebLM server. Use the left hand column to request
additional license types for this system.
6. Navigate to the Remote Server page for the Server Edition Secondary server.
7. Ensure the Licence Source is set to WebLM.
8. You can choose to enable the Enable proxy via Primary IP Office line check box.

Choice Option Choice Description

Enabled The WebLM request is sent to the WebLM server via the IP Office line
configured to the Server Edition Primary server. The line must be up and in
service
Disabled The WebLM request is sent directly to the WebLM server.

9. If Enable proxy via Primary IP Office line is enabled, enter the Server Edition Primary
server IP address in the Primary IP Address field.
10. If Enable proxy via Primary IP Office line is disabled:
a. Enter the domain name or IP address of the WebLM server in the Domain Name
(URL) field.
b. If required, change the path to the WebLM server in the Path field.
c. If required change the default Port Number.
For information on port usage see the IP Office Port Matrix document on the Avaya
support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003.
11. Click OK.
Licenses are displayed in the License | License table.
12. Repeat steps 8 to 12 for all Server Edition Expansion Systems.
Note:
In Manager, on the Solution page, you can select Set All Nodes License Source.

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Related links
Procedures for Applying Licensing on page 559

Configuring the License Server in an Enterprise Branch Deployment

Use this procedure to configure WebLM centralized licensing where a shared PLDS license file is
installed on the WebLM server. This is the recommended method for installing license files on IP
Office systems that are centrally managed by System Manager.
For a complete description of deploying Enterprise Branch, see Deploying Avaya IP Office™
Platform as an Enterprise Branch with Avaya Aura® Session Manager.
Procedure
1. Log in to IP Office Manager and select License | Remote Server.
2. Select the Enable Remote Server check box.
The Reserved Licenses information is displayed.
3. In the Domain Name (URL), field, enter the domain name or IP address of the WebLM
server or the domain name of System Manager if the system is under System Manager
control.
4. (Optional) If a secondary System Manager is configured, enter the domain name in the
Secondary Domain Name (URL) field.
5. If required, change the path to the WebLM server in the Path field.
6. If required change the default Port Number.
For information on port usage see the IP Office Port Matrix document on the Avaya support
site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003.
7. Under Reserved Licenses, the right hand column indicates which licenses will be
automatically requested from the WebLM server. Use the left hand column to request
additional licenses for this system.
Related links
Procedures for Applying Licensing on page 559

Converting from Nodal to Centralized Licensing

If you are upgrading from an earlier release, perform the procedure Migrating Licenses to
PLDS on page 565.
Note:
When upgrading from a previous release, all system must be running the same software level.
The IP Office Server Edition Solution does not support mixed versioning.

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Procedure
1. You must generate a license file using the WebLM host ID. Perform the following steps to
find the WebLM host ID.
a. In Web Manager, select Applications > Web License Manager.
b. Log in to WebLM.
c. In the navigation pane on the left, click Server Properties.
The Server Properties page displays the Host ID. The host ID is the MAC address of
the Server Edition Primary server.
Record the host ID.
2. Generate a PLDS license file using the WebLM host ID.
3. Upload the license file.
a. In Web Manager, select ApplicationsWeb License Manager.
b. In the navigation pane on the left, click Install license.
c. Click Browse to select the license file.
d. Click Install to install the license file.
4. All nodes in the solution must have the same license source. To configure centralized
licensing, all nodes must have the License Source set to WebLM. You can use Manager
to set all nodes to the same license source. On the Manager Solution page, on the right
hand side, select Set All Nodes License Source and then select WebLM.
5. If you are performing this procedure after an upgrade, you must ensure that the Domain
Name (URL) field is populated on the Server Edition Primary server.
a. In Web Manager select for the Server Edition Primary server.
b. Ensure that the Domain Name (URL) field contains the domain name or IP address
of the Server Edition Primary server.
6. Reallocate the licenses as required. See Distributing Centralized Licenses on page 558.
Note that the previously install local licenses are listed as obsolete. You can use this list to
determine which licenses to request from the WebLM server. Once licenses have been
reallocated, you can delete the obsolete licenses.

Migrating Licenses to PLDS

IP Office release 10 and higher supports only the Product Licensing and Delivery System (PLDS)
to manage license files. If you are upgrading from a previous release, you must migrate all of your
pre-R10 licenses (ADI, PLDS, mix of ADI/PLDS, virtual) to R10 PLDS licenses. The license
migration tool extracts all the licensing information from an IP Office system and saves it to a file.
This file can then be used prepare a software upgrade quote in the Avaya One Source
Configurator in order to obtain the required new PLDS R10 licenses.

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For Server Edition deployments, the License Migration tool collects licensing information from
every node in the solution.
Note:
• You must use the release 10 or higher Manager client to generate the license inventory
file.
You can install Manager before upgrading to release 10. See the procedure Installing
Manager.
• License migration is supported on all IP Office modes, release 6.0 and higher.
• The license migration tool can only be used with an online configuration. The Tools >
License Migration option is disabled for offline configurations.
• The license migration tool is not available on UCM and Application servers. When you
run the license migration tool on a Server Edition server, the tool collects licensing
information from every node in the solution.
• The generated file can be read but must not be edited. License migration will fail if the file
has been edited.
Before you begin
Ensure all licenses are loaded on the system before performing the license migration. For Server
Edition deployments, ensure all nodes are online in order to capture the current view of systems in
the solution.
The IP Office configuration must be opened online. The License Migration tool is not available in
offline mode.
Procedure
1. Log in to Manager and select Tools > License Migration.
The Save As window opens.
2. Select a location to save the file and enter a file name.
3. Click Save.
The file is saved with a .zip extension.
Next steps
Use the file to prepare a software upgrade quote in the Avaya One Source Configurator in order to
obtain the required new PLDS R10 licenses. Once you have the PLDS license files, apply them to
the system.

Certificate Management
Related links
Certificate Overview on page 567

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 Certificate Management

Certificate Support on page 571

Certificate Overview
Public key cryptography is one of the ways to maintain a trustworthy networking environment. A
public key certificate (also known as a digital certificate or identity certificate) is an electronic
document used to prove ownership of a public key. The certificate includes information about the
key, information about its owner's identity, and the digital signature of an entity that has verified the
certificate's contents are correct. If the signature is valid, and the person examining the certificate
trusts the signer, then they know they can use that key to communicate with its owner.
The system used to provide public-key encryption and digital signature services is called a public
key infrastructure (PKI). All users of a PKI should have a registered identity which is stored in a
digital format and called an Identity Certificate. Certificate Authorities are the people, processes
and tools that create these digital identities and bind user names to public keys.
There are two types of certificate authorities (CAs), root CAs and intermediate CAs. In order for a
certificate to be trusted and for a secure connection to be established, that certificate must have
been issued by a CA that is included in the trusted certificate store of the device that is
connecting. If the certificate was not issued by a trusted CA, the connecting device then checks to
see if the certificate of the issuing CA was issued by a trusted CA, and so on until either a trusted
CA is found. The trusted certificate store of each device in the PKI must contain the required
certificate chains for validation.

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IP Office Root Certificate Authority

IP Office generates a self-signed certificate. For IP500 V2 systems, a certificate is generated
automatically on the first start up. On Linux systems, a certificate is generated during the ignition
process.
The following entities can act as the certificate authority.
• The Server Edition Primary Server, an Application Server, or a Unified Communication
Module (UCM) can act as the root certificate authority for all nodes in the system.
• In Enterprise Branch deployments, the System Manager can act as the root certificate
authority.
• Identity certificates can also be purchased and issued by a third party certificate authority.
Regardless of the method used to provide the IP Office identity, the certificate authority which
signs the IP Office identity certificate must be trusted by all the clients and endpoints that need to
establish a secure connection with IP Office. They must be a part of the PKI. Therefore, the root
CA certificate must be downloaded to client devices and placed in the trusted certificate store. If
there are intermediate CAs in the certificate chain, either the intermediate CAs must be added to
the client device Trusted Certificate Store or the certificate chain must be advertised by IP Office in
the initial TLS exchange.
Certificates and TLS
Telephony signaling like SIP messaging is secured using Transport Layer Security (TLS). TLS
provides communication security using certificates to authenticate the other end of the IP Link.
The message exchange in TLS is aimed at verifying the identity of the communicating parties and
establishing the keys that will be used to encrypt the signaling data between the two parties.
Typically, the server sends its identity certificate, either self-signed or signed by the CA, to the
client. The client must have the CA certificate in its trusted certificate store.
IP Office acts as the TLS server in its interactions with SIP telephony clients. This means that the
TLS application on the IP Office must be configured to listen for client connections by enabling
TLS in the SIP Registrar on the LAN1 and LAN2 interfaces.
Note:
• Authentication of the client's certificate by the server is not a requirement. IP Office does
not support client certificate validation for all SIP endpoint types.
• The E.129 phone does not validate the IP Office identity certificate.
Related links
Certificate Management on page 566
Windows Certificate Store on page 568

Windows Certificate Store

The certificate store used by Manager to save and retrieve X509 certificates is the default one
provided by the Windows operating system. The Windows certificate store is relevant to any
application running on Windows that uses certificates for security, either TLS or HTTPS. For
example, the Avaya Communicator Client.

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 Certificate Management

Warning:
Avaya accepts no responsibility for changes made by users to the Windows operating system.
Users are responsible for ensuring that they have read all relevant documentation and are
sufficiently trained for the task being performed.
Windows Certificate Store Organization
By default, certificates are stored in the following structure:

Each of the sub folders has differing usage. The Certificates - Current User area changes with the
currently logged-in windows user. The Certificate (Local Computer) area does not change with the
currently logged-in windows user.
Manager only accesses some of the certificate sub folder:

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Certificates (Local Computer) Folder Manager Use

Personal | Certificates Folder searched by Manager 1st for matching
certificate to send to the system when requested.
Certificate matched by the subject name contained
in File | Preferences | Security | Certificate
offered to the system.
Folder accessed whenever ‘Local Machine
certificate store’ used for Security Settings.
Folder searched by Manager for matching
certificate when certificate received from the
system, and File | Preferences | Security |
Manager Certificate Checks = Medium or High.
Trusted Root Certification Authorities | Folder searched by Manager for matching parent
Certificates certificates when non-self signed certificate
received from the system, and File | Preferences |
Security | Manager Certificate Checks = Medium
or High.

Certificates – Current User Folder Manager Use

Personal | Certificates Folder searched by Manager 2nd for matching
certificate (subject name) to send to the system
when requested. Certificate matched by the subject
name contained in File | Preferences | Security |
Certificate offered to the system.
Folder accessed whenever ‘Current User
certificate store’ used for Security Settings.
Folder searched by Manager for matching
certificate when certificate received from IP Office,
and File | Preferences | Security | Manager
Certificate Checks = Medium or High.
Trusted Root Certification Authorities | Folder searched by Manager for matching parent
Certificates certificates when non-self signed certificate
received from the system, and File | Preferences |
Security | Manager Certificate Checks = Medium
or High.
Other People | Certificates Folder searched by Manager for matching parent
certificates when non-self signed certificate
received from the system, and File | Preferences |
Security | Manager Certificate Checks = Medium
or High.

Windows Certificate Store Import

In order to use certificates – either for security settings or Manager operation – they must be
present in the windows certificate store. Certificates may be placed in the store by the Certificate
Import Wizard. The Certificate Import Wizard can be used whenever a certificate is viewed. In

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 Certificate Management

order for Manager to subsequently access this certificate the Place all certificate in the
following store option must be selected:
• If the certificate is to subsequently identify the system, the Other People folder should be
used.
• If the certificate is to subsequently identify the Manager, the Personal folder should be used,
and the associated private key saved as well.
Certificate Store Export
Any certificate required outside of the Manager PC must be first saved in the Certificate store,
then exported.
If the certificate is to be used for identity checking (i.e. to check the far entity of a link) the
certificate alone is sufficient, and should be saved in PEM or DER format.
If the certificate is to be used for identification (i.e. to identify the near end of a link) the certificate
and private key is required, and should be saved in PKCS#12 format, along with a password to
access the resultant .pfx file.
Related links
Certificate Overview on page 567

Certificate Support
Related links
Certificate Management on page 566
Certificate File Naming and Format on page 571
Identity Certificate on page 572
Trusted Certificate Store on page 574
Signing Certificate on page 576
Certificate File Import on page 577

Certificate File Naming and Format

DER: Distinguished Encoding Rules (DER) format, which is a binary format used to represent a
certificate. Typically used to describe just one certificate, and cannot include a private key.
There are four main encodings/internal formats for certificate files. Note that these are encodings,
not file naming conventions.
PEM: Privacy Enhanced Mail (PEM) is a Base 64 (i.e. ASCII text) encoding of DER, one
certificate is enclosed between ‘-----BEGIN CERTIFICATE-----‘ and ‘-----END CERTIFICATE-----
‘ statements. Can contain a private key enclosed between ‘-----BEGIN PRIVATE KEY -----‘ and
‘-----END BEGIN PRIVATE KEY -----‘ statements. More than one certificate can be included. PEM
can be identified by viewing the file in a text editor. This is an unsecure format and not
recommended for private key use unless it is protected with a password.
PKCS#12: Public Key Cryptography Standard (PKCS) #12. A secure, binary format, encrypted
with a password. Typically used to describe one certificate, and its associated private key, but can
also include other certificates such as the signing certificate(s). This is the recommended format
for private key use.

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PKCS#7: A Base 64 (i.e. ASCII text) encoding defined by RFC 2315, one or more certificates are
enclosed between ‘—–BEGIN PKCS—–‘ & ‘—–END PKCS7—–‘ statements. It can contain only
Certificates & Chain certificates but not the private key. Can be identified by viewing the file in a
text editor.
There are many common filename extensions in use:
• .CRT — Can be DER or PEM. Typical extension used by Unix/Android systems’ public
certificates files in DER format.
• .CER — Can be DER or PEM. Typical extension used by Microsoft/Java systems’ public
certificates files in PEM format.
• .PEM — Should only be PEM encoded.
• .DER — Should only be DER encoded.
• .p12 — Should only be in PKCS#12 format. Typical extension used by Unix/Android systems’
identity certificates/private key pair files. Same format as .pfx hence can be simply renamed.
• .pfx — Should only be in PKCS#12 format. Typical extension used by Microsoft systems’
identity certificates/private key pair files. Same format as .p12 hence can be simply renamed.
• .pb7 — Should only be in RFC 2315 format. Typical extension used by Microsoft and Java
systems for certificate chains.
Related links
Certificate Support on page 571

Identity Certificate
Feature Support Notes
Import: Public key size Yes RSA 1024, 2048 and 4096 bit public keys must be supported. Any
other sizes are optional.
Import of RSA public key less than 1024 or greater than 4096 bits to
be rejected with an informative error.
Import of certificates with 1024 will be imported after a warning ‘The
certificate public key may not be of sufficient strength. Do you wish
to continue?’
Import: Certificate Yes SHA-1, SHA-256 SHA-384, and SHA-512 hashing algorithms must
signature algorithm be supported. Any other SHA2 algorithms are optional.
Import of certificates with SHA-1 will be imported after a warning
‘The certificate signature algorithm may not be of sufficient strength.
Do you wish to continue?’
Import of certificates with other algorithms (for example MD5, ECC)
to be rejected with an informative error.
Import: Must have Yes Must be supplied.
private key
Reject and informative error that private key has not been supplied
Table continues…

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 Certificate Management

Feature Support Notes

Import: Certificate Yes Minimum checks for:
checks
• Version (v3)
• Start + end (present)
• Subject Name (present)
• Issuer Name (present)
• Data integrity (e.g. hash)
Reject + informative error if a check fails
Import: Certificate up Yes Certificates can be varying sizes
to 4KB
Import: Formats Yes • PKCS#12 format. ‘.p12’ and ‘.pfx’ file extension. With or without
password. This shall be the preferred/default option
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• Pasted from clipboard in PEM format (optional)
NOTE that ONLY PKCS#12 file format is acceptable according to
147434–030–P1, however we cannot control what format customers
receive their certificates in, hence all should be supported
See section below for certificate file import support
Import: Up to 4 other Yes Only supported where management of TCS also available.
certificates in same file
• Any intermediate and root CA certificate included in the PKCS#12
file to be imported into the Trusted Certificate store
• The feature is intended for import of intermediate certificates, but
can include unrelated certificates.
• An informative message to the admin if any have been imported
Import: Certificate Yes Where identity certificate is signed by one or more intermediate
chain support CAs, search TCS for matching certificates and include in identity
certificate chain.
Table continues…

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Configure General System Settings

Feature Support Notes

View: Certificate Yes Minimum viewable attributes (From CEC016: 147434–030–P1):
Contents
• Serial Number
• Subject Name
• Issuer Name
• Validity Period (that includes notBefore and notAfter dates)
• Thumbprint (Hash of the certificate)
• Subject Alternative Names
• Key Usage Extensions
• Extended Key Usage
Warnings/errors as per 147434–080–P1:
• Error displayed that certificate has expired
• Warning displayed that certificate is nearing expiry (within 60
days).
View: Private Key No Private key must not be viewable
Export: Formats Yes Private key must not be exportable
Export formats:
• DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 (optional)

Related links
Certificate Support on page 571

Trusted Certificate Store

Feature Support Notes
Import: RSA Yes RSA 1024, 2048 and 4096 bit public keys must be supported. Any
1024-4096 key size other sizes are optional.
Import of RSA public key less than 1024 or greater than 4096 bits to
be rejected with an informative error.
Import: Optional Yes No private key will actually be imported.
private key
Informative message (neither warning or error) that private key has
not been imported
Table continues…

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 Certificate Management

Feature Support Notes

Import: Certificate Yes Minimum checks for:
checks
• Version (v3)
• Start + end (present)
• Subject Name (present)
• Issuer Name (present)
• Data integrity (e.g. hash)
Reject + descriptive error if a check fails
Import: Certificate up Yes Certificates can be varying sizes
to 4KB
Import: Formats Yes • DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 format. ‘.p12’ and ‘.pfx’ file extension. With or without
password.
• Pasted from clipboard in PEM format (optional)
Import: Up to 19 other Yes All included certificates, up to 20 total. More than 20 in one file can
certificates in same file be optionally supported.
View: TCS Certificate Yes Minimum viewable attributes (From CEC016: 147434–030–P1):
• Serial Number
• Subject Name
• Issuer Name
• Validity Period (that includes notBefore and notAfter dates)
• Thumbprint (Hash of the certificate)
• Subject Alternative Names
• Key Usage Extensions
• Extended Key Usage
Warnings/errors as per 147434–080–P1:
• Error displayed that a certificate has expired
• Warning displayed that a certificate is nearing expiry (within 60
days).
Export: Formats Yes Export formats:
• DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 (optional)

Related links
Certificate Support on page 571

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Configure General System Settings

Signing Certificate
Feature Support Notes
Import: RSA Yes RSA 1024, 2048 and 4096 bit public keys must be supported. Any
1024-4096 key size other sizes are optional.
Import of RSA public key less than 1024 or greater than 4096 bits to
be rejected with an informative error.
Import: Must have Yes Must be supplied.
private key
Reject and informative error that private key has not been supplied
Import: Certificate Yes Minimum checks for:
checks
• Version (v3)
• Start + end (present)
• Subject Name (present)
• Issuer Name (present)
• Data integrity (e.g. hash)
Reject and informative error if a check fails
Import: Certificate up Yes Certificates can be varying sizes
to 4KB
Import: Formats Yes • PKCS#12 format. ‘.p12’ and ‘.pfx’ file extension. With or without
password. This shall be the preferred/default option
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• Pasted from clipboard in PEM format (optional)
NOTE that ONLY PKCS#12 file format is acceptable according to
147434–030–P1, however we cannot control what format customers
receive their certificates in, hence all should be supported
Import: Other No Informative warning that other certificates have not been imported
certificates in same file
Table continues…

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 Certificate Management

Feature Support Notes

View: TCS Certificate Yes Minimum viewable attributes (From CEC016: 147434–030–P1):
• Serial Number
• Subject Name
• Issuer Name
• Validity Period (that includes notBefore and notAfter dates)
• Thumbprint (Hash of the certificate)
• Subject Alternative Names
• Key Usage Extensions
• Extended Key Usage
Warnings/errors as per 147434–080–P1:
• Error displayed that certificate has expired
• Warning displayed that certificate is nearing expiry (within 60
days).
Renew existing: Yes Regenerate CA keeping all keys and other contents same except:
• notBefore and notAfter dates
• Serial Number
• Thumbprint (Hash of the certificate)
• ??
Can this be done to imported CAs or just internally generated ones?
Create new: Yes Regenerate CA, including keys
Export: Formats Yes Private key must not be exportable
Export formats:
• DER format. ‘.cer’ ‘.der’ and ‘.crt’ file extension.
• PEM format. ‘.cer’ ‘.pem’ and ‘.crt’ file extension.
• PKCS#12 (optional)

Related links
Certificate Support on page 571

Certificate File Import

File Content Identity Certificate Trusted Signing Notes
Import Command Certificate Import Certificate Import
Command Command
DER
Table continues…

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Configure General System Settings

File Content Identity Certificate Trusted Signing Notes

Import Command Certificate Import Certificate Import
Command Command
DER: 1 certificate No – attempt Yes – attempt No – attempt
rejected with accepted with ‘N rejected with
‘Invalid certificate certificate(s) ‘Invalid certificate
format (DER)’ imported into format (DER)’
Trusted Certificate
Store’
DER: Any other No – attempt No – attempt No – attempt
content rejected with rejected with rejected with
‘Invalid content ‘Invalid content ‘Invalid content
(DER)’ (DER)’ (DER)’
PKCS#12
PKCS#12: 1 Yes – attempt No – p12/pfx Yes – attempt
certificate + private accepted with should not be accepted with
key ‘Certificate import offered for file ‘Certificate import
successful’ selection successful’
Certificate/key
imported as ID
certificate
PKCS#12: 1 Yes – attempt No – p12/pfx Yes – attempt At least 20
certificate + private accepted with should not be accepted with certificates
key, 1 or more ‘Certificate import offered for file ‘Certificate import supported in the
other certificates successful’ selection successful’ same file
Certificate/key Certificate/key
imported as ID imported as signing
certificate certificate
Other certificates Other certificates
imported into TCS ignored
with ‘N certificate(s)
imported into
Trusted Certificate
Store’
PKCS#12: Any No – attempt No – p12/pfx No – attempt
other content rejected with should not be rejected with
‘Invalid content offered for file ‘Invalid content
(PKCS#12)’ selection (PKCS#12)’
PEM: 1 Certificate No – attempt Yes – attempt No – attempt Certificate can be
rejected with accepted with ‘N rejected with encrypted on
‘Invalid certificate certificate(s) ‘Invalid certificate unencrypted
format (PEM – no imported into format (PEM – no
private key)’ Trusted Certificate private key)’
Store’
PEM
Table continues…

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 Certificate Management

File Content Identity Certificate Trusted Signing Notes

Import Command Certificate Import Certificate Import
Command Command
PEM: N Certificate No – attempt Yes – attempt No – attempt At least 20
rejected with accepted with ‘N rejected with certificates
‘Invalid certificate certificate(s) ‘Invalid certificate supported in the
format (PEM – no imported into format (PEM – no same file
private key)’ Trusted Certificate private key)’
Certificate can be
Store’
encrypted on
unencrypted
PEM: 1 Certificate Yes – attempt No – attempt Yes – attempt Certificate or Key
+ private key accepted with rejected with accepted with can be encrypted
‘Certificate import ‘Invalid certificate ‘Certificate import on unencrypted
successful’ format (PEM)’ successful’
Certificate/key Certificate/key
imported as ID imported as signing
certificate certificate
PEM: 1 Certificate Yes – attempt Yes – attempt Yes – attempt Private key must
+ private key, 1 or accepted with accepted with ‘N accepted with be before or after
more other ‘Certificate import certificate(s) ‘Certificate import the first certificate
certificates. successful’ imported into successful’
Certificate or Key
Trusted Certificate
Private key must Certificate/key Certificate/key can be encrypted
Store’
be before or after imported as ID imported as signing on unencrypted
the first certificate certificate. First certificate and certificate
private key ignored
Other certificates Other certificates
imported into TCS ignored
with ‘N certificate(s)
imported into
Trusted Certificate
Store’
PEM: Any other No – attempt No – attempt No – attempt Option to include
content rejected with rejected with rejected with more detail of the
‘Invalid content ‘Invalid content ‘Invalid content rejection cause.e.g.
(PEM)’ (PEM)’ (PEM)’ ‘Cannot detect
Identity Certificate’,
‘Too many private
keys’,
‘Unrecognized
header’ etc.

Related links
Certificate Support on page 571

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Configure General System Settings

On-boarding
On-boarding refers to the configuration of an SSL VPN service in order to enable remote
management services to customers, such as fault management, monitoring, and administration.
You must use the Web Manager client to configure on-boarding.
For full details on how to configure and administer SSL VPN services, refer to Deploying Avaya IP
Office™ Platform SSL VPN Services.
The procedure provided below configures IP Office for Avaya support services. Avaya partners
can also use an SSL VPN to provide support services. See the chapter “Configuring an Avaya
Partner SSL VPN using an SDK” in Deploying Avaya IP Office™ Platform SSL VPN Services.
Related links
Configuring an SSL VPN using an on-boarding file on page 580

Configuring an SSL VPN using an on-boarding file

The on-boarding XML file is available from Avaya. It contains the settings required to establish a
secure tunnel between IP Office and an AVG server. When you import the on-boarding XML file, it
applies the settings and installs one or multiple TLS certificates.
When you configure the SSL VPN service on a new system, you must begin by generating an
inventory file of the IP Office system. When you register your IP Office system, the inventory file
that you generated is uploaded to the GRT and the inventory data is populated in the Avaya
Customer Support (ACS) database. After you enable remote support, you can download the XML
on-boarding file from the GRT web site and upload it into your IP Office system.
The on-boarding process configures:
• SSL VPN service configuration
• short codes for enabling and disabling the SSL VPN service
• SNMP alarm traps
• one or more TLS certificates in the IP Office trusted certificate store
Perform this procedure using the Avaya IP Office Web Manager client.

Warning:
The process of 'on-boarding automatically creates an SSL VPN service in the system
configuration when the on-boarding file is uploaded to the system. Care should be taken not to
delete or modify such a service except when advised to by Avaya.
Before you begin
Before you begin, you must have the hardware codes and catalog description of your IP Office
system. For example, “IP OFFICE 500 VERSION 2 CONTROL UNIT TAA” is a hardware code
and catalog description.

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 System Date and Time

Procedure
1. Select Tools > On-boarding.
The On-boarding dialog box displays.
2. If the hardware code for your IP Office system ends with the letters TAA, select the
checkbox next to the prompt Are you using TAA series hardware?
3. Click Get Inventory File to generate an inventory of your IP Office system.
4. Click Register IP Office.
A browser opens and navigates to the GRT web site.
5. Log in to the web site and enter the required data for the IP Office system.
6. Select Remote Support for the IP Office system.
7. Click Download and save the on-boarding file.
8. Browse to the location where you saved the on-boarding file and click Upload.
A message displays to confirm that the on-boarding file has installed successfully.
Related links
On-boarding on page 580

System Date and Time

How Does the System Get The Date and Time
In most case, the system is configured during installation to automatically get the date and time
from a trusted time source.
• Linux based servers: The date and time source settings are set through the Settings |
System | Data and Time settings of the server’s Platform View menus. The possible date
and time options are:
- SNTP: Use the date and time provided by an SNTP time server. The UTC time provided
by the time server is adjusted using the server’s timezone setting. If you have a network of
servers, it is typical to set the primary server to use an external SNTP source and all other
servers are set to use SNTP from the primary server’s own address.
- Manual: Get the date and time from values entered through the platform view menu.
• IP500 V2 Servers: These systems are configured through their Time Setting Config
Source settings (System | System). The options supported are:
- Voicemail Pro/Manager: Get the date and time from the Windows PC running either the
Voicemail Pro or IP Office Manager applications. This option requires the application to be
running when the IP Office is started and for regular time updates.
- SNTP: Get the date and time from an SNTP time server in the same way as Linux based
systems above do.

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Configure General System Settings

- None: Get the date and time from values entered via a system phone. See Manually
Setting the System Time below.

Locations
In a network of systems, it may be necessary for some servers and or extensions, to have
different time and date settings in order to match where they are physically located. This can be
done by adding Location entries to the IP Office configuration <<<need link>>>. Each location
can include a time offset from the UTC time and a set of DST settings (see below) for the location.
Where necessary, system’s and extension’s can then be associated with those locations.

Automatic Daylight Saving Adjustments

It may be necessary to apply daylight saving time (DST) changes at certain times of the year. How
this is done depends on the type of IP Office server and which time source it has been configured:
• Linux-based Server: Daylight saving adjustments are applied to the SNTP time by defining
a Location for the system, see above. The location settings include the required DST
settings for that location.
• IP500 V2 Server: The method of applying DST depends on the time source being used by
the server. .
- Voicemail Pro/Manager: If the system is obtaining its time from a PC running Voicemail
Pro or IP Office Manager, then that PC needs to be configured to automatically do its own
- SNTP/None: The System | System menu includes settings for specifying when DST is
applied.
How Does the System Use the Date and Time
For files stored on memory cards the system uses the UTC time. For other activities such as call
logs, SMDR records, time display on phones; the local time (UTC + any offsets) is used.
Manually Setting the System Time
For IP500 V2 systems set to no time server source, date and time changes can be done though a
phone if the phone user has been give System Phone Rights (see System Phone Features on
page 715). The user's login code is used to restrict access to the time and date settings on the
phone. How the user accesses the date/time settings depends on the type of phone:
• 1400, 1600, 9500 and 9600 Series Phones: These phones (excluding the 1403/1603
models) can set the system time and date settings by selecting Features | Phone User |
System Administration. If the system has been configured with a time server, this option
can still used to display time and date information but cannot be used to change it.
• Other Phones: On 2410, 2420, 4412, 4424, 4612, 4624, 4610, 4620, 4621, 5410, 5420,
5610, 5620, 5621, 6412 and 6424 phone, a Self Admin 2 button can be used to set the date
and time. See Self-Administer on page 1025.

Configuring Time Profiles

Time profiles are configured on Time Profile | Time Profile

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 Configuring Time Profiles

Time Profiles are used by different services to change their operation when required. In most
areas where time profiles can be used, not setting a time profile is taken as meaning 24-hour
operation.
Time profiles consist of recurring weekly patterns of days and times when the time profile is in
effect.
Time profiles can include time periods on specified calendar days when the time profile is in effect.
Calendar records can be entered for the current and following calendar year.
For a Server Edition network, these settings can be configured at the network level and are then
automatically replicated in the configuration of all systems in the network. They can only be seen
and edited at the individual system configuration level if record consolidation is switched off.
Time profiles are used by the following record types.
Hunt Group:
A time profile can be used to determine when a hunt group is put into night service mode. Calls
then go to an alternate Night Service Fallback group if set, otherwise to voicemail if available or
busy tone if not.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
For automatic voice recording, a time profile can be used to set when voice recording is used.
User:
• Users being used for Dial In data services such as RAS can have an associated time profile
that defines when they can be used for that service.
• Users can be associated with a working hours and an out of hours user rights. A time profile
can then be used to determine which user rights is used at any moment.
• For automatic voice recording, a time profile can be used to set when that voice recording is
used.
• For mobile twinning, a time profile can be used to define when twinning should be used.
Incoming Call Route:
Incoming call routes can also use time profiles to specify when calls should be recorded. Multiple
time profiles can be associate with an incoming call route, each profile specifying a destination
and fall back destination.
ARS:
ARS forms use time profile to determine when the ARS form should be used or calls rerouted to
an out of hours route.
Account Code:
Account Codes can use automatic voice recording triggered by calls with particular account codes.
A time profile can be used to set when this function is used.
Auto Attendant :
Embedded voicemail auto attendants can use time profiles to control the different greetings played
to callers.

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Configure General System Settings

Service:
• A Service can use time profiles in the following ways:
• A time profile can be used to set when a data service is available. Outside its time profile, the
service is either not available or uses an alternate fallback service if set.
• For services using auto connect, a time profile can be used to set when that function is used.
See Service | Autoconnect.
Related links
Overriding a Time Profile on page 584

Overriding a Time Profile

You can use the Time Profile | Manual Override setting to manually override a time profile. The
override settings allow you to mix timed and manual settings.
The override options are as follows:
• Set Time Profile Active Until Next Timed Inactive
Use for time profiles with multiple intervals. Make the time profile active until the next inactive
interval.
• Set Time Profile Inactive Until Next Timed Active
Use for time profiles with multiple intervals. Make the time profile inactive until the next active
interval.
• Set Time Profile Latch Active
Set the time profile to active. Timed inactive periods are overridden and remain active.
• Set Time Profile Latch Inactive
Set the time profile to inactive. Timed active periods are overridden and remain active.
The illustration below provides an example of each override setting.

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 Working with Templates

Active
Inactive
7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Active until next Inactive

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Inactive untill next Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Inactive

A time profile can be overridden using the following methods.

• Using the Override settings on the Time Profile configuration page.
• Configure short codes for the time profile. See the description for the “Set Time Profile” short
code.
• Configure the Time Profile button action for the time profile. See the description for the “Time
Profile” button action.
Related links
Configuring Time Profiles on page 582

Working with Templates

IP Office supports a number of template options. The settings for the following types of
configuration items can be saved as template files. New records of those types can then be
created from the template file.
• User (.usr)
• Extension (H.323, SIP, IP DECT) (.ext)

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Configure General System Settings

• Group (.grp)
• Service (.ser)
• Tunnel (.tnlt)
• Firewall Profile (.fpr)
• Time Profile (.tpr)
• IP Route (.ipr)
• ARS (.ars)
• Line (H.323, SIP, IP DECT) (.lne)
Saving Template files
Standard Mode Systems: Standard mode systems export templates to a local folder on the PC
where Manager is running. Templates are stored in the default folder C:\Program Files
(x86)\Avaya\IP Office\Manager\manager_files\template.
Server Edition Systems: Server Edition system templates are stored on the Primary Server.
When the system configuration is opened, those templates are downloaded from server to the
default folder above. When the configuration is saved, the templates are uploaded back to server.
Caution:
Due to the difference in operation detailed above, any non-Server Edition templates stored in
the default folder, are liable to be overwritten when a Server Edition configuration is loaded.
Therefore, if you administer both Server Edition and non-Server Edition system, you need to
ensure that you store the non-Server Edition templates in a directory other than the default
directory.
For Server Edition, if you are working with an offline configuration, any templates created are
deleted after you close Manager.
Tested SIP Trunk Templates
The SIP trunk services from selected SIP providers are tested as part of the Avaya DevConnect
program. The results of such testing are published as Avaya Application Notes available from the
Avaya DevConnect web site (https://devconnect.avaya.com).
Related links
Importing Trunk Templates on page 586
Creating a Template in Manager on page 587
Creating a New Record from a Template in Manager on page 587
Creating an Analog Trunk Template in Manager on page 588
Creating a New Analog Trunk from a Template in Manager on page 588
Applying a Template to an Analog Trunk on page 589

Importing Trunk Templates

For Standard Mode, before you can use templates from another source, they must be placed in
the Manager \Templates directory. Use this procedure to import a template from another source.

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 Working with Templates

Procedure
1. Select Tools | Import Templates in Manager.
2. Browse to the current folder containing the templates that you want to import and select
that folder.
3. Click OK.
4. Any template files in the folder will be copied to the correct Manager sub-folder.
Related links
Working with Templates on page 585

Creating a Template in Manager

You can create a template from an existing record.
The options New From Template and Export as Template are available by:
• right clicking on the record type in the Navigation pane
• right clicking on a record in the Group pane
• using the Details Toolbar in the Details pane
This procedure uses the Group Pane.
Procedure
1. In the Navigation pane, select a record type.
2. In the Group pane, right click on the record on which you want to base your template and
select Export as Template.
3. The Save As window opens at the default template folder. Enter a name for the template.
A default extension is applied. For example, user templates are saved with the file
extension .usr and extension templates are saved with file extension .ext.
4. Click Save.
You can now create new records using the template.
Related links
Working with Templates on page 585

Creating a New Record from a Template in Manager

You can use a template to create additional records.
The options New from Template and Export as Template are available by:
• right clicking on the record type in the Navigation pane

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Configure General System Settings

• right clicking on a record in the Group pane

• using the Details Toolbar in the Details pane
This procedure uses the Group Pane.
Procedure
1. In the Navigation pane, select a record type.
2. In the Group pane, right click on the record on which you want to base your template and
select New from Template > Open from file.
Manager lists available templates stored in the default directory.
3. In the Open window, select the template file and click Open.
4. For some record types, the Import Templates window opens.
a. Enter the number of records to create in the Number of records to import field.
b. In the Start with index field, enter starting index number for the new records.
c. Click OK.
Related links
Working with Templates on page 585

Creating an Analog Trunk Template in Manager

You can create an analog trunk template from an existing trunk..
Procedure
1. In the Navigation pane, select Line.
2. In the Group pane, right click on the record on which you want to base your template and
select Generate Analog Trunk Template.
3. In the Analog Trunk Template window, you can adjust the settings if required. Click Export.
4. In the Template Type Selection window, select the Service Provider and then click Create
Template.
5. In the Browse for Folder window, select Program Files\Avaya\IP Office\Manager
\manager_files\template.
6. Click OK.
Related links
Working with Templates on page 585

Creating a New Analog Trunk from a Template in Manager

You can create a new analog trunk from a template.

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 Centralized System Directory

Procedure
1. In the Navigation pane, right click Line and select New from Template > Open.
2. In the Open window, select a template and click Open.
3. In the Template Type Selection window, select the Service Provider and then click
Create.
Related links
Working with Templates on page 585

Applying a Template to an Analog Trunk

You can apply an analogue trunk template to existing analogue trunks.
Note:
You must reboot the system for any changes to be applied.
Procedure
1. In the group pane, right click on the analogue trunk and select Copy Setting from
Template.
2. The template and trunk selection menu is displayed.
3. In the Template Type Selection window, use the Service Provider drop-down to select the
required template.
4. Select the trunks to which you want the template to be applied.
5. Click on Copy Settings.
Related links
Working with Templates on page 585

Centralized System Directory

Directory services can be used to import directory records (names and numbers) from external
sources. These sets of records are regularly re-imported.
Directory records can come from the following sources:
• LDAP Import: The system can import LDAP records for use within directories shown by user
phones and applications. LDAP import is configured through the System | Directory
Services | LDAP form. The LDAP used is LDAP Version 2.
• HTTP Import : Systems are able to import the directory records from another system using
HTTP. HTTP import is configured through the System | Directory Services | HTTP form by
specifying an IP address or multi-site network connection. The records imported can be any

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Configure General System Settings

or all of the following record types held by the system from which the records are being
imported: LDAP imported records, HTTP imported records, configuration records.
• System Directory Records (Configuration records): Records can be entered directly into
the system configuration through the Directory | Directory Entry form. System directory
records override matching LDAP/HTTP imported records.
Users with system phone rights (see System Phone Features on page 715) and a phones
with a CONTACTS button can add, delete and edit the system directory records of the
system on which they are logged in. They cannot edit LDAP or HTTP imported records.

Server Edition Directory Operation

For a Server Edition network, these settings can only be configured at the network level and they
are stored in the configuration of the Primary Server. All other systems in the network are
configured to share the directory settings of the Primary Server through the settings at System |
Directory Services | HTTP.
Directory Record Capacity
The directory capacity depends on the type of system. The figures below are applicable for
Release 10.0.
System Number of Directory Records Total Number
Configuration LDAP Import HTTP Import of Directory
Records
Standalone IP500 V2 2,500 10,000 10,000 10,000
Systems
Server Edition Primary Server 10,000 10,000 10,000 10,000
Secondary – – 10,000 10,000
Server
Table continues…

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 Centralized System Directory

System Number of Directory Records Total Number

Configuration LDAP Import HTTP Import of Directory
Records
Expansion – – 10,000 10,000
System (L)
Expansion – – 10,000 10,000
System (V2)

Directory Dialing
Directory numbers and names are displayed by user applications such as SoftConsole. The
method by which these directories are searched and used depends on the application. Refer to
the appropriate user guide.
Directory entries used for dialing can contain () and — characters in the number. Those characters
are ignored in the dialled output. Directory entries containing ? in the number (used for directory
name matching) are not included in the directory for dialing.
Directory names are also viewable through the Dir or Contacts function on many Avaya phones.
They allow the user to select the name in order to dial its associated number.
The directory function groups directory records shown to the phone user into several categories,
for example; system, personal, users and groups. Depending on the phone or application, the user
may be able to select the category currently displayed. In some scenarios, the categories
displayed may be limited to those supported for the action being performed by the user. The
typical categories are:
• External: Directory records from the system configuration. This includes HTTP and LDAP
imported records.
• Groups: Groups on the system. If the system is in a multi-site network, it will also include
groups on other systems in the network.
• Users or Index: Users on the system. If the system is in a multi-site network it will also
include users on other systems in the network.
• Personal: Available on 1400, 1600, 9500 and 9600 Series phones. These are the user's
personal directory records stored within the system configuration.
On phones that support Dir or Contacts, the user can filter the currently displayed set of directory
names by dialing on their keypad. Additional dialing applies a progressive filter. For example, if the
user presses the 5 key (JKL), only names with some part beginning with J, K or L remain listed. If
the user then presses the 2 key (ABC), only names with some part beginning with JA, JB, JC, KA,
etc. remain listed. As the users presses more keys on their phone, the number of remaining
matches reduces.
By default the letter matching is performed simultaneously against all parts of the directory name,
ie. first, middle and last name. However, this behavior can be modified for all users using a
NoUser source number.
Speed Dialing
On M-Series and T-Series phones, a Speed Dial button or dialing Feature 0 can be used to
access personal directory records using the record’s index number.
• Personal: Dial Feature 0 followed by * and the 2-digit index number in the range 01 to 99.

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Configure General System Settings

• System: Dial Feature 0 followed by 3-digit index number in the range 001 to 999.
• The Speed Dial short code feature can also be used to access a directory speed dial using
its index number from any type of phone.
Caller Directory Name Matching
Directory records are also used to associate a name with the dialled number on outgoing calls or
the received CLI on incoming calls. When name matching is being done, a match in the user's
personal directory overrides any match in the system directory. Note that some user applications
also have their own user directory.
• The ( ) and — characters are not used for directory name matching. Directory entries with
those characters are ignored for name matching.
• A ? character can be used to match any digit or digits. For example 91?3 will match 9123.
Typically a single ? is used at the end of a know dialing string such as an area code.
• The best match is used, determined by the highest number of matched digits.
• There is no minimum number of matches. For example, a directory entry of 9/External can be
used to match any external call unless it has a better match.
Other Name Sources:
• SoftConsole has its own directories which are also used for name matching. Matches in the
application directory can lead to the application displaying a different name from that shown
on the phone.
• Name matching is not performed when a name is supplied with the incoming call, for
example QSIG trunks. On SIP trunks the use of the name matching or the name supplied by
the trunk can be selected using the Default Name Priority setting (System | Telephony |
Telephony). This setting can also be adjusted on individual SIP lines to override the system
setting.
• Directory name matching is not supported for DECT handsets. For information on directory
integration, see IP Office DECT R4 Installation.
Imported Records
Imported directory records are temporary until the next import refresh. They are not added to the
system's configuration. They cannot be viewed or edited using Manager or edited by a system
phone user. The temporary records are lost if the system is restarted. However the system will
request a new set of imported directory records after a system restart. The temporary records are
lost if a configuration containing Directory changes is merged. The system will then import a new
set of temporary records without waiting for the Resync Interval. If an configuration record is
edited by a system phone user (see System Phone Features on page 715) to match the name or
number of a temporary record, the matching temporary record is discarded.
Importation Rules:
When a set of directory records is imported by HTTP or LDAP, the following rules are applied to
the new records:
• Imported records with a blank name or number are discarded.
• Imported records that match the name or number of any existing record are discarded.
• When the total number of directory records has reached the system limit, any further
imported records are discarded.

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 Advice of Charge

For capacity information, see the description for the Directory tab.

Advice of Charge
The system supports advice of charge (AOC) on outgoing calls to ISDN exchanges that provide
AOC information. It supports AOC during a call (AOC-D) and at the end of a call (AOC-E). This
information is included in the SMDR output.
AOC is only supported on outgoing ISDN exchange calls. It is not supported on incoming calls,
reverse charge calls, QSIG and non-ISDN calls. Provision of AOC signalling will need to be
requested from the ISDN service provider and a charge may be made for this service.
For users, display of AOC information is only supported on T3 phones and T3 IP phones.
The user who makes an outgoing call is assigned its charges whilst they are connected to the call,
have the call on hold or have the call parked.
If AOC-D is not available, then all indicated by AOC-E are assigned to the user who dialed the
call.
If AOC-D is available:
• If the call is transferred (using transfer, unpark or any other method) to another user, any call
charges from the time of transfer are assigned to the new user.
• If the call is manually transferred off-switch, the call charges remain assigned to the user who
transferred the call.
• If the call is automatically forwarded off switch, subsequent call charges are assigned to the
forwarding user.
• AOC-D information will only be shown whilst the call is connected. It will not be shown when
a call is parked or held.
• Call charges are updated every 5 seconds.
For conference calls all call charges for any outgoing calls that are included in the conference are
assigned to the user who setup the conference, even if that user has subsequently left the
conference.
Enabling AOC Operation
1. Set the System Currency The Default Currency (System | Telephony | Telephony) setting
is by default set to match the system locale. Note that changing the currency clears all call
costs stored by the system except those already logged through SMDR.
2. Set the Call Cost per Charge Unit for the Line AOC can be indicated by the ISDN
exchange in charge units rather than actual cost. The cost per unit is determined by the
system using the Call Cost per Charge Unit setting which needs to be set for each line.
The values are 1/10,000th of a currency unit. For example if the call cost per unit is £1.07,
a value of 10700 should be set on the line.

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Configure General System Settings

3. Applying a Call Cost Markup It may be a requirement that the cost applied to a user's
calls has a mark-up (multiplier) applied to it. This can be done using the Call Cost Markup
(User | Telephony | Call Settings) setting. The field is in units of 1/100th, for example an
entry of 100 is a markup factor of 1.
4. Enable User AOC Display By default users do not see call charges. The Display
Charges setting is used to switch this option on or off. Note that the display of AOC
information is only supported on T3 phones.
AOC Short Codes
A number of short code features exist that can be used with AOC. These features can only be
used with T3 phones.
AOC Previous Call Displays the call costs of the user's previous call if AOC information was
provided with that call.
AOC Total Display the cumulative total cost of the user's calls for which AOC information is
available.
AOC Reset Total Set the cumulative total (units and cost) for the user's calls back to zero.

Emergency Call
Manager expects that the configuration of each system should contain at least one short code that
is set to use the Dial Emergency feature. If no such short code is present in the configuration
then Manager will display an error warning. The importance of the Dial Emergency feature is that
it overrides all external call barring that may have been applied to the user whose dialing has been
matched to the short code. You must still ensure that no other short code or extension match
occurs that would prevent the dialing of an emergency number being matched to the short code.
The short code (or codes) can be added as a system short code or as an ARS record short code.
If the Dial Emergency short code is added at the solution level, that short code is automatically
replicated into the configuration of all servers in the network and must be suitable for dialing by
users on all systems. Separate Dial Emergency short codes can be added to the configuration of
an individual system. Those short codes will only be useable by users currently hosted on the
system including users who have hot desked onto an extension supported by the system.
Determining the Caller's Location
It is the installers responsibility to ensure that a Dial Emergency short code or codes are useable
by all users. It is also their responsibility to ensure that either:
the trunks via which the resulting call may be routed are matched to the physical location to which
emergency service will be despatched
or
the outgoing calling line ID number sent with the call matches the physical location from which the
user is dialing.

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 Fax Support

Hot Desking Users

In addition to the location requirements above, you must also remember that for users who hot
desk, from the networks perspective the user's location is that of the system hosting the extension
onto which the user is currently hot desked. If that is an IP extension then that location is not
necessarily the same as the physical location of the server.
Emergency call setup
Routing of emergency calls is based on a call resolving to a Dial Emergency short code. Based on
the location value for the extension making the call, routing is performed as configured in the
Emergency ARS. Emergency calls have maximum priority and are not delayed in any way.
Configuring emergency call routing
Create a Dial Emergency system short code. See Dial Emergency.
Note that the Line Group ID value in the Dial Emergency short code is the fallback route. If the
system cannot find a location or an Emergency ARS, it will try to use the Line Group ID to route
the call.
1. Create an ARS containing a Dial short code or a Dial Emergency short code. See ARS.
2. Create a Location and set the Emergency ARS to the ARS created in step 2. See
Location.
3. Open the Extn tab for an extension that will use the location defined in step 3 and set the
Location value to the location defined in step 3. Note that once you define a location, you
must set a system Location value on the System | System page.
For non-IP based extensions, the system location value is used as the default. For IP based
extensions, the location value is set to Automatic. An attempt is made to match the extension's IP
address to the subnet configured in the location. If the match is cannot be made, the location
value defaults to the system location value.
From the extension used in step 3, dial the Dial Emergency short code. IP Office checks the
location value and determines the emergency ARS set for the location. Once the emergency ARS
is found, IP Office will try to match the Telephone Number in the Dial Emergency short code to a
short code in the ARS and use it to make the emergency call.

Fax Support
Fax on IP500 V2 Systems
IP500 V2 systems can terminate T38 fax calls. For a system with an IP500 VCM, IP500 VCM V2
or IP500 Combo cards, T38 or G.711 can be used for fax transmission. Each fax call uses a VCM
channel unless it is a T38 fax call between compatibly configured call legs. A SIP line or extension
must support Re-Invite.
T38 Fallback can also be specified. On outgoing fax calls, if the called destination does not
support T38, a re-invite is sent for fax transport using G.711.
Configuring Fax on SIP Lines and Extensions:

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Configure General System Settings

To configure Fax on SIP Lines and Extensions:

1. On the VoIP page for the line or extension, set Re-Invite Supported to On in order to
enable Fax Transport Support
2. Select a value in the Fax Transport Support field.
Note the following:
• Direct media is supported.
• If Fax Transport Support is set to T38 or T38 Fallback, the T38 Fax page is available. The
T38 Fax page provides detailed T38 configuration options.
Configuring Fax on an IP Office Line:
Within a multi-site network, Fax Transport Support can also be enabled on the IP Office Lines
between the systems. This allows fax calls at one system to be sent to another system.
To configure Fax on an IP Office Line:
1. Set IP Office Line | Line Settings | Networking Level to SCN.
2. Set IP Office Line | VoIP | Fax Transport Support to Fax Relay.
Related links
Server Edition T38 Fax Support on page 597

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 Fax Support

Server Edition T38 Fax Support

PSTN
SIP
Provider

IP Office Line
Primary IP500 V2
Server Expansion

SIP Fax Analog

ATA Fax

Fax on Server Edition Linux Servers

IP Office Linux servers cannot terminate T38 fax and therefore, T38 is negotiated end-to-end.
When a SIP ATA fax is connected to an IP Office Linux server, the system directly relays
negotiation between the SIP ATA Fax and the SIP provider.
Configuring Fax on SIP Lines and Extensions:
To configure Fax on SIP Lines and Extensions, on the VoIP page for the SIP line or extension:
1. Set Re-Invite Supported to On in order to enable Fax Transport Support.
2. Select a value in the Fax Transport Support field.
Note the following.
• Direct media is supported.
• The T38 Fax page is not available.
Fax on Server Edition IP500 V2 Expansion Systems
Since an IP500 V2 system can terminate T38 fax, an analog fax can be connected to an IP500 V2
Expansion system. Fax transport is configured on the IP Office Line connecting the IP500 V2
system to the Server Edition network.

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Configuring Fax on an IP Office Line:

To configure Fax on an IP Office Line, on the Line | IP Office Line | VoIP Settings page, select a
value in the Fax Transport Support field. Fax Relay is not supported.
Note the following.
• Direct media is supported.
• The T38 Fax page is not available.
Related links
Fax Support on page 595

Caller Display
Caller display displays details about the caller and the number that they called. On internal calls,
the system provides this information. On external calls it uses the Incoming Caller Line
Identification (ICLID) received with the call. The number is also passed to system applications and
can be used for features such as call logging, missed calls and to make return calls.
Analog extension can be configured for caller display via the system configuration (Extension |
Extn | Caller Display Type).
Adding the Dialing Prefix Some systems are configured to require a dialing prefix in front of
external numbers when making outgoing calls. When this is the case, the same prefix must be
added to the ICLID received to ensure that it can be used for return calls. The prefix to add is
specified through the Prefix field of each line.
Directory Name Matching The system configuration contains a directory of names and numbers.
If the ICLID of an incoming call matches a number in the directory, the directory name is
associated with that call and displayed on suitable receiving phones.
Applications such as SoftConsole also have directories that can be used for name matching. If a
match occurs, it overrides the system directory name match for the name shown by that
application.
Extended Length Name Display
In some locales, it may be desirable to change the way names are displayed on phones in order to
maximize the space available for the called or calling name. There are two hidden controls which
can be used to alter the way the system displays calling and called information.
These controls are activated by entering special strings on the Source Numbers tab of the NoUser
user. These strings are:
LONGER_NAMES This setting has the following effects:
• On DS phones, the call status display is moved to allow the called/calling name to occupy the
complete upper line and if necessary wrap-over to the second line.
• For all phone types:

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 Parking Calls

• On incoming calls, only the calling name is displayed. This applies even to calls forwarded
from another user.
• On outgoing calls, only the called name is displayed.
HIDE_CALL_STATE This settings hides the display of the call state, for example CONN when a
call is connected. This option is typically used in conjunction with LONGER_NAMES above to
provide additional space for name display.

Parking Calls
Parking a call is an alternative to holding a call. A call parked on the system can be retrieved by
any other user if they know the system park slot number used to park the call. When the call is
retrieved, the action is known as Unpark Call or Ride Call. While parked, the caller hears music on
hold if available.
Each parked call requires a park slot number. Attempting to park a call into a park slot that is
already occupied causes an intercept tone to be played. Most park functions can be used either
with or without a specified park slot number. When parking a call without specifying the park slot
number, the system automatically assigns a number based on the extension number of the person
parking the call plus an extra digit 0 to 9. For example if 220 parks a call, it is assigned the park
slot number 2200, if they park another call while the first is still parked, the next parked call is
given the park slot number 2201 and so on.
Park slot IDs can be up to 9 digits in length. Names can also be used for application park slots.
The Park Timeout setting in the system configuration (System | Telephony | Telephony | Park
Timeout) controls how long a call can be left parked before it recalls to the user that parked it. The
default time out is 5 minutes. Note that the recall only occurs if the user is idle has no other
connected call.
There are several different methods by which calls can be parked and unparked. These are:
Using Short Codes
The short code features, Call Park and Unpark Call, can be used to create short codes to park and
unpark calls respectively. The default short codes that use these features are:
• *37*N# - Parks a call in park slot number N.
• *38*N# - Unparks the call in park slot number N.
Using the SoftConsole Application
The SoftConsole application supports park buttons. SoftConsole provides 16 park slot buttons
numbered 1 to 16 by default.
The park slot number for each button can be changed if required. Clicking on the buttons allows
the user to park or unpark calls in the park slot associated with each button. In addition, when a
call is parked in one of those slots by another user, the application user can see details of the call
and can unpark it at their extension.

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Configure General System Settings

Using Programmable Buttons

The Call Park feature can be used to park and unpark calls. If configured with a specified park slot
number, the button can be used to park a call in that slot, unpark a call from that slot and will
indicate when another user has parked a call in that slot. If configured without a number, it can be
used to park up to 10 calls and to unpark any of those calls.
Phone Defaults
Some telephones support facilities to park and unpark calls through their display menu options
(refer to the appropriate telephone user guide). In this case parked calls are automatically put into
park slots matching the extension number.

Configuring Call Admission Control

Call Admission Control (CAC) is a method of controlling system resources using defined locations.
Calls into and out of each location are allowed or disallowed based upon configured call
constraints. In Manager, use the Location tab to define a location and configure the maximum
calls allowed for the location.
Related links
Manager location tab on page 600
Assigning a network entity to a location on page 601
System actions at maximum call threshold on page 601
Example on page 602

Manager location tab

Configuring location settings
On the Manager Location tab, set the following parameters for a location:
• Location Name
• Subnet Address
• Subnet Mask
Configuring Call Admission Control settings
On the Manager Location tab, set the following CAC parameters:
• Internal Maximum Calls: Calls that pass from the location to another configured location.
• External Maximum Calls: Calls that pass from the location to an unmanaged location.
• Total Maximum Calls: The total internal and external calls permitted.
Related links
Configuring Call Admission Control on page 600

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 Configuring Call Admission Control

Assigning a network entity to a location

The Location field is a drop down list of locations defined on the Location tab. Network entities
are assigned to a location using the Location field on the following Manager tabs.
• System
• Extension
• SIP Line | VoIP
• H323 Line | VoIP
The following default settings are applied.
• Each IP Office system can be configured with a defined location. For Server Edition
deployments, the configuration of locations is done solution wide. All IP Office systems in the
solution share the same location configuration.
• Digital phones default to the system location.
• The default setting for IP phones is Automatic. Phones registering from a subnet matching
that of a location will be treated as within that location. Otherwise, the phone is assigned the
same location as the system. Cloud can be used for phones whose Location is variable or
unknown.
• IP Lines default to Cloud.
Related links
Configuring Call Admission Control on page 600

System actions at maximum call threshold

• A congestion alarm is raised.
• Calls that exceed the CAC maximum values are not allowed.
• Calls from extensions to public trunks through Alternate Route Selection (ARS) are queued
and display Waiting for Line.
• Calls from extensions to public trunks which do not route through ARS receive a fast-busy
tone and display Congestion.
• Idle phones display Emergency/Local calls only.
• Alternative routing to a local PSTN gateway follows ARS priority escalation rules.
• SIP calls that would exceed call limits and have no other targets are declined with
cause=486 or cause = 503.
Allowed calls
When CAC limits have been reached, the following calls are allowed.
• Emergency calls are always allowed.

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Configure General System Settings

• Established calls are never torn down to achieve limits.

• A phone on a remote site that parks a call is always allowed to retrieve it.
• Request Coaching Intrusion calls are allowed.
Related links
Configuring Call Admission Control on page 600

Example
The example configuration has four locations.
Location Max Calls
HQ 20
RS1 5
RS2 10
RS3 15
+Cloud unlimited

SIP Line 10 and SIP Line 11 are configured with 20 channels.

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 Configuring Call Admission Control

Notes
• Calls between locaton RS1 and SBC113 do not increment the call count for HQ.
• The HQ call count includes calls across the HQ boundary which anchor media inside HQ.
SBC113 and SBC 114 are both included.
• The HQ maximum calls value is separate and complementary to the individual trunk call
maximum.
• Incoming calls from SIP to RS1 (direct media) only need to verify that the RS1 location
maximum call value is not exceeded.
• SIP calls that are not allowed to RS1 may go to HQ voicemail if the HQ call limit is not
exceeded.
Related links
Configuring Call Admission Control on page 600

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Configure General System Settings

Ring Tones
Ring tones can be defined in the following terms.
Distinctive Ringing - Inside, Outside and Ringback:
A distinctive ring tone can be given for each of the different call types: an internal call, an external
call and a ringback calls (voicemail calls, ringback when free calls, calls returning from park, hold
or transfer).
The distinctive ringing patterns used for most non-analog phones are as follows:
• Internal Call: Repeated single-ring.
• External Call: Repeated double-ring.
• Ringback Call: Two short rings followed by a single ring.
Note:
For non-analog extensions, the ringing pattern used for each call type by the system is not
configurable.

Personalized Ringing:
This term refers to control of the ringing sound through the individual phones. For non-analog
phones, while the distinctive ringing patterns cannot be changed, the ringer sound and tone may
be personalized depending on the phone's own options. Refer to the appropriate telephone user
guide.
Analog Phone Ringing Patterns

For analog extensions, the ringing pattern used for each call type can be set using the settings on
System | Telephony | Telephony. The setting for an individual user associated with an analog
extension can be configured using the settings on User | Telephony | Call Settings.
Note that changing the pattern for users associated with fax and modem device extensions may
cause those devices to not recognize and answer calls.
The selectable ringing patterns are:
• RingNormal This pattern varies to match the Locale set in the System | System tab. This is
the default for external calls.
• RingType1: 1s ring, 2s off, etc. This is the default for internal calls.
• RingType2: 0.25s ring, 0.25s off, 0.25s ring, 0.25s off, 0.25s ring, 1.75s off, etc. This is the
default for ringback calls.
• RingType3: 0.4s ring, 0.8s off, …
• RingType4: 2s ring, 4s off, …
• RingType5: 2s ring, 2s off, …
• RingType6: 0.945s ring, 4.5s off, …
• RingType7: 0.25s ring, 0.24 off, 0.25 ring, 2.25 off, …
• RingType8: 1s ring, 3s off, …

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 Music On Hold

• RingType9: 1s ring, 4s off, …

• RingType0: Same as RingNormal for the United Kingdom locale.
• Default Ring: Shown on the User | Telephony | Call Setting tab. Indicates follow the settings
on the System | Telephony | Tones & Music tab.
Configuring Ring Tone Override for Groups and Incoming Call Routes
You can configure ring tone override for groups and incoming call routes. Ring Tone Override is
supported on 1400 and 9500 series phones.
Note that you can use short codes to configure a ring tone plan by using the “r” character as part
of the short code telephone number field. See Short Code Characters on page 790.
1. In Manager, select System | Telephony | Ring Tones.
2. In the Ring Tone Plan table, enter a Name for the ring tone. The Number field is
populated automatically.
3. Under Ring Tone, select one of the eight ring tones from the drop down list.
4. Once configured in this table, ring tone names can be selected from the Ring Tone
Override field at:
• Group | Group
• Incoming Call Route | Standard

Music On Hold
Each system can provide music on hold (MOH) from either internally stored files or from externally
connected audio inputs. Each system has one system source and then a number of alternate
sources (up to 3 alternate sources on IP500 V2 and 31 alternate sources on Server Edition).
You must ensure that any MOH source you use complies with copyright, performing rights and
other local and national legal requirements.
WAV Files
The system can use internal files that it stores in its non permanent memory. The WAV file
properties must be in the format listed below. If the file downloaded is in the incorrect format, it will
be discarded from memory after the download.
• Mono PCM 8kHz 16-bit
• maximum length 90 seconds on IP500 V2 systems, 600 seconds on Linux based systems.
• The first WAV file, for the system source, must be called HoldMusic.wav. Alternate source
WAV file names:
- can be up to 27 IA5 characters
- cannot contain spaces
- any extension is allowed
- case sensitive

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The files, when specified by the system source or an alternate source setting, are loaded as
follows:
• Following a reboot, the system will try using TFTP to download the file or files.
• The initial source for TFTP download is the system's configured TFTP Server IP Address
(System | System | LAN Settings). The default for this is a broadcast to the local subnet for
any TFTP server.
• Manager can act as a TFTP server while it is running. If Manager is used as the TFTP server,
then the wav file or files should be placed in the Manager applications working directory.
Note:
The following Manager settings are disabled by default:
- Security Settings | Unsecured Interfaces | Applications Controls | TFTP
Directory Read
- File | Preferences | Preferences | Enable BootP and TFTP Servers
• On Linux based systems, if no successful TFTP download occurs, the system automatically
looks for the files in the opt/ipoffice/tones/mohwavdir folder (disk/tones/
mohwavdir when access using file manager).
• The name of the system music .wav file should be HoldMusic.wav. The name of alternate
source .wav files should be as specified in the Alternate Sources table (System |
Telephony | Tones and Music) minus the WAV: prefix.
WAV File Download and Storage:
• If no successful TFTP download occurs:
- On IP500 V2 systems, the system automatically looks for the file in the system/primary
folder on the System SD card and downloads it from if found.
- On Linux based systems, the system automatically looks for the file in the folder opt/
ipoffice/system/primary folder (disk/system/primary when accessed using file
manager) and downloads it from there if found.
• If a music on hold file is downloaded, the system automatically write a copy of that file to its
memory card, overwriting any existing file of the same name already stored on the card.
• For files downloaded from a System SD card, the system will download the file again if the
SD card is shutdown and restarted or if files are uploaded to the card using the Embedded
File Manager.
• The system will download the file again if new files are copied to the disk or uploaded using
File Manager.
Tone
If no internal music on hold file is available and External is not selected as the System Source,
then the system provides a default tone for music on hold. The tone used is double beep tone
(425Hz repeated (0.2/0.2/0.2/3.4) seconds on/off cadence). Tone can be selected as the System
Source, overriding both the use of the external source port and the downloading of
HoldMusic.wav.

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 Music On Hold

Controlling the Music on Hold Source Used for Calls

Unless specified, the System Source is used for any calls put on hold by system users. For any
call, the last source specified for the call is the one used. The following options allow the source to
be changed.
• Hunt Group Each hunt group can specify a Hold Music Source (Group | Group). That
source is then used for calls presented to the hunt group.
In a multi system network, a hunt group member will hear the music on hold (MOH) from their
local system. For example, a call comes in to site A and rings a hunt group with members
from system A and system B. If a hunt group member from system B answers a call and puts
the call on hold, the caller hears the MOH from system B.
• Incoming Call Route Each incoming call route can specify a Hold Music Source (Incoming
Call Route | Standard). That source is then used for incoming calls routed by that incoming
call route.
• Short Code The h character can be used in the Telephone Number field of short codes to
specify the hold music to associate with calls routed by that short code. The format h( X ) is
used where X is the source number. This method can be used to specify a hold music source
for outgoing calls.
Checking Music on Hold
The system short code feature Hold Music can be used to listen to the hold music sources. Dial
*34N#, replacing N with the source number 1 (System Source) or 2 to 32 (Alternate Sources).
Related links
System Source on page 607
Alternate Source on page 607

System Source
The first source is called the System Source. This source is numbered source 1. The possible
options for this source are:
• WAV: A file called HoldMusic.wav downloaded by the system.
• External: For IP500 V2 systems, use the audio source connected to the back of the control
unit. For Linux systems, the first available USB source is used.
• Tone: A double beep tone. Used automatically if the System Source is set to WAV and the
HoldMusic.wav file has not been successfully downloaded.
Related links
Music On Hold on page 605

Alternate Source
You can specify alternate sources on the System | Telephony | Tones and Music page. The
available options depends on the system type. For IP500 V2 systems, up to 3 alternate sources

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Configure General System Settings

can be specified. For systems on a Linux based server, up to 31 alternate sources can be
specified. See the table below for details.
Alternate Option Description
WAV:<filename> • The <filename> parameter specifies the filename to be played.
• <filename>:
- can be up to 27 IA5 characters
- cannot contain spaces
- any extension is allowed
- case sensitive
• A TFTP read is attempted first, then the file location opt/ipoffice/
system/primary (Linux) or /system/primary (IP500 V2).
• When a MOH source is activated, the playback resumes from where it
left off last time, instead of starting every time from the beginning.
• At any moment, all users listening to a given MOH source will hear the
same thing (instead of every user hearing from a different file position).
• For Linux systems, this source is suitable for use with the LINE option.
XTN: <extension> Only supported on IP500 V2 systems. Any analog extension with its
Equipment Classification set as MOH Source can be entered as the
alternate source. Enter XTN: followed by the extension's Base Extension
number. For example XTN:224
WAVRST:<filename> • Not supported on IP500 V2 systems.
• The <filename> parameter specifies the filename to be played.
• A TFTP read is attempted first, then the folder opt/ipoffice/
system/primary (SSH access) (disk/system/primary (file
manager access)).
• When a MOH source is activated, the playback is started every time from
the beginning.
• At any moment, all users listening to a given MOH source will hear a
different WAV file or file position.
Table continues…

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 Music On Hold

Alternate Option Description

WAVDIR: • Not supported on IP500 V2 systems.
• No additional parameter is required.
• The directory used is opt/ipoffice/tones/mohwavdir (SSH
access) or /disk/tones/mohwavdir (file manager access).
• Up to 255 files, up to 10 minutes per file.
• The files are played in filename order (numerical, lower case, then upper
case).
• When a MOH source is activated, the playback resumes from where it
left off last time.
• At any moment, all users listening to this source will hear the same thing.
• There can only be one WAVDIR: or WAVDIRRST: entry per system.
• This is a streamed source suitable for use with the LINE option.
WAVDIRRST: As per WAVDIR above, however
• When a MOH source is activated, the playback is started every time from
the beginning (start of the first file in the folder).
• At any moment, all users listening to this source will hear a different WAV
file or file position.
• Not suitable for use with the LINE option.
Table continues…

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Alternate Option Description

USB: <number> • Not supported on IP500 V2 systems.
• The <number> parameter is the logical USB device number.
• USB:1 is the first source found and is automatically used for the System
Source when set to External.
• Additional devices are numbered sequentially. For example, USB:2,
USB:3. Up to four USB sources are supported.
• IPOffice will auto-configure USB sound devices with settings that work
well in most cases. Line input is selected and volume is set close to
maximum. If no line input is identified on the card, microphone input is
used instead.
• Supported with ALSA USB sound cards, The following USB audio
devices have been tested:
- Creative X-FI GO Pro USB
- Asus Xonar U3
• External USB sound devices are hot-pluggable. They can be added and
removed from the system at any time.
• Care should be taken when adding or removing USB sound cards as this
may change the logical number.
• When an USB MOH source becomes unavailable, the default MOH tone
will be played instead.
• A USB MOH source is not supported on virtual servers.
• This is a streamed source suitable for use with the LINE option.
Table continues…

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 Conferencing

Alternate Option Description

LINE:<X,Y> • Two parameters are supplied.
- X = SCN line number to the Linux Server (not outgoing group ID).
- Y = The MOH source number on the Linux Server.
• The Linux Server is typically the Primary, but the Secondary Server can
be used.
• The MOH Source must be a stream type (Not WAVRST: or
WAVDIRRST:)
• Centralised MOH will place a VoIP call to the MOH source when MOH is
required.
• Takes one call capacity from the trunk and therefore, can be subject to
CAC limits.
• Uses the SCN trunks’ codec preferences.
• G.729 not recommended (better results are achieved with G.711).
• Calls are dropped after 30s of no use.
• If 30s is not appropriate, it can be changed with the NoUser source
number HOLD_MUSIC_TIMEOUT=x, where x is number of seconds
(range = 0 and 600). 0 means never tear down the call (and never retry
– should not be used!)
• The status displayed in SSA
• Note that as this option can only be specified as an alternate source,
centralized MOH cannot be used as the System Source. That is, it
cannot be used for internal calls’ MOH.

Related links
Music On Hold on page 605

Conferencing
The IP Office system supports a number of conference features and allows multiple simultaneous
conferences.
Conference Types
There are 2 types of conference supported by the system:
• Ad-Hoc Conferencing An ad-hoc conference is one created on the fly, typically by holding
an existing call, making another call and then pressing a conference key on the phone. Other
people can be added to the conference by repeating the process.
• Meet Me Conferencing Conference Meet Me allows users to join or start a specific
numbered conference. This method of operation allows you to advertise a conference
number and then let the other parties join the conference themselves.

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User Personal Conference Number Each user's own extension number is treated as their own
personal conference number. Only that user is able to start a conference using that number as the
conference ID. Any one else attempting to start a conference with that number will find themselves
in a conference but on hold until the owner also joins. Personal conferences are always hosted on
the owner's system. Note, when a user calls from their mobile twinned number, the personal
conference feature will only work if they access the conference using an FNE 18 service.
Conference Notes
Other Uses of Conference Resources System features such as call intrusion, call recording and
silent monitoring all use conference resources for their operation. On IP500 V2 systems, each
Embedded Voicemail call in progress also reduces the conference capacity.
Automatically Ending Conferences The behavior for the system automatically ending a
conference varies as follows:
• A conference remains active until the last extension or trunk with reliable disconnect leaves.
Connections to voicemail or a trunk without reliable disconnect (for example an analog loop-
start trunk) will not hold a conference open.
• The Drop External Only Impromptu Conference setting controls whether a conference is
automatically ended when the last internal party exits the conference.
Analog Trunk Restriction In conferences that include external calls, only a maximum of two
analog trunk calls are supported. This limit is not enforced by the system software.
Recording Conferences If call recording is supported, conference calls can be recorded just like
normal calls. Note however that recording is automatically stopped when a new party joins the
conference and must be restarted manually. This is to stop parties being added to a conference
after any "advice of recording" message has been played.
IP Trunks and Extensions Conferencing is performed by services on the system's non-IP
interface. Therefore a voice compression channel is required for each IP trunk or extension
involved in the conference.
Call Routing A short code routing calls into a conference can be used as an Incoming Call Route
destination.
Conference Tones The system provides conference tones. These will be either played when a
party enters/leaves the conference or as a regularly repeated tone. This is controlled by the
Conferencing Tone (System | Telephony | Tones & Music) option.
Related links
Conference Phones on page 612
Ad-Hoc Conferencing on page 613
Meet Me Conferencing on page 615
Routing External Callers on page 616
Context Sensitive Conferencing on page 617

Conference Phones
The system does not restrict the type of phone that can be included in a conference call.

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 Conferencing

Use Mute When not speaking, use of the mute function helps prevent background noise from your
location being added to the conference call. This is especially important if you are attempting to
participate handsfree.
Handsfree Participation While many Avaya telephones can be used fully handsfree during a call,
that mode of operation is intended only for a single user, seated directly in front of the phone.
Attempting to use a handsfree phones for multiple people to listen to and participate in a call will
rarely yield good results. See below for details of conference phones supported by the system.
Dedicated Conference Phones
To allow multiple people in one room to speak and listen to a conference call, the system supports
the following conference phones:
• B100 Conference Phones (B149, B159 and B179).
• Audio Conferencing Unit (ACU).
Group Listen
The Group Listen function can be used via a programmable button or short code. It allows the
caller to be heard through a phones handsfree speaker while only being talked to via the phone's
handset.
Related links
Conferencing on page 611

Ad-Hoc Conferencing
Conference add controls can be used to place the user, their current call and any calls they have
on hold into a conference. When used to start a new conference, the system automatically assigns
a conference ID to the call. This is termed ad-hoc (impromptu) conferencing.
If the call on hold is an existing conference, the user and any current call are added to that
conference. This can be used to add additional calls to an ad-hoc conference or to a meet-me
conference. Conference add can be used to connect two parties together. After creating the
conference, the user can drop from the conference and the two incoming calls remain connected.
Related links
Conferencing on page 611

The methods below use the system's default system short codes
About this task
Short Code
The Conference Add short code action is used to create short codes for ad-hoc conferencing. By
default, the short code *47 is added to new systems.
Starting an ad-hoc conference using a short code:

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Procedure
1. Place your current call on hold.
2. Call the party that you want to also include in the call.
Result
• If answered and the other party wants to join the conference, put the call on hold and dial
*47.
• If not answered or diverted to voicemail or answered but the party does not want to join the
conference, put the call on hold and dial *52 to clear it.
You and the held calls are now in conference.

Conference Button
About this task
The Conference Add action can be assigned to a programmable button on phones that support
programmable buttons. The button can then be used to start an ad-hoc conference or to add
additional users to an existing conference.
On many Avaya phones, the same function is provided by a permanent Conference button.
Alternatively the phone may display a Conf soft-key option during calls. Refer to the appropriate
phone user guide.
Starting an ad-hoc conference using a button or softkey:
Procedure
1. With a current call connected, press the button.
The current call is put on hold pending the conference.
2. Call the party that you want to also include in the call.
Result
• If answered and the other part wants to join the conference, press the conference button
again.
• If not answered or diverted to voicemail or answered but the party does not want to join the
conference, end the call. Press the button representing the held call to reconnect to it.
You, the held call and the new call are now in a conference.
Adding Calls to a Conference
You can use the same processes as above to add additional calls to a conference. While you hold
a conference on your own telephone system, the existing members of the conference can still talk
to each other.

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 Conferencing

Meet Me Conferencing
Conference meet-me refers to features that allow a user or caller to join a specific conference by
using the conference's ID number (either pre-set in the control or entered at the time of joining the
conference).
IP500 V2 systems require a Preferred Edition license.
Note:
Conference Meet Me features can create conferences that include only one or two parties.
These are still conferences that are using resources from the host system's conference
capacity.
Conference ID Numbers
By default, ad hoc conferences are assigned numbers starting from 100 for the first conference in
progress. Therefore, for conference Meet Me features specify a number away from this range
ensure that the conference joined is not an ad hoc conference started by other users. It is not
possible to join a conference using conference Meet Me features when the conference ID is in use
by an ad-hoc conference.
User Personal Conference Number Each user's own extension number is treated as their own
personal conference number. Only that user is able to start a conference using that number as the
conference ID. Any one else attempting to start a conference with that number will find themselves
in a conference but on hold until the owner also joins. Personal conferences are always hosted on
the owner's system.
Note:
When a user calls from their mobile twinned number, the personal conference feature will only
work if they access the conference using an FNE 18 service.
Multi-Site Network Conferencing
Meet Me conference IDs are now shared across a multi-site network. For example, if a conference
with the ID 500 is started on one system, anyone else joining conference 500 on any system will
join the same conference. Each conference still uses the conference resources of the system on
which it was started and is limited by the available conference capacity of that system.
Previously separate conferences, each with the same conference ID, could be started on each
system in a multi-site network.
Other Features
Transfer to a Conference Button A currently connected caller can be transferred into the
conference by pressing TRANSFER, then the Conference Meet Me button and TRANSFER again
to complete the transfer. This allows the user to place callers into the conference specified by the
button without being part of the conference call themselves. This option is only support on Avaya
phones with a fixed TRANSFER button.
Conference Button Status Indication When the conference is active, any buttons associated
with the conference ID indicate the active state.

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Configure General System Settings

Short Codes
The Conference Meet Me short code action is used to create short codes for Meet Me
conferencing. There are no default short codes in a new system for this type of function. It can
also be used to transfer caller's into a Meet Me conference.
Example 1: Specific Meet Me Conference Short Code
The following example system short code allows the dialing user to join a specific conference, in
this meet-me conference 500.
Short Code: *500
Telephone Number: 500
Feature: Conference Meet Me
Example 2: General Meet Me Conference Short Code
The following example system short code allows any extension to dial *67* and then the number
of the conference which they want to join followed by #. For example dialing *67*600# will put the
user into meet-me conference 600.
Short Code: *67*N#
Telephone Number: N
Feature: Conference Meet Me
Programmable Buttons
The Conference Meet Me action can be assigned to a programmable button on phones that
support programmable buttons. The button can then be used to join a specified Meet Me
conference. It can also be used to transfer caller's into a meet-me conference.
For buttons configured with a specific conference ID, the button will indicate whether a conference
is in progress or not. For a button configured to a user's personal conference number, the button
will indicate when other people are in the conference and when the owner is also in the
conference.
Related links
Conferencing on page 611

Routing External Callers

Internal users can access Meet Me conferencing using short code and buttons. Additional
methods need to be provided for external callers. Typically this is done using a system short code
to which the external call is then directed by one of the options below. This has the advantage that
internal users can also dial the same short code to access the same conferences.
Transferring Callers
Conference Meet Me short codes and buttons can be used as the destination for call transfers by
other users. If the short code or button is not configured with a specific conference ID, that value
needs to be entered by the person transferring the call before completing the transfer.

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 Conferencing

Incoming Call Routing

The method by which calls requiring access to a conference can be identified will depend on the
customer requirement. However, once that is determined, a system short code configured for the
specific conference required can be used as the destination for the appropriate incoming call route
added to the system configuration.
If general access to more than one possible conference is required, Voicemail Pro is used. See
below.
Voicemail Pro
Voicemail Pro call flows can include transfer actions. If the target is a short code for a specific
conference, the call flow user is added to that conference. The use if a short code for this also
allow internal users to access the same conference.
This option can be used with call flow feature such as automatic attendants to provide access to
more than one specific conference. For example, the call flow can prompt the caller to enter the ID
of the conference they want to join. The call flow can then use the digits the caller dials as part of
the short code to which the call is transferred.
Related links
Conferencing on page 611

Context Sensitive Conferencing

On 1400, 1600, 9500 and 9600 Series telephones there have been changes to the display and
handling of calls put on hold pending transfer. See Context Sensitive Transfer. For those phones
there have also been changes to which calls are conferenced when a Conference button or Conf
display option is pressed on the telephone.
Previously, pressing Conference would put the user's current call and all held calls into a
conference. That included any calls that had be put on hold pending transfer by pressing
Transfer. The result of pressing Conference on the telephone now depends on which call is
currently highlighted on the phone display and what other calls are held or held pending transfer.
Which call is Other condition Result when Calls Conferenced
highlighted on (in priority Conference is Connected Call Held Calls Held Pending
the display order) pressed: Transfer
Connected call No call held Conferences –
pending the connected
transfer call and all held
calls.
Table continues…

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Configure General System Settings

Call held Conferences –

pending the connected
transfer call and the
held pending
transfer call.
Any other held
calls are
unaffected.
Held call Connected call Conferences – –
the held call
and the
connected call.
Any other held
calls including
held pending
transfer are
unaffected.
Held pending Conferences – –
transfer call the held and
held pending
transfer call. All
other held calls
are unaffected.
Held calls Conferences – –
with all other
held calls.
Held pending Connected call Conferences –
transfer call the held
pending
transfer call to a
connected call.
Any other held
calls are
unaffected.
Held calls Conferences –
the call held
pending
transfer with all
other held calls.

Note that this new behaviour only applies to conferences being initiated from the telephone. The
original behaviour of conferencing all calls still applies if the conference function is initiated from
elsewhere such as from an application like one-X Portal.
Changing which call is currently highlighted On phones with a set of cursor keys (four cursor
keys around an OK key), the up and down cursor key can be used to change the current
highlighted call (or call appearance if idle). This can be done even whilst there is a currently

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 Paging

connected call. On touchscreen phones, the cursor buttons on the right-hand edge of the screen
can be used for the same purpose. The method of highlighting is
• 1400 Series/1600 Series Telephones On these phones only details of a single call are
shown on the display at any time. The displayed call is the currently highlighted call.
• 9500 Series/9600 Series Telephones On most phones in these series, the background of
the shading is changed for the currently selected call. The exceptions are 9611, 9621, 9641,
and J179 telephones where a yellow symbol is shown on the right of the highlighted call.
Related links
Conferencing on page 611

Paging
Paging limits
Server Type Paging Group Maximum size
(Select and Non-Select)
Dell R620 256
OVA 256
DL360G7 128
HP120G7/Dell R210 II 128
IP500 V2 64

• Paging groups that include users on a V2 Expansion are limited to 64 members.

• For paging groups that include SRTP endpoints, reduce the maximum size by 50%.
Paging Scenarios
Paging Scenario Paged Device Connects to... Short Code/ Button Feature
Phone to Phone: Simple paging Digital Station and Avaya H.323 Dial Paging
to other system extensions. Phones
Mixed Paging: This refers to Analog Extension (Paging Dial Paging
simultaneous paging to phones Speaker)
and a paging speaker.
Paging Interface Device: This Analog Extension (IVR Port) Dial Extn
refers to paging to a paging Analog Trunk Dial
interface device such as a UPAM.

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Configure General System Settings

Phone to Phone Paging

• Paging is supported from all phone types. A page call can be to a single phone or a group of
phones.
- From analog and non-Avaya phones, use a Dial Paging short code.
- From Avaya feature phones, a programmable button set to Dial Paging can be used.
• Paging is only supported to Avaya phones that support auto answer.
• The page is not heard on phones that are active on another call.
• The page is not heard on phones where the user is set to Do Not Disturb or has Forward
Unconditional active.
• On Avaya phones with a dedicated Conference button, the user can answer a page call by
pressing that button. This turns the page into a normal call with the pager.
Mixed Paging

Uses an amplifier connected to an analog extension port via a 600ohm isolating transformer.
Some amplifiers include an integral transformer. Avaya/Lucent branded amplifiers are designed for
connection to special paging output ports not provided on systems. They are not suitable for
supporting mixed paging.
The transformer and amplifier must be connected when the system is restarted.
If background music is required between pages, the amplifier must support a separate background
music connection and VOX switching.
The analog extension port is set as a Paging Speaker in the system configuration (Extn | Analog |
Equipment Classification).

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 Paging

Short code/programmable button: Use DialPaging.

Paging Interface Device

Uses a paging interface device such as a UPAM or amplifier with analog trunk/extension interface.
The device can be connected to an analog trunk port or analog extension port.
If connected to a trunk port, use the short code Use Dial and the same Line Group ID as the
Outgoing Line ID set for the analog trunk.
If connected to an extension port:
• Set the analog extension as an IVR Port in the system configuration (Extn | Analog |
Equipment Classification).
• Short code/programmable button: Use Dial Extn.
Related links
Paging Via Voicemail Pro on page 621

Paging Via Voicemail Pro

Voicemail Pro can be used to deliver pre-recorded announcements. This can be useful when the
same announcement is repeated frequently. This method requires the paging port to be an analog
extension.
This method also removes the feedback loop that can occur on some sites as the page is first
recorded and then played.
Example 1
1. In Voicemail Pro, a new Module was added and named Page.

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Configure General System Settings

2. A Post Dial action was added to the module. The properties of the Specific tab were set as
shown:

3. We then saved and made live the new Voicemail Pro call flow.
4. In Manager we received the system configuration and created a new short code.
• Short Code: *80
• Telephone Number : "Page"
• Feature: VoicemailCollect.
The new system configuration was then merged.
Example 2
This example builds on example 1 by allowing the user to select which message is played from a
menu. In this example the user can press 1, 2 or 3 for different messages. They can also re-record
the message associated with option 3 by pressing #.

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 Automatic Intercom Calls

A Play List action was added and in this example set to record pagemsg3.wav. Note that just the
file name was specified as this action saves files relative to the Voicemail Server's WAVS folder.

In the Post Dial action that plays back pagemsg3.wav note that the full file path needs to be used.
In Manager, we then added a short code that triggers the module "Paging" using the
VoicemailCollect feature.
Related links
Paging on page 619

Automatic Intercom Calls

Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the

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Configure General System Settings

handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
Making Automatic Intercom Calls
The following programmable button functions can be used to make automatic intercom calls:
• Automatic Intercom
• Dial Direct
• Dial Intercom
The following short code function can be used to make automatic intercom calls:
Dial Direct
On M-Series and T-Series phones, the code Feature 66 followed by the extension number can be
used to make a direct voice (automatic intercom) call.
Deny automatic intercom calls
When enabled, any automatic intercom calls to the user's extension are automatically turned into
normal calls.
Deny automatic intercom calls can be configured per user on the User | Telephony | Supervisor
Settings tab. Deny automatic intercom call can also be enabled using the Auto Intercom Deny On
short code or the Auto Intercom Deny button action.

Wide Band Audio Support

IP Office systems support the G.722 64K codec for wide band audio. G.722 can be used with H.
323 and SIP trunks. If can also be used with some SIP and H.323 IP telephones (see below). G.
722 uses a higher speech sample rate (16KHz) than is used by most other audio codecs (8KHz).
G.722 is only supported by systems that are using IP500 VCM, IP500 VCM V2 and or IP500
Combination cards.
Avaya Phone Support
Use of G.722 is supported by the following Avaya phones on a IP Office system:
B179 1140E 9621 9650
1010 9608 9630 J169
1040 9611 9640 J179
1120E 9620 9641

Using the G.722 Codec

The G.722 codec is not available for use by default. If the codec is to be used, it must first be
selected in the system's Available Codecs list (System | Codecs). The codec can then be used in
the system's default codec preference list and or in the individual codec preferences of IP lines
and extensions.

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 Configuring Remote H.323 Extensions

The method of codec selection for specific phones will depend on the phone type. Refer to the
appropriate installation manual.
Conferencing
Where devices using G.722 are in a system conference, the system can attempt to ensures that
the speech between devices using G.722 remains wide-band even if there are also narrow-band
audio devices in the same conference. This is done if the system's High Quality Conferencing
option is enabled (System | Telephony | Telephony).
Known Limitations
The following limitations apply to G.722 wide band audio operation:
• Call recording uses G.711.
• Page calls only use G.722 when all devices being paged can use G.722.
• Fax is not supported in G.722, use G.711 or T38.
• Soft tones provided by the system use G.711.
• A maximum of 15 G.722 devices receiving wide-band audio are supported in conferences.

Configuring Remote H.323 Extensions

The configuration of remote H.323 extensions is supported without needing those extensions to be
running special VPN firmware. This option is intended for use in the following scenario:
• The customer LAN has a public IP address which is forwarded to the IP Office system. That
address is used as the call server address by the H.323 remote extensions.
• The user has a H.323 phone behind a domestic router. It is assumed that the domestic router
allows all outbound traffic from the home network to pass through and allows all symmetric
traffic. That is, if the phone sends RTP/RTCP to a public IP address and port, it will be able to
receive RTP/RTCP from that same IP address and port. If this is not the case, the
configuration of the user's router to support that is not covered by this documentation.

Other scenarios can be configured. For example one of the IP Office's LAN interfaces can be
connected to the public internet.
Supported Telephones: Currently remote H.323 extension operation is only supported with 9600
Series phones already supported by the IP Office system.

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License Requirements: For non-Server Edition systems, by default only 4 users can be
configured for remote H.323 extension usage. Additional users can be configured if those
additional users are licensed and configured with either Teleworker or Power User user profiles.
On Server Edition systems, all users can be configured for remote H.323 extension usage.
Customer Network Configuration
The corporate LAN hosting the IP Office system requires a public IP address that is routed to the
LAN interface of the IP Office system configured for remote H.323 extension support.
STUN from the IP Office system to the Internet is used to determine the type of NAT being applied
to traffic between the system and the Internet. Any routers and other firewall devices between the
H.323 phone location and the IP Office system must allow the following traffic.
Protocol Port Description
UDP 1719 UDP port 1719 traffic to the IP
Office system must be allowed.
This is used for H225 RAS
processes such as gatekeeper
discovery, registration, keepalive,
etc. If this port is not open, the
phone will not be able to register
with the IP Office system.
TCP 1720 TCP port 1720 traffic must be
allowed. This is used for H225
(call signalling).
TCP 1300 TCP port 1300 must be allowed
when using TLS.
RTP Various The ports in the range specified
RTCP by the system's RTP Port
Number Range (Remote Extn)
settings must be allowed.

User Network Configuration

It is assumed that the domestic router allows all outbound traffic from the home network to pass
through and allows all symmetric traffic. That is, if the phone sends RTP/RTCP to a public IP
address and port, it will be able to receive RTP/RTCP from that same IP address and port. If this is
not the case, the configuration of the user's router to support that is not covered by this
documentation.

System Configuration
About this task
This is a summary of the system configuration changes necessary. Additional details and
information for H.323 telephone installation are included in the H.323 IP Telephone Installation
manual. This section assumes that you are already familiar with IP Office system and H.323 IP
telephone installation.

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 Configuring Remote H.323 Extensions

Procedure
1. Licensing:
For non-Server Edition systems, by default only 4 users can be configured for remote H.
323 extension usage. Additional users can be configured if those additional users are
licensed and configured with either Teleworker or Power User user profiles. On Server
Edition systems, all users can be configured for remote H.323 extension usage.
2. System Configuration:
The following needs to be configured on the IP Office system LAN interface to which the
public IP address is routed.
a. Select System | LAN1/LAN2 | VoIP.
Check that the H.323 Gatekeeper Enable setting is selected.
b. Due to the additional user and extension settings needed for remote H.323 extension
configuration, we assume that the extension and user records for the remote H.323
extensions and users are added manually.
c. Select H.323 Remote Extn Enable.
d. Set the RTP Port Number Range (Remote Extn) range to encompass the port range
that should be used for remote H.323 extension RTP and RTCP traffic. The range
setup must provide at least 2 ports per extension being supported.
Note:
When the system is configured on an open internet connection, the standard RTP
port range is used for all H.323 calls including remote workers. In such a case the
RTP Port Number Range is used.
3. Network Topology Configuration:
STUN can be used to determine the type of NAT/firewall processes being applied to traffic
between between the IP Office system and the Internet.
a. Select the Network Topology tab.
b. Set the STUN Server IP Address to a known STUN server. Click OK. The Run
STUN button should now be enabled. Click it and wait while the STUN process is
run. The results discovered by the process will be indicated by ! icons next to the
fields
c. If STUN reports the Firewall/NAT Type as one of the following, the network must be
reconfigured if possible as these types are not supported for remote H.323
extensions: Static Port Block, Symmetric NAT or Open Internet.
4. H.323 Extension Configuration:
H.323 remote extensions use non default settings and so cannot be setup directly using
auto-create.
a. Within Manager, add a new H.323 extension or edit an existing extension.

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Configure General System Settings

b. On the Extn tab, set the Base Extension number.

c. On the VoIP tab, select H.323 Remote Extn Enable.
d. The other settings are as standard for an Avaya H.323 telephone.
• The IP Address field can be used to restrict the the source IP address that can
used by the Remote Worker. However, it should not used in the case where there is
more than one phone behind the domestic router.
• Regardless of direct media configuration, direct media is not used for remote H.323
extensions except for calls between devices behind the same NAT when Allow
Direct Media Within NAT Location is set to On.
5. User Configuration:
For non-Server Edition systems, by default only 4 users can be configured for remote H.
323 extension usage.
Additional users can be configured if those additional users are licensed and configured
with either Teleworker or Power User user profiles.
a. In the user configuration, select Enable Remote Worker.
b. If the user's Extension Number matches the Base Extension setting of an IP
extension, the H.323 Remote Extn Enable setting of that extension is automatically
changed to match the user's Enable Remote Worker setting and vice versa.

Phone Configuration
About this task
The phones do not require any special firmware. Therefore they should first be installed as normal
internal extensions, during which they will load the firmware provided by the IP Office system.
Once this process has been completed, the address settings of the phone should be cleared and
the call server address set to the public address to be used by remote H.323 extensions.
It is assumed that at the remote location, the phone will obtain other address information by DHCP
from the user's router. If that is not the case, the other address setting for the phone will need to
be statically administered to match addresses suitable for the user's home network.

Media Connection Preservation

Media Connection Preservation maintains calls that experience end-to-end signalling loss or
refresh failures but that still have an active media path.
IP Phones:

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 Media Connection Preservation

The following Avaya H.323 phones attempt to maintain calls when the signal from the host IP
Office is lost. The phones must be running the firmware release delivered with IP Office release
9.1 or higher.
• 9608
• 9611
• 9621
• 9641
When preserving a call, the phone does not attempt to reregister with their call server or attempt to
failover to a standby call server, until the call has been terminated. Softkey call actions and feature
menus do not work during this time due to the loss of signalling path. The phone display is not
updated and the only permitted action is to terminate the call.
IP Office:
When enabled for a particular IP endpoint type that supports Media Connection Preservation, the
call is put into a Preserved state and a Preservation Interval timer is started for that call at the
point the signalling loss is detected. The maximum duration of a preserved call on IP Office is two
hours. Once put into the Preserved state, a call can only transition to the Terminated state. Call
restoration is not supported.
Only the following call types are preserved:
• Connected active calls
• Two party calls where the other end is a phone, trunk, or voicemail
• Conference calls
• Calls on hold and calls to hunt groups are not preserved.
Phone Display:
When a call is in a preserved state but the phone's local signalling connection with its host IP
Office is still present, the phone call state display is prefixed with a warning icon. Hold, transfer,
and conference actions are not available.
System Configuration
When enabled on System | Telephony | Telephony, Media Connection Preservation is applied at
a system level to SCN trunks and Avaya H.323 phones that support connection preservation. All
systems in a Small Community Network (SCN) must be enabled for end to end connection
preservation to be supported.
When enabled on Line | SIP Line | SIP Advanced, Media Connection Preservation is applied to
the SIP trunk. The value of connection preservation on public SIP trunks is limited. Media
Connection Preservation on public SIP trunks is not supported until tested with a specific service
provider. Media Connection Preservation is disabled by default for SIP trunks.
When enabled on Line | SM Line | Session Manager, Media Connection Preservation is applied
to Enterprise Branch deployments. Media Connection Preservation preserves only the media and
not the call signaling on the SM Line. Media Connection Preservation does not include support for
the Avaya Aura Session Manager Call Preservation feature.

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Configure General System Settings

Configuring ARS
When a dialed number matches a short code that specifies that the number should be dialled,
there are two methods by which the routing of the outgoing call can be controlled.
Routing Calls Directly to a Line:
Every line and channel has an Outgoing Group ID setting. Several lines and channels can have
belong to the same Outgoing Group ID. Within short codes that should be routed via a line within
that group, the required Outgoing Group ID is specified in the short code's Line Group ID setting.
Routing Calls via ARS:
The short code for a number can specify an ARS form as the destination. The final routing of the
call is then controlled by the setting available within that ARS form.
ARS Features
Secondary Dial Tone:
The first ARS form to which a call is routed can specify whether the caller should receive
secondary dial tone.
Out of Service Routing:
ARS forms can be taken out of service, rerouting any calls to an alternate ARS form while out of
service. This can be done through the configuration or using short codes.
Out of Hours Routing:
ARS forms can reroute calls to an alternate ARS form outside the hours defined by an associated
time profile.
Priority Routing:
Alternate routes can be made available to users with sufficient priority if the initial routes specified
in an ARS form are not available. For users with insufficient priority, a delay is applied before the
alternate routes become available.
Line Types:
ARS can be used with all line types.
A SIP line is treated as busy and can follow alternate routes based on the SIP line setting Call
Initiation Timeout. Previously a SIP line was only seen as busy if all the configured channels
were in use.
IP lines use the NoUser Source Number setting H.323SetupTimerNoLCR to determine how long
to wait for successful connection before treating the line as busy and following ARS alternate
routing. This is set through the IP line option Call Initiation Timeout.
Multi-Site Network Calls:
Calls to multi-site extension numbers are always routed using the appropriate network trunk. ARS
can be configured for multi-site network numbers but will only be used if the network call fails due
to congestion or network failure.
Main Route:

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 Configuring ARS

The ARS form 50, named "Main" cannot be deleted. For defaulted systems it is used as a default
route for outgoing calls.
Routing Calls to ARS
1. Create the ARS form.
2. Create the required system, user or user rights short code to match the user dialing.
a. In the Telephone Number field, define the digits that will be used to match a short
code in the ARS form.
b. Use the Line Group ID field drop-down to select the ARS form required for routing
the call.
Related links
Example ARS Operation on page 631
ARS Operation on page 632

Example ARS Operation

The simplest example for ARS operation are the settings applied to a defaulted system. These
vary between U-Law systems and A-Law systems. For Server Edition systems refer to Server
Edition Outgoing Call Routing.
A-Law Systems
This set of defaults is applied to A-Law systems, typically supplied to locales other than North
America. The defaults allow any dialing that does not match an internal number to be routed off-
switch as follows:
System Short Code - ?/Dial/./50:Main:
The default system short code ? will match any dialing for which no other user, user rights or
system short code match is found. This short code is set to route all the digits dialed to ARS form
50.
ARS Form - 50:Main:
This form contains just a single short code.
?/Dial3K1/./0 This short code matches any digits passed to the ARS form. It then dials the digits
out on the first available line within line group 0 (the default outgoing line group for all lines).
U-Law Systems
This set of defaults is applied to U-Law systems, typically supplied to locales in North America.
The defaults route any dialing prefixed with a 9 to the ARS and secondary dial tone.
System Short Code - 9N/Dial/N/50:Main:
The default system short code 9N is used to match any dialing that is prefixed with a 9. It passes
any digits following the 9 to ARS form 50.
ARS Form - 50:Main:
This form has secondary dial tone enabled. It contains a number of short codes which all pass any
matching calls to the first available line within line group 0 (the default outgoing line group for all

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Configure General System Settings

lines). Whilst all these short code route calls to the same destination, having them as separate
items allows customization if required. The short codes are:
• 11/Dial Emergency/911/0 This short code matches an user dialing 911 for emergency
services.
• 911/Dial Emergency/911/0 This short code matches an user dialing 9911 for emergency
services.
• 0N;/Dial3K1/0N/0 This short code matches any international calls.
• 1N;/Dial3K1/1N/0 This short code matches any national calls.
• XN;/Dial3K1/N/0 This short code matches 7 digit local numbers.
• XXXXXXXXXX/Dial3K1/N/0 This short code matches 10 digit local numbers.
Related links
Configuring ARS on page 630

ARS Operation
The diagram below illustrates the default ARS routing applied to systems (other than Server
Edition) defaulted to the United States system locale. In summary:
• Any dialing prefixed with 9 will match the default system short code 9N.
• That short code routes calls to the default ARS form 50:Main.
• The short codes in that ARS form route all calls to an available line that has its Outgoing
Group ID set to 0.

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 Configuring ARS

The table describes in more detail the process that the system has applied to the user's dialing, in
this example 91555707392200.
The user dials...
9 The Dial Delay Count is zero, so the system
begins looking for short code matches in the system
and user's short codes immediately.
Since there is only one match, the 9N system short
code, it is used immediately.
The 9N short code is set to route the call to the ARS
form Main. It only passes those digits that match
the N part of the dialing, ie. the 9 is not passed to
the ARS, only any further digits dialed by the user.
Secondary Dial Tone is selected in the ARS form.
Since no digits for ARS short code matching have
been received, secondary dial tone is played to the
user.
1 Having received some digits, the secondary dial
tone stops.
The ARS form short codes are assessed for
matches.
The 11 and 1N; short codes are possible matches.
The 911 and 0N; short codes are not possible
matches.
The XN; and XXXXXXXXXXN; short codes are also
not matches because the 1N; short code is already
a more exact match.
Since there is more than one possible match, the
system waits for further digits to be dialed.
555 The 11 short code is no longer a possible match.
The only match is left is the 1N; short code.
The ; in the short code tells the system to wait for
the Dial Delay Time to expire after the last digit it
received before assuming that dialing has been
completed. This is necessary for line providers that
expect to receive all the routing digits for a call 'en
bloc'. The user can also indicate they have
completed dialing by pressing #.
707392200 When the dialing is completed, a line that has its
Outgoing Group ID set to 0 (the default for any
line) is seized.
If no line is available, the alternate route settings
would applied if they had been configured.

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Configure General System Settings

Related links
Configuring ARS on page 630
ARS Short Codes on page 634
Simple Alternate Line Example on page 635
Simple Call Barring on page 636
User Priority Escalation on page 637
Time Based Routing on page 638
Account Code Restriction on page 639
Tiered ARS Forms on page 640
Planning ARS on page 641

ARS Short Codes

The short codes in the default ARS form have the following roles:
Code Feature Telephone Number Line Group ID Description
11 Dial Emergency 911 0 These two short
911 Dial Emergency 911 0 codes are used to
route emergency
calls. A Dial
Emergency call is
never blocked. If
the required line is
not available, the
system will use the
first available line.
Similarly, calls
using Dial
Emergency ignore
any outgoing call
bar settings that
would be normally
applied to the user.
0N; Dial 3K1 0N 0 Matches
international
numbers.
1N; Dial 3K1 1N 0 Matches national
numbers.
XN; Dial 3K1 N 0 Matches 7 digit
local numbers.
XXXXXXXXXXN; Dial 3K1 N 0 Matches 10 digit
local numbers.

ARS Short Code Settings

Code The digits used for matching to the user dialing.

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Feature ARS short codes can use any of the Dial short code features or the Barred feature.
When a Barred short code is matched, the call will not proced any further.
Telephone Number The number that will be output to the line as the result of the short code being
used as the match for the user dialing. Short code characters can be used such as N to match any
digits dialed for N or X in the Code.
Line Group ID The line group from which a line should be seized once short code matching is
completed. Another ARS form can also be specified as the destination.
Locale Not used for outgoing external calls.
Forced Account Code If enabled, the user will be prompted to enter a valid account code before
the call can continue. The account code must match one set in the system configuration.
Related links
ARS Operation on page 632

Simple Alternate Line Example

Using the default ARS settings, despite having several short codes in the ARS form, all outgoing
calls are actually routed the same way using the same trunks. However, by having separate short
codes for different call types present, it is easy to change the routing of each call type if required.
For this example, the customer has separate sets of lines for local calls and for national/
international calls. These have been configured as follows:
• The lines for local and emergency calls have been left with the default Outgoing Group ID of
0.
• The lines for national and international calls have been set with the Outgoing Group ID of 1.
The default ARS can be configured to match this by just changing the Line Group ID settings of
the default ARS short codes to match.

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Configure General System Settings

Related links
ARS Operation on page 632

Simple Call Barring

All ARS short codes use one of the Dial short code features. The exception is the Barred short
code feature. This can be selected for ARS short codes that match dialing that is not allowed.
In the example below, any user dialing an international number will be routed to the Barred short
code. This prevents the dialing of external numbers prefixed with 0.

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 Configuring ARS

To restrict a user from making any outgoing external calls, use the user's Outgoing Call Bar option.
Related links
ARS Operation on page 632

User Priority Escalation

User priority can be used to alter call routing when the required route is not available.
In this example, international calls are initially targeted to seize a line in outgoing line group 1.
However an alternate route has been defined which will be used if no line in line group 1 is
available. The fallback ARS form allows international calls to seize a line from line group 0.
Whether this is done immediately or after a delay is set by whether the users priority is high
enough.

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Related links
ARS Operation on page 632

Time Based Routing

Time profiles can be used to switch call routing from one ARS form to another.
In the example below, a time profile has been define that sets the hours for normal operation.
Outside the times set in the time profile, the other ARS form is used. This other ARS form only
allows local and emergency calls.

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Related links
ARS Operation on page 632

Account Code Restriction

The short codes within an ARS form can be individually set to require an account code before
allowing any call that matches it to proceed.

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Configure General System Settings

In the example below, the short code for international calls has been set to require the user to
enter an account code. A valid account code must be dialed to continue with the call.

If a user should always enter an account code to make any external call, the user option Force
Account Code should be used.
Related links
ARS Operation on page 632

Tiered ARS Forms

It is possible for an ARS short code in one form to have another ARS form as its destination.
Dialing that matches the short code is then subject to further matching against the short codes in
the other ARS form.
In the example below, the user wants different routing applied to international calls based on the
country code dialed. To do that in the default ARS form would introduce a large number of short
codes in the one form, making maintenance difficult.
So the short code matching calls with the international dialing prefix 0 has been set to route
matching calls to another ARS form. That form contains short codes for the different country
dialing codes of interest plus a default for any others.

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Related links
ARS Operation on page 632

Planning ARS
Using the methods shown in the previous examples, it is possible to achieve ARS that meets most
requirements. However the key to a good ARS implementation is planning.
A number of questions need to be assessed and answered to match the system's call routing to
the customer's dialing.
What What numbers will be dialed and what needs to be output by the system. What are the
different call tariffs and the dialing codes.
Where Where should calls be routed.

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Who Which users should be allowed to use the call routes determined by the previous questions.
When When should outgoing external calls be allowed. Should barring be applied at any particular
times? Does the routing of calls need to be adjusted for reasons such as time dependant call
tariffs.
Related links
ARS Operation on page 632

Configuring IP Routes
The system acts as the default gateway for its DHCP clients. It can also be specified as the default
gateway for devices with static IP addresses on the same subnet as the system. When devices
want to send data to IP addresses on different subnets, they will send that data to the system as
their default gateway for onward routing.
The IP Route table is used by the system to determine where data traffic should be forwarded.
This is done by matching details of the destination IP address to IP Route records and then using
the Destination specified by the matching IP route. These are referred to as 'static routes'.
Automatic Routing (RIP): The system can support RIP (Routing Information Protocol) on LAN1
and or LAN2. This is a method through which the system can automatically learn routes for data
traffic from other routers that also support matching RIP options, see RIP. These are referred to as
'dynamic routes'. This option is not supported on Linux based servers.
Dynamic versus Static Routes: By default, static routes entered into the system override any
dynamic routes it learns by the use of RIP. This behavior is controlled by the Favor RIP Routes
over static routes option on the System | System tab.
Static IP Route Destinations: The system allows the following to be used as the destinations for
IP routes:
• LAN1 Direct the traffic to the system's LAN1.
• LAN2 Traffic can be directed to LAN2.
• Service Traffic can be directed to a service. The service defines the details necessary to
connect to a remote data service.
• Tunnel Traffic can be directed to an IPSec or L2TP tunnel.
Default Route: The system provides two methods of defining a default route for IP traffic that
does not match any other specified routes. Use either of the following methods:
• Default Service Within the settings for services, one service can be set as the Default
Route (Service | Service).
• Default IP Route Create an IP Route record with a blank IP Address and blank IP Mask set
to the required destination for default traffic.
RIP Dynamic Routing common
Routing Information Protocol (RIP) is a protocol which allows routers within a network to exchange
routes of which they are aware approximately every 30 seconds. Through this process, each
router adds devices and routes in the network to its routing table.

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 Creating a Virtual WAN Port

Each router to router link is called a 'hop' and routes of up to 15 hops are created in the routing
tables. When more than one route to a destination exists, the route with the lowest metric (number
of hops) is added to the routing table.
When an existing route becomes unavailable, after 5 minutes it is marked as requiring 'infinite' (16
hops). It is then advertised as such to other routers for the next few updates before being removed
from the routing table. The system also uses 'split horizon' and 'poison reverse'.
RIP is a simple method for automatic route sharing and updating within small homogeneous
networks. It allows alternate routes to be advertised when an existing route fails. Within a large
network the exchange of routing information every 30 seconds can create excessive traffic. In
addition the routing table held by each system is limited to 100 routes (including static and internal
routes).
It can be enabled on LAN1, LAN2 and individual services. The normal default is for RIP to be
disabled.
• Listen Only (Passive): The system listens to RIP1 and RIP2 messages and uses these to
update its routing table. However the system does not respond.
• RIP1: The system listens to RIP1 and RIP2 messages. It advertises its own routes in a RIP1
sub-network broadcast.
• RIP2 Broadcast (RIP1 Compatibility): The system listens to RIP1 and RIP2 messages. It
advertises its own routes in a RIP2 sub-network broadcast. This method is compatible with
RIP1 routers.
• RIP2 Multicast: The system listens to RIP1 and RIP2 messages. It advertises its own routes
to the RIP2 multicast address (249.0.0.0). This method is not compatible with RIP1 routers.
Broadcast and multicast routes (those with addresses such as 255.255.255.255 and 224.0.0.0)
are not included in RIP broadcasts. Static routes (those in the IP Route table) take precedence
over a RIP route when the two routes have the same metric.

Creating a Virtual WAN Port

Procedure
1.
Select WAN Port.
2. Click and select PPP.
3. In the Name field, enter either LINEx.y where:
• LINE must be in uppercase.
• x is the line number. For a PRI/T1 module in Slot A, this will be 1. For a PRI/T1 module
in Slot B, this will be 5.
• y is the lowest numbered channel number to be used by the WAN link minus 1. For
example, if the lowest channel to be used is channel 1 then y = 1 - 1 = 0.
4. In the Speed field, enter the total combined speed of the maximum number of channels
sets in the Service.

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Configure General System Settings

In this example, 12 channels x 64000 bits = 76800.

Note:
The maximum number of channels that can be used will be limited by the number of
data channels supported by the system Control Unit and not already in use.
5. In the RAS Name field, select the RAS name created when the new Service of that name
was created.
6. Click OK.

System Events
The system supports a number of methods by which events occurring on the system can be
reported. These are in addition to the real-time and historical reports available through the System
Status Application (SSA).
SNMP Reporting
Simple Network Management Protocol (SNMP) allows SNMP clients and servers to exchange
information. SNMP clients are built into devices such as network routers, server PC's, etc. SNMP
servers are typically PC application which receive and/or request SNMP information. The system
SNMP client allows the system to respond to SNMP polling and to send alarm information to
SNMP servers.
In order for an SNMP server application to interact with a system, the MIB files provided with the
Manager installation software must be compiled into the SNMP server's applications database.
Note:
The process of 'on-boarding' (refer to the IP Office Installation manual and the IP Office SSL
VPN Solutions Guide) may automatically configure SNMP and create a number of SNMP
alarm traps. These will override any existing SNMP configuration settings.
SMTP Email Reporting
The system can send alarms to an SMTP email server. Using SMTP requires details of a valid
SMTP email account user name and password and server address. If SMTP email alarms are
configured but for some reason the system cannot connect with the SMTP server, only the last 10
alarms are stored for sending when connection is successful. Use of SMTP alarms requires the
SMTP server details to be entered in the SMTP tab.
Syslog Reporting
The system can also send alarms to a Syslog server (RFC 3164) without needing to configure an
SNMP server. In addition Syslog output can include audit trail events.
Multiple event destinations can be created, each specifying which events and alarms to include,
the method of reporting to use (SNMP, Syslog or Email) and where to send the events. Up to 2
alarm destinations can be configured for SNMP, 2 for Syslog and 3 for SMTP email.

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 Configuring authorization codes

Related links
Configuring Alarm Destinations on page 645

Configuring Alarm Destinations

About this task
The Alarms section of the System Events tab displays the currently created alarm traps. It shows
the event destinations and the types of alarms that will trigger the send of event reports. Up to 2
alarm destinations can be configured for SNMP, 2 for Syslog and 3 for SMTP email.
Procedure
1. In the navigation pane, select System.
2. In the details pane, select System Events and then select the Alarms sub-tab.
3. Use the Add, Remove and Edit controls to alter the traps.
4. Click Add or select the alarm to alter and then click Edit.
5. For a new alarm, set the Destination to either Trap (SNMP) or Syslog or Email (SMTP).
Note that once a destination has been saved by clicking OK it cannot be changed to
another sending mode.
6. The remaining details will indicate the required destination information and allow selection
of the alarm events to include.
7. When completed, click OK.
8. Click OKagain.
Related links
System Events on page 644

Configuring authorization codes

Note:
In release 9.1, authorization codes can no longer be associated with User Rights. If an
authorization code was configured in relationship with User Rights in an earlier release
configuration, this authorization code will be lost during upgrade. The administrator must re-
configure the authorization code, after upgrade. The authorization code must be associated
with a user.
Authorization codes are enabled by default.
A user dials a number that matches a short code set to Force Authorization Code. The user is
prompted to enter an authorization code.

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Configure General System Settings

They dial their authorization code. If a matching entry is found in Authorization Codes records
the system checks the corresponding user. Note that the user checked does not necessarily need
to be connected with the user dialing or the user whose extension is being used to make the call.
The dial string is checked against the short codes with the matching user. If it matches a dial short
code or no short code the call is allowed, otherwise it is blocked. Note that the short code is not
processed, it is just checked for a match. If multi-tier authorization codes are required there must
be blocking (busy) short codes (or a wild card '?' )
Example:
A restaurant has a number of phones in publicly accessible areas and want to control what calls
can be made by staff. Staff must not be able to dial long distance numbers. staff should be able to
dial local and cell phone numbers.
ARS Table
In the Main (50) ARS table, add the following short codes:
• 044XXXXXXXXXX/Dial/044N/
• 01XXXXXXXXXX/Dial/01N/Force Auth Code checked
Authorization Codes
Configure an authorization code for each staff member that is allowed to make long distance calls. For
example, for staff members Alice and Bob:
AuthCode: 2008 - Alice
AuthCode: 1983 - Bob

It is recommended to use short codes that use X characters to match the full number of characters
to be dialed. That ensures that authorization code entry is not triggered until the full number has
been dialed rather than mid-dialing. For example 09 numbers are premium rate in the UK, so you
would create a 09XXXXXXXXX/Dial/N short code set to Forced Authorization. In the associated
user or user right short code it is recommended to use 09N type short codes.
System short codes that route to ARS will not have their Force Authorization Code setting used.
However short codes within an ARS table will have their Force Authorization Code setting used.
Forcing Authorization Codes
There are two methods to force a user to enter an authorization code in order to complete dialing
an external call.
• To Force Authorization Codes on All External Calls A user can be required to enter an
authorization code for all external calls. This is done by selecting Force Authorization Code
(User | Telephony | Supervisor Settings).
• To Force Authorization Codes on Specific Calls To require entry of an authorization code
on a particular call or call type, the Force Authorization Code option should be selected in the
short code settings. This can be used in user or system short codes in order to apply its
effect to a user or all users respectively. You need to ensure that the user cannot dial the
same number by any other method that would by pass the short code, for example with a
different prefix.
Related links
Entering an Authorization Code on page 647

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 Preventing Toll Bypass

Entering an Authorization Code

Where possible, when an authorization code is required, the user can enter it through their phones
display. However, this is not possible for all type of phone, for example it is not possible with
analog phones and Avaya XX01 or XX02 phones. The users of these device must either enter the
authorization code using a short code set to the Set Authorization Code feature immediately
before making the call.
When entry of an authorization code is triggered, the user can enter any authorization code with
which they are directly associated.
Note the following.
• If authorization code entry is setup for a particular number, calls forwarded or transferred to
that number will also trigger authorization code entry.
• On systems using line appearances to BRI trunk channels to make outgoing calls,
authorization code entry may not be triggered. This can be resolved by adding a short code
such as [9]XN;/Dial/XN/0 (adjust the prefix and line group as necessary).
Related links
Configuring authorization codes on page 645

Preventing Toll Bypass

Use this procedure to prevent toll bypass in Enterprise Branch and Small Community Network
(SCN) deployments. Toll bypass is prevented by only allowing PSTN calls where the originating
location and terminating location are the same.
The location of non-IP lines is the same as the system location. If an IP address is not resolved to
any location, then that device is assumed to be in the system location. The location of public IP
lines must be configured to same as PSTN termination location.
The Location field for extensions with simultaneous login must be automatic and the location tab
must be properly configured for the IP range.
Enterprise Branch deployments: All the distributed users must be in the same location as
system location. Users registering from a location different from the system location are not
supported.
Procedure
1. In the navigation pane on the left, select System.
2. In the details pane, click the Telephony tab.
3. Under Telephony, click the Telephony tab.

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Configure General System Settings

4. On the Telephony tab:

a. Click the check box to turn Restrict Network Interconnect on.
b. Click the check box to turn Include location specific information on.
Setting the two configuration setting on the Telephony tab adds a Network Type field
to the configuration settings for each trunk.
5. For Enterprise Branch deployments, open the SM Line | Session Manager tab. For SCN
deployments, open the IP Office Line | Line tab.
6. If the line is a PSTN trunk (includes SIP), set Network Type to Public. If the line is an
enterprise trunk, set the Network Type to Private.
7. If the Network Type is Private, the Include location specific information field is
available.
If the line is connected to an Avaya Aura® system release 7.0 or higher, or an IP Office
release 9.1 or higher, set Include location specific information to On.

Configuring unknown locations

Use this procedure to configure extensions where the location is unknown.
Procedure
1. In the navigation pane, select Location.
2. Enter a Location Name.
3. Set Parent Location for CAC to Cloud.
4. In the Extension | Extn tab, set the Location field to the location defined in step 2.

Call Barring
Related links
Applying Call Barring on page 648
Overriding call barring on page 649

Applying Call Barring

Call barring can be applied in a number of ways.
Barring a User From Receiving Any External Calls:
For each user, the Incoming Call Bar setting (User | Telephony | Supervisor Settings ) can be
selected to stop that user from receiving any external calls.

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 Call Barring

Barring a User From Making Any External Calls:

For each user, the Outgoing Call Bar setting (User | Telephony | Supervisor Settings) can be
selected to stop that user from making any external calls.
Barring Particular Numbers/Number Types:
System short codes are used to match user dialing and then perform a specified action. Typically
the action would be to dial the number to an external line. However, short codes that match the
dialing of particular numbers or types of numbers can be added and set to another function such
as Busy. Those short codes can be added to a particular user, to a User Rights associated with
several users or to the system short codes used by all users.
The system allows short codes to be set at user, user rights, system and least cost route. These
have a hierarchy of operation which can be used to achieve various results. For example a system
short code for a particular number can be set to busy to bar dialing of that number. For a specific
user, a user short code match to the same number but set to Dial will allow that user to override
the system short code barring.
Using Account Codes:
The system configuration can include a list of account codes. These can be used to restrict
external dialing only to users who have entered a valid account code.
• Forcing Account Code Entry for a User: A user can be required to enter an account code
before the system will return dialing tone. The account code that they enter must match a
valid account code stored in the system configuration. The setting for this is Force Account
Code (User | Telephony | Supervisor Settings)
• Forcing Account Code Entry for Particular Numbers: Each system short code has a
Force Account Code option. Again the account code entered must match a valid account
code stored in the system configuration. for the call to continue.
Barring External Transfers and Forwards:
A user cannot forward or transfer calls to a number which they cannot normally dial. In addition
there are controls which restrict the forwarding or transferring of external calls back off-switch. See
Off-Switch Transfer Restrictions on page 719.
Related links
Call Barring on page 648

Overriding call barring

When a system or user short code is configured to bar outgoing calls, you can override call
barring. Typically, this configuration is used for a phone in a shared or public area. By default, the
phone has outgoing calls barred. The administrator can override call barring for specific dialed
numbers by entering numbers with a record in the external directory. When the dialed number
exists in the external directory and the Directory Overrides Call Barring setting is enabled, call
barring is overridden.
The System Directory entries must use the format (shortcode)number. For example, if the number
to dial is 61234, where 6 is the shortcode used to dial externally and 1234 is the number, the
System Directory entry must be (6)1234. If the dial shortcode contains a name string rather than
digits, then Directory Overrides Call Barring will not work.

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Configure General System Settings

The Directory Overrides Barring setting is located on the System | Telephony | Telephony tab.
For information on the directory, see the description for the System | Directory Services tab.
Server Edition configuration
For Server Edition deployments, the Directory Overrides Barring must be enabled on each
node. It is not a system wide setting.
For example, if the Primary Server uses an IP500 V2 expansion system as an ISDN gateway,
Directory Overrides Barring must be enabled on the Primary Sever for Primary Server users
dialing on external ISDN lines. For the IP500 V2 expansion users, Directory Overrides Barring
must be enabled on the IP500 V2 expansion system.
It is recommend that the short code configured to dial externally on ISDN lines be the same on all
nodes. For example, if Primary Server users and IP500 V2 expansion users want to reach PSTN
number 123456789 on ISDN lines, configure the dial codes as follows.
• Primary Server: 6N/Dial/6N/XX (XX is the line group ID for the SCN line)
• IP500 V2 expansion: 6N/Dial/N/YY (YY is the line group ID for ISDN line)
• Directory Entry number defined on Primary Server: (6)123456789
Related links
Call Barring on page 648

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Chapter 13: Configure User Settings

Related links
User Management Overview on page 651
Configuring User Rights on page 653
Configuring Gmail Integration on page 657
Call Intrusion on page 658
Call Tagging on page 664
Call Waiting on page 664
Call Restriction on page 665
Centralized Call Log on page 666
Centralized Personal Directory on page 671
Account Code Configuration on page 672
Coverage Groups on page 673
DND, Follow Me and Forwarding on page 674
Hot Desking on page 688
Group Operation on page 694
Malicious Call Tracing (MCID) on page 704
Message Waiting Indication on page 704
Mobile Call Control on page 707
Twinning on page 712
Private Calls on page 715
System Phone Features on page 715
The 'No User' User on page 717
Transferring Calls on page 718

User Management Overview

Users are the people who use the system. They do not necessary have to be an extension user,
for example users are used for RAS dial in data access. In addition, more users can be created
than there are extensions, with users logging in to an extension when they want to receive calls.

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Configure User Settings

By default, a user is automatically created to match each extension. They are numbered from 201
upwards and the first 16 are placed in the hunt group Main (200), which is the default destination
for incoming calls.
Terminology
Standard User: A standard user.
Centralized User: Centralized users can be provisioned for enterprise branch deployments.
No User: Used to apply settings for extensions which currently have no associated user. The
SourceNumbers settings of the NoUser user is used to configure a number of special options.
These are then applied to all users on the system.
Remote Manager: Used as the default settings for dial in user connections.
Hot Desking User: Users with a Login Code can move between extensions by logging in and off.
Deleting a User
When a user is deleted, any calls in progress continue until completed. The ownership of the call
is shown as the NoUser user. Merging the deletion of a user causes all references to that deleted
user to be removed from the system.
Changing a User's Extension
Changing a user's extension number automatically logs the user in on the matching base
extension if available and the user doesn't have Forced Login enabled. If Forced Login is
enabled, then the user remains on the current extension being used until they log out and log in at
the new extension.
Note that changing a user's extension number affects the user's ability to collect Voicemail
messages from their own extension. Each user's extension is set up as a "trusted location" under
the Source Numbers tab of the User configuration form. This "trusted location" allows the user to
dial *17 to collect Voicemail from his own extension. Therefore if the extension number is changed
so must the "trusted location".
The following related configuration items are automatically updated when a user extension is
changed:
• User, Coverage and Bridged Appearance buttons associated with the user.
• Hunt group membership (disabled membership state is maintained).
• Forwards and Follow Me's set to the user as the destination.
• Incoming call routes to this destination.
• Dial in source numbers for access to the user's own voicemail.
• Direct call pickup buttons are updated.
• The extension number of an associated extension is updated.
Server Edition User Management
In a Server Edition network, individual users are still added to the configuration of a particular
server. Typically they are added to the configuration of the server that hosts the user's physical
extension or supports their main place of work. That server is treated as the host system for the
user. However, once a user is added to the configuration of a particular system, you can use
Manager and Web Manager to manage all users in the Server Edition solution.

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 Configuring User Rights

Centralized User Management

Centralized Users are provisioned for enterprise branch deployments. Centralized Users are
registered with Session Manager and are able to utilize telephony features provided by
Communication Manager. The Centralized User profile is applicable to both SIP and analogue
extensions. For more information, see Administering Centralized Users for an IP Office™ Platform
Enterprise Branch. The following requirements must be met when provisioning a centralized user:
• An SM line must be configured on the system.
• The user must be provisioned with an existing extension.
• The extension Base Extension value must match the centralized extension value.
• Centralized users must be configured with a password for SIP registration on Session
Manager. The password is set in User | Telephony | Supervisor Settings | Login Code field.
Related links
Configure User Settings on page 651

Configuring User Rights

For most settings in a user rights template, the adjacent drop down list is used to indicate whether
the setting is part of the template or not. The drop down options are:
• Apply User Rights Value Apply the value set in the user rights template to all users
associated with the template.
- The matching user setting is grayed out and displays a lock symbol.
- Users attempting to change the settings using short codes receive inaccessible tone.
• Not Part of User Rights Ignore the user rights template setting.
Default User Rights
For defaulted systems, the following user rights are created as a part of the default configuration.
Fields not listed are not part of the user rights.
Note:
When a user logs in as a Outbound Contact Express agent, the Outdialer user rights are
automatically applied. When the agent logs out, the previous user rights are applied.
= Set to On. = Set to Off. = Not part of the user rights.
User Call Boss Applicati Default IP Hard Mailbox Paging Outdialer
Rights Center on Phone
Agent
Priority 5 5 5 5 5 5 5 5
Voicemail
Table continues…

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Configure User Settings

User Call Boss Applicati Default IP Hard Mailbox Paging Outdialer

Rights Center on Phone
Agent
Voicemail
Ringback
Outgoing
Call Bar
No 0 0 0 0 0 0 0 0
Answer
Time
Transfer 0 0 0 0 0 0 0 0
Return
Time
Individual 10 10 10 10 10 10 10 10
Coverage
Time
Busy on
Held
Call
Waiting
Can
Intrude
Cannot
be
Intruded
Deny
Auto
Intercom
Calls
Enable
Inhibit
Off-
Switch
Forward/
Transfer
Enable
Outgoing
Call Bar
Centraliz
ed
Logging
Force
Login
Table continues…

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 Configuring User Rights

User Call Boss Applicati Default IP Hard Mailbox Paging Outdialer

Rights Center on Phone
Agent
Force
Account
Code
Button 1: a= 2: 1: a= 2: 1: a= 2: 1: a= 2: 1: a= 2:
Program b= 4: b= 3: c= b= 3: c= b= 3: c= b= 3:
ming HGEna 5: 6: 6: Dial Superviso
DNDOn DNDOn *17 r 4: Extn
6: Busy 7: Dial Logout
*17

Related links
Configure User Settings on page 651
Adding User Rights on page 655
Creating a User Right Based on an Existing User on page 655
Associating User Rights to a User on page 656
Copy User Rights Settings over a User's Settings on page 656

Adding User Rights

About this task
Procedure
1.
Select User Rights.
2. Click and select User Rights.
3. Enter a name.
4. Configure the user rights as required.
5. Click OK.
Related links
Configuring User Rights on page 653

Creating a User Right Based on an Existing User

About this task
Procedure
1.
Select User Rights.
2. In the group pane, right-click and select New User Rights from a User.

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Configure User Settings

3. Select the user and click OK.

Related links
Configuring User Rights on page 653

Associating User Rights to a User

About this task
Procedure
1.
Select User Rights or User.
2. In the group pane, right-click and select Apply User Rights to Users.
3. Select the user rights to be applied.
4. On the Members of this User Rights sub tab select the users to which the user rights
should be applied as their Working Hours User Rights.
5. On the Members when out of hours sub tab select which users should use the selected
user rights as their out of hours user rights.
6. Click OK.
Related links
Configuring User Rights on page 653

Copy User Rights Settings over a User's Settings

About this task
This process replaces a user's current settings with those that are part of the selected user rights.
It does not associate the user with the user rights.
Procedure
1.
Select User Rights.
2. In the group pane, right-click and select Copy user rights values to users.
3. Select the user rights to be applied.
4. Click OK.
Related links
Configuring User Rights on page 653

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 Configuring Gmail Integration

Configuring Gmail Integration

You can integrate the Google Gmail application into Voicemail Pro in order to use a Gmail account
for voicemail to email functions. The supported functions are:
• Forward: Voicemail messages are sent as email to the Gmail account of a user. Users can
use Gmail to retrieve and manage emails.

• Copy: Copies of voicemail messages are sent as email to the Gmail account of a user. The
message is also stored locally on the Voicemail Pro server.
• Alert: An email is sent to the Gmail account of a user indicating the arrival of a new
voicemail.
For the forwarding function:
• Up to 250 users are supported.
• The maximum message length is 7 minutes or 14 minutes when using companded.
• Messages can be accessed using Visual Voice but not one-X Communicator.

User's Gmail Account

HTTP RestFul Access

Google API Framework

Gmail Service API

Primary Server

Voicemail Pro Server

Related links
Configure User Settings on page 651

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Configure User Settings

Call Intrusion
The system supports several different methods for call intrusion. The method used affects which
parties can hear and be heard by other parties following the intrusion. Intrusion features are
supported across a multi-site network
In the scenarios below, user A is on a call with B who may be internal or external. User C invokes
one of the call intrusion methods targeting user A.
Feature Description Privacy Settings
User Target
Can Intrude Cannot Be Private Call
Intruded
Call Listen This feature allows Used Used Used
you to monitor
another user's call
without being
heard. Monitoring
can be
accompanied by a
tone heard by all
parties. Use of the
tone is controlled
by the Beep on
Listen setting on
the System |
Telephony | Tones
& Music tab. The
default for this
setting is on. If
enabled, this is the
only indication of
monitoring given to
the monitored user.
There is no phone
display indication of
monitoring.
Table continues…

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 Call Intrusion

Feature Description Privacy Settings

User Target
Can Intrude Cannot Be Private Call
Intruded
Call Intrude This feature allows Used Used Used
you to intrude on
the existing
connected call of
the specified target
user. All call parties
are put into a
conference and
can talk to and
hear each other. A
Call Intrude
attempt to a user
who is idle
becomes a Priority
Call.
Table continues…

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Configure User Settings

Feature Description Privacy Settings

User Target
Can Intrude Cannot Be Private Call
Intruded
Call Steal This function can Used Used Used
be used with or
without a specified
user target.
If the target has
alerting calls, the
function will
connect to the
longest waiting call.
If the target has no
alerting calls but
does have a
connected call, the
function will take
over the connected
call, disconnecting
the original user.
This usage is
subject to the Can
Intrude setting of
the Call Steal user
and the Cannot Be
Intruded setting of
the target.
If no target is
specified, the
function attempts to
reclaim the user's
last ringing or
transferred call if it
has not been
answered or has
been answered by
voicemail.
Table continues…

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 Call Intrusion

Feature Description Privacy Settings

User Target
Can Intrude Cannot Be Private Call
Intruded
Dial Inclusion This feature allows Used Used Used
you to intrude on
another user's call
to talk to them.
Their current call is
put on hold while
you talk and
automatically
reconnected when
you end the
intrusion. The
intruder and the
target extension
can then talk but
cannot be heard by
the other party.
This can include
intruding into a
conference call,
where the
conference will
continue without
the intrusion target.

During the intrusion

all parties hear a
repeated intrusion
tone. When the
intruder hangs-up
the original call
parties are
reconnected.
Attempting to hold
a dial inclusion call
simply ends the
intrusion. The
inclusion cannot be
parked.
Table continues…

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Configure User Settings

Feature Description Privacy Settings

User Target
Can Intrude Cannot Be Private Call
Intruded
Whisper Page This feature allows Used Used Not Used
you to intrude on
another user and
be heard by them
without being able
to hear the user's
existing call which
is not interrupted.
For example: User
A is on a call with
user B. When user
C intrudes on user
A, they can be
heard by user A but
not by user B who
can still hear user
A. Whisper page
can be used to talk
to a user who has
enabled private
call.
Coaching Intrusion This feature allows Used Used Used
the you to intrude
on another user's
call and to talk to
them without being
heard by the other
call parties to which
they can still talk.
For example: User
A is on a call with
user B. When user
C intrudes on user
A, they can hear
users A and B but
can only be heard
by user A.
Table continues…

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 Call Intrusion

Feature Description Privacy Settings

User Target
Can Intrude Cannot Be Private Call
Intruded
Request Coaching This feature allows Used Used Used
Intrusion you to request a
call intrusion. While
on a call, user A
indicates to user C
a request for
coaching support.
When user C
intrudes on user A,
they can hear
users A and B but
can only be heard
by user A.
Call Appearance In addition to Not Used Used Used
Bridged making and Not Used Used Used
Appearance answering calls,
appearance
Line Appearance buttons that Not Used Used Used
indicate 'in use
elsewhere' can be
pressed in order to
join that call.
The Can Intrude
setting of the user
is not used. The
Cannot Be
Intruded setting of
the longest present
internal party in the
call is used.

Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
Intrusion Privacy Controls
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.

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Configure User Settings

For intrusion using appearance buttons, the user's Can Intrude setting is not used. The Cannot
Be Intruded setting of the longest present internal party in the call is used.
A user who can normally be intruded on can indicate that a call is a private call by using a Private
Call short code or programmable button. While private call status is enabled, no intrusion is
allowed except for Whisper Page intrusion.
In addition to the options above, Call Listen can only be used to intrude on calls by users in the
user's Monitor Group (User | Telephony | Supervisor Settings).
For the Call Steal function, the Can Be Intruded setting is used if the call is connected.
Related links
Configure User Settings on page 651

Call Tagging
Call tagging associates a text string with a call. That string remains with the call during transfers
and forwards. That includes calls across a multi-site network.
On Avaya display phones, the text is shown whilst a call is alerting and is then replace by the
calling name and number when the call is connected. On analog phones with a caller ID display,
the tag text replace the normal caller information.
Applications such as SoftConsole display any call tag associated with a call. If the call is parked,
the tag is shown on the call park slot button used. A call tag can be added when making a call
using SoftConsole or one-X Portal. A tag can be added to a call by an Incoming Call Route or by
an Voicemail Pro Assisted Transfer action.
Related links
Configure User Settings on page 651

Call Waiting
Call waiting allows a user who is already on a call to be made aware of a second call waiting to be
answered.
User Call Waiting
Call waiting is primarily a feature for analog extension users. The user hears a call waiting tone
and depending on the phone type, information about the new caller may be displayed. The call
waiting tone varies according to locale.
For Avaya feature phones with multiple call appearance buttons, call waiting settings are ignored
as additional calls are indicated on a call appearance button if available.

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 Call Restriction

To answer a call waiting, either end the current call or put the current call on hold, and then
answer the new call. Hold can then be used to move between the calls.
Call waiting for a user can be enabled through the system configuration (User | Telephony | Call
Settings | Call Waiting On) and through programmable phone buttons.
Call waiting can also be controlled using short codes. The following default short codes are
available when using Call Waiting.
*15 - Call Waiting On Enables call waiting for the user.
*16 - Call Waiting Off Disables call waiting for the user.
*26 - Clear Call and Answer Call Waiting Clear the current call and pick up the waiting call.
Hunt Group Call Waiting
Call waiting can also be provided for hunt group calls. The hunt group Ring Mode must be
Collective Call Waiting.
On phones with call appearance buttons, the call waiting indication takes the form of an alert on
the next available call appearance button. On other phones, call waiting indication is given by a
tone in the speech path (the tone is locale specific).
The user's own Call Waiting setting is overridden when they are using a phone with call
appearances. Otherwise the user's own Call Waiting setting is used in conjunction with the hunt
group setting.
Related links
Configure User Settings on page 651

Call Restriction
Call barring can be applied in a range of ways.
Barring a User From Receiving Any External Calls For each user, Incoming Call Bar (User |
Telephony | Supervisor Settings) can be selected to stop that user from receiving any external
calls.
Barring a User From Making Any External Calls For each user, Outgoing Call Bar (User |
Telephony | Supervisor Settings) can be selected to stop that user from making any external calls.
Barring Particular Numbers/Number Types System short codes are used to match user dialing
and then perform a specified action. Typically the action would be to dial the number to an external
line. However, short codes that match the dialing of particular numbers or types of numbers can
be added and set to another function such as Busy. Those short codes can be added to a
particular user, to a User Rights associated with several users or to the system short codes used
by all users.
The system allows short codes to be set at user, user rights, system and least cost route. These
have a hierarchy of operation which can be used to achieve various results. For example a system

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Configure User Settings

short code for a particular number can be set to busy to bar dialing of that number. For a specific
user, a user short code match to the same number but set to Dial will allow that user to override
the system short code barring.
Using Account Codes The system configuration can include a list of account codes. These can
be used to restrict external dialing only to users who have entered a valid account code.
• Forcing Account Code Entry for a User A user can be required to enter an account code
before the system will return dialing tone. The account code that they enter must match a
valid account code stored in the system configuration. The setting for this is Force Account
Code (User | Telephony | Supervisor Settings).
• Forcing Account Code Entry for Particular Numbers Each system short code has a Force
Account Code option. Again the account code entered must match a valid account code
stored in the system configuration. for the call to continue.
Barring External Transfers and Forwards A user cannot forward or transfer calls to a number
which they cannot normally dial. In addition there are controls which restrict the forwarding or
transferring of external calls back off-switch. See Off-Switch Transfer Restrictions.
Related links
Configure User Settings on page 651

Centralized Call Log

The system can store a centralized call log for users. Each users' centralized call log can contain
up to 60 call records for user calls (30 on IP500 V2). When this limit is reached, each new call
records replaces the oldest previous record.
On Avaya phones with a fixed Call Log or History button (1400, 1600, 9500 and 9600 Series),
that button can be used to display the user's centralized call log. The centralized call log is also
used for M-Series and T-Series phone. The user can use the call log to make calls or to store as a
personal speed dial. They can also edit the call log to remove records. The same call log is also
used if the user logs into one-X Portal.
The centralized call log moves with the user if they log on and off from different phones. This
includes if they hot desk within a network.
Call Log Information
The following information is included in each centralized call log record:
Information Description
Table continues…

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 Centralized Call Log

Name The name, of the caller or the party called, if

available. Up to 31 characters.
This text is similar to that shown on the phone
display of phones when they receive the call. For
example, on forwarded call details of the original
target and the caller name are included, eg. Bob >
Sue.
Number The number associated with the call. Up to 31
digits.
Tag A text tag can be associated with calls by several
different methods. See Call Tagging. Up to 31
characters. The tag is not shown within the call log
display on phones.
Time and Date The time and date of the call using the system time.
Duration The call duration. For outgoing and answered calls
this is the call connection time. For missed calls this
is the call ringing time.
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Configure User Settings

Record Type Call log records can be Incoming, Outgoing or

Missed. Note that these are calls to or from the
user, not the phone, so it can include calls handled
through a twinned device such as when using
mobile call control.
Incoming Calls to the user that the user then
answered. This includes calls that the user answers
on a twinned device. This also includes outgoing
calls that are transferred to and answered by the
user.
Outgoing Calls made by the user.
Missed Calls to the user that they did not answer.
This includes calls while the user is logged off or in
Do Not Disturb state.
• Missed call records include an indication of what
happened to the missed call. Options are
Answered by Another, Answered by Voicemail
or Lost (not answered on the system).
• Missed call records are also marked as either
acknowledged or unacknowledged. If the user's
call log contains any unacknowledged call log
records, the Call Log lamp is lit when using a
1608 or 1616 phone. From the phone, viewing an
unacknowledged record changes it to
acknowledged.
• If the user has also be configured to included
missed hunt group calls in their call log, those are
also marked as acknowledged or
unacknowledged.
Count The number of times a matching call has been
logged. A matching call is one with the same name,
number and type. Only one record is kept for
matching calls, with the count increased by 1 and
using the time and date of the most recent matching
call.

If missed hunt group calls are also being logged, the system stores up to 10 call records for each
hunt group. When this limit is reached, new call records replace the oldest record.
Controlling Centralized Call Logging
The following controls exist for which users have their calls included in the centralized call log and
which calls are included.
User Setting
The user centralized call log settings can be set through the user configuration (User | Telephony |
Call Log) or through their associated user rights (User Rights | Telephony | Call Log).

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 Centralized Call Log

Centralized Call Log: Default = System Default (On) This setting allows the use of centralized
call logging to be enabled or disabled on a per user basis. The default is to match the system
setting Default Centralized Call Log On (System | Telephony | Call Log). The other options are On
or Off for the individual user. If off is selected, the call log shown on the users phone is the local
call log stored by the phone.
System Settings (System | Telephony | Call Log)
Default Centralized Call Log On: Default = On. When selected, each user is defaulted to have
the system store a call log of their calls. This call log is accessible on the phone when the user is
using a phone with a Call Log or History button. The use of centralized call logging can be
enabled/disabled on a per user basis using the Centralized Call Log user setting (User | Telephony
| Call Log).
Log Missed Calls Answered at Coverage: Default = Off. This setting controls how calls to a
user, that are answered by a covering user should be logged in the centralized call log. This option
applies for calls answered elsewhere (covered) by pickup, call coverage (call coverage buttons or
coverage group), bridged appearance button, user BLF, voicemail, etc.
Setting Targeted User Covering User
Off (Default) Nothing Answered Call
On Missed Call Answered Call

Log Missed Hunt Group Calls: Default = Off. By default, hunt group calls are not included in any
user's centralized call log unless answered by the user. If this option is selected, a separate call
log is kept for each hunt group of calls that are not answered by anyone. It includes hunt group
calls that go to voicemail.
If missed hunt group calls are also being logged, the system stores up to 10 call records for each
hunt group. When this limit is reached, new call records replace the oldest record.
Within the user call log setting (User | Telephony | Call Log), the list of hunt groups allows
selection of which hunt groups' missed call records should be displayed as part of the user's
centralized call log.
Call Scenarios
This is not a comprehensive list. However it summarizes how the user call log is used in some
common call scenarios.
Scenarios User Call Log Notes
Authorization/Account Codes Account and authorization codes used as part of a
call are not included in user call logs.
Automatic Callback If answered, they will show as an outgoing call to
the target.
Application Calls Calls made and answered using applications
(including CTI interfaces) are logged as if the user
made or answered the call using an extension.
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Configure User Settings

Scenarios User Call Log Notes

Conference Calls Conference calls are not included in the user call
log.
Hold When a user holds and then un-holds a call, the call
duration includes the time the call was on hold.
Follow-Me Calls to the user still appear in their user call log.
The follow me calls do not appear in the user call
log of the user who was the follow me destination.
Forward on Busy If the forwarded call is answered, the forwarding
Forward on No Answer user will have a Missed - Answered by Other call
log record.
If the forwarded call times out to voicemail, the user
will have a Missed - Answered by Voicemail call
log record.
Forward Unconditional When forwarding to another number, there will be
no record of forwarded calls in the forwarding users
call log.
When using the To Voicemail option, the forwarded
call will by logged as a Missed - Answered by
Voicemail call record.
Page Calls Page calls are not included in any user call logs
unless the page is answered (by pressing
Conference). When answered the page is logged as
a normal call between the two users involved.
Park Retrieving a call from Park (even if the user is the
one who parked the call) is logged as a incoming
call.
Short Codes Calls are only logged if they result in a call being
made or a call being answered. Calls made using
Break Out are not included.
Suppressed Digits Calls made with digit suppression enabled (AD
Suppress button) are not included in the users call
log.
Transfers If the user answers and accepts a supervised
transfer, they will have a incoming calls records.
One for the transfer enquiry call and one for the
transferred call.
If the user is the target of an unsupervised transfer,
they will have an Incoming or Missed call log.
Note that even if the call being transferred was
originally an outgoing call, for the user answering
the transfer it is logged as a incoming call.
Table continues…

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 Centralized Personal Directory

Scenarios User Call Log Notes

Twinning and Mobility When a user has a twinned device (either internal
twinning or mobile twinning), the user's call log
operates regardless of which device the user uses
to make or answer calls.
Calls between the twinned devices, ie. the user
transferring a call between devices, are not included
in their call log.
This includes calls made using mobile call control or
a one-X Mobile client.

Multi-Site Network
The user's call log records are stored by the system that is their home system, ie. the one on
which they are configured. When the user is logged in on another system, new call log records are
sent to the user's home system, but using the time and date on the system where the user is
logged in.
Hunt group call log records are stored on the system on which the hunt group is configured.
Related links
Configure User Settings on page 651

Centralized Personal Directory

Each system user is able to have up to 250 personal directory records stored by the system. A
user's personal directory is also usable with 1400, 1600, 9500 and 9600 Series phones with a
CONTACTS button. The user can view these records and use them to make calls.
1400, 1600, 9500 and 9600 Series phone users can edit their personal directory records through
the phone. The user personal directory records can be edited using the Manager User | Personal
Directory menu.

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Configure User Settings

When the user hot desks to another phone that supports the centralized personal directory, their
personal directory records become accessible through that phone. That also includes hot desking
to another system in the network.
Users can also use and edit their personal directory records using one-X Portal for IP Office. Note
that using one-X Portal for IP Office, users can have more personal directory records, with excess
records stored by the one-X Portal server.
Related links
Configure User Settings on page 651

Account Code Configuration

Forcing Account Code Entry for Specific Numbers
Account code can be set a being required for any dialing that matches a particular short code.
This is done by ticking the Force Account Code option found in the short code settings. Note that
the account code request happens when the short code match occurs. Potentially this can be in
the middle of dialing the external number, therefore the use of X wildcards in the short code to
ensure full number dialing is recommended.
Entering Account Codes
The method for entering account codes depends on the type of phone being used. Refer to the
relevant telephone User's Guide for details.
Account Code Button:
The Account Code Entry action (User | Button Programming | Emulation | Account Code
Entry) and Set Account Code action (User | Button Programming | Advanced | Set | Set
Account Code) can be assigned to a programmable button on some phones. They both operate

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 Coverage Groups

the same. The button can be preset with a specific account code or left blank to request account
code entry when pressed. The button can then be used to specify an account code before a call or
during a call.
Setting an Account Code using Short Codes:
The Set Account Code feature allows short codes to be created that specify an account code
before making a call.
Show Account Code Setting :
This setting on the System | Telephony | Telephony tab controls the display and listing of system
account codes.
When on and entering account codes through a phone, the account code digits are shown while
being dialed.
When off and eentering account codes through a phone, the account code digits are replaced by s
characters on the display.
Server Edition Account Code Management
Accounts codes configured on Server Edition are shared by all systems in the network.
Related links
Configure User Settings on page 651
Setting a User to Forced Account Code on page 673

Setting a User to Forced Account Code

Procedure
1. Receive the system configuration if one is not opened.
2. In the left-hand panel, click User. The list of existing user is shown in the right-hand panel.
3. Double-click the required user.
4. Select the Telephony tab.
5. Tick the Force Account Code option.
6. Click OK.
7. Merge the configuration.
Related links
Account Code Configuration on page 672

Coverage Groups
For users with a Coverage Group selected, coverage group operation is applied to all external
calls that are targeted to the user.

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Configure User Settings

For external calls:

In scenarios where an external call would normally have gone to voicemail, it instead continues
ringing and also starts alerting the members of the coverage group.
• The follow me settings of Coverage Group members are used, the forwarding settings are
not.
• If the user is not available, for example if they have logged off or set to do not disturb,
coverage group operation is applied immediately.
• If the user is configured for call forward on busy, coverage operation is applied to the user's
calls forwarded to the forward on busy destination.
Coverage group operation is not applied to the following types of call:
Hunt group calls.
Recall calls such as transfer return, hold recall, park recall, automatic callback.
The Coverage Group is set through the user's User | Telephony | Supervisor Settings or through
their associated User Rights | Telephony | Supervisor Settings. The only group settings used are:
• The list of group members. They are treated as a collective group regardless of the group's
configuration.
• If the group has Night Server Fallback Group and or Out of Service Fallback Group set,
the members of those groups are used if the coverage group is set to night service mode or
out of service mode respectively.
Related links
Configure User Settings on page 651

DND, Follow Me and Forwarding

This section contains topics looking at how users can have their calls automatically redirected. As
illustrated, there is an order of priority in which the redirect methods are used.

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 DND, Follow Me and Forwarding

Redirect priority
1. Do Not Disturb (DND) Redirect all calls to voicemail if available, otherwise return busy
tone. DND overrides all the redirect method below unless the calling number is in the
user's DND Exception Numbers List.
2. Follow Me Redirect all calls to another extension that the users is temporarily sharing.
Follow Me overrides Forward Unconditional. The Follow Me destination is busy or does not
answer, the user's Forward on Busy or Forward on No Answer options can be used if set.
3. Forward Unconditional Redirect the user's external calls to another number. That number
can be any number the user can normally dial including external numbers. Forwarding of
hunt group and internal calls is optional. Forward Unconditional overrides Forward on Busy
and Forward on No Answer.
If the destination is an internal user on the same system, they are able to transfer calls
back to the user, overriding the Forward Unconditional.
4. Forward on Busy Redirects the user's external calls when the system sees the user as
being busy. Uses the same number as Forward Unconditional unless a separate Forward
on Busy Number is set. Forwarding internal calls is optional. Forward on Busy overrides
Forward on No Answer.
5. Forward on No Answer Redirects the user's external calls when they ring for longer than
the user's No Answer Time. Uses the same number as Forward Unconditional unless a
separate Forward on Busy Number is set. Forwarding internal calls is optional.

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Configure User Settings

Retrieving Externally Forwarded Calls:

Where a call is forwarded to an external destination and receives busy or is not answered within
the forwarding user's No Answer Time, the system will attempt to retrieve the call. If forwarded on
a trunk that does not indicate its state the call is assumed to have been answered, for example
analog loop start trunks.
Off-Switch Forwarding Restrictions:
User forwarding is subject to the same restrictions as transferring calls. To bar a user from
forwarding calls to an external number, the Inhibit Off-Switch Forward/Transfers (User |
Telephony | Supervisor Settings) option. To bar all users from forwarding calls to external
numbers the Inhibit Off-Switch Forward/Transfers option can be used.
When transferring a call to another extension that has forwarding enabled, the type of call being
transferred is used. For example, if transferring an external call, if the transfer target has
forwarding of external calls enabled then the forward is used.
Block Forwarding:
The Block Forwarding setting is used for enforcing predictable call routing, where the call should
always go to the same destination. This setting was implemented for contact center applications.
Block Forwarding can be set for a user on the User | Forwarding page or as a user rights setting
on the User Rights | Forwarding page.
Related links
Configure User Settings on page 651
Do Not Disturb (DND) on page 676
Follow Me on page 678
Forward Unconditional on page 680
Forward on Busy on page 682
Forward on No Answer on page 684
Determining a User's Busy Status on page 686
Chaining on page 687

Do Not Disturb (DND)

Summary: Redirect all calls to busy tone or to voicemail if available except those in your DND
exceptions list.
Do Not Disturb (DND) is intended for use when the user is present but for some reason does not
want to be interrupted. Instead calls are sent to voicemail if available, otherwise they receive busy
tone.
Exceptions Specific numbers can be added to the user's Do Not Disturb Exception List. Calls
from those numbers override DND. N and X wildcards can be used at the end of exception
numbers to match a range of numbers. For external numbers, this uses the incoming caller line ID
(ICLID) received with the call.

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Priority Enabling DND overrides any Follow Me or forwarding set for the user, except for calls in
the user's Do Not Disturb Exception List.
Phone When enabled, the phone can still be used to make calls. An N is displayed on many
Avaya phones. When a user has do not disturb in use, their normal extension will give alternate
dialtone when off hook.
Applied to
Call Types Blocked Call Treatment
Internal Voicemail if available, otherwise
busy tone.
External Voicemail if available, otherwise
busy tone.
Hunt Group Call not presented (DND
exceptions are not used).
Page Call not presented.
Follow Me Rings.
Forwarded Busy.
VM Ringback Rings
Automatic Callback Rings
Transfer Return Rings.
Hold Return Rings.
Park Return Rings.
Twinning Voicemail if available, otherwise
busy tone.

Do Not Disturb and Twinning

Mobile Twinning Selecting DND disables mobile twinning.
Internal Twinning
• Logging out or setting do not disturb at the primary stops twinned calls alerting at the
secondary also.
• Logging out or setting do not disturb at the secondary only affects the secondary.
Do Not Disturb Exceptions List For both types of twinning, when DND is selected, calls from
numbers entered in the user's Do Not Disturb Exception List are presented to both the primary
and secondary phones.
Do Not Disturb Controls
Do Not Disturb
Manager A user's DND settings can be viewed and changed
through the User | DND tab within the system
configuration settings.
Table continues…

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Configure User Settings

Controls The following short code features/button

programming actions can be used:

Voicemail If voicemail is available, it is used instead of busy

tone for callers not in the users exceptions list.
For Voicemail Pro, the Play Configuration Menu
action can be used to let callers switch DND on or
off.
SoftConsole A SoftConsole user can view and edit a user's DND
settings except exception numbers. Through the
directory, select the required user. Their current
status including DND is shown. Double-click on the
details to adjust DND on or off.

Feature/Action Short Code Default Button

Do Not Disturb On *08 - Toggles.
Do Not Disturb Off *09
Do Not Disturb *10*N#
Exception Add
Do Not Disturb *11*N#
Exception Delete
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 674

Follow Me
Summary: Have your calls redirected to another user's extension, but use your coverage,
forwarding and voicemail settings if the call receives busy tone or is not answered.
Follow Me is intended for use when a user is present to answer calls but for some reason is
working at another extension such as temporarily sitting at a colleague's desk or in another office
or meeting room. Typically you would use Follow Me if you don't have a Hot Desking log in code
or if you don't want to interrupt your colleague from also receiving their own calls, ie. multiple users
at one phone.
Priority Follow Me is overridden by DND except for callers in the user's DND Exception Numbers
List. Follow Me overrides Forward Unconditional but can be followed by the user's Forward on
Busy or Forward on No Answer based on the status of the Follow Me destination.
Destination The destination must be an internal user extension number. It cannot be a hunt group
extension number or an external number.

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 DND, Follow Me and Forwarding

Duration The Follow Me user's no answer timeout is used. If this expires, the call either follows
their Forward on No Answer setting if applicable, or goes to voicemail is available. Otherwise the
call continues to ring at the destination.
Phone When enabled, the phone can still be used to make calls. When a user has follow me in
use, their normal extension will give alternate dialtone when off hook.
Exceptions
• The Follow Me destination extension can make and transfer calls to the follow me source.
• The call coverage settings of the user are applied to their Follow Me calls. The call coverage
settings of the destination are not applied to Follow Me calls it receives.
Calls Forwarded
Call Types Redirected
Internal Redirected.
External Redirected.
Hunt Group Redirected*.
Page Redirected.
Follow Me Not redirected.
Forwarded Redirected.
VM Ringback Not redirected.
Automatic Callback Not redirected.
Transfer Return Not redirected.
Hold Return Not redirected.
Park Return Not redirected.

*Except calls for "Longest Waiting" type hunt groups.

Follow Me Controls
Follow Me
Manager A user's Follow Me settings can be viewed and
changed through the User | Forwarding tab within
the system configuration settings. Note that on this
tab, entering a Follow Me Number also enables
Follow Me.
Controls The following short code features/button
programming actions can be used:

Table continues…

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Configure User Settings

Voicemail For calls initially targeted to the user but then

redirected, when voicemail is invoked the mailbox of
the user is used and not the mailbox of the
destination.
For Voicemail Pro, the Play Configuration Menu
action can be used to let callers alter or set their
current Follow Me destination.
SoftConsole A SoftConsole user can view and edit a user's
Follow Me settings. Through the directory, select
the required user. Their current status including
Follow Me is shown. Double-click on the details and
select Forwarding to alter their forwarding settings
including Follow Me.

Feature/Action Short Code Default Button

Follow Me Here *12*N#
Follow Me Here Cancel *13*N#
Follow Me To *14*N#
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 674

Forward Unconditional
Summary: Have your calls redirected immediately to another number including any external
number that you can dial.
Priority This function is overridden by DND and or Follow Me if applied. Forward Unconditional
overrides Forward on Busy.
Destination The destination can be any number that the user can dial. If external and Inhibit Off-
Switch Transfers is applied, the caller is directed to voicemail if available, otherwise they receive
busy tone. If the destination is an internal user on the same system, they are able to transfer calls
back to the user, overriding the Forward Unconditional.
Duration After being forwarded for the user’s no answer time, if still unanswered, the system can
apply additional options. It does this if the user has forward on no answer set for the call type or if
the user has voicemail enabled.
• If the user has forward on no answer set for the call type, the call is recalled and then
forwarded to the forward on no answer destination.
• If the user has voicemail enabled, the call is redirected to voicemail.
• If the user has both options set, the call is recalled and then forwarded to the forward on no
answer destination for their no answer time and then if still unanswered, redirected to
voicemail.

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• If the user has neither option set, the call remains redirected by the forward unconditional
settings.
Note that for calls redirected via external trunks, detecting if the call is still unanswered requires
call progress indication. For example, analog lines do not provide call progress signalling and
therefore calls forwarded via an analog lines are treated as answered and not recalled.
Phone When enabled, the phone can still be used to make calls. An D is displayed on DS phones.
When a user has forward unconditional in use, their normal extension will give alternate dialtone
when off hook.
Calls Forwarded Once a call has been forwarded to an internal destination, it will ignore any
further Forward No Answer or Forward on Busy settings of the destination but may follow
additional Forward Unconditional settings unless that creates a loop.
Call Types Forwarded
Internal Optional.
External Forwarded.
Hunt Group Optional.*
Page Not presented.
Follow Me Rings.
Forwarded Forwarded.
VM Ringback Rings.
Automatic Callback Rings.
Transfer Return Rings.
Hold Return Ring/hold cycle.
Park Return Rings.

*Optional only for calls targeting sequential and rotary type groups. Includes internal call to a hunt
group regardless of the forward internal setting.
To Voicemail: Default = Off. If selected and forward unconditional is enabled, calls are forwarded
to the user's voicemail mailbox. The Forward Number and Forward Hunt Group Calls settings
are not used. This option is not available if the system's Voicemail Type is set to None. 1400,
1600, 9500 and 9600 Series phone users can select this setting through the phone menu. Note
that if the user disables forward unconditional the To Voicemail setting is cleared.
Forward Unconditional Controls
Forward Unconditional
Manager A user's forwarding settings can be viewed and
changed through the User | Forwarding tab within
the system configuration settings.
Controls The following short code features/button
programming actions can be used:

Table continues…

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Configure User Settings

Voicemail For calls initially targeted to the user but then

redirected, when voicemail is invoked the mailbox of
the user is used and not the mailbox of the
destination.
For Voicemail Pro, the Play Configuration Menu
action can be used to let callers set their current
forwarding destination and switch Forwarding
Unconditional on/off.
SoftConsole A SoftConsole user can view and edit a user's
forwarding settings. Through the directory, select
the required user. Their current forwarding status is
shown. Double-click on the details and select
Forwarding to alter their forwarding settings.

Feature/Action Short Code Default Button

Forward Number *07*N#
Forward Unconditional *01 - Toggles.
On
Forward Unconditional *02
Off
Forward Hunt Group - Toggles.
Calls On
Forward Hunt Group
Calls Off
Disable Internal
Forwards
Enable Internal
Forwards
Disable Internal
Forwards Unconditional
Enable Internal
Forwards Unconditional
Set No Answer Time
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 674

Forward on Busy
Summary: Have your calls redirected when you are busy to another number including any external
number that you can dial.

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 DND, Follow Me and Forwarding

The method by which the system determines if a user is 'busy' to calls depends on factors such as
whether they have multiple calls appearance buttons or Call Waiting and or Busy on Held set. See
Busy.
Priority This function is overridden by DND and or Forward Unconditional if applied. It can be
applied after a Follow Me attempt. It overrides Forward on No Answer.
Destination The destination can be any number that the user can dial. The Forward Unconditional
destination number is used unless a separate number Forward on Busy Number is set. If Inhibit
Off-Switch Transfers is applied, the caller is directed to voicemail if available, otherwise they
receive busy tone.
Duration The destination is rung using the forwarding user's No Answer Time. If this expires, the
call goes to voicemail is available. Calls to an external destination sent on trunks that do not signal
their state are assumed to have been answered, for example analog loop start trunks.
Phone Forward on Busy is not indicated and normal dial tone is used.
Calls Forwarded Once a call has been forwarded to an internal destination, it will ignore any
further Forward No Answer or Forward on Busy settings but may follow additional Forward
Unconditional settings.
Call Types Forwarded
Internal Optional.
External Forwarded.
Hunt Group Not presented.
Page Not presented.
Follow Me Rings.
Forwarded Forwarded.
VM Ringback Rings.
Automatic Callback Rings.
Transfer Return Rings.
Hold Return Ring/hold cycle.
Park Return Rings.

Forward on Busy Controls

Forward on Busy
Software Level A user's forwarding settings can be viewed and
changed through the User | Forwarding tab within
the system configuration settings.
Controls The following short code features/button
programming actions can be used:

Table continues…

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Configure User Settings

Voicemail For calls initially targeted to the user but then

redirected, when voicemail is invoked the mailbox of
the user is used and not the mailbox of the
destination.
For Voicemail Pro, the Play Configuration Menu
action can be used to let callers set the forward
destination.
SoftConsole A SoftConsole user can view and edit a user's
forwarding settings. Through the directory, select
the required user. Their current forwarding status is
shown. Double-click on the details and select
Forwarding to alter their forwarding settings.

Feature/Action Short Code Default Button

Forward Number *07*N#
Forward on Busy *57*N#
Number
Forward on Busy On *03 - Toggles.
Forward on Busy Off *04
Disable Internal
Forwards
Enable Internal
Forwards
Disable Internal
Forwards Busy or No
Answer
Enable Internal
Forwards Busy or No
Answer
Set No Answer Time
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 674

Forward on No Answer
Summary: Have your calls redirected another number if it rings without being answered.
Priority This function is overridden by DND and Forward on Busy if applied. It can be applied after
a Follow Me attempt. Forward Unconditional overrides Forward on Busy and Forward on No
Answer.

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Destination The destination can be any number that the user can dial. The Forward Unconditional
destination number is used unless a separate number Forward on Busy Number is set. If Inhibit
Off-Switch Transfers is applied, the caller is directed to voicemail if available, otherwise they
receive busy tone.
Duration The destination is rung using the forwarding user's No Answer Time. If this expires, the
call goes to voicemail is available. Otherwise the call continues to ring at the destination. Calls to
an external destination sent on trunks that do not signal their state are assumed to have been
answered, for example analog loop start trunks.
Phone Forward on No Answer is not indicated and normal dial tone is used.
Calls Forwarded Once a call has been forwarded to an internal destination, it will ignore any
further Forward No Answer or Forward on Busy settings but may follow additional Forward
Unconditional settings.
Call Types Forwarded
Internal Optional.
External Forwarded.
Hunt Group Not applicable.
Page Not applicable.
Follow Me Rings.
Forwarded Forwarded.
VM Ringback Rings.
Automatic Callback Rings.
Transfer Return Rings.
Hold Return Ring/hold cycle.
Park Return Rings.

Forward on No Answer Controls

Forward on No Answer
Manager A user's forwarding settings can be viewed and
changed through the User | Forwarding tab within
the system configuration settings.
Controls The following short code features/button
programming actions can be used:

Table continues…

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Configure User Settings

Voicemail For calls initially targeted to the user but then

redirected, when voicemail is invoked the mailbox of
the user is used and not the mailbox of the
destination.
For Voicemail Pro, the Play Configuration Menu
action can be used to let callers set the forward
destination. It cannot however be used to enable
Forward on Busy or set a separate Forward on
Busy number.
SoftConsole A SoftConsole user can view and edit a user's
forwarding settings. Through the directory, select
the required user. Their current forwarding status is
shown. Double-click on the details and select
Forwarding to alter their forwarding settings.

Feature/Action Short Code Default Button

Forward Number *07*N#
Forward on Busy *57*N#
Number
Forward on No Answer *05 - Toggles.
On
Forward on No Answer *06
Off
Enable Internal
Forwards
Disable Internal
Forwards
Enable Internal
Forwards Busy or No
Answer
Disable Internal
Forwards Busy or No
Answer
Set No Answer Time
Cancel All Forwarding *00

Related links
DND, Follow Me and Forwarding on page 674

Determining a User's Busy Status

Various system features allow users to handle more than one call at a time. Therefore the term
"busy" has different meanings. To other users it means whether the user is indicated as being
busy. To the system it means whether the user is not able to receive any further calls. The latter is

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 DND, Follow Me and Forwarding

used to trigger 'busy treatment', either using a user's Forward on Busy settings or redirecting
calls to voicemail or just returning busy tone.
Busy Indication - In Use The user busy indication provided to programmable buttons and to user
applications, is based on the monitored user's hook switch status. Whenever the user is off-hook,
they will be indicated as being busy regardless of call waiting or call appearance settings.
Busy to Further Calls Whether a user can receive further calls is based on a number of factors
as described below.
• Logged In and Present Is the user logged into an extension and is that extension physically
connected to the system.
• Busy on Held If a user enables their Busy on Held setting, whenever they have a call on
hold, they are no longer available to any further incoming calls.
• Appearance Buttons A user's call appearance button are used to receive incoming calls.
Normally, whilst the user has any free call appearance buttons, they are available to receive
further calls. Exceptions are:
- Reserve Last Appearance Users with appearance buttons require a free call appearance
button to initiate transfers or conferences. Therefore it is possible through the user's
configuration settings to reserve their last call appearance button for outgoing calls only.
- Other Appearance Buttons Calls may also be indicated on line, call coverage and
bridged appearance buttons.
Call Waiting Users of phones without appearance buttons can use call waiting. This adds an
audio tone, based on the system locale, when an additional call is waiting to be answered. Only
one waiting call is supported, any further calls receive busy treatment.
Hunt Group Calls A user's availability to receive hunt group calls is subject to a range of other
factors. See Member Availability.
Related links
DND, Follow Me and Forwarding on page 674

Chaining
Chaining is the process where a call forward to an internal user destination is further forwarded by
that user's own forwarding settings.
Follow Me Calls Follow Me calls are not chained. They ignore the forwarding, Follow Me and Do
Not Disturb settings of the Follow Me destination.
Voicemail If the call goes to voicemail, the mailbox of the initial call destination before forwarding
is used.
Looping When a loop would be created by a forwarding chain, the last forward is not applied. For
example the following are scenarios where A forwards to B, B forwards to C and C forwards to A.
In each case the final forward is not used as the destination is already in the forwarding chain.

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Hunt Group Loop If a user forwards a call to a hunt group of which they are a member, the group
call is not presented to them but is presented to other members of the hunt group.
Maximum Number of Forwards A maximum of 10 forwarding hops are supported for any call.
Calls Forwarded Once a call has been forwarded to an internal destination, it will ignore any
further Forward No Answer or Forward on Busy settings but may follow additional Forward
Unconditional settings.
Related links
DND, Follow Me and Forwarding on page 674

Hot Desking
Hot desking allows users to log in at another phone. Their incoming calls are rerouted to that
phone and their user settings are applied to that phone. There are a number of setting and
features which affect logging in and out of system phones.
In order to hot desk, a user must be assigned a Login Code (User | Telephony | Supervisor
Settings) in the system configuration.
By default, each system extension has an Base Extension setting. This associates the extension
with the user who has the matching Extension settings as being that extension's default
associated user.
• By leaving the Base Extension setting for an extension blank, it is possible to have an
extension with no default associated user. This is only supported for non-IP/CTI extensions.
Extensions in this state use the settings of a special user named NoUser. On suitable
phones the display may show NoUser.
• You can create users whose Extension directory number is not associated with any physical
extension. These users must have a log in code in order to log in at a phone when they need

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 Hot Desking

to make or receive calls. In this way the system can support more users than it has physical
extensions.
When another user logs in at an extension, they take control of that phone. Any existing user,
including the default associated user, is logged out that phone.
• Any user settings not applicable to the type of phone on which the user has logged in
become inaccessible. For example some programmable button features will become
inaccessible if the phone at which a user logs in does not have a sufficient number of
programmable buttons.
• Note that settings that are stored by the phone rather than by the system remain with the
phone and do not move when a user hot desks.
1400 Series, 1600 Series, 9500 Series, 9600 Series, M-Series and T-Series telephones all use the
centralized call log and centralized personal directory features that move those settings with the
user as they hot desk.
Other Avaya H.323 IP telephones can be configured to backup and restore user settings to a file
server when a user hot desks between phones. The range of settings supported depends on the
particular phone model. Refer to the IP Office H.323 IP Telephone Installation Manual.
For all other features and phone types, it must be assumed that any settings and data shown by
the phone is stored by the phone and are still accessible after logging off.
When a user logs off or is logged out by someone else logging in, they are automatically logged
back in at the extension for which they are the default associated user if no one else is logged in
at that extension. However this does not happen for users set to Forced Login (User | Telephony |
Supervisor Settings).
For each user, you can configure how long the extension at which they are logged in can remain
idle before they are automatically logged out. This is done using the Login Idle Period option. This
option should only be used in conjunction with Force Login.
Logged in users who are members of a hunt group can be automatically logged out if they do not
answer hunt group calls presented to them. This is done by selecting Logged Off as the user's
Status on No Answer (User | Telephony | Supervisor Settings) setting.
Calls to a logged out user are treated as if the user is busy until the user logs in.
Logging in and out at a phone can be done either using system short codes or programmable
buttons.
• The default system short code for logging in, is *35*N# where the user replaces N with their
extension number and then log in code separated by a *. This uses the short code feature
ExtnLogin. If the user dials just a log in code as N, it is checked against the user with the
same extension number as the extension's base extension number.
• The default system short code for logging out is *36. This uses the short code feature
ExtLogout.
• The ExtnLogin and ExtnLogout features can be assigned to programmable buttons on
suitable Avaya phones. The ExtnLogin button will then prompt the user to enter their details.

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Configure User Settings

Related links
Configure User Settings on page 651
Remote Hot Desking on page 690
Call Center Agents on page 691
Hot Desking Examples on page 691
Automatic Log Out on page 693

Remote Hot Desking

The system supports hot desking between systems within a network.
In the descriptions below, the system on which the user is configured is termed their 'home'
system, all other systems are 'remote' systems.
When a user logs in to a remote system:
• The user's incoming calls are rerouted to that system.
• The user's outgoing calls uses the settings of the remote system.
• The user's license privileges move with them, for example their user profile setting is
retained. The host system does not need to be licensed for the user.
• The user's own settings are transferred. However, some settings may become unusable or
may operate differently.
• User rights are not transferred to the remote system but the name of any user rights
associated with the user are transferred. If user rights with the same name exist on the
remote system, then they will be used. The same applies for user rights applied by time
profiles, if time profiles with the same name exist on the remote system .
• Appearance buttons configured for users on the home system will no longer operate.
• Various other settings may either no longer work or may work differently depending on the
configuration of the remote system at which the user has logged in.
• The rights granted to the user by their Profile settings are retained by the user. There is no
requirement for the remote system to have the appropriate licenses for the Profile.
If the user's home system is disconnected while the user is remotely hot desked, the user will
remain remotely hot desked. They can remain in that state unless the current host system is
restarted. They retain their license privileges as if they were on their home system. Note however
that when the user's home system is reconnected, the user may be automatically logged back
onto that system.
Break Out Dialing In some scenarios a hot desking user logged in at a remote system will want to
dial a number using the system short codes of another system. This can be done using either
short codes with the Break Out feature or a programmable button set to Break Out. This feature
can be used by any user within the multi-site network but is of most use to remote hot deskers.
Related links
Hot Desking on page 688

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 Hot Desking

Call Center Agents

On systems with a call center application such as Compact Contact Center (CCC) or Compact
Business Center (CBC), logging in and logging out is a key part of tracking and reporting on call
center agents. It also controls call distribution as, until the agent logs in, their hunt group
membership is seen as disabled.
For CCC, CBC and Delta Server, an agent is defined as being a user with a Login Code and set to
Forced Login. Those users consume a CCC agent license.
Related links
Hot Desking on page 688

Hot Desking Examples

The following are example of different ways that the hot desking settings can be used.
Related links
Hot Desking on page 688

Scenario 1: Occasional Hot Desking

About this task
In this scenario, a particular user, for this example extension 204, needs to occasionally work at
other locations within the building.
Procedure
1. A Login Code is added to the user's configuration settings, for this example 1234.
2. The user can now log in when needed at any other phone by dialing *35*204*1234#.
The phone's default associated user is logged out by this and their calls get busy
treatment. User 204 is also logged out their normal phone and their calls now rerouted to
the phone at which they have logged in.
3. When finished, the user can dial *36 to log out.
4. This logs the phone's normal default user back on.
Its also logs the hot desking user back on at their normal extension.

Scenario 2: Regular Hot Desking

About this task
This scenario is very similar to the one above. However, the user doesn't want to be automatically
logged back in on their normal phone until they return to its location.

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Configure User Settings

Procedure
1. A Login Code is added to the user's configuration settings, for this example 1234.
2. The Forced Login option is selected.
3. When the user logs out of the phone that they are currently using, they are no longer
automatically logged in on their normal extension.
When they return to it they must dial *35*204*1234# to log in.
4. Whilst not logged in anywhere, calls to the user receive busy treatment.

Scenario 3: Full Hot Desking

About this task
Similar to the scenarios above but this time the user doesn't have a regular phone extension that
they use. In order to make and receive calls they must find a phone at which they can log in.
Procedure
1. The user is given an Extension directory number that is not matched by the extension
directory number setting of any existing extension.

2. They are also given a Login Code and a Login Idle Period is set, for this example 3600
seconds (an hour).
Forced Login isn't required as the user has no default extension at which they might be
automatically logged in by the system.
3. The user can now log in at any available phone when needed.
4. If at the end of the business day they forget to log out, the Login Idle Period will eventually
log them off automatically.

Scenario 4: Call Center Hot Desking

About this task
In this scenario, the phone extensions have no default extension number. Several phones set like
this might be used in a call center where the agents use whichever desk is available at the start of
their shift. Alternatively a set of desks with such phones might be provided for staff that are
normally on the road but occasionally return to the office and need a temporary desk area to
complete paper work.
Procedure
1. For the extensions, the Extension setting is left blank.
This means that those phones will be associated with the NoUser user's settings and
display NOT LOGGED ON.
2. The call center agents or road-warrior users are configured with Extension directory
numbers that also don't match any existing physical extensions.

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 Hot Desking

They are all given Login Code numbers.

3. The users can log in at any of the extensions when required.
When they log out or log in elsewhere, the extensions return to the NoUser setting.

Automatic Log Out

Normally a user can either log themselves out or be logged out by another user logging in. The
following methods can be used by the system to automatically log out a user, so long as that user
has a Login Code and is set to Forced Login.
Note: A remote hot desking user whose home system can no longer be seen by the remote
system at which they are logged in is automatically logged out after 24 hours.
Idle Timeout:
The user Login Idle Period (User | Telephony | Supervisor Settings) can be used to
automatically log out the user after a set period of phone inactivity. The period can be set between
1 to 99999 seconds and is based on call inactivity other than ringing calls.

Unanswered Calls:
Users who are members of hunt groups are presented with hunt group calls when they are logged
in and not already on a call. If the user is logged in but not actually present they will continue to be
presented with hunt group calls. In this scenario it can be useful to log the user off.
• For the hunt group On the Hunt Group | Hunt Group tab, use the Agent's Status on No
Answer Applies to setting to select which types of unanswered hunt group calls should
change the user's status. The options are:
- None
- Any Calls
- External Inbound Calls Only
• For the user The Status on No Answer setting (User | Telephony | Supervisor Settings)
can be used. This sets what the user's status should be changed to if they do not answer a
hunt group call. The options are:
- Logged In If this option is selected, the user's status is not changed.
- Busy Wrap-Up If this option is selected, the user's membership status of the hunt group
triggering the action is changed to disabled. The user can still make and receive calls and
will still continue to receive calls from other hunt groups to which they belong.
- Busy Not Available If this option is selected, the user's status is changed to do not
disturb. This is the equivalent of DND and will affect all calls to the user.
- Logged Off If this option is selected, the user’s status is changed to logged out. In that
state the cannot make calls and cannot receive calls. Hunt group calls go to the next
available agent and personal calls treat the user as being busy.

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Configure User Settings

Related links
Hot Desking on page 688

Group Operation
A group is a collection of users accessible through a single directory number. Calls to that group
can be answered by any available member of the group. The order in which calls are presented
can be adjusted by selecting different group types and adjusting the order in which group
members are listed.

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 Group Operation

Fallback
checks

Availability
checks

Call
presentation

Queuing

Overflow

Voicemail

• Call Presentation: The order in which the available members of the group are used for call

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Configure User Settings

presentation is selectable.
• Availability: There are a range of factors which control whether group calls are presented to
a user in addition to that user being a member of the group.
• Queuing: This optional feature allows calls to be queued when the number of calls to be
presented exceeds the number of available group members to which call can be presented.
• Announcements: On systems with a voicemail server (Voicemail Pro or Embedded
Voicemail), announcements can be played to callers waiting to be answered. That includes
calls that are ringing and calls that are queued.
• Overflow: This optional feature can be used to include additional agents from an overflow
group or groups when a call is not answered.
• Fallback: A group can be taken out of operation manually or using a time profile. During
fallback, calls can be redirected to a fallback group or sent to voicemail or just receive busy
tone. Two types of fallback are supported; night service and out of service.
• Voicemail: Calls can be redirected to voicemail. The system allows selection of whether
group calls remain in the group mailbox or are copied (broadcast) to the individual mailboxes
of the group members. When messages are stored in the group's own mailbox, selection of
who receives message waiting indication is possible.
Group Editing
Changing the name of a group has the following effects:
• A new empty mailbox is created on voicemail with the new group name.
• Records in other groups' Overflow lists will be updated.
• Out-of-Service and Night-Service fallback references are updated.
Modifying the extension number of a group updates the following:
• Group buttons.
• Overflow, Out of Service Fallback and Night Service Fallback group records.
• Incoming call route records.
When a group is deleted, all references to the deleted group will be removed including:
• Records in Incoming call routing tables.
• Transfer target in internal auto-attendant.
• Overflow, Night-Service or Fallback-Service on other groups.
• DSS keys monitoring group status.
Server Edition Group Management
Groups can be stored in the configuration of any system in the network. Groups created at the
solution level on Manager and Web Manager are stored on the Primary Server. All groups can
include users from anywhere in the network and are automatically advertised to and diallable on
any of the systems in the network.
Groups configured on the Server Edition Primary by default fail over to the Server Edition
Secondary. Groups configured on a Server Edition Expansion System can be configured to fail
over to the Server Edition Primary, the Server Edition Secondary, or another Server Edition
Expansion System.

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 Group Operation

Groups in a Multi-Site Network

In a multi-site network, the extension numbers of users are automatically shared between systems
and become diallable from other systems without any further programming.
The following features are available for groups.
Advertised Groups:
Each group can be set as being 'advertised'. The group can then be dialed from other systems
within the multi-site network. The groups extension number and name must be unique within the
network. Non-advertised group numbers remain local only to system hosting the group.
Distributed Groups:
Groups on a system can include users located on other systems within the network. Distributed
groups are automatically advertised to other systems within the network. Note that distributed
groups can only be edited on the system on which they were created.
Related links
Configure User Settings on page 651
Group Types on page 697
Call Presentation on page 698
Group Member Availability on page 699
Example Hunt Group on page 701
CBC/CCC Agents and Hunt Groups on page 703

Group Types
At its most basic, a group’s settings consist of a group name, an extension number, a list of group
members and a hunt type selection. It is the last two settings which determine the order in which
incoming calls are presented to hunt group members.
The available group types are; Collective, Sequential, Rotary and Longest Waiting. These work
are follows:
Collective Group (formerly Group Group)
An incoming call is presented simultaneously to all
the available group members.

Sequential Group (formerly Hunt or Linear Group)

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Configure User Settings

An incoming call is presented to the first available

member in the list. If unanswered, it is presented to
the next available member in the list.
The next incoming call uses the same order. It is
presented to the available members starting again
from the top of the list.
Rotary Hunt Type (formerly Circular Group)
This hunt type operates similarly to Sequential.
However the starting point for call presentation is
the first available member after the last member to
answer a call.

Longest Waiting Hunt Type (formerly Idle or Most Idle)

This hunt type does not present calls to hunt group members in the order that they are listed. It presents
calls using the order of how long the available hunt group members have been idle.
An incoming call is first presented to the available member who has been idle the longest. If unanswered it
is presented to the next longest idle member.
Release 4.2+: Where hunt group calls are being presented to a twinned extension, the longest waiting
status of the user can be reset by calls answered at either their master or twinned extension.

Related links
Group Operation on page 694

Call Presentation
Summary: Calls are presented to each available hunt group member in turn. If having been
presented to all the available members, none answers, the call is redirected to voicemail if
available, otherwise it continues to be presented to the next available member.
In addition to the summary, options exist to have calls queued or to have calls also presented to
agents in an overflow group or groups.
First and Next Available Members The first available member to which a call is presented and
the order of the next available members to which a call is presented are determined by the hunt
group's Hunt Type setting.
Additional Calls When additional calls are waiting to be presented, additional available hunt
group members are alerted using the hunt group type. When any member answers a call it will be
the first waiting call that is answered.
No Available Members If the number of incoming calls exceeds the number of available
members to which calls can be presented, the following actions are usable in order of precedence.
Queuing If queuing has been enabled for the hunt, it is applied to the excess calls up to the limits
specified for the number of queued calls or length of time queued.

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 Group Operation

Voicemail If voicemail has been enabled for the hunt group, excess calls are directed to
voicemail.
Busy Tone Busy tone is returned to the excess calls (except analog and T1 CAS calls which
remain queued).
No Answer Time This value is used to determine how long a call should ring at a hunt group
member before being presented to the next available hunt group member. The System |
Telephony | Telephony | No Answer Time setting is used unless a specific Hunt | Hunt Group |
No Answer Time is set.
Voicemail If voicemail is being used, if having been presented to all the available group members
the call is still not answered then it goes to voicemail.
The call will also go to voicemail when the hunt group's Voicemail Answer Time is exceeded. the
mailbox of the originally targeted hunt group is used even if the call has overflowed or gone to a
night server hunt group.
Calls Not Being Answered Quick Enough - Overflow In addition to ringing at each available
member for the No Answer Time, a separate Overflow Time can be set. When a calls total ring
time against the group exceeds this, the call can be redirected to an overflow group or groups.
No Available Member Answers If a call has been presented unanswered to all the available
members, either of two actions can be applied. If voicemail is available, the call is redirected to
voicemail. If otherwise, the call will continue being presented to hunt group members until
answered or, if set, overflow is used.
Call Waiting For hunt groups using the Group hunt type, call waiting can be used.
Related links
Group Operation on page 694

Group Member Availability

Summary: Details when a hunt group member is seen as being available to be presented a hunt
group call.
The Hunt Group settings within Manager list those users who are members of the hunt group and
therefore may receive calls directed to that hunt group. However there are a range of factors that
can affect whether a particular hunt group member is available to take hunt group calls at any
time.
Existing Connected Call Users with an existing connected call are not available to further hunt
group calls. This is regardless of the type of connected call, whether the user has available call
appearance buttons or is using call waiting.
Hunt Group Call Waiting For Collective hunt groups call waiting can be enabled using the Ring
Type of Collective Call Waiting.
Logged In/Logged Out The system allows user's to log in and out extensions, a process known
as 'hot desking'. Whilst a user is logged out they are not available to receive hunt group calls.

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Configure User Settings

Mobile Twinning users with both Hunt group calls eligible for mobile twinning and Twin when
logged out selected will still receive hunt group calls unless they switch off twinning.
Membership Enabled/Disabled The system provides controls to temporarily disable a users'
membership of a hunt group. Whilst disabled, the user is not available to receive calls directed to
that hunt group.
Do Not Disturb This function is used by users to indicate that they do not want to receive any
calls. This includes hunt group calls. In call center environments this state is also known as 'Busy
Not Available'. See Do Not Disturb.
Busy on Held When a user has a held call, they can receive other calls including hunt group calls.
The Busy on Held settings can be used to indicate that the user is not available to further calls
when they have a held call.
Forward Unconditional Users set to Forward Unconditional are by default not available to hunt
group calls. The system allows the forwarding of hunt group calls to be selected as an option.
Idle /Off Hook The hunt group member must be idle in order to receive hunt group call ringing.
No Available Members If queuing has been enabled, calls will be queued. If queuing has not
been enabled, calls will go to the overflow group if set, even if the overflow time is not set or is set
to 0. If queuing is not enabled and no overflow is set, calls will go to voicemail. If voicemail is not
available, external calls go to the incoming call routes fallback destination while internal calls
receive busy indication.
Hunt Group Member Availability Settings
Manager Forwarding and do not disturb controls for a user
are found on the User | Forwarding and User | DND
tabs.
Enabling and disabling a users hunt group
membership is done by ticking or unticking the user
entry in the hunt group's extensions list on the Hunt
Group | Hunt Group tab.
Controls The following short code features/button
programming actions can be used:

SoftConsole A SoftConsole user can view and edit a user's

settings. Through the directory, select the required
user. Their current status including DND, Logged In
and hunt group membership states are shown and
can be changed. Forwarding settings can be
accessed by then selecting Forwarding.

Feature/Action Short Code Default Button

Hunt Group Enable HGEna - Toggles.
Hunt Group Disable HGDis
Forward Hunt Group On -*50 FwDH+ - Toggles
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 Group Operation

Feature/Action Short Code Default Button

Forward Hunt Group Off -*51 FwDH-
Busy on Held BusyH
Do Not Disturb On -*08 DNDOn - Toggles
Do Not Disturb Off -*09 DNDOf
Extn Login -*35*N# Login
Extn Logout -*36 Logof

Related links
Group Operation on page 694

Example Hunt Group

The follow are simple examples of how a department might use the facilities of a hunt group.
1. Basic Hunt Group
Scenario The Sales department want all sales related calls to
be presented first to Jane, then Peter and finally
Anne.
Actions 1. Create a hunt group named Sales and assign it
an extension number.
2. Set the Hunt Type to Sequential.
3. Add Jane, Peter and Ann to the User L ist in
that order.
4. Turn off queuing on the Queuing tab and
voicemail on the Voicemail tab.
5. Route relevant calls to the Sales group by
selecting it as the destination in the appropriate
Incoming Call Routes.
Results Any call received by the Sales hunt group is first
presented to Jane if she is available. If Jane is not
available or does not answer within 15 seconds the
call is presented to Peter. If Peter is not available or
does not answer within 15 seconds the call goes
Anne. Since voicemail is not on, the call will
continue to be presented around the group
members in that order until it is answered or the
callers hangs up.

2. Adding Voicemail Support

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Configure User Settings

Scenario A voicemail server has now been added to the

system. The Sales department wants to use it to
take messages from unanswered callers. When
messages are left, they want Jane to receive
message waiting indication.
Actions 1. Open the Sales hunt group settings and select
Voicemail On on the Voicemail tab.
2. Select the User settings for Jane. On the
Source Numbers tab, add the entry HSales.
Results Once a call to the Sales group has been presented
to all the available members, if it is still unanswered
then the call will be redirected to the group's
voicemail mailbox to leave a message. When a
message has been left, the message waiting
indication lamp on Jane's phone is lit.

3. Using the Queuing Facility

Scenario The Sales department now wants calls queued
when no one is available to answer. However if the
number of queued calls exceeds 3 they then want
any further callers directed to voicemail.
Actions 1. Open the Sales hunt group settings and select
Queuing On on the Queuing tab.
2. Set the Queue Limit to 3.
Results When the Sales group are all on calls or ringing,
any further calls to the group are queued and
receive queuing announcements from the voicemail
server. When the number of queued calls exceeds
3, any further calls are routed to the group's
voicemail mailbox.

4. Using Out of Service Fallback

Scenario During team meetings, the Sales department want
their calls redirected to another group, for this
example Support.
Actions 1. Open the Sales hunt group settings and select
the Fallback tab. In the Out of Service
Fallback Group field select the Support
group.
2. Create a system short code *88/Set Hunt
Group Out of Service/300.
3. Create a system short code *89/Clear Hunt
Group Out of Service/300.
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 Group Operation

Results Prior to team meetings, dialing *88 puts the Sales

group into out of service mode. Its calls are then
redirected to the Support group. Following the
meeting, dialing *89 puts the Sales group back In
Service.

5. Using a Night Service Time Profile

Scenario Outside their normal business hours, the Sales
department want their group calls automatically sent
to voicemail. This can be done using a time profile
and leaving the Night Service Fallback Group
setting blank.
Actions 1. Create a Time Profile called Sales Hours and
in it enter the times during which the Sales
department are normally available.
2. Open the Sales hunt group settings and select
the Fallback tab.
3. In the Time Profile field select Sales Hours.
Results Outside the normal business hours set in the time
profile, the Sales hunt group is automatically put
into Night Service mode. Since no Night Service
Fallback Group has been set, calls are redirected to
voicemail.

Related links
Group Operation on page 694

CBC/CCC Agents and Hunt Groups

The use of and reporting on hunt groups is a key feature of call center operation. For IP Office,
reporting is provided through the Compact Business Center (CBC) or Compact Contact Center
(CCC) applications.
In order for these applications to provide hunt group and hunt group user (agent) reports, the
following rules apply:
• The hunt group names must be restricted to a maximum of 12 characters.
• The hunt group and user extension numbers should be a maximum of 4 digits.
• Hunt group members should be given a Login Code and set to Force Login.
• The agent state Busy Not Available is equivalent to Do Not Disturb. The agent state Busy
Wrap Up is equivalent to hunt group disable.
Related links
Group Operation on page 694

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Configure User Settings

Malicious Call Tracing (MCID)

MCID (Malicious Caller ID) is an ISDN feature. It is supported on BRI and PRI trunks to ISDN
service provider who provide MCID.
When used, it instructs the ISDN exchange to perform a call trace on the user's current call and to
keep a record of the call trace at the exchange for the legal authorities. Trace information is not
provided to or displayed by the system or system phones.
The use of MCID is subject to local and national legal requirements that will vary. The feature may
also not be enabled until specifically requested from the service provider. You should consult with
your ISDN service provider and with appropriate legal authorities before attempting to use MCID.
Note:
Currently, in Server Edition network, MCID is only supported for users using an MCID button and
registered on the same IP500 V2 Expansion system as the MCID trunks.
Activating MCID
1. Liaise with the ISDN Service Provider MCID should not be used without first confirming
its usage with the ISDN service provider.
2. Enabling MCID Call Tracing on a Line BRI and PRI lines include a Support Call Tracing
Option which by default is off.
3. Enabling MCID Call Tracing for a User Each user has a Can Trace Calls (User |
Telephony | Supervisor Settings) option. This option is off by default.
4. Providing an Active MCID Control The user needs to be provided with a mechanism to
trigger the MCID call trace at the exchange. This can be done using either a short code or
a programmable button.
• MCID Activate Button The action MCID Activate (Advanced | Miscellaneous | MCID
Activate) can be assigned to a programmable buttons. It allows a malicious call trace to be
triggered during a call.
• MCID Activate Short Codes The feature MCID Activate can be used to create a short code
to triggering a malicious call trace.
Related links
Configure User Settings on page 651

Message Waiting Indication

Message waiting indication (MWI) or a message lamp is supported for a wide variety of phones. It
is used to provide the user with indication of when their voicemail mailbox contains new
messages. It can also be configured to provide them with indication when selected hunt group
mailboxes contain new messages.

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 Message Waiting Indication

Avaya digital and IP phones all have in-built message waiting lamps. Also for all phone users, the
one-X Portal for IP Office application provides message waiting indication.
Related links
Configure User Settings on page 651
Message Waiting Indication for Analog Phones on page 705
Message Waiting Indication for Analog Trunks on page 706

Message Waiting Indication for Analog Phones

For analog phones, the system supports a variety of analog message waiting indication (MWI)
methods. The method used for an individual analog extension is set for the Extn | Analog |
Message Waiting Lamp Indication Type field. Those methods are
• 101V
• 51V Stepped
• 81V
• Bellcore FSK
• Line Reversal A
• Line Reversal B
• None
• On
The 101V method is only supported when using a Phone V2 expansion module.
81V is typically used in European countries. 51V Stepped is used in most other countries.
However the actual method used for a particular model of analog phone should be confirmed with
the phone manufacturer's documentation.
The Message Waiting Lamp Indication Type field also provides options for None (no MWI
operation) and On. On selects a default message waiting indication method based on the system
locale.
'On' Method Locale
81V Belgium, Denmark, Finland, France, Germany,
Greece, Hungary, Iceland, Italy, Netherlands,
Norway, Poland, Portugal, Russia, Saudi Arabia,
Sweden, Switzerland, United Kingdom.
51V Stepped Argentina, Australia, Brazil, Canada, Chile, China,
Colombia, Japan, Korea, Mexico, New Zealand,
Peru, South Africa, Spain, United States.

For the United Kingdom system locale (eng), the default Caller Display Type (UK) allows updates
of an analog phone's ICLID display whilst the phone is idle. The system uses this facilities to
display the number of new messages and total number of messages in the users own mailbox.
This feature is not supported with other Caller Display Types.

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Configure User Settings

Hunt Group Message Waiting Indication

By default no message waiting indication is provided for hunt group voicemail mailboxes. Message
waiting indication can be configured by adding an H entry followed by the hunt groups name to the
Source Numbers tab of the user requiring message waiting indication for that hunt group. For
example, for the hunt group Sales, add HSales. Hunt group message waiting indication does not
require the user to be a member of the hunt group.
Related links
Message Waiting Indication on page 704

Message Waiting Indication for Analog Trunks

IP Office can provide a MWI for analog trunks from the PSTN network that terminate on an
ATM4U-V2 card. Multiple users can be configured to receive a MWI from a single analog line.
Users can receive an MWI from multiple lines. Configuring a user for MWI includes configuraton of
a button for automatically dialing the message center.
Note the following conditions.
• Only supported for analog trunks terminating on the ATM4U-V2 card.
• When Analog Trunk MWI is selected as the Voicemail Type, no other voicemail system is
active. As a result, hunt group queue announcements are not supported, since they require
Embedded Voice Mail or Voicemail Pro.
• All analog trunks configured for MWI must use the same message center number. Multiple
message centers are not supported.
• Not supported in One-X Portal.
• No TAPI is provided for analog trunk MWI status.
• Not supported across multiple IP Office systems. If the analog line is on a different node than
the user’s phone, that phone cannot receive an MWI for the line.
• Mobile twinning is not supported. Analog trunk MWI is displayed only on the master set.
• Internal twinning is not supported automatically. However, the twinned set can be configured
to receive the same analog trunk MWI as the master set.
Configuring MWI for an Analog Trunk
1. Go to System | Voicemail. In the Voicemail field, select Analog Trunk MWI.
2. In the Destination field, enter the message center telephone number.
3. Select the Line you want to configure for Analog MWI, and then select the Analog
Options tab.
4. In the MWI Standard field, select Bellcore FSK.
5. Select the User you want to configure for MWI and then select the Button Programming
tab.
6. Select the button you want to configure and then click Edit.
7. In the Action field click the browse (...) button and select Advanced > Voicemail >
Monitor Analog Trunk MWI.

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 Mobile Call Control

8. In the Action Data field, enter the line appearance ID of the analog line.
Related links
Message Waiting Indication on page 704

Mobile Call Control

Mobile call control is only supported on digital trunks, including SIP trunks. It allows a user
receiving a call on their twinned device to access system dial tone and then perform dialing action
including making calls and activating short codes.
After answering a twinned call, the Mobile Call Control user can dial ** (within 1 second of each
other) to place that call on hold and instead get dial tone from the system. Any dialing is now
interpreted as if the user is logged into a basic single line extension on the system using their user
settings. That also include user BLF status indication.
To use these features the user must be configured to support mobile call control.

Warning:
This feature allows external callers to use features on your phone system and to make calls
from the phone system for which you may be charged. The only security available to the
system is to check whether the incoming caller ID matches a configured users' Twinned
Mobile Number setting. The system cannot prevent use of these features by caller's who
present a false caller ID that matching that of a user configured for access to this feature.
Trunk Restrictions:
Mobile call control is only supported on systems with trunk types that can give information on
whether the call is answered. Therefore, mobile call control is not supported on analog or T1
analog trunks. All other trunk types are supported (ISDN PRI and BRI, SIP (RFC2388), H323).
Routing via trunks that do not support clearing supervision (disconnect detection) should not be
used.
DTMF detection is applied to twinned calls to a user configured for this feature. This will have the
following effects:
DTMF dialing is muted though short chirps may be heard at the start of any DTMF dialing.
DTMF dialed by the user will not be passed through to other connected equipment such as IVR or
Voicemail.
Mobile Call Control Features and FNE Services:
Mobile call control uses a short code set to invoke an FNE service. The codes relevant to mobile
call control are summarized below.
• FNE 31 = Mobile Call Control This code allows a user called or calling the system to
invoke mobile call control and to then handle and make calls as if they were at their system
extension.

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Configure User Settings

• FNE 32 = Mobile Direct Access Mobile direct access FNE32 immediately redials on switch
the DDI digits received with the call rather than returning dial tone and waiting for DTMF
digits as with FNE31 .
• FNE 33 = Mobile Callback Mobile callback allows the user to call the system and then
hang up. The system will then make a call to the user's CLI and when answered, provide
them with dial tone from the system to make calls.
• FNE 35 = Simplified Mobile Call Control: In addition to the Mobile Call Control feature that
enables your mobile to make and handle calls as if your are using your extension, this
Simplified Mobile Call Control FNE 35 clears the dial tone when the call recipient ends the
call. The dial tone is provided on the mobile phone for fresh calls after the current call is
cleared.
• FNE 36 = Simplified Mobile Direct Access: In addition to the Mobile Direct Access feature,
the Simplified Mobile Direct Access FNE36 clears the dial tone when the call recipient ends
the call.
• FNE 37 = Simplified Mobile Callback: In addition to the Mobile Callback feature that
enables your mobile to get call back from the system and lets you use the dial tone for
making and handling calls, this Simplified Mobile Callback FNE 37 clears the dial tone when
the call recipient ends the call. The dial tone is provided on the mobile phone for fresh calls
after the current call is cleared.
Using Mobile Call Control:
In addition to using ** to access mobile call control, the user has access to the following additional
controls:
• Clearing a Call: *52 It may be necessary to clear a connected call, for example after
attempting a transfer and hearing voicemail or ringing instead. To do this dial ** for dial tone
and then *52 (this is a default system short code and can be changed if required).
• Return to Dial Tone: ## Return to dial tone after getting busy, number unobtainable or short
code confirmation tones from the system.
Enabling Outgoing Mobile Call Control:
1. Configure the user for Mobile Twinning and Mobile Call Control On the User | Mobility
tab do the following:
• Enable Mobility Features for the user.
• Set the Twinned Mobile Number for the user's twinned calls destination.
1. Digits are matched from right to left.
2. The match must be at least 6 digits. If either the CLI or the Mobile Twinned Number is less
than 6 digits no match will occur.
3. Matching is done for up to 10 digits. Further digits are ignored. If either the CLI or Mobile
Twinned Number is less than 10 digits, matching stops at that shorter length.

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 Mobile Call Control

4. If multiple matches occur the first user in the configuration is used. Manager will warn
against configuration where such a conflict may exist.
• Select Can do Mobile Call Control.
On systems with some unsupported trunk types, further changes such as Outgoing Group ID,
system shorts codes and ARS may be necessary to ensure that calls to the mobile twinned
numbers are only routed via trunks that support mobile call control.
Incoming Mobile Call Control:
The system can be configured to allow Mobile Call Control users to use this function when making
an incoming call to the system. This requires the user to make the incoming call from the same
CLI as their Mobile Twinning Number (even if they do not actually use Mobile Twinning).
The call will be rejected:
• If the caller ID is blank or withheld.
• If the caller ID does not match a Twinned Mobile Number of a user with Can do Mobile Call
Control enabled.
• If the call is received on a trunk type that does not support Mobile Call Control.
Enabling Incoming Mobile Call Control:
On the User | Mobility tab do the following:
1. Enable Mobility Features for the user.
2. Set the Twinned Mobile Number to match the CLI of the device from which the user will
be making calls.
3. Select Can do Mobile Call Control.

Add a FNE Short Code In the system short codes section of the configuration add a short
code similar to the following. Key points are the use of the FNE Service feature and the
Telephone Number value 31.
• Short Code: *89
• Feature: FNE Service
• Telephone Number: 31

Add an Incoming Call Route for the user Create an incoming call route that matches the
user's CLI and with the FNE short code created above as its destination.
On systems with some unsupported trunk types, further changes such as Incoming Group ID
changes may be necessary to ensure that only calls received on trunks that support Mobile Call
Control are routed to this short code.
Related links
Configure User Settings on page 651
Mobile Direct Access (MDA) on page 710
Mobile Callback on page 711

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Configure User Settings

Mobile Direct Access (MDA)

For a Mobile Call Control or one-X Mobile client user, FNE32 immediately redials on switch the
DDI digits received with the call rather than returning dial tone and waiting for DTMF digits as with
FNE31. This is called Mobile Direct Access (MDA).
MDA requires the user's external telephony provider to provide a direct trunk with DDI to the
system (ie. an ISDN or SIP trunk). By assigning a specific incoming line group ID to the trunk, an
incoming call route can be created for the same line group ID with blanks incoming number and
incoming CLI fields. The destination is a short code set to FNE32.
User validation is performed using the CLI in the same way as for normal Mobile Call Control. In
addition the call will be rejected no DDI digits are provided. Once connected the user can use the
other Mobile Call Control features such as **.

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 Mobile Call Control

Related links
Mobile Call Control on page 707

Mobile Callback
Mobile callback allows the user to call the system and then hang up. The system will then make a
call to the user's CLI and when answered, provide them with dial tone from the system to make
calls.

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Configure User Settings

Mobile callback is subject to all the normal trunk type and user licensing restrictions of mobile call
control. In addition the user must have the Mobile Callback (User | Mobility)setting enabled in
the system configuration.
When the user makes a call using a DDI that is routed to an FNE33 short code, the system will not
connect (answer) the call but will provide ringing while it waits for the user to hang up (after 30
seconds the system will disconnect the call).
• The system will reject the call if the CLI does not match a user configured for Mobile Callback
or does not meet any of the other requirements for mobile call control.
• The system will reject calls using FNE33 if the user already has a mobile twinning or mobile
call control call connected or in the process of being connected. This includes a mobile
callback call in the process of being made from the system to the user.
If the CLI matches a user configured for mobile callback and they hang up within the 30 seconds,
the system will within 5 seconds initiate a callback to that user's CLI.
• If the call is answered after the user's Mobile Answer Guard time and within the user's No
Answer Time, the user will hear dial tone from the system and can begin dialling as if at their
system extension.
• If the call is not answered within the conditions above it is cleared and is not reattempted.
Related links
Mobile Call Control on page 707

Twinning
Twinning allows a user's calls to be presented to both their current extension and to another
number. The system supports two modes of twinning:
Internal Mobile
Twinning Destination Internal extensions only External numbers only.
Supported in All locales. All locales.
License Required No No

User BLF indicators and application speed dials set to the primary user will indicate busy when
they are connected to a twinned call including twinned calls answered at the mobile twinning
destination.
Do Not Disturb and Twinning
Mobile Twinning
Selecting DND disables mobile twinning.
Internal Twinning
• Logging out or setting do not disturb at the primary stops twinned calls alerting at the
secondary also.

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 Twinning

• Logging out or setting do not disturb at the secondary only affects the secondary.
Do Not Disturb Exceptions List
For both types of twinning, when DND is selected, calls from numbers entered in the user's Do Not
Disturb Exception List are presented to both the primary and secondary phones.
Internal Twinning
Internal twinning can be used to link two system extensions to act as a single extension. Typically
this would be used to link a users desk phone with some form of wireless extension such as a
DECT or WiFi handset.
Internal twinning is an exclusive arrangement, only one phone may be twinned with another. When
twinned, one acts as the primary phone and the other as the secondary phone. With internal
twinning in operation, calls to the user's primary phone are also presented to their twinned
secondary phone. Other users cannot dial the secondary phone directly.
• If the primary or secondary phones have call appearance buttons, they are used for call
alerting. If otherwise, call waiting tone is used, regardless of the users call waiting settings. In
either case, the Maximum Number of Calls setting applies.
•
• Calls to and from the secondary phone are presented with the name and number settings of
the primary.
• The twinning user can transfer calls between the primary and secondary phones.
• Logging out or setting do not disturb at the primary stops twinned calls alerting at the
secondary also.
• Logging out or setting do not disturb at the secondary only affects the secondary.
• User buttons set to monitor the status of the primary also reflect the status of the secondary.
• Depending on the secondary phone type, calls alerting at the secondary but then answered
at the primary may still be logged in the secondary's call log. This occurs if the call log is a
function of the phone rather than the system.
• Call alerting at the secondary phone ignoring any Ring Delay settings applied to the
appearance button being used at the primary phone. The only exception is buttons set to No
Ring, in which case calls are not twinned.
The following applies to internal twinned extensions:
If using a 1400, 1600, 9500 or 9600 Series phone as the secondary extension:
• The secondary extension's directory/contacts functions access the primary user's Centralized
Personal Directory records in addition to the Centralized System Directory.
• The secondary extension's call Log/call List functions access the primary user's Centralized
Call Log.
• The secondary extension's redial function uses the primary users Centralized Call Log. Note
that the list mode or single number mode setting is local to the phone.
It is also shown on 3700 Series phones on a DECT R4 system installed using system
provisioning .
For all phone types, changing the following settings from either the primary or secondary
extension, will apply the setting to the primary user. This applies whether using a short code,

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Configure User Settings

programmable button or phone menu. The status of the function will be indicated on both
extensions if supported by the extension type.
• Forwarding settings.
• Group membership status and group service status.
• Voicemail on/off.
• Do Not Disturb on/off and DND Exceptions Add/Delete.
Mobile Twinning
This method of twinning can be used with external numbers. Calls routed to the secondary remain
under control of the system and can be pulled back to the primary if required. If either leg of an
alerting twinned call is answered, the other leg is ended.
Mobile twinning is only applied to normal calls. It is not applied to:
• Intercom, dial direct and page calls.
• Calls alerting on line appearance, bridged appearance and call coverage buttons.
• Returning held, returning parked, returning transferred and automatic callback calls.
• Follow me calls.
• Forwarded calls except if the user's Forwarded Calls Eligible for Mobile Twinning setting
is enabled.
• Hunt group calls except if the user's Hunt Group Calls Eligible for Mobile Twinning setting
is enabled.
• Additional calls when the primary extension is active on a call or the twinning destination has
a connected twinned call.
A number of controls are available in addition to those on this tab.
Button Programming Actions:
The Emulation | Twinning action can be used to control use of mobile twinning. Set on the
primary extension, when that extension is idle the button can be used to set the twinning
destination and to switch twinning usage on/off. When a twinned call has been answered at the
twinned destination, the button can be used to retrieve the call at the primary extension.
Mobile Twinning Handover:
When on a call on the primary extension, pressing the Twinning button will make an unassisted
transfer to the twinning destination. This feature can be used even if the user's Mobile Twinning
setting was not enabled.
• During the transfer process the button will wink.
• Pressing the twinning button again will halt the transfer attempt and reconnect the call at the
primary extension.
• The transfer may return if it cannot connect to the twinning destination or is unanswered
within the user's configured Transfer Return Time (if the user has no Transfer Return Time
configured, a enforced time of 15 seconds is used).
Short Code Features:

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 Private Calls

The following short code actions are available for use with mobile twinning.
• Set Mobile Twinning Number.
• Set Mobile Twinning On.
• Set Mobile Twinning Off.
• Mobile Twinned Call Pickup.
Related links
Configure User Settings on page 651

Private Calls
This feature allows users to mark a call as being private.
When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private call.
Note that use of private calls is separate from the user's intrusion settings. If the user's Cannot be
Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching private calls off
does not affect that status. To allow private calls to be used to fully control the user status, Cannot
be Intruded (User | Telephony | Supervisor Settings) should be disabled for the user.
Use of private calls can be changed during a call. Enabling privacy during a call will stop any
current recording, intrusion or monitoring. Privacy only applies to the speech part of the call. Call
details are still recorded in the SMDR output and other system call status displays.
Button Programming The button programming action Advanced | Call | Private Call can be
used to switch privacy on/off. Unlike the short code features it can be used during a call to apply or
remove privacy from current calls rather than just subsequent calls. On suitable phones the button
indicates the current status of the setting.
Short Codes A number of short code features are available for privacy.
• Private Call Short codes using this feature toggle private status on/off for the user's
subsequent calls.
• Private Call On Short codes using this feature enable privacy for all the user's subsequent
calls until privacy is turn off.
• Private Call Off Short codes using this feature switch off the user's privacy if on.
Related links
Configure User Settings on page 651

System Phone Features

The user option System Phone Rights (User | User) can be used to designate a user as being a
system phone user. System phone users can access a number of additional function not available

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Configure User Settings

to other phone users. Note that if the user has a login code set, they will be prompted to enter that
code in order to access these features..
• None The user cannot access any system phone options.
• Level 1 The user can access all system phone options supported on the type of phone they
are using except system management and memory card commands.
• Level 2 The user can access all system phone options supported on the type of phone they
are using including system management and memory card commands. Due to the nature of
the additional commands a login code should be set for the user to restrict access.
System Phone Functions:
The following functions are supported:
• MENU to set date/time Restricted to 4412, 4424, 4612, 4624, 6408, 6416 and 6424 phones
where supported by the system. Note 4612 and 4624 not support for 4.1+. On these phones,
a system phone user can manually set the system date and time by pressing Menu | Menu |
Func | Setup.
• SoftConsole Send Message If the system phone user is using SoftConsole, they can
access the SoftConsole function Send Message to send a short text message (up to 16
characters) to a display phone. Refer to the SoftConsole documentation for details. Note that
this is no longer required for 4.0+.
• Change Login Code of Other Users Using a short code with the Change Login Code
feature, system phone users can change the login code of other users on the system.
• Outgoing Call Bar Off Using a short code with the Outgoing Call Bar Off feature, system
phone users can switch off the outgoing call bar status of other users on the system. .
The following commands are only supported using 1400, 1600, 9500 and 9600 Series phones.
Due to the nature of the commands a login code should be set for the user to restrict access. The
commands are accessed through the Features | Phone User | System Administration menu.
For full details refer to the appropriate phone user guide.
• Edit System Directory Records Using a 1400, 1600, 9500 or 9600 Series phone, a system
phone user can edit system directory records stored in the configuration of the system on
which they are hosted. .They cannot edit LDAP and/or HTTP imported records.
• System Management (IP500 V2 only) Allows the user to invoke a system shutdown
command.
• Memory Card Management Allows the user to shutdown, startup memory cards and to
perform actions to move files on and between memory cards.
• System Alarms (IP500 V2 only) For certain events the system can display an S on the
user's phone to indicate that there is a system alarm. The user can then view the full alarm
text in the phone's Status menu. The possible alarms in order of priority from the highest first
are:
1. Memory Card Failure.
2. Expansion Failure.

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 The 'No User' User

3. Voicemail Failure.
4. Voicemail Full.
5. Voicemail Almost Full.
6. License Key Failure.
7. System Boot Error.
8. Corrupt Date/Time.
• Date/Time Programmable Button: Allows system phone users to manually set the system
date and time through a programmable button (see System Date and Time on page 581).
Related links
Configure User Settings on page 651

The 'No User' User

It is possible to have an extension which has no default associated user. This can occur for a
number of reasons:
• The extension has no Base Extension setting associating it with a user who has the same
setting as their Extension to indicate that they are the extension's default associated user.
• The extension's default associated user has logged in at another extension. Typically they
will be automatically logged back in at their normal extension when they log out the other
phone.
• The extension's default associated user cannot be automatically logged in as they are set to
Forced Login.
Phones with no current user logged in are associated with the setting of the NoUser user in the
system configuration. This user cannot be deleted and their Name and Extension setting cannot
be edited. However their other settings can be edited to configure what functions are available at
extensions with no currently associated user.
By default the NoUser user has Outgoing Call Bar enabled so that the extension cannot be used
for external calls. The users first programmable button is set to the Login action.
Avaya 1100 Series, 1200 Series, M-Series and T-Series phones, when logged out as No User, the
phones are restricted to logging in and dial emergency calls only.
NoUser Source Numbers
The SourceNumbers tab of the NoUser user is used to configure a number of special options.
These are then applied to all users on the system. For details refer to the User | Source Numbers
section.
Related links
Configure User Settings on page 651
Suppressing the NoCallerId alarm on page 718

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Configure User Settings

Suppressing the NoCallerId alarm

Use this procedure to suppress the NoCallerId alarm for all users on the system. Once the task is
completed, the NoCallerID alarm is not raised in SysMonitor, SNMP traps, email notifications,
SysLog or System Status.
Procedure
1. In Manager, in the navigation pane on the left, select User.
2. In the list of users, select NoUser.
3. In the details pane, select the Source Numbers tab.
4. Click Add.
5. In the Source Number field, enter SUPPRESS_ALARM=1.
6. Click OK.
Related links
The 'No User' User on page 717

Transferring Calls
The following are some of the methods usable to transfer calls.
• Supervised Transfer: This is a transfer where the user waits for the transfer destination to
answer and talks to that party before completing the transfer, this is referred to as a
consultation call. They then either complete the transfer or drop the call and return to the held
for transfer call. The call details, display, ringing and forwarding applied are appropriate to the
type of call (internal or external) being transferred.
• Unsupervised Transfer: This is a transfer completed whilst the destination is still ringing.
• Automatic Transfer - Forwarding: The system allows users to automatically transfer calls
using forwarding options such as Forward on Busy, Forward on No Answer and Forward
Unconditional. For full details see DND, Follow Me and Forwarding.
• Transfers to a Forwarded Extension: When transferring a call to another extension that
has forwarding enabled, the type of call being transferred is used. For example, if transferring
an external call, if the transfer target has forwarding of external calls enabled then the
forward is used.
• Transfer Return Time (secs):Default = Blank (Off), Range 1 to 99999 seconds. Sets the
delay after which any call transferred by the user, which remains unanswered, should return
to the user. A return call will continue ringing and does not follow any forwards or go to
voicemail.
- Transfer return only occurs if the user has an available call appearance button.

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 Transferring Calls

- Transfer return is not applied if the transfer is to a hunt group that has queuing enabled.
Tool Unsupervised Transfer Supervised Transfer Reclai
m
Analog Phone/ 1. Press R. Note that broken dial 1. Press R. *46
Single Line Phones tone is heard while a call is on
2. Dial the transfer destination
hold.
number.
2. Dial the transfer destination
3. If the destination answers and
number.
accepts the call, hang-up.
3. Hang-up.
4. If the called party does not
answer or does not want to
accept the call, press Ragain.
5. To return to the original caller
press R.
Avaya DS Phone 1. Press Transfer. 1. Press Transfer. *46
2. Dial the transfer destination 2. Dial the transfer destination
number. number.
3. Press Transfer again to 3. If the destination answers and
complete the transfer. accepts the call, press
Transfer again to complete
the transfer.
4. If the called party does not
answer or does not want to
accept the call, press
Drop.
5. To return to the original caller
press it’s call appearance
button.

Related links
Configure User Settings on page 651

Off-Switch Transfer Restrictions

Users cannot transfer calls to a destination that they cannot normally dial. This applies to manual
transfers and also to automatic transfers (forwarding). In addition to call barring applied through
short codes, the following system settings may restrict a users ability to transfer calls.
User Specific Controls
Outgoing Call Bar: Default = Off (User | Telephony | Supervisor Settings) When enabled, this
setting stops a user from making any external calls. It therefore stops them making any external
transfers or forwards.

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Configure User Settings

Inhibit Off-Switch Forward/Transfer: Default = Off (User | Telephony | Supervisor Settings).

When enabled, this setting stops the specific user from transferring or forwarding calls externally.
This does not stop another user transferring the restricted users calls off-switch on their behalf.
When either system or user Inhibit Off-Switch Forward/Transfer is enabled, it affects the
operation of the user's phone and applications. User attempts to set an external forward
destination via a short code will receive error tone. User attempt to set an external forward
destination via a programmable button on their phone will not have a Next option allowing the
number to be saved.
Line Specific Control
Analog Trunk to Trunk Connection: Default = Off (Line | Analog Line | Analog Options) When
not enabled, users cannot transfer or forward calls on one analog trunk back off-switch using
another analog trunk.
System Wide Controls
Inhibit Off-Switch Forward/Transfer: Default = Off (System | Telephony) When enabled, this
setting stops any user from transferring or forwarding calls externally.
Restrict Network Interconnect: Default = Off (System | Telephony). When this option is
enabled, each trunk is provided with a Network Type option that can be configured as either
Public or Private. The system will not allow calls on a Public trunk to be connected to a Private
trunk and vice versa, returning busy indication instead.
Due to the nature of this feature, its use is not recommended on systems also using any of the
following other system features: multi-site networks, VPNremote, application telecommuter mode.
Conference Control
Users can use conference controls to effectively transfer calls. This includes transferring an
external call to another external number. The use of conferencing to effect off-switch transfers can
be restricted using the Inhibit External Only Impromptu Conference setting (System |
Telephony).

Context Sensitive Transfer

Calls and Button Status Indication The status indication for a call on hold pending transfer has
changed to differentiate such calls from standard held calls:
• On phones with both dual lamp buttons, both the green and red lamps fast flash (flutter)
when the button represents a call on hold pending transfer.
• On phones with single lamp buttons or status icons, Xfer: is now shown in front of the caller
ID information rather than the button name. For example Xfer:Extn299 is shown rather than
a = Extn299.
• The call status information shown when the button of a call on hold pending transfer is the
currently highlight line is now prefixed with On-Hold-Xfer rather than On-Hold.
Switching Between Calls Switching from a connected call to an existing call on hold pending
transfer puts the connected call on hold pending transfer. The following table is an example of the
resulting operation .

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 Transferring Calls

Call or answer A Connected to A

Press Transfer A on hold pending transfer
Call or answer B A on hold pending transfer. Connected to B.
Reconnect to A Connected to A. B on hold pending transfer
Press Transferor Complete*. A transferred to B.

Requirement for a Free Call Appearance Before Starting a Transfer When the user already
has a call or calls on hold, they can now put their current call on hold pending transfer even if
there are no free call appearances available. Previously an available call appearance was
required in order to then make a consultation call to the potential transfer destination.
Conferencing Calls For these phone there have also been changes to which calls are
conferenced in different scenarios including when there is a call on hold pending transfer. See
Context Sensitive Conferencing.

Dial Tone Transfer

Release 5.0+: A user who is not able to make external calls to any or some external numbers, can
be transferred to dial tone by a user who is able to make external calls.
• The restricted user wanting to make the external call, dials the unrestricted user and requests
dial tone.
• The unrestricted user initiates a transfer and dials the prefix for an ARS form configured to
provide secondary dial tone.
The prefix is a short code set up to access the required ARS form. While this can be a system
short code, using a user or user rights short code will allow control over who can provide dial tone
transfer for restricted users.
• When they hear the secondary dial tone, the unrestricted user completes the transfer.
• The restricted user now hears the secondary dial tone and is now able to make an external
call.
• The restricted user is now able to make calls as permitted by the short codes in the ARS
form.
• The restricted user is not able to transfer the dial tone to another user.
The ARS form being used can still contain short codes that restrict the dialing that can be
attempted after the restricted user hears secondary dial tone. Other ARS features can also be
used such as alternate routing or time profiles to provide out of hours routing. The ARS form
timers are run from when the unrestricted caller dials the ARS form. They are not reset when the
restricted user is transferred to the ARS form.
Multiple prefixes and ARS forms can be used if required to create more complex scenarios. For
example, one where the unrestricted user can transfer the restricted users to an ARS forms that
allows international calls or to an ARS form that only allows national dialing.
Example Configuration:

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Configure User Settings

The example below is a simple configuration that allows the unrestricted user to use 8 as a
transfer destination that provides secondary dial tone.
Create an ARS Form for Secondary Dial Tone The ARS form needs to be created before short
codes can be added to route callers to it.
• Enter a Route Name to identify the ARS form, for example Dial Tone Trans.
• Select Secondary Dial Tone.
• Select either System Tone (this matches locale specific normal dial tone) or Network Tone
(this matches locale specific secondary dial tone). For some locales both tones are the
same.
• Enter short codes that will take any digits dialed by the restricted user and process them for
external dialing to an outgoing line group. For this example we will allow any digits dialed to
be presented to the first trunk seized in outgoing line group 0.
Code N
Telephone Number N
Feature Dial
Line Group ID 0

• Other short codes can be used to allow or bar the dialing of specific numbers or types of
numbers.
• Configure the rest of the ARS form as required. For full details on ARS form configuration see
ARS.
Create a Short Code for Dial Tone Transfer For this example we will allow the prefix 8 to be
used to access an ARS form created above.
In the user short codes of the unrestricted user, create a short code that invokes the ARS form
created above. For example:
Code 8
Telephone Number
Feature Dial
Line Group ID 51 Dial Tone Trans

• It is important that the short code does not pass any digits to the ARS form. Once the ARS
form receives any digits, it starts short code matching and ends secondary dial tone.
• The short code could also be setup as a system or user rights short code.
The unrestricted user is now able to provide secondary dial tone to other users by on request by
pressing Transfer, dialing 8 and then pressing Transfer again.
Account and Authorization Codes:
If the restricted user enters an account or authorization code while calling the unrestricted user to
request dial tone, that value is not carried forward with their external call once they have been
provided with secondary dial tone.

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 Transferring Calls

If the unrestricted user enters an account or authorization code while dialing the ARS form, that
value remains associated with the call made by the restricted user.
If the ARS form short code used to route the restricted users call requires an account or
authorization code, the value already entered is used, otherwise the restricted user is prompted to
enter a value.
Call Logging:
The restricted user's outgoing call log will include the call to the unrestricted user and the outgoing
external call they subsequently make. The outgoing external call record will include the prefix
dialed by the unrestricted user to access the ARS form.
The unrestricted users call log will include just an incoming call from the restricted user.
Within the SMDR output, the calls by the restricted user are included. The call by the unrestricted
user is not included.

Handsfree Announced Transfers

This feature allows the enquiry call part of a supervised transfer to be answered handsfree. In
addition the system can be optionally configured to allow both the enquiry call and completed
transfer call to be auto-answered.
Example:
1. User 201 answers a call that they then want to transfer to user 203.
2. They press Transfer to put the call on hold pending transfer.
3. They then press a Dial Direct button and dial 203.
4. The transfer enquiry call is auto answered by User 203's phone. User 201 is able to
announce the pending transfer and hear if User 203 wants to accept the call.
The auto-answer only occurs if the target user's extension is idle. If the target is already connected
to a call, the transfer enquiry will be presented as normal call.
If the transfer is accepted, User 201 can press TRANSFER again to complete the transfer
process.
The transferred call will then ring at the target. However, if required the system can be configured
to also auto-answer the completed transfer.
Configuration:
Handsfree announced transfers are supported when using one of the following features after
having pressed TRANSFER.
Button Features Short Code Features
Table continues…

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Configure User Settings

Dial Direct Dial Direct

Automatic Intercom
Dial Intercom

User Button Usability:

Following the use of any of the buttons above, if the button has not been programmed with a
specific target, a User button can be used to indicate the target for the enquiry call. This gives the
advantage of being able to see the target user's status before attempting the transfer.
• For Automatic Intercom and Dial Intercom buttons without a pre-specified target, the User
button must be on a button module.
• For Dial Direct buttons without a pre-specified target, the User button can be on the phone
or button module. Due to this and the support for Dial Direct across a network of systems,
we recommend that a Dial Direct button is used for handsfree announced transfers.
Phone Support:
Handsfree announced transfer is supported for calls being transferred to the following phones:
Full Support Partial Support Not Supported
The following system phones The following phone can auto- Announced transfer is not
support full announced transfer answer announced transfers but supported for any phones not
operation. require the user to use the listed in the other column.
handset to respond.
1603, 1608, 1616, 2410, 2420, On unsupported phones the
5410, 5420, 4610, 4621, 4625, 2402, 4601, 4602, 5402, 5601, transfer eqnuiry consultation call
5610, 5620, 5621. 5602. will be presented as a normal call.
Analog Off-Hook Stations (See
notes below).

Notes:
• On supported phones, if the target user's phone is not idle when the enquiry call attempt is
made, the enquiry call is turned into a normal transfer attempt, eg. alerting on an available
call appearance.
• Enabling the extension specific setting Disable Speakerphone will turn all auto-answer calls,
including handsfree announced transfers to the extension, into normal calls.
• Off-Hook Station Analog Phones Analog phone extensions configured as Off-Hook Station
can auto-answer transfers when off-hook and idle.
• Headset Users The following applies to users on supported phones with a dedicated
HEADSET button. These users, when in headset mode and idle will auto-answer the
announced transfer enquiry call through the headset after hearing 3 beeps. The transfer
completion will require them to press the appropriate call appearance unless they are set to
Headset Force Feed.
• Twinning Handsfree announced transfer calls to users with twinning enabled will be turned
into normal calls.

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 Transferring Calls

• Multi-site network Support Dial Direct is supported to targets across a multi-site network,
therefore allowing handsfree announced transfers to remote users.
Full Handsfree Transfer Operation:
If required the system can be configured to allow the full handsfree announced transfer process,
ie. both the enquiry call and the transfer, to be auto-answered on supported phones. This is done
by entering FORCE_HANDSFREE_TRANSFER into the Source Numbers of the NoUser user and
rebooting the system

One Touch Transferring

This feature allows selected users to transfer calls to each other using a reduced number of key
presses.
With this option, a call can be transferred by simply selecting the transfer destination and then
hanging up (or pressing Transfer if working handsfree).
Without this option the normal sequence is to press Transfer, dial the destination and then
hanging up (or pressing Transfer if working handsfree).
For one touch transfer the transfer destination number must be selected using a button
programmed to one of the following features:
• User
• Dial
• Abbreviated Dial
• Automatic Intercom
• Dial Intercom
• Dial Direct
This feature is enabled on a per user basis by adding Enable_OTT to the Source Number
settings of the user. This feature is supported on all Avaya phones that support the programmable
button features.

Centrex Transfer
Centrex Transfer is a feature provided by some line providers on external analog lines. It allows
the recipient of a calls on such a line to transfer that call to another external number. The transfer
is performed by the line provider and the line is freed. Without Centrex Transfer, transferring an
external call to another external number would occupy both a incoming and outgoing line for the
duration of the call.

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Configure User Settings

The following are the supported controls and usages for Centrex Transfer:
• Centrex Transfer Button Operation The action Flash Hook can be assigned to a
programmable button. This button can be configured with or without a telephone number for
an automatic or manual transfer.
- Manual Transfer If the programmable button is setup without a target telephone number,
pressing the button returns dial tone to the user. They can then dial the required transfer
number and when they hear ringing or an answer, hang up to complete the Centrex
Transfer.
- Automatic Transfer If the programmable button is setup with a target telephone number,
pressing the button performs the Centrex Transfer to the number as a single action.
• Centrex Transfer Short Code Operation The Flash Hook short code feature can be used
with system short codes. It can be setup with or without a telephone number in the same way
as a Flash Hook programmable button above. The line group must be the group of analog
lines from the Centrex service line provider.
- Centrex Transfer Operation for Analog Extensions Most analog phones have a button
that performs the action of sending a hook flash signal. The marking of the button will vary
and for example may be any of R, H, Recall or Hold. Pressing this button sends a hook
flash to the system to hold any current call and return dial tone.
• To perform a Centrex Transfer, pressing the analog extension's hook flash button should
be followed by the dialing of a Flash Hook short code.
• For analog extension users with call waiting enabled, pressing the hook flash button
during a call will hold the current call and connect any call waiting. Therefore it is
recommend that analog extension users wanting to use Centrex Transfer should not
also have call waiting enabled.
• Auto Attendant Transfer System’s using embedded voicemail can select Centrex Transfer
as an action. For system using Voicemail Pro, the equivalent can be achieved by transferring
calls to a Flash Hook short code.
Additional Notes
• Networked Systems In networked systems, Centrex Transfer is only supported using Flash
Hook or Centrex Transfer features on the system which hosts the Centrex analog trunks.
• Addition Prefix Dialing In some cases the Centrex service provider may require a prefix for
the transfer number. If that is the case, that prefix must be inserted in the button
programming or the short code used for the Centrex Transfer.
• Application Transfers Centrex Transfer is not supported for calls being held and transferred
through applications such as SoftConsole.
• Conference Calls Centrex Transfer is not supported with conference calls.

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Chapter 14: Configure Server Edition
system settings

Synchronizing Server Edition passwords in Web Manager

In order to open IP Office Manager for a Server Edition solution, all IP Office systems in the
solution must have a service user with common credentials. If the security settings on any system
are reset, service user passwords are reset to the default value. In this case, when a system does
not have a service user with common credentials, launch of IP Office Manager fails.
Before you begin
You must know the user ID and password of the service user that is common to all systems in the
solution.
Procedure
1. For the system where the security settings were reset, open Web Manager using the
address https://<ip_address>:7070/WebManagement/WebManagement.html.
2. Log on as Administrator.
3. In Web Manager, select Security Manager > Service Users.
4. Create the common service user.
5. Log out of this Web Manager session.
6. Start another Web Manager session on the system using the address https://
<ip_address>/index.html.
7. Log on as the common service user.
8. In Web Manager, select Security Manager > Service Users.
9. Click Synchronize Service User and System Password.
10. Select Applications > IP Office Manager.

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Configure Server Edition system settings

Shared Administration User Account

About this task
When managing multiple systems, it may be useful to create a common user name and password
on all the systems for configuration access. This tool can be used to create a new service user
account, SCN_Admin, for configuration access.
This process requires you to have a user name and password for security configuration access to
each of the systems.
Managing a Common Configuration Administration Account
Procedure
1. Select Tools | Server Edition Service User Management.
2. The Select IP Office menu displays the list of discovered systems.
3. Select the systems for which you want to create a common configuration account.
Typically this should be all systems. Click OK.
4. A user name and password for security configuration access to each system is requested.
Enter the values and click OK. If the same values can be used for all systems enter those
values, select Use above credentials for all remaining, selected IPOs. If each system
requires a different security user names and password, deselect Use above credentials
for all remaining, selected IPOs.
5. The systems will be listed and whether they already have an SCN_Admin account is
shown.
6. To create the SCN_Admin account on each system and set the password for those
account click on Create Service User.
7. Enter the common password and click OK.
8. The password can be changed in future using the Change Password option.
9. Click Close.

Voicemail Administration
If the Voicemail Pro client application is installed on the same PC as Manager, it can be launched
from Manager. If not already installed, the Voicemail Pro client application can be downloaded
from the Primary Server via its web control menus. Refer to the Server Edition Deployment Guide.
Starting the Voicemail Pro Client

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 Server Edition Resiliency

Using the following method automatically starts the Voicemail Pro client with information about the
system to be administered.
1. In the Server Edition Solution View, select the server for which you want to administer the
voicemail application that the server hosts. This can be either the Primary Server or
Secondary Server. If Solution is selected, it is assumed that the voicemail server on the
Primary Server is being administered.
2.
Click on the Voicemail Administration link on the right-hand edge of the menu.
Alternate Method
The Voicemail Pro client can . When started this way, it will be necessary to manually enter the IP
address and other details of the system you want to monitor after starting the System Status
Application.
Click File | Advanced | Launch Voicemail Pro Client.

Server Edition Resiliency

Related links
Voicemail Pro Resiliency on page 731
Avaya one-X Portal resiliency on page 733
Phone Resiliency on page 735
Configuring Resiliency on page 738

Resilience
A single Server Edition Primary server supports redundant hard disk drives and power supply
units. You can also configure Alternate Route Selection.
Add a Server Edition Secondary to provide resilience at any level. The Server Edition Secondary
Server provides resilience for the Server Edition Primary Server users, H.323 and SIP extensions,
hunt groups, and voicemail without any administration. The Server Edition Secondary server can
provide resilience for Avaya one-X® Portal for IP Office.
A Server Edition Expansion System can be backed up to either the Server Edition Primary, Server
Edition Secondary, or another Server Edition Expansion System. The dual star Multi-Site Network
topology when a Server Edition Secondary Server is present supports diverse routing between all
nodes.
For Server Edition Select deployments, IP Office Lines (SCN trunks) can be configured between
Server Edition Expansion Systems. Hunt groups can be configured local to the Expansion system
and resiliency for hunt groups and phones can be configured, with failover to the Server Edition
Primary, Server Edition Secondary, or another Server Edition Expansion System.
At all times, no server hardware is forced to be idle, enabling you decide whether to provide true
redundancy, or shared resource resilience.

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Configure Server Edition system settings

The IP Office Server Edition Solution provides resilience for supported H.323 phones, SIP
endpoints, and DECT R4 deployments. IP Office Lines between systems can be configured to
allow control to be automatically passed to a backup IP Office when the home system is not
available.
Resilient components
The following components of IP Office Server Edition Solution are resilient:
• IP Office Server Edition
• Voicemail Pro server
• Avaya one-X® Portal server
• H.323 telephones
• SIP endpoints
• DECT R4
• Hunt groups
• Interdevice links
• Trunks
• Incoming Call Routes
• Management
Multisite network
A multisite network enhances resilience by providing the following capabilities:
• Transparency for most of the features
• Resilience of users and hunt groups
• Back up system for Voicemail Pro
• Network topology provides resilience
• None of the hardware is idle
• Simple to activate resilience

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 Server Edition Resiliency

Figure 3: Resiliency Architecture

Resilient management
You can continue to administer and manage the failure of IP Office Server Edition server and
devices in an IP Office Server Edition Solution network through the Server Edition Secondary
Server. This provides management without the offline capability, and a facility to realign the
configuration after the outages have been resolved. The resynchronization feature highlights the
time and source of configuration change and enables the administrator to decide which change
set to retain. In addition, you can directly manage each device and application to allow
configuration whilst isolated. You can use the resynchronization capability to realign the
configurations after the devices are reconnected.

Voicemail Pro Resiliency

One Active Voicemail Pro server
Server Edition supports one active Voicemail Pro server on the Server Edition Primary server. A
backup Voicemail Pro server is supported on the Server Edition Secondary server for resiliency.

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Configure Server Edition system settings

Voicemail Pro
Primary Server Failover
Secondary Server

Voicemail Pro Voicemail Pro Sync Voicemail Pro

Master Backup

one-X Portal

IP Office Line
IP Office IP Office

IP Office Line

Voicemail Pro
Leave/Collect

Voicemail Pro
Leave/Collect Voicemail Pro
Leave/Collect
IP Office IP Office IP Office

Figure 4: One active Voicemail Pro server

Dual Active Voicemail Pro servers

Server Edition deployments with Select licensing support two active Voicemail Pro servers,
doubling the maximum channel capacity and dual processing locations. Each expansion system
and all contained users can be configured to use one or the other. Each Voicemail Pro server
provides backup for the other. The two Voicemail Pro servers are both active for a configured
subset of users. They share a common configuration and message store. Each can support all
mailboxes, message waiting indicators (MWI) and call flows under failure conditions.

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 Server Edition Resiliency

Voicemail Pro
Primary Server Failover
Secondary Server

Voicemail Pro Voicemail Pro Sync Voicemail Pro

Master Master

Voicemail Pro
one-X Portal Failover one-X Portal

IP Office Line
IP Office IP Office

IP Office Line

Voicemail Pro
Voicemail Pro
Leave/Collect
Leave/Collect

IP Office IP Office IP Office

Figure 5: Dual Active Voicemail Pro servers

Supported Configurations
The following configuration are supported.
• Dual Voicemail Pro, each acting as a backup
• Dual Voicemail Pro, no backup operation
• Single master Voicemail Pro on Primary, no backup (non-Select)
• Single master Voicemail Pro on Primary and backup on Secondary (non-Select)
• Single master Voicemail Pro on Secondary and backup on Primary
Related links
Server Edition Resiliency on page 729

Avaya one-X® Portal resiliency

Server Edition Select deployments support a backup Avaya one-X® Portal server, providing
resiliency for Unified Communication (UC) features. The resilient one-X Portal server is installed
by default on the Server Edition Secondary server. Resilient one-X Portal servers can also be
located on standalone application servers.

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Configure Server Edition system settings

Resilient one-X Portal on the Server Edition Primary and Server Edition Secondary
Servers

Primary Server Secondary Server

Voicemail Pro Sync

Voicemail Voicemail

Active
Connection

one-X Portal Sync

one-X Portal one-X Portal
Passive Passive
Connection Connection
Active Passive
Connection Connection

IP Office Sync
IP Office IP Office

Resilient Avaya one-X® Portal on Standalone Application Servers

Primary Server Secondary Server

Voicemail Pro Sync

Voicemail Voicemail

IP Office Sync
IP Office IP Office
Passive Passive
Connection Connection

Active Active Passive

Connection Connection Connection

one-X Portal Sync

one-X Portal one-X Portal

Primary Application Secondary Application

Server Server

Avaya one-X® Portal Failover and Failback

When the primary Avaya one-X® Portal server is not available, failover occurs and the backup
server becomes active. Clients automatically recognize that the primary Avaya one-X® Portal

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 Server Edition Resiliency

server is not available and log in to the backup server. Logged in users are automatically logged in
to the backup Avaya one-X® Portal server.
When the primary Avaya one-X® Portal server is once again available, users automatically failback
to it. The backup Avaya one-X® Portal server redirects login requests to the primary Avaya one-X®
Portal server.
If connectivity between the primary and backup Avaya one-X® Portal servers is lost for an
extended time, both Avaya one-X® Portal servers become active and accept login requests. Once
connectivity is reestablished, the two servers are synchronized and all users are logged in to the
primary one-X Portal server.
Note:
When both Avaya one-X® Portal servers are active, any changes made by users logged in to
the backup server will be lost when failback occurs.
one-X Portal Resiliency with IP Office and Voicemail Pro
If the Voicemail Pro server on the Server Edition Primary server is not available, the primary
Avaya one-X® Portal server fails over to the Voicemail Pro server on the Server Edition Secondary
server. When the Voicemail Pro server on the Server Edition Primary server is once again
available, the primary Avaya one-X® Portal server automatically performs a failback to the
Voicemail Pro server on the Server Edition Primary server.
Related links
Server Edition Resiliency on page 729

Phone Resiliency
Phone Failover
When phone resiliency is configured, the home system shares information about the registered
phones and users on those phones with the backup system. If the home system is no longer
visible to the phones, failover occurs and the phones register with the backup system.
Phone Failback
If the phone’s home system has been up for more than 10 minutes, the system causes idle
phones to perform a failback to the home system. If the phone is unable to connect to the home
system, there is a five minute grace period, referred to as homeless prevention, where the phone
can be logged in to either the home or backup system.
Automatic failback to the home system is the default mode. Failback can be configured to operate
manually. This may be desired if for example, the home system will be unavailable for some time.
In manual mode, failback does not occur until the phone has been logged out or rebooted.
Note:
Manual failback is not supported for SIP phones.

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Configure Server Edition system settings

Configuration Options
Resiliency is configured on IP Office Line > Line under SCN Resiliency Options. The options
are:
• Backs up my IP Phones
• Backs up my Hunt Groups
• Backs up my Voicemail
• Backs up my IP DECT Phones
• Backs up my one-X Portal
Notes on Phone Resiliency Behavior
• Failover handover takes a minimum of 3 minutes (longer for larger networks). This ensures
that failover is not invoked when it is not required; for example, when the home system is
simply being rebooted to complete a non-mergeable configuration change.
• Failover is only intended to provide basic call functionality while the cause of failover
occurring is investigated and resolved. If users make changes to their settings while in
failover, for example changing their DND mode, those changes will not apply after failback.
• Calls anchored on the home system lose all voice paths during failover. Direct media calls in
a stable state might maintain voice paths until the next call event, but this is not guaranteed.
Media preservation is not supported on SIP phones.
• If the failover system is rebooted while it is providing failover services, the failover services
are lost.
• Failover features require that the phones local to each system are still able to route data to
the backup system when the home system is not available. This will typically require each
system site to be using a separate data router.
• When an IP phone fails over, the backup system allows it to operate indefinitely as a “guest”,
but only until the system resets. Licenses will never be consumed for a guest phone.
• Hot desking users are automatically logged out. When their base extension fails back to the
home system, the user is automatically logged in on their base extension.
• The media security configuration should be the same on all systems. For example, if an
extensions home system is set to Best Effort, the failover system should also be set to Best
Effort.
• For secure communication using TLS/SRTP, all IP Office systems must have an identity
certificate that has been signed by the same trusted root CA.
Note:
- Authentication of the client's certificate by the server is not a requirement. IP Office
does not support client certificate validation for all SIP endpoint types.
Supported Network Configurations
Phone resiliency is supported between any IP Office systems linked through an IP Office Line with
Networking Level set to SCN. This includes failover from an IP500 V2 system to another IP500
V2 system.
For Server Edition deployments, failover from one node to any other node in the solution is
supported.

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 Server Edition Resiliency

Note:
Resiliency can be configured by specifying a Location with a unique IP address for the
backup system. For cloud deployments, some systems cannot be configured as a Location.
See Configuring Location Based Resiliancy on page 742.
Supported Phones
Phone resiliency is supported on the phones listed below. Each IP Office system can be the initial
or backup system for a mixture of resilient phone types.
H.323 Phones:
• Resiliency is supported on 1600 and 9600 series phones.
SIP Phones:
The following SIP phones are supported:
• 1120
• 1140
• 1220
• 1230
• E129
• B179
• H175
SIP Soft Clients:
The following SIP soft clients are supported. Note that for these client, resiliency is also dependent
on Avaya one-X® Portalresilience.
• Avaya Equinox™ for Windows
• one-X Mobile Preferred for Android
• one-X Mobile Preferred for iOS
Note:
Resiliency is not supported for:
• Avaya Lync Plug-in
• IP Office SoftConsole
Notes on Supported Phones:
• E129, 1100 Series, 1200 Series, B179 and Avaya Equinox™ for Windows always attempt to
register to multiple servers. The homeless prevention feature is not applicable to these
phones.
• The B179 phone does not have a configuration file setting for the backup system. You must
use the phone web interface to configure the Secondary SIP Server setting and the
Fallback Account settings.
• For SIP phone resiliency, all IP Office systems in the Server Edition solution or SCN cluster
must have the same System > LAN > VoIP > SIP Domain setting.

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Configure Server Edition system settings

The SIP Domain is not the same as the IP Office fully qualified domain name (FQDN). They
are not synonymous but can be related through DNS SRV A records. A single SIP Domain
can include multiple SIP servers.
• On Avaya Communicator, the System > LAN > VoIP > SIP Domain value must be
configured in the Domain name field under Settings.
• Remote worker SIP phones will not failover when the failover server is CPE and their home
system is cloud based.
Related links
Server Edition Resiliency on page 729

Configuring Resiliency
Related links
Server Edition Resiliency on page 729

Configuring Resiliency Using the Resiliency Administration Tool

This process adjusts the settings of the IP Office lines between systems to indicate which lines are
being used to give/receive fallback options and what fallback options.
Note:
You can activate the resilience for an expansion system either on a primary server or a
secondary server only. You cannot activate the resilience for an expansion system on both the
primary server and secondary server.
Procedure
1. In the Server Edition Solution View, select the Resilience Administration link on the right.
2. The Resiliency Administration page opens.
Select the options required.
• Backup Primary Server IP Phones, Hunt Groups, Voicemail and one-X Portal on
Secondary Server When selected, the Secondary Server will support hunt group
operation during any failure of the Primary Server. Also when selected, the Secondary
Server will support the continued operation of Avaya IP phones normally registered to
the Primary Server.
• Backup Secondary Server IP Phones on Primary Server When selected, the Primary
Server will support the continued operation of Avaya IP phones normally registered to
the Secondary Server.
• Backup Expansion Systems
- For each expansion system, you can select to Backup Phones and Backup Hunt
Groups by clicking the check boxes.
- In the Resilient To field, select the IP Office system that will act as the backup.

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 Server Edition Resiliency

3. Click OK.

Linking Server Edition Expansion Systems

You can link Server Edition Select expansion systems with an IP Office Line in order to enable
resiliency capability. When you link expansion systems using the Link Expansions tool, an IP
Office Line is configured with default settings. You must select the security level for the line.
Procedure
1. Open Manager and log in to the Primary Server.
2. On the Solution page, on the right hand side, click Link Expansions.
3. In the Link Expansions window, in the First Expansion field, select an expansion system
from the list.
4. In the Second Expansion field, select an expansion system from the list.
5. Under Select Link Type, select one of the following options:
• SCN Websocket (Secure): Recommended for security and NAT traversal.
• SCN Websocket: Supports NAT traversal with limited security.
• SCN: Legacy SCN line. Not recommended for new deployment.
6. If the Link Type is set to SCN Websocket (Secure) or SCN Websocket, you must
configure a password.

Configuring Voicemail Pro Resiliency

Supported configurations are listed in the table below. A dual Voicemail Pro configuration is
supported on Server Edition deployments with Select licensing.
Configuration Voicemail IP Backup Voicemail Voicemail IP Backup Voicemail
Type Address on IP Address on Address on IP Address on
Primary Primary Secondary Secondary
Dual Voicemail Pro, Primary Server IP Secondary Server Secondary Server Primary Server IP
each acting as a Address IP Address IP Address Address
backup
Dual Voicemail Pro, Primary Server IP Secondary Server
no backup Address IP Address
operation
Single master Primary Server IP
Voicemail Pro on Address
Primary, no backup
[1]
Table continues…

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Configure Server Edition system settings

Configuration Voicemail IP Backup Voicemail Voicemail IP Backup Voicemail

Type Address on IP Address on Address on IP Address on
Primary Primary Secondary Secondary
Single master Primary Server IP Secondary Server
Voicemail Pro on Address IP Address
Primary and
backup on
Secondary [1]
Single master Secondary Server Primary Server IP
Voicemail Pro on IP Address Address
Secondary and
backup on Primary

Notes:
1. Non-Select.
2. For resiliency operation, you must also ensure that the default SMTP Sender setting of the
voicemail server is set to be the server’s fully qualified domain name. Within the voicemail
server preferences, select System Preferences | Email | SMTP Sender. The Domain
and Server fields of the first entry must be set to the fully-qualified domain name of the
voicemail server, not local host. This needs to be done on both the primary and secondary
voicemail servers.
Procedure
1. Log in to Manager.
2. In the navigation pane on the left, open the Primary server configuration and select
System.
3. In the details pane, select the Voicemail tab.
4. In the Voicemail Type field, select Voicemail Lite/Pro.
5. Set the Voicemail IP Address and Backup Voicemail IP Address as required based on
the table above.
6. Repeat for the Secondary server configuration.

Configuring one-X Portal Resiliency

Use this procedure to turn on one-X Protal Resiliency. Note that if resiliency is not turned on, the
one-X Portal server on the Server Edition Secondary remains in a stopped state.
Procedure
1. Log in to Manager.
2. In the navigation pane on the left, select the Primary Server configuration and click Line.
3. In the Group pane, select the IP Office Line connection to the Secondary Server.
4. On the Line tab, under SCN Resiliency Options, check the box Supports Resiliency.

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 Server Edition Resiliency

5. Check the box Backs up my one-X Portal.

6. Repeat on the Secondary server for the IP Office Line connection to the Primary server.
Next steps
You must configure resiliency on one-X Portal. See the section “Resilience” in Administering
Avaya one-X® Portal for IP Office™ Platform .

Configuring Phone Resiliency

Procedure
1. Using Manager, log in to the backup system that phones will failover to.
2. In the navigation pane on the left, select Line.
3. In the group pane, select the IP Office Line that connects to the home system.
In the details pane, the Line tab is displayed.
4. Under SCN Resiliency Options, check the box for Supports Resiliency.
5. Check the boxes for the desired resiliency options.
The options are:
• Backup my IP phones — Note that changes to the resilience destination for phones
registered with the system are not picked up by those phones until they are restarted.
• Backup my Hunt Groups
• Backup my Voicemail
• Backup my IP DECT Phones
6. If you are configuring resiliency for SIP phones.
a. Navigate to System > LAN > VoIP > SIP Domain and record the SIP domain value.
b. Log in to the home system for the resilient phones and navigate to System > LAN >
VoIP > SIP Domain.
c. Ensure the values are the same on both systems.

Configuring Manual Phone Failback

Automatic failback to the home system is the default mode. Failback can be configured to operate
manually. This may be desired if for example, the home system will be unavailable for some time.
In manual mode, failback does not occur until the phone has been logged out or rebooted.
Note:
SIP endpoints do not support manual phone failback.
Procedure
1. Using Manager, log in to the home system for the resilient phones.
2. In the navigation pane on the left, select System.

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Configure Server Edition system settings

3. In the details pane, click the Telephony tab.

The Telephony | Telephony tab is displayed.
4. In the Phone Fallback field, select Manual.

Configuring Location Based Extension Resiliency

Server Edition Select location based resiliency allows you to create a group of extensions by
applying a common Location value to each extension. You can then specify the fallback IP Office
server for the group. The location based fallback overrides the system fallback configuration.
Using location based resiliency
• Phones on the Primary and Secondary server can fall back to an Expansion system
• Phones on an Expansion system can fall back to another Expansion system
Location based resiliency is supported on Avaya 1600 and 9600 series phones and all SIP
endpoints.
Procedure
1. In Manager, open the Location page and define a location for the phone group.
2. If the phone group is in a remote location, you can define the Time Settings for the group.
3. In the Fallback System field, select the system where the phone group will fall back to.
Note that the Fallback System list only contains systems where an IP Office Line has
been configured.
4. Save the location.
5. For all phones that will be part of the group, open the extension page for the phone and
select the new Location.
Note that you can also set an IP address and subnet mask at the location level to match
the phone IP addresses.
6. On the system where the extension is configured, open the Line | IP Office Line page.
7. Under SCN Resiliency Options, enable Supports Resiliency.

Synchronizing the Configurations

About this task
Normally during configuration of the Server Edition solution, records that are shared (Incoming
Call Route, Time Profile, Account Code and User Rights) are automatically synchronized with the
configuration of the individual servers as they are edited. However, when new servers are added
to the network or systems have their configuration individually edited, it is possible that some
share records may become out of synch with the Primary Server. This process can be used to re-
establish the correct shared records.

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 Starting Web Control

Synchronizing the Configurations

Procedure
1. In the Server Edition Solution View, right-click on Solution.
2. Select Synchronize Configurations.
3. Select Yes to confirm the removal.

Starting Web Control

About this task
Web control is the term used for a set of web based administration menus used by Linux based
servers. That includes the Primary Server, Secondary Server and Expansion System (L) in a
Server Edition solution. The menus provide functions such as stopping and starting individual
services being run by the server. The menus for the Primary Server provide special network
functions such as backing up and upgrading the whole network.
Procedure
1. In the Server Edition Solution View, select the system for which you want to display its web
control menus.
The option is not available for Expansion System (V2).
2.
Click on the Web Control link on the right-hand edge of the menu.
3. The PCs default web browser is started with the address to the system.
4. When the login menu is displayed, login using the same configuration name and password
as used for Manager configuration access.

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Chapter 15: Configuring SIP Trunks

Related links
Overview on page 744
Configuring a SIP Trunk on page 745
SIP Line Requirements on page 746
SIP Incoming Call Routing on page 748
SIP Prefix Operation on page 749
SIP messaging on page 750
IP Office SIP trunk specifications on page 764

Overview
A growing number of service providers now offer PSTN access to businesses via public SIP trunk
connections, either to extend their reach beyond their typical copper based network coverage
areas, or so that multiple services (voice and internet access) can be bundled into a single
network connection. Although detailed public SIP trunk service offerings vary depending on the
exact nature of the offer from the specific service provider, SIP trunks can potentially provide
several advantages compared to traditional analog or digital trunks. These advantages include:
• cost savings resulting from reduced long distance charges, more efficient allocation of trunks,
and operational savings associated with managing a consolidated network
• simplified dialing plans and number portability
• geographic transparency for local accessibility creating a virtual presence for incoming calls
• trunk diversity and redundancy
• multi-media ready to roll out future SIP enabled applications
• fewer hardware interfaces to purchase and manage, reducing cost and complexity
• faster and easier provisioning
IP Office delivers functionality that enhances its ability to be deployed in multi-vendor SIP-based
VoIP networks. While this functionality is primarily based on the evolving SIP standards, there is
no guarantee that all vendors, interpret and implement the standards in the same way. To help the
SIP service provider, Avaya operates a comprehensive SIP Compliance Testing Program referred
to as GSSCP. Avaya's DevConnect program validates the operation of the IP Office solution with
the service provider’s SIP trunk offering.

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 Configuring a SIP Trunk

Related links
Configuring SIP Trunks on page 744

Configuring a SIP Trunk

This procedure provides the basic steps for configuring a SIP trunk between two IP Office
systems.
Before you begin
• You must know the IP address of both ends of the trunk.
• You must have valid licenses on both IP Office systems.
• On Server Edition, make sure you have a non-zero value in the SIP Trunk Sessions field on
the License | Remote Server tab. If you do not, you will see Monitor messages about
insufficient licenses.
Procedure
1. In the Manager navigation pane, right click Line and select New > SIP Line.
2. Record the Line Number value that appears on the SIP Line page for use later.
3. In the ITSP Domain Name field, enter the domain name required by the far end.
If nothing is configured in this field, then IP Office inserts the far end's ITSP Proxy
Address from the Transport tab as the ITSP domain in the SIP messaging.
4. Use the default values for the remaining fields.
5. Select the Transport tab.
6. In the ITSP Proxy Address field, enter the IP Address of the far end.
7. Select the SIP URI tab.
8. Click Add.
9. Enter values for the Incoming Group and Outgoing Group fields.
You can use the Line Number from the SIP Line tab for both values.
10. In the Manager navigation page, select Incoming Call Route.
11. On the Standard tab, in the Line Group ID field, enter the Line Number from the SIP
Line tab.
12. Select the Destinations tab.
13. In the Destination column, replace the value with a period (“.”).
14. In the Manager navigation pane, select Short Code.
15. Add a short code to dial the trunk you have just added.
16. One end of the trunk is now configured. Save the configuration to the IP Office.

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Configuring SIP Trunks

17. Using Manager, open the configuration for the IP Office at the other end of the SIP trunk
and repeat the steps.
Related links
Configuring SIP Trunks on page 744

SIP Line Requirements

Use of SIP requires the following:
• SIP Service Account An account or accounts with a SIP internet service provider (ITSP).
The method of operation and the information provided will vary. The key requirement is a SIP
URI, a web address of the form [email protected]. This is the equivalent of a SIP
telephone number for making and receiving calls via SIP.
• Voice Compression Channels SIP calls use system voice compression channels in the
same way as used for standard IP trunks and extensions. For an IP500 V2 system, these are
provided by the installation of VCM modules within the control unit. RTP relay is applied to
SIP calls where applicable.
• Licensing SIP trunks require licenses in the system configuration. These set the maximum
number of simultaneous SIP calls supported by the system.
• Firewall Traversal Routing traditional H.323 VoIP calls through firewalls often fails due to the
effects of NAT (Network Address Translation). For SIP a number of ways to ensure
successful firewall traversal can be used. The system does not apply any firewall between
LAN1 and LAN2 to SIP calls.
- STUN (Simple Traverse of UDP NAT) UDP SIP can use a mechanism called STUN to
cross firewalls between the switch and the ITSP. This requires the ITSP to provide the IP
address of their STUN server and the system to then select from various STUN methods
how to connect to that server. The system can attempt to auto-detect the required settings
to successfully connect. To use STUN, the line must be linked to the Network Topology
settings of a LAN interface using the line's Use Network Topology Info setting.
- TURN (Traversal Using Relay NAT) TCP SIP can use a mechanism called TURN
(Traversal Using Relay NAT). This is not currently supported.
- Session Border Control STUN does not have to be used for NAT traversal when SBC is
between IP Office and the ITSP, since the SBCE will be performing NAT traversal.
• SIP Trunks These trunks are manually added to the system configuration. Typically a SIP
trunk is required for each SIP ITSP being used. The configuration provides methods for
multiple URI's from that ITSP to use the same trunk. For each trunk at least one SIP URI
entry is required, up to 150 SIP URI's are supported on the same trunk. Amongst other things
this sets the incoming and outgoing groups for call routing.
• Outgoing Call Routing The initial routing uses any standard short code with a dial feature.
The short code's Line Group ID should be set to match the Outgoing Group ID of the SIP URI
channels to use. However the short code must also change the number dialed into a

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 SIP Line Requirements

destination SIP URI suitable for routing by the ITSP. In most cases, if the destination is a
public telephone network number, a URI of the form [email protected] is suitable.
For example:
- Code: 9N#
- Feature: Dial
- Telephone Number: N"@example.com"
- Line Group ID: 100
While this can be done in the short code, it is not an absolute necessity. The ITSP Proxy
Address or ITSP Domain Name will be used as the host/domain part.
• Incoming Call Routing Incoming SIP calls are routed in the same way as other incoming
external calls. The caller and called information in the SIP call header can be used to match
Incoming CLI and Incoming Number settings in normal system Incoming Call Route records.
• DiffServ Marking DiffServ marking is applied to calls using the DiffServer Settings on the
System | LAN | VoIP tab of the LAN interface as set by the line's Use Network Topology
Info setting.
SIP URIs
Calls across SIP require URI's (Uniform Resource Identifiers), one for the source and one for the
destination. Each SIP URI consists of two parts, the user part (for example name) and the domain
part (for example example.com) to form a full URI (in this case [email protected]). SIP URI's
can take several forms:
• [email protected]
• [email protected]
• [email protected]
Typically each account with a SIP service provider will include a SIP URI or a set of URI's. The
domain part is then used for the SIP trunk configured for routing calls to that provider. The user
part can be assigned either to an individual user if you have one URI per user for that ITSP, or it
can also be configured against the line for use by all users who have calls routed via that line.
If the wildcard * is used in the SIP trunk's Local URI, Contact and Display fields, that SIP trunk
will accept any incoming SIP call. The incoming call routing is still performed by the system
incoming call routes based on matching the values received with the call or the URI's incoming
group setting. For outgoing calls using this SIP URI, all valid short code CLI manipulations are
used (transforming calling party number to ISDN will be ignored). For a full list of valid CLI
manipulations, see “Telephone Number Field Characters” under Short Code Characters on
page 790. For example, character ‘i’ is not supported since it sets calling party number plan to
isdn and number type to national.
Resource Limitation
A number of limits can affect the number of SIP calls. When one of these limits is reached the
following occurs: any further outgoing SIP calls are blocked unless some alternate route is
available using ARS; any incoming SIP calls are queued until the required resource becomes
available. Limiting factors are:
• the number of licensed SIP sessions.

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Configuring SIP Trunks

• the number of SIP sessions configured for a SIP URI.

• the number of voice compression channels.
- SIP Line Call to/from Non-IP Devices Voice compression channel required.
- Outgoing SIP Line Call from IP Device No voice compression channel required.
- Incoming SIP Line Call to IP Device If using the same codec, voice compression
channel reserved until call connected. If using differing codecs then 2 channels used.
SIP Information Display
The full from and to SIP URI will be recorded for use by SMDR, CBC and CCC. For all other
applications and for telephone devices, the SIP URI is put through system directory matching the
same as for incoming CLI matching. First a match against the full URI is attempted, then a match
against the user part of the URI. Directory wildcards can also be used for the URI matching.
Related links
Configuring SIP Trunks on page 744

SIP Incoming Call Routing

Incoming SIP calls are routed using Incoming Call Routes in the same way as call arriving on
other external trunks. The following Incoming Call Route fields are used to determine which route
is the best match for a call.
Line Group ID This field is matched against the Incoming Group settings of the SIP URI (Line |
SIP URI). This must be an exact match.
Incoming Number This field can be used to match the called details (TO) in the SIP header of
incoming calls. It can contain a number, SIP URI or Tel URI. For SIP URI's the domain part of the
URI is removed before matching by incoming call routing occurs. For example, for the SIP URI
[email protected] , only the user part of the URI, ie. mysip, is used for matching.
Incoming CLI This field can be used to match the calling details (FROM) in the SDP header of
incoming SIP calls. It can contain a number, SIP URI, Tel URI or IP address received with SIP
calls. For all types of incoming CLI except IP addresses, a partial entry can be used to achieve the
match, records being read from left to right. For IP addresses only full entry matching is supported.
The fields Bearer Capability and Incoming Sub Address are not used for matching of incoming
SIP calls. The remaining Incoming Call Route fields, including those on the Voice Recording
page, are used as for all call types..
If the wildcard * is used in the SIP trunk's Local URI, Contact and Display fields, that SIP trunk
will accept any incoming SIP call. The incoming call routing is still performed by the system
incoming call routes based on matching the values received with the call or the URI's incoming
group setting. For outgoing calls using this SIP URI, all valid short code CLI manipulations are
used (transforming calling party number to ISDN will be ignored). For a full list of valid CLI
manipulations, see “Telephone Number Field Characters” under Short Code Characters on
page 790. For example, character ‘i’ is not supported since it sets calling party number plan to
isdn and number type to national.

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 SIP Prefix Operation

Related links
Configuring SIP Trunks on page 744

SIP Prefix Operation

The prefix fields Prefix, National Prefix, Country Code and International Prefix are available
with the SIP Line settings. These fields are used in the following order:
1. If an incoming number (called or calling) starts with the + symbol, the + is replaced with the
International Prefix.
2. If the Country Code has been set and an incoming number begins with that Country
Code or with the International Prefix and Country Code, they are replaced with the
National Prefix.
3. If the Country Code has been set and the incoming number does not start with the
National Prefix or International Prefix, the International Prefix is added.
4. If the incoming number does not begin with either the National Prefix or International
Prefix, then the Prefix is added.
For example, if the SIP Line is configured with prefixes as follows:
• Prefix: 9
• National Prefix: 90
• International Prefix: 900
• Country Code: 44
Number Received Processing Resulting Number
+441707362200 Following rule 1 above, the + is 901707362200
replace with the International
Prefix (900), resulting in
900441707362200.
The number now matches the
International Prefix (900) and
Country Code (44).Following rule
2 above they are replace with the
National Prefix (90).
00441707362200 Following rule 2 above the 90107362200
International Prefix (900) and
the Country Code (44) are
replaced with the National Prefix
(90).
441707362200 Following rule 2 above, the 901707362200
Country Code (44) is replace
with the National Prefix (90).
Table continues…

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Configuring SIP Trunks

Number Received Processing Resulting Number

6494770557 Following rule 3 above the 9006494770557
International Prefix (900) is
added.

OPTIONS Operation

Options are not sent only when active SIP registration is present. In all other cases, OPTIONS are
sent.
The interval is determined as by the No User source number SIP_OPTIONS_PERIOD=X as
follows.
• If no SIP_OPTIONS_PERIOD parameter is defined and the LAN1 | Network Topology |
Binding Refresh Time is 0, then the default value of 300 seconds is used.
• To establish a period less than 300 seconds, do not define a SIP_OPTIONS_PERIOD
parameter and set the Binding Refresh Time to a value less than 300 seconds. The
OPTIONS message period will be equal to the Binding Refresh Time.
• To establish a period greater than 300 seconds, a SIP_OPTIONS_PERIOD parameter must
be defined. The Binding Refresh Time must be set to a value greater than 300 seconds.
The OPTIONS message period will be the smaller of the Binding Refresh Time and the
SIP_OPTIONS_PERIOD.
Related links
Configuring SIP Trunks on page 744

SIP messaging
SIP trunk prerequisites
Before any calls can be made, the system must have sufficient SIP trunk licenses for the
maximum number of simultaneous SIP trunk calls expected.
On Server Edition systems, the System | Telephony | Telephony | Maximum SIP Sessions
value must match the total number of SIP extension and trunk calls that can occur at the same
time.
Related links
Configuring SIP Trunks on page 744
Outgoing call message details on page 751
Incoming call message details on page 755
Codec selection on page 760
DTMF transmission on page 761
Fax over SIP on page 761
Hold scenarios on page 761
SIP REFER on page 763

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 SIP messaging

Outgoing call message details

Related links
SIP messaging on page 750
Destination URI on page 751
From field content on page 751
To field content on page 752
Contact field content on page 752
P-Asserted Identity field content on page 752
Typical outgoing call scenarios on page 753

Destination URI
The destination URI in an INVITE message has the general format of an e-mail address. Specific
rules have been defined for expressing telephone numbers in this format. These rules are defined
in RFC 2806 and RFC 3261 (section 19.1.6). A sample URI for a call on a SIP trunk is:
sip: 12125551234@ITSP_Domain SIP/2.0
The ITSP_Domain in the following headers is taken from the SIP Line | ITSP Domain Name field.
If that is empty, the IP Address of the IP Office LAN interface is used or the public address of that
interface if topology discovery is used.
Related links
Outgoing call message details on page 751

From field content

If the call is originated from an IP Office endpoint, the settings on the SIP line | SIP URI tab
determine whether the information should be taken from the trunk’s SIP credentials, or from the
User | SIP tab.
• If the channel’s Local URI is set to * then the extension number is used for the User part of
the identity.
From: "SipDisplayNameAlice" <sip: 311@ITSP_Domain>;tag=8a9fed65b
• If the channel’s Local URI is set to ‘Use Internal Data’ then the User | SIP | SIP Name will be
used for the User part of the identity, and the ITSP Domain for the host part.
From: "SipDisplayNameAlice" <sip: SipName@ITSP_Domain>;tag=8a9fed65b
• If the SIP Name field also contains a domain (indicated by the presence of @) then that
domain will be used.
From: "SipDisplayNameAlice" <sip: SipName@USER_Domain>;tag=8a9fed65b
• If Call-ID is blocked either by short code or if the User | SIP | Anonymous checkbox is
checked then the From: header will be anonymous, unless the SIP Line | Send From in
Clear checkbox is checked.
From: "Anonymous" <sip:[email protected]>;tag=8a9fed65b

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• If the channel’s Local URI is set to Use Credentials … then there must first be at least one
set of SIP Credentials defined, and that account selected in the channel’s Registration
dropdown selection box. The corresponding field from the SIP Line | SIP Credentials tab will
be used for the User part of the identity.
From: "Line17Cred2" <sip:Line17Cred2@ITSP_Domain>;tag=8a9fed65b
• The contact identity is populated similarly to the From: header. If Call-Id blocking is invoked:
via W in a short code, or by checking the User | SIP | Anonymous checkbox then the
Contact: field becomes semi-anonymous:
Contact: <sip:[email protected]:5060;transport=udp
Related links
Outgoing call message details on page 751

To field content
Since the identity of the called party is not known at the time of the initial INVITE, the To: field
shows only the information necessary to route the call, which is the dialed digits after any short
code and ARS manipulation, prefix manipulation, and removal of any end-of-dial digits (# in North
America).
To: <sip: 12125551234@ITSP_Domain>
Related links
Outgoing call message details on page 751

Contact field content

The contact identity is populated similarly to the From: header. If Call-Id blocking is invoked: via W
in a short code, or by checking the User | SIP | Anonymous checkbox then the Contact: field
becomes semi-anonymous:
Contact: <sip:[email protected]:5060;transport=udp
Related links
Outgoing call message details on page 751

P-Asserted Identity field content

Without Call-Id blocking, this field essentially mirrors the From: field.
P-Asserted-Identity: " SipDisplayName " <sip: SipName@ITSP_Domain>
Note:
You can enter the wildcard character “*”. Entering this value populates the SIP PAI header
with the caller information available to IP Office.
Call-Id blocking: using W in a short code, or by checking the User | SIP | Anonymous checkbox
results in the P-Asserted field being the only header that carries the calling party information, and
so is unchanged from the non-blocked case above.

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 SIP messaging

P-Asserted-Identity: " SipDisplayName" <sip:SipName@ITSP_Domain>

Related links
Outgoing call message details on page 751

Typical outgoing call scenarios

INVITE with SDP, local ringback

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Configuring SIP Trunks

INVITE with SDP, early media

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 SIP messaging

INVITE with SDP, early media re-directed by destination

Related links
Outgoing call message details on page 751

Incoming call message details

Related links
SIP messaging on page 750
Incoming call routing on page 755
Media path connection on page 756
Typical incoming call scenarios on page 757

Incoming call routing

When a SIP INVITE is received by IP Office, its origin is compared to the known IP addresses of
the SIP lines configured. If a match is not found, then the INVITE is presented internally to the

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Configuring SIP Trunks

extension interface to determine if it matches any of the registered terminals. SIP messages from
unknown endpoints are discarded, and solicit no response from IP Office.
SIP lines have incoming and outgoing groups associated with them, which are configured on the
SIP line | SIP URI tab. In the example below, the incoming and outgoing groups are both 19, and
the Local URI specifies Use Internal Data. With this Local URI setting, to route a call to a user,
the User | SIP | SIP Name field is used to match against the incoming digits.

The incoming group indicates the identity of an Incoming Call Route, which is used to match the
incoming digits in the Request-URI to a target. That target could be an extension, a hunt group,
another trunk, or an ARS entry.
Due to this grouping, calls incoming to several different trunks or trunk types can use the same
Incoming Call Route, but in order for this to work,the Local URI must be manually set to <*>.
Incoming Call Routes are identified by the Line Group ID or optionally, an Incoming Number
may be specified to match against in the received digits. Then a Destination specified, which may
be a specific target, or may contain only a <.> to indicate that the digits are to be matched against
known system targets.
Related links
Incoming call message details on page 755

Media path connection

IP Office does not provide in-band ringback to incoming SIP trunk calls. This is different from what
is done for H.323. The only scenario in which an incoming SIP trunk call will hear in-band ringback
occurs when the call terminates on an analog trunk. With analog trunks, the media path is cut
through immediately because IP Office has no way of determining the state (ringing, busy,
answered) of the trunk.

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 SIP messaging

IP Office connects “early” media before the call is answered by sending a 183 Session Progress
response only if the following two conditions are met:
• A PROGRESS (in-band tone indication OR 183 Session Progress with SDP) message is
received from the destination (this can only happen in a SIP-to-PRI or SIP-to-SIP tandem
scenario).
• The INVITE message contains SDP.
IP Office does not attempt to connect early media on PROGRESS when there is no SDP in the
initial INVITE, since this is unlikely to succeed. The reason there is no SDP in INVITE is probably
that the originating system does not know the originator’s media address yet. A typical scenario
where this is the case occurs when the call on the originating system comes from an H.323
SlowStart trunk.
Related links
Incoming call message details on page 755

Typical incoming call scenarios

INVITE with SDP, local ringback
If the destination is an analog trunk, the 180 Ringing will be replaced with a 183 Progress with
SDP followed immediately by a “fake” answer in order that the media will be connected right away
so that the originator hears whatever in-band tones are present on the analog trunk (ringback or
busy). If the target is an extension that is unconditionally call forwarded over an analog trunk, then
there will be a 180 Ringing without SDP, followed immediately by the “fake” answer.

INVITE with SDP, early media

If the SIP Trunk receives a FAR_PROGRESS (in-band) message from its peer in the core (e.g.
from a tandem PRI or SIP trunk), it sends a 183 Session Progress message with SDP to the far
end. IP Office will connect the media on receipt of 180 or 183 with SDP.

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Configuring SIP Trunks

INVITE without SDP, local ring back

IP Office does not attempt to send early media in this scenario.

INVITE without SDP, early media

In this scenario, the far end attempts to connect media before the call is answered. IP Office does
not provide early media when receiving an empty INVITE, but rather 180 Ringing instead. There is

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 SIP messaging

no requirement to provide an SDP in the 180 Ringing provisional response, as that response is not
sent reliably using the PRACK mechanism.

INVITE without SDP, call terminates on H.323 endpoint

If the destination of the call is an H.323 trunk, the destination media address is not known when
the call is answered. Therefore, the SDP offer in 200 OK will contain a null port number (and IP
address). Once the logical channels are opened on the H.323 side, IP Office sends a re-INVITE
using the real media address.

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Configuring SIP Trunks

Related links
Incoming call message details on page 755

Codec selection
Codec selection is based on the Offer/Answer model specified in RFC 3264. The endpoint that
issues the offer includes the list of codecs that it supports. IP Office offers the codecs set on the
SIP line | VoIP tab, not those that are set on the extension.
The other endpoint sends an answer that should normally contain a single codec. If there are
multiple codecs in the answer, IP Office only considers the first codec. If the SIP Line is configured
to do Codec Lockdown (Re-Invite Supported is a prerequisite) then it will send another INVITE
with the single chosen codec.
Related links
SIP messaging on page 750

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 SIP messaging

DTMF transmission
DTMF over RTP (RFC 2833) can be used in IP Office. Asymmetric dynamic payload negotiation is
supported when it is necessary to bridge multiple SIP endpoints that do not support payload
negotiation. The value used for an initial offer is configured on the System | Codecs tab. The
default value is 101. Upon receipt of an offer with an RFC2833 payload type, IP Office will
automatically use the proposed value rather than its own configured value. This helps to support
networks that do not negotiate payload types.
There are cases in which direct media is desirable between SIP trunks and endpoints that do not
support RFC2833. To allow for this, if key presses are indicated from the extension, the IP Office
will ‘shuffle’ the media in. This connects its own media engine to the endpoint and to the SIP trunk,
and injects the digits in-band using the negotiated dynamic payload. After fifteen seconds of no
key presses, the media will be shuffled back out to re-establish a direct connection again.
Related links
SIP messaging on page 750

Fax over SIP

T.38 Fax over SIP is supported on the IP500 V2 platform deployed as standalone or as an
expansion gateway. G.711 fax is also supported, and is supported on Linux servers. For networks
that do or do not support T.38, IP Office allows both G3 and Super G3 fax machines to
interoperate.
There are configuration parameters that control the behavior in different networks. If T.38 is
supported in a network, then it may make sense to select T.38 as the Fax Transport preference in
order to make use the inherent quality provided by the redundancy mechanisms. On the other
hand, if all of the fax machines in the network are Super G3 capable, there may be a need to take
advantage of the increased speed that this encoding provides. Since T.38 is not capable of
encoding Super G3, G.711 may be a better choice for the Fax Transport. In either case, IP Office
will accept codec change requests from the far SIP endpoint to switch to T.38 or to G.711.
T.38 Fax Transport and Direct Media are mutually exclusive on a given SIP Line. IP Office keeps
itself in the media path so that it can detect fax tones to make the switch to T.38.
Related links
SIP messaging on page 750

Hold scenarios
Hold originated by IP Office
When an IP Office DS extension or non-IP trunk puts a SIP trunk on hold, there is no indication to
the network. The voice path is merely switched in the TDM domain to the appropriate hold

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Configuring SIP Trunks

treatment source, be it tones, silence or music. For IP extensions and trunks, be they H.323 or
SIP, if the call uses direct media, there will be a re-INVITE sent to redirect the media source from
the extension or trunk endpoint to a port on the IP Office in order to connect hold treatment. When
the call is then unheld, another INVITE will go out to connect the extension with the far end.

Hold originated by far end

The far end of a SIP trunk can put the IP Office on hold by sending it re-INVITE with an SDP Offer
containing:
• A sendonly attribute. IP Office replies with an SDP Answer containing the recvonly attribute.
• An inactive attribute. IP Office replies with inactive.
• A zero media connection address (c=0.0.0.0). IP Office replies with inactive.
Unhold
A held call is unheld by means of an SDP Offer with the sendrecv attribute (or no direction
attribute, since sendrecv is assumed if not specified).

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 SIP messaging

Unhold from mutual hold

Either end can un-hold the other end, i.e., allow it to transmit again, by sending a new Offer with
the sendrecv or recvonlyattribute. The other end replies with sendonly if the call is still on hold
at its end.
Related links
SIP messaging on page 750

SIP REFER
After a SIP call has been established between two parties (the “Primary” call), the SIP REFER
method is used by the TransferOR end of the call to transfer the TransferEE end to a Transfer
Target. The REFER message provides the Transfer Target’s contact information in the Refer-To
header. This causes the TransferEE to establish the Secondary call to the Transfer Target, thus
completing the transfer.
For public SIP trunks, IP Office supports only consultative call transfer using REFER. Consultative
transfer is also known as Attended. With consultative transfer, the TransferOR puts the Primary
call on hold and establishes a Consult call to another party. After the consultation, the TransferOR
completes the transfer, causing the TransferEE to connect to the Transfer Target, thereby
replacing the Transfer Target’s call with the TransferOR.
REFER can be configured to accept incoming, reject incoming, or decide based on the presence
of REFER in the Allow: header in responses to OPTIONS messages. Similarly, there is the same
configuration for outgoing REFER.
Although the TransferOR and TransferEE must be SIP endpoints, the Transfer Target may be a
TDM, PRI, H.323 or SIP terminal on the same IP Office, or an endpoint reachable over the same
SIP line as the REFER request is received from.

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Configuring SIP Trunks

Related links
SIP messaging on page 750

IP Office SIP trunk specifications

This section outlines the SIP trunk capabilities supported by IP Office.
Related links
Configuring SIP Trunks on page 744
RFCs on page 766
Transport protocols on page 767

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 IP Office SIP trunk specifications

Request methods on page 767

Response methods on page 767
Headers on page 768

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Configuring SIP Trunks

RFCs
• ITU-T T.38 Annex D, Procedures for real-time • RFC 3515 – The Session Initiation Protocol (SIP)
Group 3 facsimile communication over IP Refer method
networks
• RFC 3550 - RTP: A Transport Protocol for Real-
• RFC 1889 - RTP: A Transport Protocol for Real- Time Applications
Time Applications
• RFC 3551 - RTP Profile for Audio and Video
• RFC 2327 - SDP: Session Description Protocol Conferences with Minimal Control
• RFC 2617 - HTTP Authentication: Basic and • RFC 3665 - Session Initiation Protocol Basic Call
Digest Access Authentication Flow Examples
• RFC 2833/RFC 4733 - RTP Payload for DTMF • RFC 3666 - Session Initiation Protocol PSTN Call
Digits, Telephony Tones and Telephony Signals Flows
• RFC 2976 - The SIP INFO Method • RFC 3725 - Best Current Practices for Third Party
Call Control (3pcc) in the Session Initiation
• RFC 3087 - Control of Service Context using SIP
Protocol (SIP)
Request-URI
• RFC 3824 - Using E.164 numbers with the
• RFC 3261 - Session Initiation Protocol
Session Initiation Protocol (SIP)
• RFC 3262 - Reliability of Provisional Responses
• RFC 3842 - A Message Summary and Message
in the Session Initiation Protocol (SIP)
Waiting Indication Event Package for the Session
• RFC 3263 - Session Initiation Protocol (SIP): Initiation Protocol
Locating SIP Servers
• RFC 3891 - The Session Initiation Protocol (SIP)
• RFC 3264 - An Offer/Answer Model with the "Replaces" Header
Session Description Protocol (SDP)
• RFC 3960 - Early Media and Ringing Tone
• RFC 3311 - The Session Initiation Protocol (SIP) Generation in the Session Initiation Protocol (SIP)
UPDATE Method
• RFC 4028 - Session Timers in the Session
• RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP)
Initiation Protocol (SIP)
• RFC 4566 - SDP: Session Description Protocol
• RFC 3325 - Private Extensions to the Session
• RFC 5359 - Session Initiation Protocol Service
Initiation Protocol (SIP) for Asserted Identity within
Examples
Trusted
• RFC 5379 - Guidelines for Using the Privacy
• RFC 3326 - The Reason Header Field for the
Mechanism for SIP
Session Initiation Protocol (SIP)
• RFC 5806 - Diversion Indication in SIP
• RFC 3398 - Integrated Services Digital Network
(ISDN) User Part (ISUP) to Session Initiation • RFC 5876 - Updates to Asserted Identity in the
Protocol (SIP) Mapping Session Initiation Protocol (SIP)
• RFC 3407 - Session Description Protocol (SDP) • RFC 6337 - Session Initiation Protocol (SIP)
Simple Capability Usage of the Offer/Answer Model
• RFC 3489 - STUN - Simple Traversal of User • RFC 6432 - Carrying Q.850 Codes in Reason
Datagram Protocol (UDP) Through Network Header Fields in SIP (Session Initiation Protocol)
Address Translators (NATs) Responses

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 IP Office SIP trunk specifications

Related links
IP Office SIP trunk specifications on page 764

Transport protocols
• UDP
• TCP
• RTP
• RTCP
Related links
IP Office SIP trunk specifications on page 764

Request methods
• INVITE • NOTIFY
• ACK • PRACK
• BYE • OPTIONS
• CANCEL • UPDATE
• INFO • PUBLISH
• REFER • MESSAGE
• REGISTER • PING
• SUBSCRIBE

Related links
IP Office SIP trunk specifications on page 764

Response methods
• 100 Trying • 202 ACCEPTED
• 180 Ringing • 3XX
• 181 Call Is Being Forwarded • 4XX
• 182 Call Queued • 5XX
• 183 Session progress • 6XX
• 200 OK

Related links
IP Office SIP trunk specifications on page 764

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Configuring SIP Trunks

Headers
• Accept • P-Early-Media
• Alert-Info • P-Preferred-Identity
• Allow • Privacy
• Allow-Event • Proxy-Authenticate
• Authorization • Proxy-Authorization
• Call-ID • Proxy-Require
• Contact • Require
• Content-Length • Remote-Party-ID
• Content-Type • Server
• CSeq • Session-Timers
• Diversion • Supported
• From • To
• History-Info • User-Agent
• Max-Forwards • Via
• P-Asserted-Identity • WWW-Authenticate

Related links
IP Office SIP trunk specifications on page 764

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Chapter 16: Configuring Small Community
Networking

Systems linked by IP Office Line IP trunks can enable voice networking across those trunks to form
a multi-site network. Within a multi-site network, the separate systems automatically learn each
other's extension numbers and user names. This allows calls between systems and support for a
range of internal call features, see Supported Small Community Network Features.
Capacity
The following are the supported capacity limits for a Small Community Network system.
Maximum Number of Systems 32
Maximum Number of Users 1000
Maximum H.323 Line Hops Between Systems 5
Star H.323 Line Layout
Serial H.323 Line Layout
Mesh H.323 Line Layout

Configuration Summary
To set up a Small Community Network, the following are required:
A working IP Office Line trunk between the systems, that has been tested for correct voice and data
traffic routing.
• The arrangement the IP Office Line trunks must meet the requirements detailed in Supported
Small Community Network Layouts.
• Within a particular system, all SCN trunks should be on the same LAN interface.
• VCM channels are required in all systems.
• The extension, user and group numbering on each system must be unique.
• The user and group names on each system must be unique.
• We also recommend that all names and numbers (line, services, etc) on the separate systems
are kept unique. This will reduce potential maintenance confusion.
• The Outgoing Group ID on the Small Community Network lines should be changed to a
number other than the default 0.
• All systems should use the same set of telephony timers, especially the Default No Answer
Time.

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• Check that all systems in the network are configured to use the same Codecs.
• Only one system should have its Voicemail Type set to Voicemail Pro/Lite. All other systems
must be set to either Centralized Voicemail or Distributed Voicemail. No other settings are
supported.
Software Level Interoperation
Small Community Networks is supported between systems with the same major software level or
one level of difference in major software level. For example between 9.1 and 9.0 (same major level)
and between 8.0 and 9.0 (one major level of difference).
This option is intended mainly to allow the phased upgrading of sites within a Small Community
Network. It is still recommended that all systems within a network are upgraded to the same level
where possible. Within a Small Community Network including differing levels of software, the
network features and capacity will be based on the lowest level of software within the network.
Related links
Supported Small Community Network Network Layouts on page 770
Telephone Features Supported Across Server Edition and SCN Networks on page 103
Voicemail Support on page 772
Enabling Small Community Networking on page 773
Small Community Network Management on page 775
Small Community Network Remote Hotdesking on page 784
Small Community Network Fallback on page 785
SCN Short Code Programming on page 786

Supported Small Community Network Network Layouts

The allowed arrangement of IP Office Lines between the systems depends on the lowest software
level of any system in the network. Note that we are referring to IP Office Lines configured in the
system configurations. The actual IP network configuration, including IP routes in the system
configurations, can differ as per the customer network requirements.
Star/Serial Layouts The following are examples of star and serial layouts.

---- = IP network, | = IP Office Line.

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 Telephone Features Supported Across Server Edition and SCN Networks

Mesh Layout A mesh layout is one where there is more than one possible IP Office Line route
between any two systems. The following are examples of mesh layouts. Mesh, star and serial
layouts can be combined.

Small Community Network Signalling Small Community Network uses a signalling similar to
RIP is order to update each other of there presence. This traffic can be seen in the System
Monitor application as AVRIP packets. This traffic is is sent to port 50795 on which each system
listens.
Each system in the Small Community Network transmits an update every 30 seconds. Additionally
BLF updates are transmitted when applicable up to a maximum of every 0.5 seconds. Typically
the volume is less than 1Kbps per system.
Related links
Configuring Small Community Networking on page 769

Telephone Features Supported Across Server Edition and

SCN Networks
Each system running IP Office in a multi-site network acts as a self-contained IP Office telephone
system. In addition to the remote systems sharing knowledge of user and hunt group extension
numbers, the following additional telephony features are supported between systems in a multi-
site network. Features not listed are not supported across the multi-site network.
• Absence Text
• Advertised Hunt Groups Hunt groups set to advertised can be dialed by users on other
systems
• Anti-tromboning Calls routed across the multi-site network and back to the originating
system are turned back into internal calls on the originating system only.
• Break Out Dialing
• Call Park / Unpark Call
• Call Pick-up Extension
• Call Tagging
• Callback When Free
• Centralized Call Log

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Configuring Small Community Networking

• Centralized Personal Directory

• Conference
• Distributed Hunt Groups
• Distributed Voicemail Server Support When using Vociemail Pro, each system can support
its own Voicemail Pro server.
• Enable ARS / Disable ARS
• Extension Dialing Each system automatically learns the user extension numbers available
on other systems and routes calls to those numbers.
• Resiliency Options
• Fax Relay
• Follow Me Here / Follow Me To
• Forwarding
• Hold Held calls are signalled across the network.
• Internal Twining
• Intrusion Features
• Mobile Call Control Licensed mobile call control users who remote hot desk to another
system take their licensed status with them.
• Music On Hold Source Selection
• Remote Hot Desking
• Set Hunt Group Out of Service / Clear Hunt Group Out of Service
• Transfer Calls can be transferred to network extensions.
• User DSS/BLF Monitoring of user status only. The ability to use additional features such as
call pickup via a USER button will differ depending on whether the monitored user is local or
remote. Indication of new voicemail messages provided by SoftConsole user speed dial icon
is not supported.
• User Profile Resilence When a user hot desks to another system, they retain their Profile
settings and rights.
Related links
Working with the Server Edition Manager User Interface on page 98
Configuring Small Community Networking on page 769

Voicemail Support
Within a Small Community Network, a single Voicemail Pro can be used to provide voicemail
services for all the systems. For full details of installation and setup refer to the Voicemail Pro
documentation. The Voicemail Pro is licensed and hosted by a chosen central system and

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 Enabling Small Community Networking

provides full operation for that system. The voicemail features supported for the other remote
systems are listed below:
The use of additional Voicemail Pro servers is supported. The distributed severs provide call
recording and auto attendant functions to their local system. The central Voice Pro server is still
used as the message store for all messages. Refer to the Voicemail Pro documentation.
• User mailboxes.
• Call recording. Recording of incoming call routes is only supported for destinations on the
same system, not for remote Small Community Network destinations.
• Dial by Name.
• Auto Attendants.
• Breakout Requires that the numbers used are routable by the system hosting the voicemail
server.
• Announcements
• UMS Web Services Users for UMS Web Services (IMAP and or web voicemail) are licensed
through the UMS Web Services license on their host system. This applies even if the user
remote hot desks to another system in the Small Community Network.
Related links
Configuring Small Community Networking on page 769

Enabling Small Community Networking

The process below adds an IP Office Line to the system configuration. It is assumed that data
routing between the systems has already been configured and tested. Adding Small Community
Network connections between systems can also be done using Manager's Small Community
Network Management mode.
Related links
Configuring Small Community Networking on page 769

Setup the VoIP Line from System A to System B

About this task
Receive the system configuration for System A. Prepare the system for addition to the Small
Community Network:

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Configuring Small Community Networking

Procedure
1. Change all extensions numbers and names to values that will be unique within the multi-
site network.
• For users and extensions this can be done using the Extension Renumber tool. That
will adjust all users and extension and all items using those numbers, for example hunt
group memberships and incoming call routes.
• For hunt groups, each hunt group must be change individually.
2. Click Line to display a list of existing lines.
3. Right-click on the displayed list and select New and then IP Office Line.
4. Select the Line tab and set the following:
• In the Transport Type field, select Proprietary.
• In the Networking Level field, select SCN.
• In the Description field, enter a description of the link. For example System B Small
Community Network.
• Set the Outgoing Group ID to a unique value. For example match the automatically
assigned Line Number value.
5. Under Gateway, set the following:
• For the Gateway IP Address, enter the IP address of the remote System B.
• Use of IP Office SCN - Fallback is detailed in Small Community Network Fallback.
6. Click the VoIP Settings tab.
•
• Select the preferred Compression Mode. The same mode must be used by all VoIP
lines and extensions within the network.
• The other option can be configured as required but must be matched by the other IP
Office Lines in the network. For example the Silence Suppression settings on all the
network trunks must match.
7. Select System | Voicemail.
a. Only one system should have its Voicemail Type set to Voicemail Pro/Lite.
The Voicemail IP Address will be the IP address of the central voicemail server PC.
b. Any other system with its own Voicemail Pro server PC should have its Voicemail
Type set to Distributed Voicemail.
The Voicemail IP Address should be the IP address of the distributed voicemail
server PC. The Voicemail Destination should be set to the Outgoing Group ID used
for the Small Community Network line to the system that is set as Voicemail Pro/Lite.
c. All other systems should have their Voicemail Type set to Centralized Voicemail.

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 Small Community Network Management

The Voicemail Destination should be set to the Outgoing Group ID used for the
Small Community Network line to the system that is set as Voicemail Pro/Lite.
8. Save the configuration and reboot System A.
Next steps
Set up the IP Office Line from B to A.

Setup the VoIP Line from System B to System A

Procedure
1. On the remote system, repeat the previous steps to create an IP Office Line to System A.
As stated above, where possible the line settings, especially the VoIP settings, must match
those used for other IP Office Lines in the network.
2. Load the configuration and reboot the remote system.
Next steps
Test by making calls between extensions on the different systems

Small Community Network Management

Manager supports the ability to load and manage the configurations of the systems in a Small
Community Network at the same time. Manager must be enabled for Small Community Network
discovery.
When the configurations of the systems in a Small Community Network are loaded, Manager
switched to Small Community Network management mode. This differs from normal system
configuration mode in a number of ways:
• A network viewer is available. In addition to giving a graphical view of the Small Community
Network, the view can be used to add and remove links between the systems in the Small
Community Network.
• In the configuration tree, the records for users and hunt groups on all systems are grouped
together.
• Time Profiles and User Right common to all systems are grouped together.
• The configuration settings for each system in the Small Community Network can be
accessed and edited.
Related links
Configuring Small Community Networking on page 769
Enabling SCN Discovery on page 776
Creating a Common Admin Account on page 776
Loading a Small Community Network Configuration on page 777

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Configuring Small Community Networking

Editing a Small Community Network Configuration on page 778

Using the Network Viewer on page 779
System Inventory on page 784

Enabling SCN Discovery

About this task
In order for the Select IP Office menu to groups systems in a Small Community Network and
allow loading of all the Small Community Network configurations, Manager must be enabled for
SCN discovery.
Procedure
1. Select File | Preferences.
2. Select the Discovery tab.
3. Select the SCN Discovery option.
4. Check that the other discovery setting are sufficient to discover all the systems in the Small
Community Network.
5. Click OK.
Related links
Small Community Network Management on page 775

Creating a Common Admin Account

About this task
When managing multiple systems, it may be useful to create a common user name and password
on all the systems for configuration access. This tool can be used to create a new service user
account, SCN_Admin, for configuration access.
This process requires you to have a user name and password for security configuration access to
each of the systems.
Select Tools | SCN Service User Management.
The option is not shown if a Basic Mode system configuration is loaded. If no configuration is
loaded, and the option is not shown, select View | Advanced View.
Procedure
1. The Select IP Office menu displays the list of discovered systems.
2. Select the systems for which you want to create a common configuration account.
Click OK.
3. A user name and password for security configuration access to each system is requested.

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 Small Community Network Management

Enter the values and click OK. If the same values can be used for all systems enter those
values, select Use above credentials for all remaining, selected IPOs. If each system
requires a different security user names and password, deselect Use above credentials
for all remaining, selected IPOs.
4. The systems will be listed and whether they already have an SCN_Admin account is
shown.
5. To create the SCN_Admin account on each system and set the password for those
account click on Create Service User.
6. Enter the common password and click OK.
7. The password can be changed in future using the Change Password option.
8. Click Close.
Related links
Small Community Network Management on page 775

Loading a Small Community Network Configuration

About this task
If Manager is configured with SCN Discovery enabled, the Select IP Office menu will display any
SCNs it discovers.
Procedure
1. With no configuration loaded, click on or select File | Open Configuration.
2. The Select IP Office menu is displayed.
Any systems in a Small Community Network will be grouped together.
3. To load the configuration of all the systems in the network, click the check box next to the
network name and then click OK.

If a warning icon is displayed next to the SCN check box, it indicates that not all the
systems known to be in the Small Community Network were discovered. Hovering the
cursor over the icon will display details of the missing systems. Loading the network
configuration at this time would not include the configuration of the missing system or
systems. The missing systems:
• May be disconnected
• The discovery settings for the Manager PC may be incorrect.
• The data routing between the Manager PC and the missing systems may be incorrect or
blocked.
4. Enter the name and password for configuration access to each system.
If the systems all have a common user name and password (see Common Administrator
Access below), select Use above credentials for all remaining selected IPOs. Click OK.

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5. Manager will load and display the combined configurations in Small Community Network
Management mode.
Related links
Small Community Network Management on page 775

Editing a Small Community Network Configuration

When the configuration of a Small Community Network is loaded, Manager displays the
configuration in a different way from when the configuration of a single system is loaded. The main
differences are in how configuration records are grouped in the configuration tree.

Clicking on the Small Community Network icon displays the Network Viewer which shows the
lines between the systems in the Small Community Network.
Small Community Network Configuration Records Certain records from each of the systems in
the Small Community Network are grouped together in the configuration tree differently from when
just a single system configuration is loaded. There are two types, unique Small Community
Network records and shared Small Community Network records:
Unique Records They can be edited here and the system to which they belong is indicated in the
group pane and in the title bar of the details pane. However, to add or delete these types of record
must be done within the configuration records of the particular system that will host the entry's
configuration details.
•
All user in the Small Community Network are shown under the User icon.
•
All hunt groups in the Small Community Network are shown under the Hunt Group icon.
Shared Records Shared records are configuration items that exist on all systems in the Small
Community Network, having the same name and settings on each system. Editing the shared
record updates the matching copy in the configuration of each system. Similarly, adding or
deleting a shared record adds or deletes from the individual system configurations. If the copy of
the shared record within an individual configuration is edited, it is no longer a shared record for the
Small Community Network though the individual records on other system will remain. Changing
the individual records back to matching will turn the records back into a shared record.
•
Shared time profiles are shown under the Time Profile icon.
•
Shared user rights are shown under the User Rights icon.

Individual System Configurations The full configuration for each system in the Small
Community Network can be accessed and edited as required. It is possible to copy and paste
configuration records between systems using the configuration tree.
Saving Changes

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 Small Community Network Management

When the save icon or File | Save Configuration is selected, the menu for multiple
configuration saves is displayed. It provides similar options are for a normal single configuration
save. Note that when working in Small Community NetworkManagement mode, after saving
configuration changes the Manager will always close the displayed configuration.

Change Mode If Manager thinks the changes made to the configuration settings are mergeable, it
will select Merge by default, otherwise it will select Reboot.
• Merge Send the configuration settings without rebooting the system. This mode should only
be used with settings that are mergeable. Refer to Mergeable Settings.
• Reboot Send the configuration and then immediately reboot the system.
• Reboot When Free Send the configuration and reboot the system when there are no calls in
progress. This mode can be combined with the Call Barring options.
• Timed The same as When Free but waits for a specific time after which it then wait for there
to be no calls in progress. The time is specified by the Reboot Time. This mode can be
combined with the Call Barring options.
Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for the
system reboot. If the time is after midnight, the system's normal daily backup is canceled.
Call Barring These settings can be used when the reboot mode Reboot When Free is selected.
They bar the sending or receiving of any new calls.
Error Status The warning will appear if the configuration being sent contains any validation errors
indicated by a icon in the error pane. The configuration can still be sent if required.
Related links
Small Community Network Management on page 775

Using the Network Viewer

Clicking on Small Community Network in the configuration tree displays the Network Viewer. This
shows each of the systems in the Small Community Network and the links between each of the
systems. Systems with attached Voicemail Pro servers are also indicated.

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Green System with Voicemail Pro system.

Black Small Community Network line between two systems.
Red Incorrect Small Community Network line between systems (probably one-way connection).
Right-click on the line and select Repair.
You can use the Network Viewer to perform a range of functions:
• Arrange the View
• Launch System Status
• Launch Voicemail Pro
• Add an IP Office Line
• Add a system
• Remove an IP Office Line
• Remove a system from the Small Community Network
• Repairing an IP Office Line
• Add a Background Image

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Related links
Small Community Network Management on page 775

Arranging the View

About this task
Procedure
You can click and drag items around in order to position them where required.
Alternatively if you right click on the view you can select Auto Arrange.
Note:
The position of elements in the network view are stored as part of the system configuration.
Therefore changes to the view will require the configuration to be saved.

Adding a Line Within the Small Community Network

About this task
You can use the network viewer to add a Small Community Network link between two systems in
the Small Community Network that are currently linked. This process will add new H.323 Small
Community Network line records to the configurations of each of the systems.
Note:
Adding a line between systems will require those systems to reboot when the changes are
saved.
Procedure
1. Right click on the starting system for the link.
Select Connect To and select the name of the other system in the Small Community
Network to which you want to link.
2. Select the type of line, IP Office SCN or IP Office SCN-Fallback.
3. Click OK.
If Small Community Network-Fallback is selected, the actual backup function still need to
be configured.
a. The newly added line is displayed in the network viewer.
b. Click OK.

Repairing a Line Within the Small Community Network

About this task
A red line in the network viewer indicates a incorrectly configured line between two systems in the
Small Community Network. Typically this will be a line configured in one of the systems but not
matched by a line configured in the other system. The network viewer can be used to correct this
error.

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Procedure
1. Note that adding a line between systems will require those systems to reboot when the
changes are saved.
2. Right click on the red line and select Repair Line.
3. The line is changed to black.
4. Click OK.

Adding a System to the Small Community Network

About this task
You can use the network viewer to add a Small Community Network line to a system not yet in the
Small Community Network. This process will add new H.323 Small Community Network line
records to the configurations of each of the systems.
Procedure
1. Note that adding a line between systems will require those systems to reboot when the
changes are saved.
2. Right click on the starting system for the link.
Select Connect To and select Discovery.
3. The Select IP Office menu will display any discoverable systems not already in the Small
Community Network.
If the discovery includes systems already in another Small Community Network it will not
indicate such. If you want to add such a system in order to join the SCNs you can do so.
However after adding the system, you should immediately save the configuration and
reload the Small Community Network.
a. Select the required system and click OK.
b. Enter the name and password for configuration access to the selected system and
click OK.
c. The newly added system is displayed in the network viewer.
d. Click OK.
The configuration of the newly added system is now included in the configuration tree.
e. If the Error List is visible (View | Error Pane), check that none of the error are Small
Community Network specific errors, for example duplicate names or extension
numbers.

Removing a Small Community Network Line

About this task
You can use the network viewer to remove the Small Community Network lines between two
systems in the Small Community Network.

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Procedure
1. Note that removing a link between systems will require those systems to reboot when the
changes are saved.
2. Right click on the link and select Delete Line.
3. The line is removed from the network viewer.
4. Click OK.

Removing a System
About this task
You can use the network viewer to remove a system from the Small Community Network.
Procedure
1. Note that removing a system will require previous linked systems to reboot when the
changes are saved.
2. Right click on the system and select Remove From Small Community Network.
3. Any lines to other system in the Small Community Network are removed.
4. Click OK.

Start System Status

About this task
If the System Status Application is also installed on the Manager PC, you can start it for a
particular system.
Procedure
1. Right click on the system and select System Status.
2. The application is started and the login form pre filled with the IP address of the system.

Start Voicemail Pro

About this task
If the Voicemail Pro client is also installed on the Manager PC, you can start it for the any system
with an associated Voicemail Pro server.
Procedure
Right click on the voicemail server icon and select Launch VMPro Client.

Add a Background Image

About this task
You can select an image file to be displayed in the background of the Network Viewer display. This
file is not saved as part of the configuration in any way, ie. if the image file is moved or deleted it is
not longer used by Manager.

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Procedure
1. Right click on the general background area of the network viewer and select Background
Image.
2. Select Set Background Image to browse to the location of the file to be used.
3. The Visible option can be used to switch the display of the background image on or off.

System Inventory
When working in Small Community Network Management mode, clicking on the System icon for a
particular system displays a system inventory page for that system.

Related links
Small Community Network Management on page 775

Small Community Network Remote Hotdesking

The system supports hotdesking between systems within a network.
In the descriptions below, the system on which the user is configured is termed their 'home'
system, all other systems are 'remote' systems.
When a user logs in to a remote system:
• The user's incoming calls are rerouted to that system.

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• The user's outgoing calls uses the settings of the remote system.
• The user's license privileges move with them, for example their user profile setting is
retained. The host system does not need to be licensed for the user.
• The user's own settings are transferred. However, some settings may become unusable or
may operate differently.
• User rights are not transferred to the remote system but the name of any user rights
associated with the user are transferred. If user rights with the same name exist on the
remote system, then they will be used. The same applies for user rights applied by time
profiles, if time profiles with the same name exist on the remote system .
• Appearance buttons configured for users on the home system will no longer operate.
• Various other settings may either no longer work or may work differently depending on the
configuration of the remote system at which the user has logged in.
• The rights granted to the user by their Profile settings are retained by the user. There is no
requirement for the remote system to have the appropriate licenses for the Profile.
If the user's home system is disconnected while the user is remotely hot desked, the user will
remain remotely hot desked. They can remain in that state unless the current host system is
restarted. They retain their license privileges as if they were on their home system. Note however
that when the user's home system is reconnected, the user may be automatically logged back
onto that system.
Break Out Dialing In some scenarios a hot desking user logged in at a remote system will want to
dial a number using the system short codes of another system. This can be done using either
short codes with the Break Out feature or a programmable button set to Break Out. This feature
can be used by any user within the multi-site network but is of most use to remote hot deskers.
Related links
Configuring Small Community Networking on page 769

Small Community Network Fallback

Each system in the Small Community Network can include one IP Office Line where the SCN
Resiliency Options is set to Supports Resiliency. The system to which the IP Office Line
connects is then requested to provide fallback support for selected options for the local system.
• Note that both ends of the SCN trunk connection must be set to fallback.
• On the system requesting backup, the required SCN Resiliency Options are selected,
indicating that it is requesting backup. A single system can only request backup from one
other system.
• A system providing backup can provide backup for up to 7 other systems.

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Configuring Small Community Networking

Resiliency is supported on Server Edition systems for Avaya 1600 and 9600 series H.323 phones.
IP500 V2 systems also support 4600 and 5600 series phones. Resiliency is configured on Line |
IP Office Line | Line under SCN Resiliency Options. The options supported are:
• Backs up my IP Phones
• Backs up my Hunt Groups
• Backs up my Voicemail
• Backs up my IP DECT Phones
Phone Resiliency
When Backs up my IP Phones is selected, the local system shares information about the
registered phones and users on those phones with the other system. If the local system is no
longer visible to the phones, the phones reregisters with the other system.
Failback Recovery: If the setting System | Telephony | Telephony | Phone Failback is set to
Automatic, and the phone’s primary gatekeeper has been up for more than 10 minutes, the
system causes idle phones to perform a failback recovery to the original system.

Notes
• Fallback handover takes approximately 3 minutes. This ensure that fallback is not invoked
when it is not required, for example when the local system is simply being rebooted to
complete a non-mergeable configuration change.
• Fallback is only intended to provide basic call functionality while the cause of fallback
occurring is investigated and resolved. If users make changes to their settings while in
fallback, for example changing their DND mode, those changes will not apply after fallback.
• If the fallback system is rebooted while it is providing fallback services, the fallback services
are lost.
• Fallback features require that the IP devices local to each system are still able to route data
to the fallback system when the local system is not available. This will typically require each
system site to be using a separate data router from the system.
• When an IP Phone re-registers to a secondary IP Office on the failure of the primary control
unit, the second system will allow it to operate indefinitely as a “guest”, but only until the
system resets. Licenses will never be consumed for a guest IP phone.
• Remote hot desking users on H323 extensions are automatically logged out.
Related links
Configuring Small Community Networking on page 769

SCN Short Code Programming

In a multi-site network, the systems automatically learn each others extension numbers and route
calls appropriately. However the same does not apply to dialing of other numbers. Using short
codes it is possible to have the dialing of numbers on one system to be redirected to another
system and dialed there. The dialing is then matched against the short codes available on the
remote system.

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 SCN Short Code Programming

Scenario
We want a short code on System A which will correctly route any 3000 range number to System
B. This will allow System B group numbers to be dialed from System A. To achieve the above
scenario, we will add a new system short code. By using a system short code it becomes available
to all users.
Example Short Code
In the configuration for System A.
1. Click Short codeto display a list of existing system short codes.
2. Right-click on the displayed list and select New.
3. Enter the short code settings as follows:
• Short Code: 3XXX This will match any four-digit number beginning with 3.
• Telephone Number: . The . indicates that the short code should output the digits as dialed.
• Line Group ID: 99999 This should match the Outgoing Group ID given to the IP Office Line
connected to System B.
• Feature: Dial
Click OK.
A similar system short code can be added to System B's configuration to route 2XXX dialing to
System A.
Related links
Configuring Small Community Networking on page 769

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Chapter 17: Short Code Overview

Whenever the system receives a set of digits to process, if those digits do not match a user or group
extension number, the system will look for a short code match. The matching short code then
defines what action (short code feature) should be applied to the call, where it should be routed and
which of the dialed digits, if any, should be used in the subsequent action.
This applies to digits dialled by a telephone user, sent by a user selecting a directory contact or
speed dial, and in some cases to digits received with an incoming call on a line.
This section provides an overview of short codes configuration and use.

Warning:
The dialing of emergency numbers must not be blocked. Whenever short codes are edited, you
must ensure that the ability of users to dial emergency numbers is tested and maintained. This
is typically done by ensuring that the dialing of an emergency number always matches a user or
system short code set to the Dial Emergency feature. If the system uses external dialing
prefixes, you should also ensure that the dialing of emergency numbers including the prefix is
not blocked.

Short Code Fields:

Each short code has the following fields:
• Short Code: The digits which, if proved to be a best match to the dialed digits, trigger use of
the short code. In addition to the normal dialing digits (0 to 9 plus * and #), characters can also
be used as follows:
- Some characters have special meaning. For example, the wildcard X to match any single
digit or N to match any set of digits. See Short Code Characters on page 790
- Using characters also allows the creation of short codes which cannot be dialed from a
phone but can be dialed from some applications.
• Telephone Number: The number used by the short code feature if needed, for example the
outgoing number for a call to be passed to an external telephone line. Again special characters
can be used in this field, see Short Code Characters on page 790.
• Line Group ID: This field is used for short codes that result in a number to be dialed, that is
any short code set to one of the various Dial short code features. When that is the case, this
field specifies the outgoing line group or ARS form to be used for the call. .
• Feature: This sets the action to performed by the short code. See Short Code Features on
page 809.

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 • Locale: Features that transfer the call to voicemail indicate the language required. If the
required set of language prompts is not available, the voicemail system will fallback to another
appropriate language if possible (refer to the appropriate voicemail installation manual for
details). The locale sent to the voicemail server by the system is determined in the following
order of priority:
1. Short Code Locale: The short code locale, if set, is used if the call is routed to voicemail
using the short code.
2. Incoming Call Route Locale: The incoming call route locale, if set, is used if caller is
external.
3. User Locale: The user locale, if set, is used if the caller is internal.
4. System Locale: If no user or incoming call route locale is set, the system locale is used
unless overridden by a short code locale. Systems using Embedded Voicemail, if the
required set of upgraded language prompts to match the locale is not present on the
system SD card, Manager will display an error. The required prompt set can be uploaded
from Manager using the Add/Display VM Locales option.
• Force Account Code: When selected, if the short code results in the dialing of an external
number, the user is prompted to enter a valid account code before the call is allowed to
continue. See Account Code Configuration on page 672.
• Force Authorisation Code: When selected, if the short results in the dialing of an external
number, the user is prompted to enter a valid authorisation code before the call is allowed to
continue. See Configuring authorization codes on page 645.

Short Code Descriptions:

The short method for describing short codes in this manual, for example 9N/Dial/./0, indicates the
settings of main short code fields, eacg separated by a / as follows:
• Code: In this case 9N.
• Feature: In this case Dial.
• Telephone Number: In this case the symbol . representing all dialed digits.
• Line Group ID: In this case the call is sent to outgoing line group 0.

Examples:

• *17/VoicemailCollect/?U A user dialing *17 is connected to their own mailbox to collect

messages.
• *14*N#/FollowMeTo/N If a user dials *14*210# at their own extension, their calls are redirected
to extension 210.

Types of Short Code:

In addition to different short code features, there are different types of short code:
• Dialing Short Codes: The following types of short code applied to on-switch dialing. The result
may be an action to be performed by the system, a change to the user's settings or a number

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Short Code Overview

to be dialed. The order below is the order of priority in which they are used when applied to
user dialing.
- User Short Codes: These are usable by the specific user only. User short codes are
applied to numbers dialled by that user and to calls forwarded via the user.
- User Rights Short Codes: These are usable by any users associated with the user rights in
which they are set. User Rights short codes are only applied to numbers dialed by that user.
For example they are not applied to calls forwarded via the user.
- System Short Codes: These are available to all users on the system. They can be
overridden by user or user rights short codes.
• Post-Dialing Short Codes: When any the short code above result in a number to be dialed,
further short code can be applied to that number to be dialed. This is done using the following
types of short codes.
- ARS (Alternate Route Selection) Short Codes: The short code that matches dialing can
specify that the resulting number should be passed to an ARS form. The ARS form can
specify which routes should be used for the call by using further short code matches and
also provide option to use other ARS forms based on other factors such as time and
availability of routes.
- Transit Network Selection (TNS) Short Codes: Used on T1 ISDN trunks set to use AT&T
as the Provider. Applied to the digits presented following any other short code processing.
• Incoming Number Short Codes: On certain types of trunks short codes can be applied to the
incoming digits received with calls.
- Line Short Codes: These short codes are used to translate incoming digits received with
calls. The stage at which they are applied varies between different line types and may be
overridden by an extension number match.
Related links
Short Code Characters on page 790
User Dialing on page 793
Application Dialing on page 796
Secondary Dial Tone on page 796
? Short Codes on page 798
Short Code Matching Examples on page 798
Default System Short Code List on page 803

Short Code Characters

The short code fields Short Code and Telephone Number can contain the normal diallable digits
*, # and 0 to 9. In addition they can also use a range of special characters as listed below.

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 Short Code Characters

Short Code Field Characters

• ? = Default Match This character can be used on its own to create a short code match in the
absence of any other short code match. See ? Short Codes on page 798.
• ?D = Default Number Dialing This character combination makes a call to the defined phone
number as soon as the user goes off-hook. See ? Short Codes on page 798.
• ?D(x) = Default Number Dialing Timeout The character x represents time in seconds. If a
phone is off-hook or speaker is enabled and no number is dialed for x seconds, the phone
makes a call to the defined phone number. A maximum of 30 seconds is used for x though
system accepts values more than 30 on the interface.
• N =Match Any Digits Matches any dialed digits (including none). The Dial Delay Time or a
following matching character is used to resolve when dialing is complete.
• X = Match a Digit Matches a single digit. When a group of X's is used, the short code
matches against the total number of X's.
• [ ] =Secondary Dial Tone Trigger For pre-4.0 IP Office systems used to trigger secondary
dial tone. Not used for Release 4.0+. See Secondary Dial Tone.
• ; =Receive Sending Complete When used this must be the last character in the short code
string.
- If the Dial Delay Count is 0, a ; instructs the system to wait for the number to be fully
dialed, using the Dial Delay Time or the user dialing #, to indicate completion and then
acting on the short code.
- If the Dial Delay Count is not zero, the dialing is only evaluated when # is pressed. The
majority of North-American telephony services use en-bloc dialing. Therefore the use of a ;
is recommended at the end of all dialing short codes that use an N before routing those
calls to a trunk or ARS. This is also recommended for all dialing where secondary dial tone
short codes are being used.

Telephone Number Field Characters

• A = Allow Outgoing CLI Allow the calling party number sent with the call to be used. This
character may be required by service providers in some locales.
• C = Use Called Number Field Place any following digits in the outgoing call's Called number
field rather than the Keypad field.
• D = Wait for Connect Wait for a connect message before sending any following digits as
DTMF.
• E = Extension Number Replace with the extension number of the dialing user. Note that if a
call is forwarded this will be replaced with the extension number of the forwarding user.
• h = Hold Music Source When used as part of the short code telephone number field, this
character allows the source for music on hold to be selected. Enter h(X)where X indicates
the required hold music source if available. This overrides any previous hold music selection
that may have been applied to the call.
- For IP500 V2 systems, the value of X can be 1 to 4.
- For systems on a Linux based server, the value of X can be 1 to 32.
- When used with Park Call short codes, the h(X) should be entered before the park slot
number part of the telephone number.

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Short Code Overview

• I = Use Information Packet Send data in an Information Packet rather than Set-up Packet.
• K = Use Keypad Field Place any following digits in the outgoing call's Keypad field rather
than the Called Number field. Only supported on ISDN and QSIG.
• l = Last Number Dialed (lower case L) Use the last number dialed.
• L = Last Number Received Use the last number received.
• N = Dialed Digit Wildcard Match Substitute with the digits used for the N or X character
match in the Short Code number field.
• p = Priority The priority of a call is normally assigned by the Incoming Call Route or else is
1-Low for all other calls. Dial Extn short codes can use p( x ) as a suffix to the Telephone
Number to change the priority of a call. Allowable values for x are 1, 2 or 3 for low, medium
or high priority respectively.
• In situations where calls are queued, high priority calls are placed before calls of a lower
priority. This has a number of effects:
- Mixing calls of different priority is not recommended for destinations where Voicemail Pro
is being used to provided queue ETA and queue position messages to callers since those
values will no longer be accurate when a higher priority call is placed into the queue. Note
also that Voicemail Pro will not allow a value already announced to an existing caller to
increase.
- If the addition of a higher priority call causes the queue length to exceed the hunt group's
Queue Length Limit, the limit is temporarily raised by 1. This means that calls already
queued are not rerouted by the addition of a higher priority call into the queue.
• r = Ring Tone Plan When used as part of the short code telephone number field, this
character can specify a Ring Tone Plan number. Enter r(X)where X is 1 to 8 indicating the
Ring Tone Plan number to use.
• S = Calling Number Place any following digits into the outgoing call's calling number field.
Using S does not alter any allow or withhold CLI setting associated with the call, the short
code characters A or W should be used respectively.
- On mobile twinned calls, if the original party information is used or a specific calling party
information CLI is set, that number overrides setting the outgoing CLI using short codes.
- Note that for SIP trunks, the SIP URI configuration options override this setting.
- Outgoing CLI Warning Changing the outgoing CLI for calls requires the line provider to
support that function. You must consult with your line provider before attempting to change
the outgoing CLI, failure to do so may result in loss of service. If changing the outgoing CLI
is allowed, most line providers required that the outgoing CLI used matches a number
valid for return calls on the same trunks. Use of any other number may cause calls to be
dropped or the outgoing CLI to be replaced with a valid number. On mobile twinned calls, if
the original party information is used or a specific calling party information CLI is set, that
number overrides setting the outgoing CLI using short codes.
• SS = Pass Through Calling Number Pass through the Calling Party Number. For example,
to provide the incoming ICLID at the far end of a VoIP connection, a short code ? with
telephone number .SS should be added to the IP line.
• i = National Both the S and SS characters can be followed by an i, that is Si and SSi. Doing
this sets the calling party number plan to ISDN and number type to National. This may be
required for some network providers.

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 User Dialing

• t = Allowed Call Duration Set the maximum duration in minutes for a call plus or minus a
minute. Follow the character with the number of minutes in brackets, for example t(5).
• U = User Name Replace with the User Name of the dialing user. Used with voicemail.
• W = Withhold Outgoing CLI Withhold the sending of calling ID number. Operation is service
provider dependent.
• Y = Wait for Call Progress Message Wait for a Call Progress or Call Proceeding message
before sending any following digits as DTMF. For example, the Y character would be
necessary at a site where they have signed up with their telephone service provider to
withhold international dialing until a DTMF pin/account number is entered that initiates the
call progress/proceeding message.
• Z = Calling Party Name This option can be used with trunks that support the sending of
name information. The Z character should be followed by the name enclosed in " " quotation
marks. Note that their may be name length restrictions that vary between line providers. The
changing of name information on calls being forwarded or twinned may also not be supported
by the line provider.
• @ = Use Sub Address Field Enter any following digits into the sub-address field.
• . = Dialed Digits Replace with the full set of dialed digits that triggered the short code match.
• , = One Second Pause Add a one second pause in DTMF dialing.
• ; = Receive Sending Complete When used this must be the last character in the short code
string. If the Dial Delay Count is 0, a ; instructs the system to wait for the number to be fully
dialed, using the Dial Delay Time or the user dialing #, to indicate completion and then
acting on the short code. If the Dial Delay Count is not zero, the dialing is only evaluated
when # is pressed.
• " " = Non Short Code Characters Use to enclose any characters that should not be
interpreted as possible short code special characters by the system. For example characters
being passed to the voicemail server.
Related links
Short Code Overview on page 788

User Dialing
The following rules are used when short code matching is performed for user dialing:
• A short code is used immediately an exact match is found unless followed by a ;.
• If no match is found but partial matches exist, the user can continue dialing.
• If no match or partial matches are found, incompatible is returned.
• The following precedence is used to determine which short codes are used:
- Extension number matches override all short codes.
- User short codes override user rights and system short codes.

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Short Code Overview

- User Rights short code matches override system short codes.

• When multiple exact matches occur,
- The match with the most specified digits rather than wildcards is used.
- If there are still more than one match, the match with the most exact length is used. This
means X wildcards will override N when both match.
• The rules above are applied even if the number is dialled by selection from a directory or
using any other method of stored number dialing.

User Digit Dialing Settings:

The following system settings influence user dialing.
Dial Delay Count: Default = 0 (US/Japan), 4 (ROW). This value sets the number of digits dialed
before the system looks for a short code match.
Dial Delay Time: Default = 4 seconds (US/Japan), 1 second (ROW). This value sets the
maximum allowed interval between the dialing of each digit. If exceeded, the system looks for a
short code match even if the Dial Delay Count has not been reached.
Off-Hook Timer: When a user goes off-hook, the system starts a 30 second off-hook timer (10
seconds in Italy). If the off-hook timer expires before a short code match occurs, the user is
disconnected.

User Dialing Flowchart:

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 User Dialing

Related links
Short Code Overview on page 788

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Short Code Overview

Application Dialing
Numbers speed dialed by system applications such as SoftConsole are treated differently. Since
the digits are received en bloc as a single group, they can override some short code matches. The
same applies to short codes used within system configuration settings such as Incoming Call
Route destinations.
Example:
• Telephone Number: 12345678
• Short Code 1: 1234XX/Dial/Extn/207
• Short Code 2: 12345678/Dial Extn/210
If dialed manually by the user, as soon as they have dialed 123456 a match to short code 1
occurs. They can never dial short code 2.
If dialed using an application, 12345678 is sent as a string and a match to short code 2 occurs.
Partial Dialing
If the application dialing does not trigger an exact match, the user can dial additional digits through
their extension. The processes for normal user dialing are applied.
Non-Digit Short Codes
Short codes can be created that use alphabetic characters instead of numbers. While these short
codes cannot be dialed from a phone, they can be dialed using application speed dials and
settings. However characters that are interpreted as special short code characters will still be
interpreted as such.
Related links
Short Code Overview on page 788

Secondary Dial Tone

Some locales prefer to provide users with secondary dial tone once they have started dialing
external calls. This dial tone is heard by the user until they have completed dialing and a trunk is
seized at which point call progress tones are provided by the trunk, or camp on/busy tone is
provided by the system if the required trunk cannot be seized.
Release 4.0 and Higher
The use of secondary dial tone is provided through the Secondary Dial Tone check box option on
the ARS form to which the call is routed. When on, this setting instructs the system to play
secondary dial tone to the user.
The tone used is set as either System Tone (normal dial tone) or Network Tone (secondary dial
tone). Both tone types are generated by the system in accordance with the system specific locale
setting. Note that in some locales normal dial tone and secondary dial tone are the same.

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 Secondary Dial Tone

When Secondary Dial Tone is selected, the ARS form will return tone until it receives digits with
which it can begin short code matching. Those digits can be the result of user dialing or digits
passed by the short code which invoked the ARS form. For example with the following system
short codes:
In this example, the 9 is stripped from the dialed number and is not part of the telephone number
passed to the ARS form. So in this case secondary dial tone is given until the user dials another
digit or dialing times out.
• Code: 9N
• Telephone Number: N
• Line Group ID: 50 Main
In this example, the dialed 9 is included in the telephone number passed to the ARS form. This will
inhibit the use of secondary dial tone even if secondary dial tone is selected on the ARS form.
• Code: 9N
• Telephone Number: 9N
• Line Group ID: 50 Main
Pre-4.0 IP Office Secondary Dial Tone
Pre-4.0 systems provided dial tone through the use of the short code feature Secondary Dial Tone
and the [ ] special characters. For example, on a system where 9 is used as a prefix for external
dialing, the system short code 9/./Secondary Dial Tone/0 will trigger secondary dial tone when
users dial a number prefixed with 9. This method is not supported by Release 4.0 which provides
ARS forms for the control of outgoing calls.
In order to allow further digit matching, the digits dialed are put back through short code matching
against any short codes that start with [n] where n is the digit used to trigger the system secondary
dial tone short code.
On all systems where secondary dial tone is used, a ; should also be used in dialing short codes
that contain N.
For example:
System Short Codes
• 9/SecondaryDialTone/.
• [9]0N;/Dial/0
User Short Code
[9]0N;/Busy/0
The user dials 90114445551234. The 9 is matches the system secondary dial tone short code and
unlike other short codes this is applied immediately. The user's dialing is put through short code
matching again using the normal order of precedence but matched to possible short codes
beginning [9]. In this case the user's [9]0N; short code would take precedence over the system
[9]0N; short code.

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Short Code Overview

Related links
Short Code Overview on page 788

? Short Codes
The ? character can be used in short codes in the following ways:
Default Short Code Matching:
? short codes are used in short code matching in the following way. If no user or system short
code match is found, the system will then look for a ? short code match. It will look first for a user ?
short code and then, if not found, a system ? short code.
Example: On systems outside North America, the system short code ?/Dial/./0 is added as a
default short code. This short code provides a match for any dialing to which there is no other
match. Therefore, on systems with this short code, the default is that any unrecognized number
will be dialed to Outgoing Line Group 0.
Hot-Line Dialing:
A user short code ?D can be used to perform a short code action immediately the user extension
goes off-hook. This is supported with Dial type short code features. Typically it is used with door,
lift and lobby phones to immediately connect the phone to a number such as the operator or
reception.
Voicemail Collect Short Codes:
The ? character can appear in the Telephone Number field of a short code. This is done with
short codes using the VoicemailCollect feature. In this instance the ? character is not interpreted
by the system, it is used by the voicemail server.
Related links
Short Code Overview on page 788

Short Code Matching Examples

The following examples are not meant as practical examples. However they are simple to
implement and test on real system without conflicting with its normal operation. They illustrate the
interaction between different short codes in resolving which short code is an exact match. They
assume that extension numbers are in the 200 to 299 range.
The term 'dials' means dialing the indicated digit or digits without the inter-digit Dial Delay Time
expiring.
The term 'pause' means a wait that exceeds the inter-digit Dial Delay Time.

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 Short Code Matching Examples

Scenario 1
Short Code 1 = 60/Dial Extn/203
Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 No possible match, incompatible
returned immediately
2 6 No exact match but there is a
potential match, so the system
waits. When the Dial Delay Time
expires, no exact match is found
so incompatible is returned.
3 60 Exact match to Short Code 1.
Extension 203 called immediately.
4 61 No possible match, the system
returns incompatible.

Scenario 2
Short Code 1 = 60/Dial Extn/203
Short Code 2 = 601/Dial Extn/210
Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 No possible match, incompatible
returned immediately
2 60 Exact match to Short Code 1.
Extension 203 called immediately.
3 601 Exact match to Short Code 1 as
soon as the 0 is dialed. The user
cannot manually dial 601.

Scenario 3
Short Code 1 = 60/Dial Extn/203
Short Code 2 = 601/Dial Extn/210
Dial Delay Count = 3. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 Insufficient digits to trigger
matching. The system waits for
additional digits or for Dial Delay
Time to expire. When Dial Delay
Time expires, no possible match
is found so incompatible is
returned.
Table continues…

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Short Code Overview

2 60 Insufficient digits to trigger

matching. The system waits for
additional digits or for Dial Delay
Time to expire. When Dial Delay
Time expires, matching started
and exact match to Short Code 1
occurs. .
3 601 Third digit triggers matching.
Exact match to Short Code 2.
Extension 210 dialed immediately.
4 60# # is treated as a digit and as the
third digit triggers matching. No
exact match found. The system
returns incompatible.

Scenario 4
Short Code 1 = 60;/Dial Extn/203
Short Code 2 = 601/Dial Extn/210
Dial Delay Count = 3. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 8 Insufficient digits to trigger
matching. The system waits for
additional digits or for Dial Delay
Time to expire. When Dial Delay
Time expires, no possible match
is found so incompatible is
returned.
2 6 Insufficient digits to trigger
matching. The system waits for
additional digits or for the
interdigit Dial Delay Time to
expire. If the Dial Delay Time
expires, a potential match exists
to a short code that uses ; so the
system waits for an additional
digit until the off-hook timer
expires.
Table continues…

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 Short Code Matching Examples

3 60 As above but an additional digit

now may create a match.
If 1 is dialed, it creates an exact
match to Short Code 2 and is
used immediately.
If 0, * or 2 to 9 is dialed, no
possible match exists. The
system returns incompatible.
If the next digit is a #, it is treated
as signaling dialing complete
rather than being a digit. Short
code 1 becomes an exact match
and is used immediately.
4 601 Third digit triggers matching.
Exact match to Short Code 2.
Extension 210 dialed immediately.

Scenario 5
Short Code 1 = 601/Dial Extn/203
Short Code 2 = 60N/Dial Extn/210
Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 6 No exact match but there is a
potential match, so the system
waits for additional dialing. If the
Dial Delay Time expires, no exact
match is found so incompatible is
returned.
2 60 Potential match to both short
codes. The system waits for
additional dialing. If the Dial Delay
Time expires, Short Code 2
becomes an exact match with N
blank.
3 601 Exact match to Short Code 1.
Used immediately
4 602 Exact match to Short Code 2.
Used immediately.

Scenario 6
Table continues…

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Short Code Overview

Short Code 1 = 601/Dial Extn/203

Short Code 2 = 60N/Dial Extn/210
Short Code 3 = 60X/Dial Extn/207
Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 6 No exact match but there are
potential matches so the system
waits for additional dialing. If the
Dial Delay Time expires, no exact
match has occurred so
incompatible is returned.
2 60 Potential match to all short codes.
System waits for additional
dialing. If the Dial Delay Time
expires, Short Code 2 becomes
an exact match with N blank. If a
digit is dialed, Short Code 3
becomes a more exact match and
is used.
3 601 Exact match all short code,
however Short Code 1 is treated
as being more exact (more
matching digits) and is used
immediately
4 602 Exact match to short codes 2 and
3, however the Short Code 3 is
treated as being more exact
(length match) and is used
immediately.

Scenario 7
Short Code 1 = 601/Dial Extn/203
Short Code 2 = 60N/Dial Extn/210
Short Code 3 = 6XX/Dial Extn/207
Dial Delay Count = 0. Dial Delay Time = 4 seconds.
Test Dialing Effect
1 6 No exact match but there are
potential matches so the system
waits for additional dialing. If the
Dial Delay Time expires, no exact
match has occurred so
incompatible is returned.
Table continues…

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 Default System Short Code List

2 60 Potential match to all short codes.

System waits for additional
dialing. If the Dial Delay Time
expires, Short Code 2 becomes
an exact match with N blank. If a
digit is dialed, Short Code 3
becomes an more exact match
and is used.
3 601 Exact match all short code,
however Short Code 1 is treated
as being more exact (more
matching digits) and is used
immediately
4 602 Exact match to short codes 2 and
3, however the Short Code 2 is
treated as being more exact
(more matching digits) and is
used immediately.
5 612 Exact match to Short Code 3.

Related links
Short Code Overview on page 788

Default System Short Code List

Most control units are available in A-Law and U-Law models. Typically U-Law models are supplied
to North American locales, A-Law models are supplied to the rest of the world. In addition to the
using different default companding for digital lines and phone, A-Law and U-Law models support
different default short codes. The following table lists the default system short codes present in a
system's configuration.
Standard Mode
Short Code Telephone Feature A-Law ULAW
Number
*00 Blank Cancel All
Forwarding
*01 Blank Forward
Unconditional On
*02 Blank Forward
Unconditional Off
*03 Blank Forward On Busy
On
Table continues…

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Short Code Overview

Short Code Telephone Feature A-Law ULAW

Number
*04 Blank Forward On Busy
Off
*05 Blank Forward On No
Answer On
*06 Blank Forward On No
Answer Off
*07*N# N Forward Number
*08 Blank Do Not Disturb On
*09 Blank Do Not Disturb Off
*10*N# N Do Not Disturb
Exception Add
*11*N# N Do Not Disturb
Exception Del
*12*N# N Follow Me Here
*13*N# N Follow Me Here
Cancel
*14*N# N Follow Me To
*15 Blank Call Waiting On
*16 Blank Call Waiting Off
*17 ?U Voicemail Collect
*18 Blank Voicemail On
*19 Blank Voicemail Off
*20*N# N Set Hunt Group
Night Service
*21*N# N Clear Hunt Group
Night Service
*22*N# N Suspend Call
*23*N# N Resume Call
*24*N# N Hold Call
*25*N# N Retrieve Call
*26 Clear CW
*27*N# N Hold CW
*28*N# N Suspend CW
*29 Blank Toggle Calls
*30 Blank Call Pickup Any
*31 Blank Call Pickup Group
*32*N# N Call Pickup Extn
Table continues…

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 Default System Short Code List

Short Code Telephone Feature A-Law ULAW

Number
*33*N# N Call Queue
*34N; N Hold Music
*35*N# N Extn Login
*36 Blank Extn Logout
*37*N# N Call Park
*38*N# N Unpark Call
*39 1 Relay On
*40 1 Relay Off
*41 1 Relay Pulse
*42 2 Relay On
*43 2 Relay Off
*44 2 Relay Pulse
*45*N# N Acquire Call
*46 Blank Acquire Call
*47 Blank Conference Add
*48 Blank Voicemail Ringback
On
*49 Blank Voicemail Ringback
Off
*50 Blank Forward Huntgroup
On
*51 Blank Forward Huntgroup
Off
*52 Blank Cancel or Deny
*53*N# N Call Pickup
Members
*55 Blank Stamp Log
*57*N# N Forward On Busy
Number
*70 Blank Call Waiting
Suspend
*70*N# N Dial Physical Extn
by Number
*71*N# N Dial Physical Extn
by Id
*9000* "MAINTENANCE" Relay On
*91N; N".1" Record Message
Table continues…

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Short Code Overview

Short Code Telephone Feature A-Law ULAW

Number
*92N; N".2" Record Message
*99; "edit_messages" Voicemail Collect
9N N Dial
? . Dial

Server Edition
Short Code Telephone Feature A-Law ULAW
Number
*00 Blank Cancel All
Forwarding
*01 Blank Forward
Unconditional On
*02 Blank Forward
Unconditional Off
*03 Blank Forward On Busy
On
*04 Blank Forward On Busy
Off
*05 Blank Forward On No
Answer On
*06 Blank Forward On No
Answer Off
*07*N# N Forward Number
*08 Blank Do Not Disturb On
*09 Blank Do Not Disturb Off
*10*N# N Do Not Disturb
Exception Add
*11*N# N Do Not Disturb
Exception Del
*12*N# N Follow Me Here
*13*N# N Follow Me Here
Cancel
*14*N# N Follow Me To
*17 ?U Voicemail Collect
*18 Blank Voicemail On
*19 Blank Voicemail Off
*20*N# N Set Hunt Group
Night Service
Table continues…

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 Default System Short Code List

Short Code Telephone Feature A-Law ULAW

Number
*21*N# N Clear Hunt Group
Night Service
*29 Blank Toggle Calls
*30 Blank Call Pickup Any
*31 Blank Call Pickup Group
*32*N# N Call Pickup Extn
*33*N# N Call Queue
*34N; N Hold Music
*35*N# N Extn Login
*36 Blank Extn Logout
*37*N# N Call Park
*38*N# N Unpark Call
*44 2 Relay Pulse
*45*N# N Acquire Call
*46 Blank Acquire Call
*47 Blank Conference Add
*48 Blank Voicemail Ringback
On
*49 Blank Voicemail Ringback
Off
*50 Blank Forward Huntgroup
On
*51 Blank Forward Huntgroup
Off
*52 Blank Cancel or Deny
*53*N# N Call Pickup
Members
*55 Blank Stamp Log
*57*N# N Forward On Busy
Number
*66*N# N Conference Meet
Me
*70 Blank Call Waiting
Suspend
*70*N# N Dial Physical Extn
by Number
*71*N# N Dial Physical Extn
by Id
Table continues…

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Short Code Overview

Short Code Telephone Feature A-Law ULAW

Number
*99; "edit_messages" Voicemail Collect
9N N Dial [1]
? . Dial [1]

For U-Law systems, a 9N is a default short code on the Primary Server while a ? short code is a
default on all other servers.
Additional short codes of the form *DSSN, *SDN, *SKN, these are used by the system for internal
functions and should not be removed or altered. Short codes *#N and **N may also visible, these
are used for ISDN functions in Scandinavian locales.
The default *34 short code for music on hold has changed to *34N;.
Related links
Short Code Overview on page 788

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Chapter 18: Short Code Features

Note that this document describes all existing short codes. The short codes available in the
Manager application depends on the software release.
Related links
Auto Attendant on page 812
Auto Intercom Deny Off on page 813
Auto Intercom Deny On on page 813
Break Out on page 814
Barred on page 814
Busy On Held on page 815
Call Intrude on page 816
Call Listen on page 816
Call Park on page 818
Call Park and Page on page 819
Call Pickup Any on page 819
Call Pickup Extn on page 820
Call Pickup Group on page 820
Call Pickup Line on page 821
Call Pickup Members on page 822
Call Pickup User on page 822
Call Queue on page 823
Call Record on page 824
Call Steal on page 824
Call Waiting On on page 825
Call Waiting Off on page 826
Call Waiting Suspend on page 826
Cancel All Forwarding on page 827
Cancel Ring Back When Free on page 827
Change Login Code on page 828
Clear After Call Work on page 829
Clear Call on page 829
Clear CW on page 830
Clear Hunt Group Night Service on page 830
Clear Hunt Group Out Of Service on page 831

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Short Code Features

Clear Quota on page 832

Coaching Intrusion on page 832
Conference Add on page 833
Conference Meet Me on page 834
CW on page 835
Dial on page 836
Dial 3K1 on page 837
Dial 56K on page 837
Dial 64K on page 838
Dial CW on page 838
Dial Direct on page 839
Dial Direct Hot Line on page 839
Dial Emergency on page 840
Dial Extn on page 840
Dial Fax on page 841
Dial Inclusion on page 842
Dial Paging on page 842
Dial Physical Extension by Number on page 843
Dial Physical Extension By ID on page 844
Dial Speech on page 844
Dial V110 on page 845
Dial V120 on page 845
Dial Video on page 845
Disable ARS Form on page 846
Disable Internal Forwards on page 846
Disable Internal Forward Unconditional on page 847
Disable Internal Forward Busy or No Answer on page 847
Display Msg on page 847
Do Not Disturb Exception Add on page 849
Do Not Disturb Exception Delete on page 849
Do Not Disturb On on page 850
Do Not Disturb Off on page 851
Enable ARS Form on page 851
Enable Internal Forwards on page 852
Enable Internal Forward Unconditional on page 852
Enable Internal Forward Busy or No Answer on page 852
Extn Login on page 853
Extn Logout on page 854
Flash Hook on page 855
FNE Service on page 855
Follow Me Here on page 856

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 Follow Me Here Cancel on page 856
Follow Me To on page 857
Forward Hunt Group Calls On on page 858
Forward Hunt Group Calls Off on page 858
Forward Number on page 859
Forward On Busy Number on page 860
Forward On Busy On on page 860
Forward On Busy Off on page 861
Forward On No Answer On on page 861
Forward On No Answer Off on page 862
Forward Unconditional On on page 863
Forward Unconditional Off on page 863
Group Listen Off on page 864
Group Listen On on page 864
Headset Toggle on page 865
Hold Call on page 866
Hold CW on page 866
Hold Music on page 867
Hunt Group Disable on page 867
Hunt Group Enable on page 868
Last Number Redial on page 869
MCID Activate on page 869
Mobile Twinned Call Pickup on page 870
Off Hook Station on page 870
Outgoing Call Bar Off on page 871
Outgoing Call Bar On on page 872
Private Call Off on page 872
Private Call On on page 873
Priority Call on page 873
Record Message on page 874
Relay On on page 875
Relay Off on page 876
Relay Pulse on page 876
Resume Call on page 877
Retrieve Call on page 877
Ring Back When Free on page 878
Secondary Dial Tone on page 879
Set Absent Text on page 879
Set Account Code on page 881
Set Authorization Code on page 881
Set Fallback Twinning Off on page 882

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Short Code Features

Set Fallback Twinning On on page 882

Set Hunt Group Night Service on page 883
Set Hunt Group Out Of Service on page 883
Set Inside Call Seq on page 884
Set Mobile Twinning Number on page 885
Set Mobile Twinning On on page 885
Set Mobile Twinning Off on page 886
Set No Answer Time on page 886
Set Outside Call Seq on page 887
Set Ringback Seq on page 887
Set Time Profile on page 888
Set Wrap Up Time on page 889
Speed Dial on page 890
Shutdown Embedded Voicemail on page 891
Stamp Log on page 892
Startup Embedded Voicemail on page 892
Suspend Call on page 893
Suspend CW on page 893
Start After Call Work on page 894
Toggle Calls on page 894
Unpark Call on page 895
Voicemail Collect on page 895
Voicemail Node on page 898
Voicemail On on page 898
Voicemail Off on page 899
Voicemail Ringback On on page 899
Voicemail Ringback Off on page 900
Whisper Page on page 900

Auto Attendant
This feature is used with Embedded Voicemail. It is not supported by Server Edition. It allows the
recording of the greetings used by auto-attendant services and the transfer of calls to that auto
attendant. This feature was previously called Record Greeting.
Details
Telephone Number:
Four system short codes (*81XX, *82XX, *83XX and *84XX) are automatically added for use with
all auto attendants, for the morning, afternoon, evening and menu options greetings respectively.

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 Auto Intercom Deny Off

These use a telephone number of the form "AA:" N" . Y " where N is the replaced with the auto
attendant number dialed and Y is 1, 2, 3 or 4 for the morning, afternoon, evening or menu option
greeting.
• An additional short code of the form (for example) *80XX/Auto Attendant/"AA:"N can be
added manual if internal dialed access to auto attendants is required.
• To add a short code to access a specific auto attendant, the name method should be used.
• For IP Office deployed in a Enterprise Branch environment, the short codes *800XX,
*801XX…*809XX, *850XX, and *851XX are automatically created for recording a Page
prompt.
Default Short Code: See Configuration Settings | Auto Attendant.
Programmable Button Control:
Release: 2.0+.
Related links
Short Code Features on page 809

Auto Intercom Deny Off

Details
Telephone number:
Default short code:
Programmable Button Control: Auto Intercom Deny Off
Related links
Short Code Features on page 809

Auto Intercom Deny On

Details
Telephone number:
Default short code:
Programmable Button Control: Auto Intercom Deny On
Related links
Short Code Features on page 809

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Short Code Features

Break Out
This feature is usable within a system multi-site network. It allows a user on one system in the
network to specify that the following dialing be processed by another system on the network as if
the user dialed it locally on that other system. Pre-Release 5.0: This feature requires the IP Offices
to have Advanced Small Community Networking licenses.
Details
Telephone Number: The IP Address or Name of the system, using * characters in place of .
characters.
Default Short Code:
Programmable Button Control: BkOut
Release: 4.0+.
Example On a system, to break out via a system called RemoteSwitch with the IP Address
192.168.42.3, either of the following short codes could be used.
Example 1
Code: *80*N#
Telephone Number: N
Feature: Break Out
Example 2
Code: *81
Telephone Number: RemoteSwitch
Feature: Break Out
Example 1 allows break out using any remote switch by dialing its IP address, for example
*80*192*168*42*3#. Example 2 does this for a specific remote system by dialing just *81.
Related links
Short Code Features on page 809

Barred
This short code feature can be used for call barring by using the short code as the call destination.
This short code feature was previously called Busy. It has been renamed but its function has not
changed.
When used in an ARS form that has been configured with an Alternate Route, for callers whose
dialing has matched the short code no further routing is applied.

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 Busy On Held

Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 1.0+.
Related links
Short Code Features on page 809

Busy On Held
When on, busy on held returns busy to new calls when the user has an existing call on hold. This
short code feature is useful when a user does not want to be distracted by an additional incoming
call when they have a call on hold.
Details
Telephone Number: Y or 1 for on, N or 0 for off.
Default Short Code:
Programmable Button Control: BusyH
Release: 1.0+.
Example: Turning Busy on Held on
If on, when the user has a call on hold, new calls receive busy tone (ringing if analog) or are
diverted to Voicemail if enabled, rather than ringing the user.
Note:
This overrides call waiting when the user has a call on hold.
Short Code: *12
Telephone Number: Y
Feature: BusyOnHeld
Example: Turning Busy on Held off
Another short code must be created to turn the Busy on Held feature off. If off, when the uses has
a call on hold, new calls will still get directed to the user.
Short Code: *13
Telephone Number: N
Feature: BusyOnHeld
Related links
Short Code Features on page 809

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Short Code Features

Call Intrude
This feature allows you to intrude on the existing connected call of the specified target user. All call
parties are put into a conference and can talk to and hear each other. A Call Intrude attempt to a
user who is idle becomes a Priority Call.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
Note that this feature requires conference resources from the system for the duration of the
intrusion.
Users can use privacy features that to indicate that a call cannot be intruded on. See Private Calls.
Intruding onto a user doing silent monitoring (Call Listen on page 816) is turned into a silent
monitoring call.
The system support a range of other call intrusion methods in addition to this feature.
Details
Telephone Number: Target extension number.
Default Short Code:
Programmable Button Control: Intru
See also: Call Listen on page 816, Coaching Intrusion on page 832, Dial Inclusion on
page 842, Whisper Page on page 900.
Release: 1.0+.
Related links
Short Code Features on page 809

Call Listen
This feature allows you to monitor another user's call without being heard. Monitoring can be
accompanied by a tone heard by all parties. Use of the tone is controlled by the Beep on Listen
setting on the System | Telephony | Tones & Music tab. The default for this setting is on. If
enabled, this is the only indication of monitoring given to the monitored user. There is no phone
display indication of monitoring.

Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must

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 Call Listen

ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
The use of call listen is dependant on:
The target being a member of the group set as the user's Monitor Group (User | Telephony |
Supervisor Settings). The user does not have to be a member of the group.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
This feature uses system conference resources. If insufficient conference resource are available it
will not be possible to use this feature.
A number of features are supported for call listening:
• Users can be given privacy features that allow them to indicate that a call cannot be
monitored. See Private Calls.
• IP extensions can be monitored including those using direct media. Previously the monitoring
of IP extensions could not be guaranteed.
• The monitoring call can be initiated even if the target user is not currently on a call and
remains active until the monitoring user clears the monitoring call.
• The user who initiated the call listen can also record the call.
Intruding onto a user doing silent monitoring (Call Listen) is turned into a silent monitoring call.
1400, 1600, 9500 and 9600 Series phones with a user button can initiate listening using that
button if the target user meets the criteria for listening.
The system support a range of other call intrusion methods in addition to this feature.
Details
Telephone Number: Target extension number (extension must be local).
Default Short Code:
Programmable Button Control: Listn
See also: Call Intrude on page 816, Coaching Intrusion on page 832, Dial Inclusion on
page 842, Whisper Page on page 900.
Release: 1.0+.
Example
User 'Extn205' wants to be able to monitor calls received by members of the Hunt Group 'Sales'.
1. For user 'Extn205', in the Monitor Group (User | Telephony | Supervisor Settings) list box
select the hunt group.

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Short Code Features

2. Ensure that Can Intrude is checked.

3. Create a user short code to allow Extn205 to start monitoring.
• Short Code: *89*N#
• Telephone Number: N
• Line Group ID: 0.
• Feature: CallListen
4. For each member of the hunt group, check that their Cannot be Intruded setting is
unchecked.
5. Now when a member of the 'Sales' hunt group is on a call, Extn205 can replace N in the
short code with the extension number of that member and monitor their call.
Related links
Short Code Features on page 809

Call Park
Parks the user's current call into the specified park slot number. The call can then be retrieved by
other extensions (refer to the appropriate telephone user guide). While parked the caller hears
music on hold if available. The 'Unpark Call' feature can be used to retrieve calls from specific
park slots.
Park Timeout (System | Telephony | Telephony) controls how long a call will remain parked. When
this expires the call will recall to the parking user if they are idle or when they next become idle.
The recall call will continue ring and does follow any forwards or go to voicemail.
Details
Telephone Number: Park slot number. If no park slot number is specified when this short code
is used, the system automatically assigns a park slot number based on the extension number of
the user parking the call plus one digit 0 to 9.
Park slot IDs can be up to 9 digits in length. Names can also be used for application park slots.
Default Short Code: *37*N#
Programmable Button Control: Call Park
See also: Unpark Call.
Release: 1.0+.
Example
This short code is a default within the system configuration. This short code can be used to toggle
the feature on/off. N represents the park slot number in which the call will be parked. For example,
if a user wants to park a call to slot number 9, the user would dial *37*9#. The call will be parked
there until retrieved by another extension or the original extension.

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 Call Park and Page

Short Code: *37*N#

Telephone Number: N
Feature: ParkCall
Related links
Short Code Features on page 809

Call Park and Page

Parks the user's current call into the highest park slot number within the range specified on the
System | Telephony | Park & Page tab, in the Central Park Range field. For example, if the
specified Central Park Range is 1XX, then the Park & Page short code would attempt to Park on
199. If the range is 567XX, then the call would attempt to Park on 56799.
Call Park and Page via short code is primarily useful for phones with no display, or phones on
which a Call Park operation is seldom performed. It provides a way for the user to Central Park in
a pre-known location. If the highest Central Park slot is already in use, then the short code Call
Park and Page attempt will not be successful.
In order to perform a Page after a successful Call Park via short code, the user must enter a valid
Page short code.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: Call Park and Page
Release: 9.0+.
Related links
Short Code Features on page 809

Call Pickup Any

Pick up the first available ringing call.
Details
Telephone Number:
Default Short Code: *30
Programmable Button Control: PickA

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Short Code Features

See also: Call Pickup Extn, Call Pickup Group, Call Pickup Members, Acquire Call, Call Pickup
Line, Call Pickup User.
Release: 1.0+.
Example
Below is an example of the short code setup:
• Short Code: *30
• Feature: CallPickupAny
Related links
Short Code Features on page 809

Call Pickup Extn

Pick up a ringing call from a specific extension.
Details
Telephone Number: Target extension number.
Default Short Code: *32*N#
Programmable Button Control: CpkUp
See also: Call Pickup Any, Call Pickup Group, Call Pickup Members, Acquire Call, Call Pickup
Line, Call Pickup User.
Release: 1.0+.
Example
This short code is a default within the system configuration. N represents the specific extension.
For example, if a user dials *32*201#, they will pick up the call coming into extension 201.
Short Code: *32*N#
Telephone Number: N
Feature: CallPickupAny
Related links
Short Code Features on page 809

Call Pickup Group

Pick up a call ringing any hunt group of which the user is a member. The user can use this feature
even if their membership of the group is currently set as disabled.

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 Call Pickup Line

Details
Telephone Number:
Default Short Code: *31
Programmable Button Control: PickG
See also: Call Pickup Any, Call Pickup Extn, Call Pickup Members, Acquire Call, Call Pickup Line,
Call Pickup User.
Release: 1.0+.
Example
Below is an example of the short code setup.
Short Code: *31
Feature: CallPickupGroup
Related links
Short Code Features on page 809

Call Pickup Line

Pick up an incoming call which is alerting, parked or held. The pickup uses the Line Appearance
ID specified in Telephone Number field of the short code. It cannot be used to pickup conferenced
calls. The normal user intrusion features are not applied to this pickup feature.
Details
Telephone Number: Target Line Appearance ID.
Default Short Code:
Programmable Button Control:
See also: Call Pickup Any, Call Pickup Extn, Call Pickup Group, Call Pickup Members, Acquire
Call, Call Pickup User.
Release: 4.0+ (Added in the Release 4.0 Q2 2007 maintenance release).
Example
This short code is a default within the system configuration. N represents the specific Line
Appearance ID.
Short Code: *89*N#
Telephone Number: N
Feature: CallPickupLine

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Short Code Features

Related links
Short Code Features on page 809

Call Pickup Members

This feature can be used to pick up a ringing or queuing call at an extension that is a member of
the Hunt Group specified. The call picked up does not have to be a hunt group call. The function
includes group members even if their membership of the group is currently disabled.
Details
Telephone Number: Group number or "Group name".
Default Short Code: *53*N#
Programmable Button Control: PickM
See also: Call Pickup Any, Call Pickup Extn, Call Pickup Group, Acquire Call, Call Pickup Line,
Call Pickup User.
Release: 1.0+.
Example
Below is an example of the short code setup. N represents the extension number of the Hunt
Group. For example, if a user dials *53*500#, they will pick up the call coming into extension 500
(the hunt group's extension).
Short Code: *53*N#
Telephone Number: N
Feature: CallPickupMembers
Related links
Short Code Features on page 809

Call Pickup User

Pick up an incoming call which is alerting, parked or held. The pickup uses the user extension
number specified in Telephone Number field of the short code. If there are multiple calls, priority is
given to picking up alerting, then parked and then held in that order of priority. It cannot be used to
pickup conferenced calls. The normal user intrusion features are not applied to this pickup feature.
Details
Telephone Number: Target user extension number.
Default Short Code:

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 Call Queue

Programmable Button Control:

See also: Call Pickup Any, Call Pickup Extn, Call Pickup Group, Call Pickup Members, Acquire
Call, Call Pickup Line.
Release: 4.0+.
Example
N represents the specific user.
Short Code: *89*N#
Telephone Number: N
Feature: CallPickupUser
Related links
Short Code Features on page 809

Call Queue
Queue the current call to the destination phone, even when the destination phone is busy. This is
the same as a transfer except it allows you to transfer to a busy phone.
Details
Telephone Number: Target extension number.
Default Short Code: *33*N#
Programmable Button Control: Queue
Release: 1.0+.
Example
Below is an example of the short code setup. N represents the extension the caller wishes to
queue for. For example, if a user dials *33*201# while connected to a caller, this caller will be
queued for extension 201.
Short Code: *33*N#
Telephone Number: N
Feature: CallQueue
Related links
Short Code Features on page 809

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Short Code Features

Call Record
This feature allows you to record a conversation. To use this requires Voicemail Pro. Refer to your
local regulations in relation to the recording of calls.
This feature uses system conference resources. If insufficient conference resource are available it
will not be possible to use this feature.
Release 4.0+: The system provides privacy features that allow users to indicate that a call should
not be recorded. See Private Calls.
Details
Telephone Number: Target extension number.
Default Short Code:
Programmable Button Control: Recor
Release: 1.0+.
Example: Record your own extension's call
To use this short code, the user should place the call on hold and dial *55. They will automatically
be reconnected to the call when recording begins.
Short Code: *55
Telephone Number: None
Feature: CallRecord
Related links
Short Code Features on page 809

Call Steal
This function can be used with or without a specified user target.
If the target has alerting calls, the function will connect to the longest waiting call.
If the target has no alerting calls but does have a connected call, the function will take over the
connected call, disconnecting the original user. This usage is subject to the Can Intrude setting of
the Call Steal user and the Cannot Be Intruded setting of the target.
If no target is specified, the function attempts to reclaim the user's last ringing or transferred call if
it has not been answered or has been answered by voicemail.
Details
Telephone Number: Target extension number or blank for last call transferred.
Default Short Code: *45*N# and *46

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 Call Waiting On

Programmable Button Control: Aquire

Release: 2.1+
Example: Taking Over a Call
In this example, N represents the extension to be taken over. For example, if a user dials
*45*201#, they will take over the current call on extension 201.
Short Code: *45*N#
Telephone Number: N
Feature: Call Steal
Example: Reclaiming a Call
This short code reclaims the last call from your extension. This function is useful when you want to
catch a call you have just missed that has gone off to Voicemail.
Short Code: *46
Feature: Call Steal
Related links
Short Code Features on page 809

Call Waiting On
Enables call waiting on the user's extension. When on, if the user receives a second calls when
already on a call, they hear a call waiting tone in the speech path.
Call waiting settings are ignored for users with multiple call appearance buttons. In this case the
appearance buttons are used to indicate additional calls. Call waiting is automatically applied for
users with 'internal twinned' phones.
Details
Telephone Number:
Default Short Code: *15 (not on Server Edition)
Programmable Button Control: CWOn
See also: Call Waiting Off, Call Waiting Suspend.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *15
Feature: CallWaitingOn

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Short Code Features

Related links
Short Code Features on page 809

Call Waiting Off

Disables call waiting on the user's extension. Call waiting may be applied for users with internal
twinned phones regardless of their call waiting settings.
Details
Telephone Number:
Default Short Code: *16 (not on Server Edition)
Programmable Button Control: CWOff
See also: Call Waiting On, Call Waiting Suspend.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *16
Feature: Call Waiting Off
Related links
Short Code Features on page 809

Call Waiting Suspend

For phones using call waiting, this feature temporarily disables call waiting for the duration of the
user's next call.
Details
Telephone Number:
Default Short Code: *70 (A-Law only)
Programmable Button Control: CWSus
See also: Call Waiting On, Call Waiting Off.
Release: 1.0+.
Example

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 Cancel All Forwarding

Below is a sample of the short code setup. This short code is a default within the system
configuration.
Short Code: *70
Feature: CallWaitingSuspend
Related links
Short Code Features on page 809

Cancel All Forwarding

This feature cancels all forms of forwarding on the user's extension including "Follow Me" and "Do
Not Disturb".
Details
Telephone Number:
Default Short Code: *00
Programmable Button Control: FwdOf
See also: Forward On Busy On, Forward On Busy Off, Forward On No Answer On, Forward On
No Answer Off, Forward Unconditional On, Forward Unconditional Off, Do Not Disturb On, Do Not
Disturb Off.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *00
Feature: CancelCallForwarding
Related links
Short Code Features on page 809

Cancel Ring Back When Free

Cancels any existing ring back (also known as callback) set by the user.
Details
Telephone Number:
Default Short Code:

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Short Code Features

Programmable Button Control: RBak-

See also: Ring Back When Free.
Release: 1.0+.
Example: Cancel Ring Back When Free
This example Short Code will cancel Ring Back When Free on the specified extension. N
represents the target extension from which you have set a ring back. For example, if Paul has set
a ring back on extension 201, he must dial *84*201# to cancel that ring back request.
Short Code: *84*N#
Telephone Number: N
Feature: CancelRingBackWhenFree
Related links
Short Code Features on page 809

Change Login Code

Allows a user to change their login code. The login code must meet the Login Code Complexity
requirements defined on the Manager System | Telephony tab.
Details
Telephone Number: The user's current and new log in codes separated by a *, see the
examples below.
Default Short Code:
Programmable Button Control:
Example
The user has a Login Code of 1234 and wants to change it to 5678. To use the short code below,
the user must dial *60*1234*5678#.
Short Code: *60*N#
Telephone Number: N
Feature: Change Login Code.
Example
For a user with no login code currently set, they can still use the short code to set a login code.
For example using the short code created above to set their login code to 1234 they should dial
*60**1234#.
Example

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 Clear After Call Work

System phone users (see System Phone Features on page 715) can also use this short code to
change the login code of an other user. For example 403 is configured as a system phone with a
login code of 1234. User 410 has forgotten their login code and needs it changed. User 403 can
do this by dialing the following:
*60*410*1234*<new code>#
Related links
Short Code Features on page 809

Clear After Call Work

This feature can be users who have been configured as CCR agents. It allows them to dial a short
code to exit the After Call Work (ACW) state as reported by the Customer Call Reporter (CCR)
application.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: ACWrk
See also: Start After Call Work.
Release: 4.2 4Q 2008 Maintenance release+.
Related links
Short Code Features on page 809

Clear Call
This feature can be used to end the current call.
Details
Telephone Number:
Default Short Code: *52
Programmable Button Control: Clear
Release: 1.0+.
Example
Below is a sample of the short code setup. This example could be used in a situation where you
are doing a supervised transfer and the party to be transferred to does not want to take the call. In

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Short Code Features

this scenario, you can put the call on hold and dial *52. This will clear the last connected call (for
example the party who has just refused the transfer), and retrieve the original call or dial tone.
Short Code: *52
Feature: Deny/ClearCall
Related links
Short Code Features on page 809

Clear CW
This feature is most commonly used to end the user's current call and answer the waiting call.
Note:
Call waiting settings are ignored for users with multiple call appearance buttons.
Details
Telephone Number:
Default Short Code: *26 (A-Law only) (not on Server Edition)
Programmable Button Control: ClrCW
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *26
Feature: ClearCW
Related links
Short Code Features on page 809

Clear Hunt Group Night Service

This feature changes the specified hunt group from Night Service mode to In Service mode.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is currently not supported between systems in a multi-site network. It can only be
used by a user currently logged onto the same system as hosting the hunt group.
Details

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 Clear Hunt Group Out Of Service

Telephone Number: Hunt group extension number. If left blank, the short code will affect all
hunt groups of which the user is a member.
The Set Hunt Group Night Service and Clear Hunt Group Night Service short code and button
features can be used to switch an SSL VPN service off or on respectively. The service is indicated
by setting the service name as the telephone number or action data. Do not use quotation marks.
Default Short Code: *21*N#
Programmable Button Control: HGNS-
See also: Clear Hunt Group Out Of Service, Set Hunt Group Night Service, Set Hunt Group Out
Of Service.
Release: 1.0+.
Example
Below is a sample of the short code setup. N represents the telephone number of the hunt group
to be taken out of "Night Service" mode and placed into "In Service" mode. For example, when
*21*201# is dialed, the hunt group associated with extension 201 will be taken out of "Night
Service" mode.
Short Code: *21*N#
Telephone Number: N
Feature: ClearHuntGroupNightService
Related links
Short Code Features on page 809

Clear Hunt Group Out Of Service

This feature changes the specified hunt group from Out of Service mode to In Service mode. This
will not override a hunt group in night service due to a time profile.
Details
Telephone Number: Hunt group extension number. If left blank, the short code will affect all
hunt groups of which the user is a member.
Default Short Code:
Programmable Button Control: HGOS-
See also: Clear Hunt Group Night Service, Set Hunt Group Night Service, Set Hunt Group Out Of
Service.
Release: 1.0+.
Example

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Short Code Features

Below is a sample short code using the Clear Hunt Group Out Of Service feature. N represents
the telephone number of the hunt group to be taken out of "Out of Service" mode. For example,
when *55*201# is dialed, the hunt group associated with extension 201 will be placed into "In
Service" mode.
Short Code: *55*N#
Telephone Number: N
Feature: ClearHuntGroupOutOfService
Related links
Short Code Features on page 809

Clear Quota
This feature refreshes the time quota for all services or a specific service.
Details
Telephone Number: "Service name" or "" (all services).
Default Short Code:
Programmable Button Control: Quota
Release: 1.0+.
Related links
Short Code Features on page 809

Coaching Intrusion
This feature allows the you to intrude on another user's call and to talk to them without being
heard by the other call parties to which they can still talk. For example: User A is on a call with
user B. When user C intrudes on user A, they can hear users A and B but can only be heard by
user A.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.

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 Conference Add

Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
The system support a range of other call intrusion methods in addition to this feature.
Details
Telephone Number: Target extension number.
Default Short Code:
Programmable Button Control: Coach.
See also: Call Intrude, Call Listen, Dial Inclusion, Whisper Page.
Release:9.0+
Related links
Short Code Features on page 809

Conference Add
Conference add controls can be used to place the user, their current call and any calls they have
on hold into a conference. When used to start a new conference, the system automatically assigns
a conference ID to the call. This is termed ad-hoc (impromptu) conferencing.
If the call on hold is an existing conference, the user and any current call are added to that
conference. This can be used to add additional calls to an ad-hoc conference or to a meet-me
conference. Conference add can be used to connect two parties together. After creating the
conference, the user can drop from the conference and the two incoming calls remain connected.
For further details refer to the Conferencing section.
Details
Telephone Number:
Default Short Code: *47
Programmable Button Control: Conf+
See also: Conference Meet Me.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *47

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Short Code Features

Feature: ConferenceAdd
Related links
Short Code Features on page 809

Conference Meet Me
Conference meet-me refers to features that allow a user or caller to join a specific conference by
using the conference's ID number (either pre-set in the control or entered at the time of joining the
conference).
IP500 and IP500 V2 systems require a Preferred Edition license.
Note:
Conference Meet Me features can create conferences that include only one or two parties.
These are still conferences that are using resources from the host system's conference
capacity.
Conference ID Numbers
By default, ad hoc conferences are assigned numbers starting from 100 for the first conference in
progress. Therefore, for conference Meet Me features specify a number away from this range
ensure that the conference joined is not an ad hoc conference started by other users. It is no
longer possible to join a conference using conference Meet Me features when the conference ID is
in use by an ad-hoc conference.
User Personal Conference Number Each user's own extension number is treated as their own
personal conference number. Only that user is able to start a conference using that number as the
conference ID. Any one else attempting to start a conference with that number will find themselves
in a conference but on hold until the owner also joins. Personal conferences are always hosted on
the owner's system.
Note:
When a user calls from their mobile twinned number, the personal conference feature will only
work if they access the conference using an FNE 18 service.
Multi-Site Network Conferencing
Meet Me conference IDs are now shared across a multi-site network. For example, if a conference
with the ID 500 is started on one system, anyone else joining conference 500 on any system will
join the same conference. Each conference still uses the conference resources of the system on
which it was started and is limited by the available conference capacity of that system.
Previously separate conferences, each with the same conference ID, could be started on each
system in a multi-site network.
Other Features

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 CW

Transfer to a Conference Button A currently connected caller can be transferred into the
conference by pressing TRANSFER, then the Conference Meet Me button and TRANSFER again
to complete the transfer. This allows the user to place callers into the conference specified by the
button without being part of the conference call themselves. This option is only support on Avaya
phones with a fixed TRANSFER button.
Conference Button Status Indication When the conference is active, any buttons associated
with the conference ID indicate the active state.
For further details refer to the Conferencing section.
Details
Details
• Telephone Number: Conference number. This can be an alphanumeric value up to 15
characters.
- The number can be prefixed with H(x) where x is the number of the music-on-hold source
that should be played to the first caller to enter the conference.
• Default Short Code: / *66*N# on Server Edition systems.
• Programmable Button Control: CnfMM
• See also: Conference Add.
• Release: 1.0+.
Related links
Short Code Features on page 809

CW
Pick up the waiting call. This feature provides same functionality as pressing the Recall or Hold
key on the phone. Unlike the Clear CW feature, this feature does not disconnect you from the
existing call when the second call is picked up.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 1.0+.
Related links
Short Code Features on page 809

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Short Code Features

Dial
This short code feature allows users to dial the number specified to an outside line.
Details
Telephone Number: Telephone number.
Default Short Code: Various depending on locale and system type.
Programmable Button Control: Dial
See also: Dial Direct, Dial Emergency, Dial Extn, Dial Inclusion, Dial Paging.
Release: 1.0+.
Example: Creating a Speed Dial
In this example, users entering 401 on their telephone key pad will dial the New Jersey Office on
212 555 0000.
Short Code: 401
Telephone Number: 2125550000
Example: Replace Outgoing Caller ID
This short code is useful in a "call center" environment where you do not want customers to have
access to the number of your direct line; you want the general office number displayed. The
sample short code below will force the outgoing caller ID to display 123.
Note:
The usability of this feature is dependent upon your local service provider.
Short Code: ?
Telephone Number: .s123
Example: External Dialing Prefix
The short code is for dialing a prefix for an outside line N represents the external number you want
to call.
Short Code: 9N
Telephone Number: N
Example: Blocking Caller ID
This is for blocking Caller ID for external calls. This feature can be applied to specific external
numbers or to all out going calls. In most situations, the company will choose to block the caller ID
for all external calls or leave it available for all external calls.
Short Code: 9N
Telephone Number: NW
Example: Maximum Call Length

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 Dial 3K1

The character t can be used in dialing short codes to set the maximum allowed duration of a call.
For example, the following short code will dial a number but then disconnect the call after 20
minutes (plus or minus a minute).
Short Code: 9N
Telephone Number: Nt(20)
Related links
Short Code Features on page 809

Dial 3K1
Sets the ISDN bearer capabilities to 3.1Khz audio call.
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: D3K1
Release: 1.0+.
Related links
Short Code Features on page 809

Dial 56K
Sets the ISDN bearer capabilities to 56Kbps data call.
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: D56K
Release: 1.0+.
Related links
Short Code Features on page 809

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Short Code Features

Dial 64K
Sets the ISDN bearer capabilities to 64Kbps data call.
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: D64K
Release: 1.0+.
Related links
Short Code Features on page 809

Dial CW
Call the specified extension number and force call waiting indication on if the extension is already
on a call.
If the user has call appearance buttons programmed, call waiting will not get activated. The next
incoming call will appear on an available call appearance button. When there are no available call
appearance buttons, the next incoming call will receive busy tone.
Details
Telephone Number: Extension number.
Default Short Code:
Programmable Button Control: DCW
Release: 1.0+.
Example
N represents the extension number to be dialed. For example, a user dialing *97*201# will force
call waiting indication on at extension 201 if extension 201 is already on a call.
Short Code: *97*N#
Telephone Number: N
Feature: DialCW
Related links
Short Code Features on page 809

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 Dial Direct

Dial Direct
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
Details
Telephone Number: Extension number
Default Short Code:
Programmable Button Control: Dirct
See also: Dial Paging.
Release: 1.0+.
Example
This allows the extension specified to be automatically answered. N represents the extension that
will be forced to automatically answer. For example, when a user dials *83*201#, extension 201
will be forced to automatically answer the call.
Short Code: *83*N#
Telephone Number: N
Feature: DialDirect
Related links
Short Code Features on page 809

Dial Direct Hot Line

When the line appearance button is mapped to a short code using the DialDirectHotLine short
code feature, no secondary dial tone is generated and the number is dialed directly. This feature
should not be confused with the hot line feature enabled using ?D short codes.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 3.0 to 4.0, 8.0+
Example

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Short Code Features

Below is a sample short code using the DialDirectHotLine feature. The short code *83* should
then be set as the prefix for the particular line required.
Short Code: *83*
Telephone Number: .
Feature: DialDirectHotLine
Related links
Short Code Features on page 809

Dial Emergency
Dials the number specified regardless of any call barring applicable to the user.
On all systems, regardless of locale; system and or ARS short codes using the Dial Emergency
feature should be created for any required emergency service numbers. Those short codes should
be usable by all users from all extensions. Those short codes should route the calls to suitable
lines. If the system uses prefixes for external dialing, the dialing of emergency numbers with and
without the prefix should be allowed.
The blocking of emergency calls or the rerouting of emergency calls to a intermediate destination
other than the central office may be against local and nation laws.
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: Emrgy
Release: 1.0+.
Related links
Short Code Features on page 809

Dial Extn
This feature can be used to dial an internal extension number (user or hunt group).
Details
Telephone Number: Extension number.
p( x ) can be added as a suffix to the Telephone Number to change the priority of a call.
Allowable values for x are 1, 2 or 3 for low, medium or high priority respectively. For example
Np(1).

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 Dial Fax

Default Short Code:

Programmable Button Control:
See also: Dial Direct, Dial Paging, DialPhysicalExtensionByNumber, DialPhysicalNumberByID.
Release: 1.0+.
Example: Dial on Pick up
The following user short code dials the extension specified the moment the user's handset is
picked up.
Short Code: ?D
Telephone Number: 201
Line Group ID: 0
Feature: Dial Extn
Related links
Short Code Features on page 809

Dial Fax
This feature is used to route fax calls via Fax Relay.
Details
Telephone Number: Fax destination number.
Default Short Code:
Programmable Button Control:
Release: 5.0+.
Example
In this example, the line group ID matches the URI configured on a SIP line that has been
configured for Fax Relay.
Short Code: 6N
Telephone Number: N"@192.16.42.5"
Line Group ID: 17
Feature: Dial Fax
Related links
Short Code Features on page 809

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Short Code Features

Dial Inclusion
This feature allows you to intrude on another user's call to talk to them. Their current call is put on
hold while you talk and automatically reconnected when you end the intrusion. The intruder and
the target extension can then talk but cannot be heard by the other party. This can include
intruding into a conference call, where the conference will continue without the intrusion target.
During the intrusion all parties hear a repeated intrusion tone. When the intruder hangs-up the
original call parties are reconnected. Attempting to hold a dial inclusion call simply ends the
intrusion. The inclusion cannot be parked.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
The system support a range of other call intrusion methods in addition to this feature.
Details
Telephone Number: Target extension number.
Default Short Code:
Programmable Button Control: Inclu.
See also: Call Intrude, Call Listen, Coaching Intrusion, Whisper Page.
Release: 1.4+.
Example
N represents the extension to be intruded upon. For example, if a user dials *97*201# while
extension 201 is on a call, then the user is intruding into extn. 201's current call.
Short Code: *97*N#
Telephone Number: N
Feature: DialInclusion
Related links
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Dial Paging
This feature makes a page call to an extension or group. The target extension or group members
must support page calls (that is be able to auto-answer calls).
Details

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 Dial Physical Extension by Number

Telephone Number: Extension or group number.

Default Short Code:
Programmable Button Control: Page
See also: Dial Direct.
Release: 1.0+.
When paging, always use only one codec (the preferred). It is the system administrator's
responsibility to ensure all the phones in the paging group support the codec.
Related links
Short Code Features on page 809

Dial Physical Extension by Number

Dial a specified extension number regardless of the current user logged in at that extension and
any forwarding, follow me or do not disturb settings applied by the current extension user. Note
that the extension number used is the Base Extension number set against the extension
configuration settings.
Details
Telephone Number: Base Extension number.
Default Short Code: *70*N# (U-Law only) (not on Server Edition)
Programmable Button Control: PhyEx
See also: Dial Physical Extension By Id, Priority Call.
Release: 1.4+.
Example
The example below allows the extension with the base extension number 201 to be called
regardless of the extension number of the user currently logged in at that extension.
Short Code: *97
Telephone Number: 201
Feature: DialPhysicalExtnByNumber
Related links
Short Code Features on page 809

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Short Code Features

Dial Physical Extension By ID

Dial a specific extension using its system ID. This may be necessary in hot desking environments
where some extensions have been created with no default extension number. Without an
extension number, a call can not be made to that extension unless a short code is created.
Details
Telephone Number: Extension ID
Default Short Code: *71*N# (U-Law only)
Programmable Button Control: DialP
See also: DialPhysicalExtensionByNumber, Priority Call.
Release: 1.4+.
Example
In the above example, if the telephone at extension ID 16 is not associated with an extension
number, a user can dial *97 to connect to that phone. This may be useful in hot desking
environments where some extensions may not have a dedicated base extension number.
Short Code: *97
Telephone Number: 16
Feature: DialPhysicalNumberByID
Related links
Short Code Features on page 809

Dial Speech
This feature allows a short code to be created to force the outgoing call to use the Speech bearer
capability.
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: DSpch
Release: 1.0+.
Related links
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 Dial V110

Dial V110
Sets the ISDN bearer capabilities to V110. The call is presented to local exchange as a "Data
Call".
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: DV110
Release: 1.0+.
Related links
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Dial V120
Sets the ISDN bear capabilities using V.120.
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: DV120
Release: 1.0+.
Related links
Short Code Features on page 809

Dial Video
The call is presented to the local exchange as a "Video Call".
Details
Telephone Number: Telephone number.
Default Short Code:
Programmable Button Control: Dvide
Release: 1.0+.

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Short Code Features

Related links
Short Code Features on page 809

Disable ARS Form

This feature can be used to put an ARS form out of service. It can be used with ARS forms for
which an Out of Service Route has been configured in Manager. The short code feature Enable
ARS Form can be used to return an ARS form to in service.
Details
Telephone Number: ARS form number.
Default Short Code:
Programmable Button Control:
See also: Enable ARS Form
Release: 4.0+.
Related links
Short Code Features on page 809

Disable Internal Forwards

This feature turns off the forwarding of internal calls for the user. It applies to Forward
Unconditional, Forward on Busy and Forward on No Answer.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Disable Internal Forwards Unconditional, Disable Internal Forwards Busy or No Answer,
Cancel All Forwarding, Enable Internal Forwards, Enable Internal Forwards Unconditional, Enable
Internal Forwards Busy or No Answer.
Release: 3.2+.
Related links
Short Code Features on page 809

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 Disable Internal Forward Unconditional

Disable Internal Forward Unconditional

This feature turns off the forwarding of internal calls for the user. It applies to Forward
Unconditional only.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Disable Internal Forwards, Disable Internal Forwards Busy or No Answer, Cancel All
Forwarding, Enable Internal Forwards, Enable Internal Forwards Unconditional, Enable Internal
Forwards Busy or No Answer.
Release: 3.2+.
Related links
Short Code Features on page 809

Disable Internal Forward Busy or No Answer

This feature turns off the forwarding of internal calls for the user. It applies to Forward on Busy and
Forward on No Answer.
Telephone Number: No
Default Short Code: No
Programmable Button Control: No
See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Cancel All
Forwarding, Enable Internal Forwards, Enable Internal Forwards Unconditional, Enable Internal
Forwards Busy or No Answer.
Related links
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Display Msg
Allows the sending of text messages to digital phones on the local system.
Telephone Number: The telephone number takes the format N";T" where:
• N is the target extension.

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Short Code Features

• T is the text message. Note that the "; before the text and the " after the text are required.
Default Short Code: No
Programmable Button Control: Displ
Example
Below is a sample of the short code setup. When used, the target extension will hear a single ring
and then see the message. If the target extension is on a call then may need to scroll the display
to a free call appearance in order to see the text message.
Short Code: *78*N#
Feature: Display Msg
Telephone Number: N";Visitor in Reception"
SIP Extension Message Waiting Indicator
You can use the Display Msg short code to turn an extension’s message waiting indicator (MWI)
on or off.
Telephone Number: The telephone number takes the format N";T" where:
• N is the target extension.
• T is the text message. Note that the "; before the text and the " after the text are required.
To turn MWI on, the telephone number must be N";Mailbox Msgs=1".
To turn MWI off, the telephone number must be N";Mailbox Msgs=0".
Default Short Code: No
Example
Below is a sample of the short code setup to turn MWI on. When used, the target extension will
receive a message directing it to turn the MWI on.
Short Code: *99*N#
Feature: Display Msg
Telephone Number: N";Mailbox Msgs=1"
Example
Below is a sample of the short code setup to turn MWI off. When used, the target extension will
receive a message directing it to turn the MWI off.
Short Code: *98*N#
Feature: Display Msg
Telephone Number: N";Mailbox Msgs=0"
Related links
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 Do Not Disturb Exception Add

Do Not Disturb Exception Add

This feature adds a number to the user's "Do Not Disturb Exception Numbers List". This can be an
internal extension number or external ICLID. Calls from that number, except hunt group calls, will
ignore the user's Do Not Disturb setting. For further details see Do Not Disturb (DND).
Telephone Number: Telephone number or ICLID. Up to 31 characters. For ICLID numbers any
prefix added by the system must also be included.
Default Short Code: *10*N#
Programmable Button Control: DNDX+
See also: Do Not Disturb Exception Delete, Do Not Disturb On, Do Not Disturb Off.
Example
N represents the number to be added to the user's "Do Not Disturb Exception List". For example,
when a user has DND turned on and dials *10*4085551234#, incoming calls from telephone
number (408) 555-1234. All other calls, except those numbers on the exception list hear busy
tones or are redirected to voicemail if available.
Short Code: *10*N#
Telephone Number: N
Feature: DoNotDisturbExceptionAdd
Example
In this example, the last number received by the user is added to their exception list.
Short Code: *89
Telephone Number: L
Feature: DoNotDisturbExceptionAdd
Related links
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Do Not Disturb Exception Delete

This feature removes a number from the user's "Do Not Disturb Exception List". For further details
see Do Not Disturb (DND).
Details
Telephone Number: Telephone number or ICLID.
Default Short Code: *11*N#
Programmable Button Control: DNDX-
See also: Do Not Disturb Exception Add, Do Not Disturb On, Do Not Disturb Off.

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Short Code Features

Release: 1.0+.
Example
N represents the number to be deleted from the user's "Do Not Disturb Exception List". For
example, when a user has DND turned on and the telephone number (408) 555-1234 in their "Do
Not Disturb Exception List", dialing *10*4085551234# will remove this phone number from the list.
Incoming calls from (408) 555-1234 will no longer be allowed through; instead they will hear busy
tone or be redirected to voicemail if available.
Short Code: *11*N#
Telephone Number: N
Feature: DoNotDisturbExceptionDel
Related links
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Do Not Disturb On
This feature puts the user into 'Do Not Disturb' mode. When on, all calls, except those from
numbers in the user's exception list hear busy tones or are redirected to voicemail if available. For
further details see Do Not Disturb (DND).
Details
Telephone Number:
Default Short Code: *08
Programmable Button Control: DNDOn
See also: Do Not Disturb Off, Do Not Disturb Exception Add, Do Not Disturb Exception Delete.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *08
Feature: DoNotDisturbOn
Related links
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 Do Not Disturb Off

Do Not Disturb Off

Cancels the user's 'do not disturb' mode if set. For further details see Do Not Disturb (DND).
Details
Telephone Number:
Default Short Code: *09
Programmable Button Control: DNDOf
See also: Do Not Disturb On, Do Not Disturb Exception Add, Do Not Disturb Exception Delete.
Release: 1.0+.
Example
This short code is a default within the system configuration. Below is a sample of the short code
setup.
Short Code: *09
Feature: DoNotDisturbOff
Related links
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Enable ARS Form

This feature can be used to put an ARS form in service. It can be used with ARS forms that have
been put out of service through Manager or the use of a Disable ARS Form short code.
Details
Telephone Number: ARS form number.
Default Short Code:
Programmable Button Control:
Release: 4.0+
Related links
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Short Code Features

Enable Internal Forwards

This feature turns on the forwarding of internal calls for the user. It applies to Forward
Unconditional, Forward on Busy and Forward on No Answer.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Disable Internal
Forwards Busy or No Answer, Cancel All Forwarding, Enable Internal Forwards Unconditional,
Enable Internal Forwards Busy or No Answer.
Release: 3.2+.
Related links
Short Code Features on page 809

Enable Internal Forward Unconditional

This feature turns on the forwarding of internal calls for the user. It applies to Forward
Unconditional only.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Disable Internal
Forwards Busy or No Answer, Cancel All Forwarding, Enable Internal Forwards, Enable Internal
Forwards Busy or No Answer.
Release: 3.2+.
Related links
Short Code Features on page 809

Enable Internal Forward Busy or No Answer

This feature turns on the forwarding of internal calls for the user. It applies to Forward on Busy and
Forward on No Answer.

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 Extn Login

Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Disable Internal Forwards, Disable Internal Forwards Unconditional, Disable Internal
Forwards Busy or No Answer, Cancel All Forwarding, Enable Internal Forwards, Enable Internal
Forwards Unconditional.
Release: 3.2+.
Related links
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Extn Login
Extn Login allows a user who has been configured with a Login Code (User | Telephony |
Supervisor Settings) to take over ownership of any extension. That user's extension number
becomes the extension number of the extension while they are logged. This is also known as ‘hot
desking’.
Hot desking is not supported for H175, E129 and J129 telephones.
When used, the user will be prompted to enter their extension number and then their log in code.
Login codes of up to 15 digits are supported with Extn Login buttons. Login codes of up to 31
digits are supported with Extn Login short codes.
When a user logs in, as many of their user settings as possible are applied to the extension. The
range of settings applied depends on the phone type and on the system configuration.
By default, on 1400 Series, 1600 Series, 9500 Series and 9600 Series phones, the user's call log
and personal directory are accessible while they are logged in. This also applied to M-Series and
T-Series telephones.
On other types of phone, those items such as call logs and speed dials are typically stored locally
by the phone and will not change when users log in and log out.
If the user logging in was already logged in or associated with another phone, they will be
automatically logged out that phone.
Details
Telephone Number: Extension Number*Login Code. If just a single number is dialed containing
no * separator, the system assumes that the extension number to use is the physical extension's
Base Extension number and that the number dialed is the log in code.
Default Short Code: *35*N#
Programmable Button Control: Login

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Short Code Features

See also: Extn Logout.

Release: 1.0+.
Example: Individual Hot Desking
Based on the above sample short code, Paul (extension 204) can go to another phone (even if it is
already logged in by another user) and log in as extension 204 by simply dialing 299. Once Paul
has logged into this phone, extension 204 is logged out at Paul's original phone. For Paul to make
use of this short code, his log in code must match that configured in the above short code. When
Paul logs out of the phone he has "borrowed", his original extension will automatically be logged
back in.
Short Code: 299
Telephone Number: 204*1234
Feature: Extnlogin
Example: Log In
The default short code for logging into a phone is configured as shown below. N represents the
users extension number followed by a * and then their log in code, for example *35*401*123#.
Short Code: *35*N#
Telephone: N
Feature: ExtnLogin
Related links
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Extn Logout
This feature logs the user off the phone at which they are logged in. This feature cannot be used
by a user who does not have a log in code or by the default associated user of an extension
unless they are set to forced log in.
Details
Telephone Number:
Default Short Code: *36
Programmable Button Control: Logof
See also: Extn Login.
Release: 1.0+.
Example

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 Flash Hook

Below is a sample short code using the Extn Logout feature. This short code is a default within the
system configuration.
Short Code: *36
Feature: ExtnLogout
Related links
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Flash Hook
This feature sends a hook flash signal to the currently connected line if it is an analog line.
Details
Telephone Number: Optional The telephone number field can be used to set the transfer
destination number for a Centrex Transfer. In this case the use of the short code Forced Account
Code and Forced Authorization Code are not supported and the Line Group Id must match the
outgoing line to the Centrex service provider.
Default Short Code:
Programmable Button Control: Flash
Release: 1.4+.
Example
Below is a sample short code using the Flash Hook feature.
Short Code: *96
Feature: FlashHook
Related links
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FNE Service
This short code feature is used for Mobile Call Control and one-X Mobile Client support.
Details
Telephone Number: This number sets the required FNE function.
Default Short Code:
Programmable Button Control:

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Short Code Features

Release: 4.2+.
Related links
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Follow Me Here
Causes calls to the extension number specified to be redirected to the extension initiating the
'Follow Me Here'. If the redirected call receives a busy tone or is not answered, then the call
behaves as though the User's extension had failed to answer. For further details see Follow Me.
Details
Telephone Number: Extension to redirect to the dialing extension.
Default Short Code: *12*N#
Programmable Button Control: Here+
See also: Follow Me Here Cancel, Follow Me To.
Release: 1.0+.
Example
This feature is used at the Follow Me destination. N represents the extension number of the user
wanting their calls redirected to that destination. For example, User A's extension is 224. However
they are working at extension 201 and want their calls redirected there. If the following short code
is available, they can do this by dialing *12*224# at extension 201.
Short Code: *12*N#
Telephone Number: N
Feature: FollowMeHere
Related links
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Follow Me Here Cancel

Cancels any Follow Me set on the specified extension. This action can only be performed at the
extension to which the Follow Me Here is targetted. For further details see Follow Me.
Details
Telephone Number: Extension being redirected to the dialing extension.
Default Short Code: *13*N#

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 Follow Me To

Programmable Button Control: Here-

See also: Follow Me Here, Follow Me To.
Release: 1.0+.
Example
This feature is used at the Follow Me destination. N represents the extension number of the user
whose calls are being redirected to that destination. For example, User A's extension is 224.
However they are working at extension 201 and so have set a Follow Me on their own extension
to redirect their calls to 201. If the following short code is available, they can cancel the Follow Me
by dialing *13*224# at extension 201.
Short Code: *13*N#
Telephone Number: N
Feature: FollowMeHereCancel
Related links
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Follow Me To
Causes calls to the extension to be redirected to the Follow Me destination extension specified.
For further details see Follow Me.
Details
Telephone Number: Target extension number or blank (cancel Follow Me To)
Default Short Code: *14*N#
Programmable Button Control: FolTo
See also: Follow Me Here, Follow Me Here Cancel.
Release: 1.0+.
Example
This feature is used at the extension that wants to be redirected. N represents the extension
number to which the user wants their calls redirected. For example, User A's extension is 224.
However they are working at extension 201 and want their calls redirected there. If the following
short code is available, they can do this by dialing *14*201# at extension 224.
Short Code: *14*N#
Telephone Number: N
Feature: FollowMeTo

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Short Code Features

Related links
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Forward Hunt Group Calls On

Forward the user's hunt group calls (internal and external) to their forward number when the user
has Forward Unconditional active. For further details see Forward Unconditional.
This option is only applied for calls to Sequential and Rotary type hunt groups. Calls from other
types of hunt group types are not presented to the user when they have Forward Unconditional
active. Note also that hunt group calls cannot be forwarded to another hunt group.
Details
Telephone Number:
Default Short Code: *50
Programmable Button Control: FwdH+
See also: Forward Hunt Group Calls Off, Forward Unconditional On, Forward Unconditional Off.
Release: 1.0+.
Example
This short code is useful if the hunt group member temporarily uses another workstation and so
does not require a permanent extension change.
Short Code: *50
Feature: ForwardHuntgroupCallsOn
Related links
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Forward Hunt Group Calls Off

This feature cancels the forwarding of the user's hunt group calls. For further details see Forward
Unconditional.
Details
Telephone Number:
Default Short Code: *51
Programmable Button Control: FwdH-
See also: Forward Hunt Group Calls On, Forward Unconditional On, Forward Unconditional Off.

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 Forward Number

Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *51
Feature: ForwardHuntgroupCallsOff
Related links
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Forward Number
Sets the number to which the user's calls are redirected. This can be an internal or external
number. The number is still subject to the user's call barring settings. For further details see
Forward Unconditional.
This feature does not activate forwarding; it only sets the number for the forwarding destination.
This number is used for all forward types; Forward Unconditional, Forward on Busy and Forward
on No Answer, unless the user has a separate Forward on Busy Number set for forward on busy
and forward on no answer functions.
Details
Telephone Number: Telephone number.
Default Short Code: *07*N#
Programmable Button Control: FwdNo
See also: Forward On Busy Number.
Release: 1.0+.
Example
N represents the forward destination. For example, if extension 224 wants to set the forwarding
number to extension 201, the user can dial *07*201#.
Short Code: *07N*#
Telephone Number: N
Feature: ForwardNumber
Related links
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Short Code Features

Forward On Busy Number

Sets the number to which the user's calls are forwarded when Forward on Busy or Forward on No
Answer are on. If no Forward on Busy Number is set, those functions use the Forward Number.
This feature does not activate the forwarding, it only sets the number for the forwarding
destination.
Details
Telephone Number: Telephone number.
Default Short Code: *57*N#
Programmable Button Control: FwBNo
See also: Forward Number.
Release: 1.0+.
Example
N represents the extension number to be forwarded to. For example, if Paul (whose extension is
224) wants to set the forwarding number for his 'Forward on Busy' and/or 'Forward on No Answer'
feature to extension 201, Paul can dial *57*201# followed by the short code for the forwarding
function.
Short Code: *57N*#
Telephone Number: N
Feature: ForwardOnBusyNumber
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Forward On Busy On
This feature enables forwarding when the user's extension is busy. It uses the Forward Number
destination or, if set, the Forward on Busy Number destination. If the user has call appearance
buttons programmed, the system will not treat them as busy until all the call appearance buttons
are in use.
Release 3.2+: Forward Internal (User | Forwarding) can also be used to control whether internal
calls are forwarded.
Details
Telephone Number:
Default Short Code: *03

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 Forward On Busy Off

Programmable Button Control: FwBOn

See also: Forward On Busy Off, Cancel All Forwarding, Enable Internal Forward Busy or No
Answer.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *03
Feature: ForwardOnBusyOn
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Forward On Busy Off

This feature cancels forwarding when the user's extension is busy.
Details
Telephone Number:
Default Short Code: *04
Programmable Button Control: FwBOf
See also: Forward On Busy On, Cancel All Forwarding.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *04
Feature: ForwardOnBusyOff
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Forward On No Answer On
This feature enables forwarding when the user's extension is not answered within the period
defined by their No Answer Time. It uses the Forward Number destination or, if set, the Forward
on Busy Number destination.

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Short Code Features

Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Details
Telephone Number:
Default Short Code: *05
Programmable Button Control: FwNOn
See also: Forward On No Answer Off, Cancel All Forwarding.
Release: 1.0+.
Example
Below is a sample of the short code setup. Remember that the forwarding number for this feature
uses the 'Forward on Busy Number'.
Short Code: *05
Feature: ForwardOnNoAnswerOn
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Forward On No Answer Off

This feature cancels forwarding when the user's extension is not answered.
Details
Telephone Number:
Default Short Code: *06
Programmable Button Control: FwNOf
See also: Forward On No Answer On.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *06
Feature: ForwardOnNoAnswerOff
Related links
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 Forward Unconditional On

Forward Unconditional On
This feature enables forwarding of all calls, except group calls, to the Forward Number set for the
user's extension. To also forward hunt group calls, Forward Hunt Group Calls On must also be
used. For further details see Forward Unconditional.
Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: FwUOn
See also: Forward Unconditional Off.
Release: 1.0+.
Example
Remember that this feature requires having a forward number configured.
Short Code: *01
Feature: ForwardUnconditionalOn
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Forward Unconditional Off

This feature cancels forwarding of all calls from the user's extension.
Note:
This does not disable Forward on No Answer and or Forward on Busy if those functions are
also on. For further details see Forward Unconditional.
Details
Telephone Number:
Default Short Code: *02
Programmable Button Control: FwUOf
See also: Forward Unconditional On.
Release: 1.0+.

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Short Code Features

Example
Example
Below is a sample of the short code setup.
Short Code: *02
Feature: ForwardUnconditionalOff
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Group Listen Off

Disables the group listen function for the user’s extension. See Group Listen On.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: GroupListenOn
Release: 4.1+.
Example
Below is a sample short code using the Group Listen Off feature.
Short Code: *27
Feature: GroupListenOff
Related links
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Group Listen On
Using group listen allows callers to be heard through the phone's handsfree speaker but to only
hear the phone's handset microphone. When group listen is enabled, it modifies the handsfree
functionality of the user’s phone in the following manner
When the user’s phone is placed in handsfree/speaker mode, the speech path from the connected
party is broadcast on the phone speaker but the phone's base microphone is disabled.
The connected party can only hear speech delivered via the phone's handset microphone.
Group listen is not supported for IP phones or when using a phone's HEADSET button.

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 Headset Toggle

Currently connected calls are not affected by changes to this setting. If group listen is required it
must be selected before the call is connected.
This enables listeners at the user’s phone to hear the connected party whilst limiting the
connected party to hear only what is communicated via the phone handset.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: GroupListenOn
Release: 4.1+.
Example
Below is a sample short code using the Group Listen Off feature.
Short Code: *28
Feature: GroupListenOn
Related links
Short Code Features on page 809

Headset Toggle
Toggles between the use of a headset and the telephone handset.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: HdSet
Release: 1.4+.
Example
Below is a sample short code using the Headset Toggle feature. This short code can be used to
toggle the feature on/off. If an Avaya supported headset is connected to your telephone, this short
code can be used to toggle between using the headset and the telephone handset.
Short Code: *55
Feature: HeadsetToggle
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Short Code Features

Hold Call
This uses the Q.931 Hold facility, and "holds" the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The Hold Call feature "holds" the current call to a slot. The current call is
always automatically placed into slot 0 if it has not been placed in a specified slot. Only available if
supported by the ISDN exchange.
Details
Telephone Number: Exchange hold slot number or blank (slot 0).
Default Short Code:
Programmable Button Control: Hold
See also: Hold CW, Hold Music, Suspend Call.
Release: 1.0+.
Example
Below is a sample short code using the Hold Call feature. This short code is a default within the
system configuration. N represents the exchange hold slot number you want to hold the call on.
For example, while connected to a call, dialing *24*3# will hold the call onto slot 3 on the ISDN.
Short Code: *24*N#
Telephone Number: N
Feature: HoldCall
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Hold CW
This uses the Q.931 Hold facility, and "holds" the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The Hold CW feature "holds" the current call to an exchange slot and
answers the call waiting. The current call is always automatically placed into slot 0 if it has not
been placed in a specified slot. Only available if supported by the ISDN exchange.
Details
Telephone Number: Exchange slot number or blank (slot 0).
Default Short Code: *27*N# (A-Law only) (not on Server Edition)
Programmable Button Control: HoldCW
See also: Hold Call, Suspend Call.
Release: 1.0+.

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 Hold Music

Example
Below is a sample short code using the Hold CW feature.
Short Code: *27*N#
Feature: HoldCW
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Hold Music
This feature allows the user to check the system's music on hold. See Music On Hold for more
information.
Details
Telephone Number: Optional. If no number is specified, the default system source is assumed.
The system supports up to 4 hold music sources, numbered 1 to 4. 1 represents the System
Source. 2 to 4 represent the Alternate Sources.
Default Short Code:
*34N; where N is the number of the hold music source required.
Programmable Button Control: Music
Release: 1.0+.
Example
Below is a sample short code using the Hold Music feature. This short code is a default within the
configuration.
Short Code: *34N;
Feature: HoldMusic
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Hunt Group Disable

This feature disables the user's membership of the specified hunt group. They will no longer
receive call to that hunt group until their membership is enabled again. To use this feature, you
must already belong to the hunt group. See also Hunt Group Enable.
Details

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Short Code Features

Telephone Number: Group number.

Default Short Code:
Programmable Button Control: HGDis
See also: Hunt Group Enable.
Release: 1.0+.
Example
N represents the hunt group number from which the user wants to be disabled from. For example,
if Paul wants to be disabled from the Sales hunt group (extn. 500), he needs to dial *90*500#.
Short Code: *90*N#
Telephone Number: N
Feature: HuntGroupDisable
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Hunt Group Enable

This feature enables the user's membership of a hunt group so they can begin to receive calls to
the specified hunt group. To use this feature, the user must already belong to the hunt group. This
short code can not be used to add someone to a hunt group, that must be done within Manager's
Hunt Group form.
Details
Telephone Number: Group number.
Default Short Code:
Programmable Button Control: HGEna
See also: Hunt Group Disable.
Release: 1.0+. Previously in Release 3.2 the Set Hunt Group Night Service, Set Hunt Group
Out of Service and Hunt Group Enable short code features toggled. That behaviour is not
supported in 4.0 and higher.
Example
This short code can be used to turn the feature on. N represents the hunt group number for which
the user wants to start receiving calls. For example, if Paul is already a member of the sales hunt
group (extn. 500) but has changed his availability status for that hunt group using hunt group
disable, he can make himself available for receiving calls to the Sales hunt group again by dialing
*91*500#.

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 Last Number Redial

Short Code: *91*N#

Telephone Number: N
Feature: HuntGroupEnable
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Last Number Redial

This feature allows an extension to redial the last number they dialed.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 3.0+.
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MCID Activate
This feature should only be used in agreement with the ISDN service provider and the appropriate
local legal authorities. It allows users with Can Trace Calls (User | Telephony | Supervisor
Settings) set to trigger a malicious call trace of their previous call at the ISDN exchange. Refer to
Telephone Features Malicious Call Tracing for further details.
Note:
Currently, in Server Edition network, MCID is only supported for users using an MCID button and
registered on the same IP500 V2 Expansion system as the MCID trunks.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: Advanced | Miscellaneous | MCID Activate.
Release: 4.0+.

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Short Code Features

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Mobile Twinned Call Pickup

This short code feature allows the user to pickup a call ringing or connected at the destination of
their mobile twinning number. This short code can only be used from the primary extension which
is being used for the twinning operation.
Note that the use of mobile twinning requires entry of a Mobile Twinning license and may be
subject to a time profile.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Set Mobile Twinning Number, Set Mobile Twinning On, Set Mobile Twinning Off.
Release: 3.2+.
Related links
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Off Hook Station

Enables or disables whether the user's extension acts as a fully handsfree unit. Typically this is
used when the answering and clearing of calls is done through an application. For more details
see Off Hook Station (User | Telephony | Call Settings).
Details
Telephone Number: "Y" for on or "N" for off.
Default Short Code:
Programmable Button Control: OHStn
Release: 1.0+.
Example: Turning the off hook station off
Short Code: *89
Telephone Number: N
Feature: OffHookStation

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 Outgoing Call Bar Off

Example: Turning the off hook station on

Short Code: *98
Telephone Number: Y
Feature: OffHookStation
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Outgoing Call Bar Off

Allows a user to switch off their outgoing call bar status. The short code user must enter their log
in code, if set, in order to be successful.
If you add a short code using this feature to a system it is recommended that you also assign a
login code to the No User user to prevent the short code being used to change the status of that
user.
Details
Telephone Number: The user's log in code.
System phone users (see System Phone Features on page 715) can use <target user>*<system
phone user's login code>.
Default Short Code:
Programmable Button Control:
Release: 4.1+ (Added to Release 4.1 2008Q2 Maintenance release).
Example
The user has a Login Code of 1234. To use the short code below below, the user must dial
*59*1234#.
Short Code: *59*N#
Telephone Number: N
Feature: Outgoing Call Bar Off.
Example
A user set as a system phone can also switch off the Outgoing Call Bar status of another user.
This is done using their own login code. For example the system phone 401 with login code 1234
can switch off the outgoing call bar status of extension 403 as follows:
*59*403*1234
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Short Code Features

Outgoing Call Bar On

Allows a user to switch on their outgoing call bar status.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 4.1+ (Added to Release 4.1 2008Q2 Maintenance release).
Example
To use the short code below below, the user must dial *58.
Short Code: *58
Telephone Number: <blank>
Feature: Outgoing Call Bar On.
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Private Call Off

Short codes using this feature turn off private call status for the user if set. The short code features
Private Call and Private Call On can be used to turn private call on.
When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private call.
Note that use of private calls is separate from the user's intrusion settings. If the user's Cannot be
Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching private calls off
does not affect that status. To allow private calls to be used to fully control the user status, Cannot
be Intruded (User | Telephony | Supervisor Settings) should be disabled for the user.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: Advanced | Call | Private Call.
Release: 4.0+.
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 Private Call On

Private Call On
Short codes using this feature turn on the private call settings for the user regardless.
When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private call.
Note that use of private calls is separate from the user's intrusion settings. If the user's Cannot be
Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching private calls off
does not affect that status. To allow private calls to be used to fully control the user status, Cannot
be Intruded (User | Telephony | Supervisor Settings) should be disabled for the user.
Private call status can be switched off using a short code with the Private Call Off feature or a
programmed button set to the Private Call action. To enable private call status for a single
following call only the Private Call short code feature should be used.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: Advanced | Call | Private Call.
Release: 4.0+.
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Priority Call
This feature allows the user to call another user even if they are set to 'do not disturb'. Priority calls
to a user without DND will follow forwarding and follow me settings but will not go to voicemail.
Details
Telephone Number: Extension number.
Default Short Code:
Programmable Button Control: PCall
See also: DialPhysicalExtensionByNumber, DialPhysicalNumberByID.
Release: 1.0+.
Example
N represents the extension number to be called, despite the extension being set to 'do not disturb'.
For example, if extension 201 has 'do not disturb' enabled, a user can dial *71*201# and still get
through. This short code is useful for companies that frequently use the 'do not disturb' feature and

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Short Code Features

can be given to Managing Directors or people who may need to get through to people regardless
of their 'do not disturb' status.
Short Code: *71*N#
Telephone Number: N
Feature: PriorityCall
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Record Message
This short code feature is used to record hunt group announcements on Embedded Voicemail,
see Hunt Group | Announcements. Release 5.0+: It is also used to record mailbox user name
prompts for the auto attendant Dial by Name function.
Details
Telephone Number:
For a hunt group queue announcement, use the hunt group extension number followed by ".1".
For a hunt group still queue announcement, use the hunt group extension number followed by ".
2".
For a mailbox user name prompt, use the user extension number followed by ".3".
Default Short Code: *91N; and *92N; (not on Server Edition)
Programmable Button Control:
Release: 4.0+.
Example
For a hunt group with extension number 300, the default short codes *91N;/Record Message/N".
1" and *92N;/Record Message/N".2" can be used to allow recording of the announcements by
dialing *91300# and *92300#.
To allow users to record their own name prompt, the short code *89#/Record Message/E."3" can
be used. The E is replace by the extension number of the dialing user.
Related links
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 Relay On

Relay On
This feature closes the specified switch in the system's external output (EXT O/P) port.
This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Details
Telephone Number: Switch number (1 or 2).
Default Short Code: *39 (Switch 1), *42 (Switch 2), *9000*.
Programmable Button Control: Rely+
See also: Relay Off, Relay Pulse.
Release: 1.0+.
Example
This short code is a default within the system configuration. This short code is useful for
companies that have external devices, such as door controls, connected to the system. Based on
this sample short code, a user dialing *42 is closing switch number 2 to activate an external
device.
Short Code: *42
Telephone Number: 2
Feature: RelayOn
Analog Modem Control
On systems with an analog trunk card in the control unit, the first analog trunk can be set to
answer V.32 modem calls. This is done by either selecting the Modem Enabled option on the
analog line settings or using the default short code *9000* to toggle this service on or off. This
short code uses the RelayOn feature with the Telephone Number set to "MAINTENANCE". Note
that the short code method is always returned to off following a reboot or if used for accessing the
system date and time menu.
IP500 ATM4 Uni Trunk Card Modem Support It is not required to switch the card's modem port
on/off. The trunk card's V32 modem function can be accessed simply by routing a modem call to
the RAS service's extension number. The modem call does not have to use the first analog trunk,
instead the port remains available for voice calls.
Related links
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Short Code Features

Relay Off
This feature opens the specified switch in the system's external output (EXT O/P) port.
Details
Telephone Number: Switch number (1 or 2).
Default Short Code: *40 (Switch 1), *43 (Switch 2)
Programmable Button Control: Rely-
See also: Relay On, Relay Pulse.
Release: 1.0+.
Example
This short code is a default within the system configuration. This short code is useful for
companies that have external devices, such as door controls, connected to the system. Based on
this sample short code, a user dialing *43 is opening switch number 2 to activate an external
device.
Short Code: *43
Telephone Number: 2
Feature: RelayOff
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Relay Pulse
This feature closes the specified switch in the system's external output (EXT O/P) port for 5
seconds and then opens the switch.
Details
Telephone Number: Switch number (1 or 2).
Default Short Code: *41 (Switch 1), *44 (Switch 2)
Programmable Button Control: Relay
See also: Relay On, Relay Off.
Release: 1.0+.
Example
This short code is a default within the system configuration. This short code is useful for
companies that have external devices, such as door controls, connected to the system. Based on

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 Resume Call

this sample short code, a user dialing *44 is opening switch number 2 to activate an external
device.
Short Code: *44
Telephone Number: 2
Feature: RelayPulse
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Resume Call
Resume a call previously suspended to the specified ISDN exchange slot. The suspended call
may be resumed from another phone/ISDN Control Unit on the same line.
Details
Telephone Number: Exchange suspend slot number.
Default Short Code: *23*N# (A-Law only) (not on Server Edition)
Programmable Button Control: Resum
See also: Suspend Call.
Release: 1.0+.
Example
Below is sample short code using the Resume Call feature. N represents the exchange slot
number from which the call has been suspended. For example, if a user has suspended a call on
slot number 4, this user can resume that call by dialing *23*4#.
Short Code: *23*N#
Telephone Number: N
Feature: ResumeCall
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Retrieve Call
Retrieves a call previously held to a specific ISDN exchange slot.
Details

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Short Code Features

Telephone Number: Exchange hold slot number.

Default Short Code: *25*N# (A-Law only) (not on Server Edition)
Programmable Button Control: Retriv
See also: Hold Call.
Release: 1.0+.
Example
Below is sample short code using the Retrieve Call feature. N represents the exchange slot
number from which the call has been placed on hold. For example, if a user has placed a call hold
on slot number 4, the user can resume that call by dialing *25*4#.
Short Code: *25*N#
Telephone Number: N
Feature: RetrieveCall
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Ring Back When Free

This feature sets a ringback on the specified extension. This sets a 'ringback when free' on an
extension currently on a call or a 'ringback when next used' for an extension that is free but does
not answer.
When the target extension is next used or ends its current call, the users is rung and when they
answer a call is made to the target extension.
Details
Telephone Number: Target extension number.
Default Short Code:
Programmable Button Control: RBak+
See also: Cancel Ring Back When Free.
Release: 1.0+.
Example
N represents the target extension from which you want to receive the callback. For example, if you
call extension 201 but the line is busy, hang up and then dial *71*201#. When extension 201
disconnects from its current call, your phone will ring. Once you pick up the phone, extension
201's line will start ringing to indicate an incoming call.
Short Code: *71*N#

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 Secondary Dial Tone

Telephone Number: N
Feature: RingBackWhenFree
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Secondary Dial Tone

Secondary dial tone is a system feature to generate a secondary dial tone after the user has
begun dialing an external number. This dial tone is then played until the number dialing and an
external trunk seized.
Pre-Release 4.0: Secondary dial tone is triggered through the use of the secondary dial tone short
code feature.
Release 4.0+: The use of this short code feature has been replaced by the Secondary Dial Tone
check box option on ARS forms.
Details
Telephone Number: Digit which triggers secondary dial tone.
Default Short Code: 9 (U-Law only)
Programmable Button Control:
Release: 1.0+.
Example
For pre-4.0 systems secondary dial tone works in two parts. The following system short code will
trigger secondary dial tone. To use it to trigger secondary dial tone and then continue dialing, other
user, user rights and system short codes should begin with [9].
Short Code: 9
Telephone Number: .
Feature: Secondary Dial Tone
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Set Absent Text

This feature can be used to select the user's current absence text. This text is then displayed to
internal callers who have suitable display phones or applications. It doesn't changes the users

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Short Code Features

status. The absence text message is limited to 128 characters. Note however that the amount
displayed will depend on the caller's device or application.
The text is displayed to callers even if the user has forwarded their calls or is using follow me.
Absence text is supported across a multi-site network.
Details
Telephone Number: The telephone number should take the format "y,n,text" where:
• y = 0 or 1 to turn this feature on or off.
• n = the number of the absent statement to use, see the list below:
0 = None. 4 = Meeting until. 8 = With cust. til.
1 = On vacation until. 5 = Please call. 9 = Back soon.
2 = Will be back. 6 = Don't disturb until. 10 = Back tomorrow.
3 = At lunch until. 7 = With visitors until. 11 = Custom.

text = any text to follow the absent statement.

Default Short Code:
Programmable Button Control: Absnt
Release: 1.0+.
Example
The following short code can be used to turn an absent text message on:
• Short Code: *88
• Telephone Number: "1,5,me on 208"
• Line Group ID: 0
• Feature: SetAbsentText
Example
The following short code could be used to turn this facility off. In the Telephone Number the first 0
is used to turn this facility off and the second 0 is used to select the absent statement "None".
Short Code: *89
Telephone Number: "0,0"
Line Group ID: 0
Feature: SetAbsentText
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 Set Account Code

Set Account Code

This short code feature is used to allow system users to enter a valid account code prior to making
a phone call. Once this short code is set up, any existing account code in the system configuration
can be used in conjunction with it.
This short code feature is essential for allowing analog phone users to enter account codes as
they cannot enter account code through the phone during a call or after dial a number.
Details
Telephone Number: A valid account code.
Default Short Code:
Programmable Button Control: Acct.
Release: 2.1+.
Example
In this example, N represents any valid account code. For the purpose of this example, we will
imagine the account code to be 1234. Once this short code is created, a user can dial 11*1234# to
get a dial tone for dialing the restricted telephone number or the phone number needing to be
tracked for billing purposes.
Short code: 11*N#
Telephone Number: N
Feature: SetAccountCode
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Set Authorization Code

This short code feature is only available on systems configured to use authorization codes. See
Authorization Codes. The feature is used to allow a user to enter a valid authorization code prior to
making a phone call.
This short code feature is essential for allowing analog phone users to enter authorization codes.
Note that the authorization code must be associated with the user or the user rights to which the
user belongs.
Details
Telephone Number: A valid authorization code.
Default Short Code:
Programmable Button Control:

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Short Code Features

Release: 3.2+.
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Set Fallback Twinning Off

This feature can be used by users to disable fallback twinning operation. This feature requires the
user to have a mobile twinning number set.
Fallback twinning redirects calls to the user’s configured mobile twinning number when the system
cannot detect a connection to the user’s normal registered extension. This feature can be used
without mobile twinning itself being enabled.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
Related links
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Set Fallback Twinning On

This feature can be used by users to enable fallback twinning operation. This feature requires the
user to have a mobile twinning number set.
Fallback twinning redirects calls to the user’s configured mobile twinning number when the system
cannot detect a connection to the user’s normal registered extension. This feature can be used
without mobile twinning itself being enabled.
Details
• Telephone Number:
• Default Short Code:
• Programmable Button Control:
Related links
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 Set Hunt Group Night Service

Set Hunt Group Night Service

This feature puts the specified hunt group into Night Service mode.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Details
Telephone Number: Hunt group extension number. If left blank, the short code will affect all
hunt groups of which the user is a member.
The Set Hunt Group Night Service and Clear Hunt Group Night Service short code and button
features can be used to switch an SSL VPN service off or on respectively. The service is indicated
by setting the service name as the telephone number or action data. Do not use quotation marks.
Default Short Code: *20*N#
Programmable Button Control: HGNS+
See also: Set Hunt Group Out Of Service, Clear Hunt Group Night Service, Clear Hunt Group Out
Of Service.
Release: 1.0+.
Example
This short code is a default within the system configuration. N represents the telephone number of
the hunt group to be placed into "Night Service" mode. For example, when *20*201# is dialed, the
hunt group associated with extension 201 will be placed into "Night Service" mode.
Short Code: *20*N#
Telephone Number: N
Feature: SetHuntGroupNightService
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Set Hunt Group Out Of Service

This feature manually puts the specified hunt group into Out of Service mode. If a time profile has
also been defined to control hunt group night service, the action may vary:
Set Hunt Group Out of Service can be used to override a time profile and change a hunt group
from night service to out of service.

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Short Code Features

Details
Telephone Number: Hunt group extension number. For Release 4.0+, if left blank, the short
code will affect all hunt groups of which the user is a member.
Default Short Code:
Programmable Button Control: HGOS+
Release: 1.0+.
Example
Below is a sample short code using the Set Hunt Group Out Of Service feature. N represents
the telephone number of the hunt group to be placed into "Out of Service" mode. For example,
when *56*201# is dialed, the hunt group associated with extension 201 will be placed into "Out of
Service" mode.
Short Code: *56*N#
Telephone Number: N
Feature: SetHuntGroupOutOfService
Related links
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Set Inside Call Seq

This feature allows the user to select the ringing used on their analog extension for internal calls.
The number entered corresponds to the ring pattern required. This is 0 for Default Ring, 1 for
RingNormal, 2 for RingType1, etc. For more information on selectable ringing patterns, see Ring
Tones. Use of this short code function is applicable to analog phone users only. The distinctive
ringing pattern used for other phones is set by the phone type.
Details
Telephone Number: Number corresponding to the desired ring pattern. See Ring Tones.
Default Short Code:
Programmable Button Control: ICSeq
See also: Set Ringback Seq, Set Inside Call Seq.
Release: 1.0+.
Example
This Short Code allows a user to change their inside call pattern. N represents the number
corresponding to the Call Sequence the user wishes to choose, the numbering starts at 0
selecting Default Ring, 1 selects Ring Normal, 2 selects RingType1, etc. For example, if a user

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 Set Mobile Twinning Number

wants to set her/his internal ring pattern to RingType1, the user would dial *80*2# because 2
corresponds to RingType1. This short code is useful for distinguishing an external call from an
internal one simply by the ring tone.
Short Code: *80*N#
Telephone Number: N
Feature: SetInsideCallSeq
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Set Mobile Twinning Number

This short code feature can be used to set a mobile twinning number. The destination can be any
external number the user is able to dial normally. It should include any prefix if necessary.
Details
Telephone Number: Twinning destination.
Default Short Code:
Programmable Button Control:
See also: Set Mobile Twinning On, Set Mobile Twinning Off, Mobile Twinned Call Pickup.
Release: 3.2+.
Related links
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Set Mobile Twinning On

This short code feature turns on the user's mobile twinning. It requires a mobile twinning number
to have been set for the user. That can be done through using the Set Mobile Twinning Number
short code feature or through the User | Twinning tab within Manager.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Set Mobile Twinning Off, Set Mobile Twinning Number, Mobile Twinned Call Pickup.
Release: 3.2+.

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Short Code Features

Related links
Short Code Features on page 809

Set Mobile Twinning Off

This short code feature turns off the user's mobile twinning.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
See also: Set Mobile Twinning On, Set Mobile Twinning Number, Mobile Twinned Call Pickup.
Release: 3.2+.
Related links
Short Code Features on page 809

Set No Answer Time

This short code feature allows the user to change their No Answer Time (User | Telephony | Call
Settings).
Details
Telephone Number: Time in seconds.
Default Short Code:
Programmable Button Control: NATim
See also: Set Wrap Up Time.
Release: 1.0+.
Example
This short code allows a user to change the length of time they have to answer the phone before it
goes to divert or voicemail. N represents the number of seconds. For example, if a user wants to
set the no answer time to 15 seconds, the following information needs to be entered: *81*15#.
Short Code: *81*N#
Telephone Number: N
Feature: SetNoAnswerTime

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 Set Outside Call Seq

Related links
Short Code Features on page 809

Set Outside Call Seq

This feature allows the user to select the ringing used on their analog extension for external calls.
The number entered corresponds to the ring pattern required. This is 0 for Default Ring, 1 for
RingNormal, 2 for RingType1, etc. For more information on selectable ringing patterns, see Ring
Tones. Use of this short code function is applicable to analog phone users only. The distinctive
ringing pattern used for other phones is set by the phone type.
Details
Telephone Number: Number corresponding to the desired ring pattern. See Ring Tones.
Default Short Code:
Programmable Button Control: OCSeq
See also: Set Ringback Seq, Set Outside Call Seq.
Release: 1.0+.
Example
This short code allows a user to change the ringing tone for an external call. N represents the
number corresponding to the Call Sequence the user wishes to choose, the numbering starts at 0
selecting Default Ring, 1 selects RingNormal, 2 selects RingType1, etc. For example, if a user
wants to set her/his ring pattern for external calls to RingType1, the user would dial *81*2#
because 2 corresponds to RingType1. This short code is useful for distinguishing an external call
from an internal one simply by the ring tone.
Short Code: *81*N#
Telephone Number: N
Feature: SetOutsideCallSeq
Related links
Short Code Features on page 809

Set Ringback Seq

This feature allows the user to select the ringing used on their analog extension for ringback calls.
The number entered corresponds to the ring pattern required. This is 0 for Default Ring, 1 for
RingNormal, 2 for RingType1, etc. For more information on selectable ringing patterns, see Ring

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Short Code Features

Tones. Use of this short code function is applicable to analog phone users only. The distinctive
ringing pattern used for other phones is set by the phone type.
Telephone Number: Number corresponding to the desired ring pattern. See Ring Tones.
Default Short Code:
Programmable Button Control: RBSeq
See also: Set Outside Call Seq, Set Inside Call Seq.
Example
This short code allows a user to change the ringing tone for a ringback call. N represents the
number corresponding to the ring tone the user wishes to choose, the numbering starts at 0
selecting Default Ring, 1 selects RingNormal, 2 selects RingType1, etc. For example, if a user
wants to set her/his ring pattern for ringback calls to RingType1, the user would dial *81*2#
because 2 corresponds to RingType1. This short code is useful for distinguishing a ringback call
from any other call simply by the ring tone.
Short Code: *81*N#
Telephone Number: N
Feature: SetRingbackSeq
Related links
Short Code Features on page 809

Set Time Profile

You can manually override a time profile. The override settings allow you to mix timed and manual
settings.
Five short codes can be configured.
Short Code Name Description
Set Time Profile Timed Operation No override. The time profile operates as
configured.
Set Time Profile Active Until Next Timed Inactive Use for time profiles with multiple intervals. Select
to make the current timed interval active until the
next inactive interval.
Set Time Profile Inactive Until Next Timed Active Use for time profiles with multiple intervals. Select
to make the current active timed interval inactive
until the next active interval.
Set Time Profile Latch Active Set the time profile to active. Timed inactive periods
are overridden and remain active.
Set Time Profile Latch Inactive Set the time profile to inactive. Timed active periods
are overridden and remain active.

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 Set Wrap Up Time

Active
Inactive
7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Active until next Inactive

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Inactive untill next Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Inactive

Telephone Number: Time profile name.

Default Short Code: No.
Programmable Button Control: Yes: Time Profile
Related links
Short Code Features on page 809

Set Wrap Up Time

Allows users to change their Wrap-up Time (User | Telephony | Call Settings) setting.
Other phones or applications monitoring the user's status will indicate the user as still being busy
(on a call).
Hunt group calls will not be presented to the user.
If the user is using a single line set, direct calls also receive busy treatment. If the user is using a
mutli-line set (multiple call appearances), direct calls to them will ring as normal.

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Short Code Features

It is recommended that this option is not set to less than the default of 2 seconds. 0 is used to
allow immediate ringing.
For users set as an CCR Agent, the After Call Work Time (User | Telephony | Supervisor Settings)
setting should be used.
Details
Telephone Number: Time in seconds.
Default Short Code:
Programmable Button Control: WUTim
See also: Set No Answer Time.
Release: 1.0+.
Example
N represents the number of seconds. For example, if a user wants to set her/his wrap up time to 8
seconds, this user would dial *82*5#. This short code is useful in a "call center" environment
where users may need time to log call details before taking the next call. If set to 0 the user does
not receive any calls. It is recommended that this option is not set to less than the default of 2
seconds.
Short Code: *82*N#
Telephone Number: N
Feature: SetWrapUpTime
Related links
Short Code Features on page 809

Speed Dial
Each system directory and personal directory number stored in the configuration can be optionally
assigned an index number. That index number can then be used by M-Series and T-Series phone
users to dial the directory number. This short code feature allows the creation of short codes to
perform the same function. However, the short code is diallable from any type of telephone
extension on the system.
For example:
• If Feature 0 is followed by a 3-digit index number in the range 000 to 999, the system
directory record with the matching index number is dialed.
• If Feature 0 is followed by * and a 2-digit index number in the range 00 to 99, the personal
directory record with the matching index number is dialed. Alternatively Feature 0 can be
followed by 00# to 99#. Note: Release 10.0 allows users to have up to 250 personal directory
entries. However, only 100 of those can be assigned index numbers.

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 Shutdown Embedded Voicemail

Details
Telephone Number: System directory entry index number (000 to 999) or personal directory
entry index number (00 to 99).
Default Short Code:
Programmable Button Control:
Release: 8.1.
Example
Using the example below, a user is able to dial *0 and then either a 2 digit code for an indexed
personal directory entry or a 3 digit code for an indexed system directory entry.
Short Code: *0N#
Telephone Number: N
Feature: Speed Dial
Related links
Short Code Features on page 809

Shutdown Embedded Voicemail

Allows the Embedded Voicemail service provided by an Avaya memory card in a control unit to be
shut down. To restart the service, a Startup Embedded Voicemail short code should be used.
The short code has the following effects:
1. Immediately disconnect all current users within Embedded Voicemail. This is not a polite
shutdown.
2. Mark the Embedded Voicemail as inactive so that it will not receive any new calls.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 4.0+ (Added in the Release 4.0 Q2 2007 maintenance release).
Related links
Short Code Features on page 809

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Short Code Features

Stamp Log
The stamp log function is used to insert a line into any System Monitor trace that is running. The
line in the trace indicates the date, time, user name and extension plus additional information. The
line is prefixed with LSTMP: Log Stamped and a log stamp number. When invoked from a Avaya
phone with a display, Log Stamped# is also briefly displayed on the phone. This allows users to
indicate when they have experienced a particular problem that the system maintainer want them
to report and allows the maintainer to more easily locate the relevant section in the monitor trace.
The log stamp number is set to 000 when the system is restarted. The number is then
incremented after each time the function is used in a cycle between 000 and 999. Alternately if
required, a specific stamp number can be assigned to the button or short code being used for the
feature.
Details
Telephone Number: Optional. If not set, a number in the sequence 000 to 999 is automatically
used. If set, the number set is used.
Default Short Code: *55
Programmable Button Control: Stamp Log
Release: 8.1+
Related links
Short Code Features on page 809

Startup Embedded Voicemail

Restarts the Embedded Voicemail service provided by an Avaya Memory in a control unit.
Details
Telephone Number:
Default Short Code:
Programmable Button Control:
Release: 6.0+
Related links
Short Code Features on page 809

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 Suspend Call

Suspend Call
This feature uses the Q.931 Suspend facility. It suspends the incoming call at the ISDN exchange,
freeing up the ISDN B channel. The call is placed in exchange slot 0 if a slot number is not
specified.
Details
Telephone Number: Exchange slot number or blank (slot 0).
Default Short Code:
Programmable Button Control: Suspe
See also: Resume Call.
Release: 1.0+.
Related links
Short Code Features on page 809

Suspend CW
This feature uses the Q.931 Suspend facility. Suspends the incoming call at the ISDN exchange
and answer the call waiting. The call is placed in exchange slot 0 if a slot number is not specified.
Only available when supported by the ISDN exchange.
Details
Telephone Number: Exchange slot number or blank (slot 0).
Default Short Code: *28*N# (A-Law only) (not on Server Edition)
Programmable Button Control: SusCW
See also: Resume Call.
Release: 1.0+.
Example
Sample short code using the Suspend CW feature.
Short Code: *28*N#
Feature: Suspend CW
Related links
Short Code Features on page 809

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Short Code Features

Start After Call Work

This feature can be users who have been configured as CCR agents. It allows them to dial a short
code to enter the After Call Work (ACW) state as reported by the Customer Call Reporter (CCR)
application.
Details
Telephone Number:
Default Short Code:
Programmable Button Control: ACWrk
See also: Clear After Call Work.
Release: 4.2 4Q 2008 Maintenance release+.
Related links
Short Code Features on page 809

Toggle Calls
This feature cycles through each call that the user has on hold on the system. This feature is
useful when a user with a single-line telephone has several calls on hold and needs to respond to
each one in turn.
Details
Telephone Number:
Default Short Code: *29
Programmable Button Control: Toggl
Release: 1.0+.
Example
Below is sample short code using the Toggle Calls feature.
Short Code: *29
Feature: ToggleCalls
Related links
Short Code Features on page 809

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 Unpark Call

Unpark Call
Retrieve a parked call from a specified system park slot.
Details
Telephone Number: System park slot number.
Default Short Code: *38*N#
Programmable Button Control: Ride
See also: Call Park.
Release: 1.0+.
Example
Below is a sample short code using the Unpark Call feature. N represents the park slot number in
which the call you want to retrieve was parked. For example, if a user parked a call to slot number
9, you can retrieve that call by dialing *38*9#.
Short Code: *38*N#
Telephone Number: N
Feature: Unpark Call
Related links
Short Code Features on page 809

Voicemail Collect
This feature connects to the voicemail system. Normally the telephone number field is used to
indicate the name of the mailbox to be accessed, for example "?Extn201" or "#Extn201".
? indicates 'collect messages'.
# indicates 'leave a message'. It also instructs the voicemail server to give a brief period of ringing
before connecting the caller. This is useful if the short code is used for functions like call transfers
as otherwise the voicemail server can start playing prompts before the transfer is completed.
However, the # can be omitted for immediate connection if required.
" " quotation marks must be used to enclose any information that needs to be sent to the voicemail
server as is. Any text not enclosed by quote marks is checked by the telephone system for short
code character matches which will be replaced before being sent to the voicemail server.
Manager will automatically add quotation marks to the Telephone Number field if there are no
manually added quotation marks. Care should be taken to ensure that special characters that you
want replaced by the telephone system, such as U, N or X, are not enclosed by the quotation

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Short Code Features

marks. For scenarios where the telephone number only contains short code characters, an empty
pair of quotation marks, for example ""N.
When using Voicemail Pro, names of specific call flow start points can directly access those start
points via a short code. In these cases, ? is not used and # is only used if ringing is required
before the start point's call flow begins.
Short codes using the Voicemail Collect feature, with either "Short Codes.name" and "#Short
Codes.name" records in the Telephone Number field are automatically converted to the Voicemail
Node feature and name.
Note:
CallPilot voicemail is used for IP Office Branch deployments with CS 1000.
Users can access their CallPilot voicemail by dialing the Voicemail Collect short code. Access
to CallPilot voicemail from Auto Attendant cannot be enabled by setting a Normal Transfer
action to point to the Voicemail Collect short code. If desired, it can be enabled by setting a
Normal Transfer action to point to the CallPilot number.
Details
Telephone Number: See the notes above.
Default Short Code: *17
Programmable Button Control: VMCol
See also: Voicemail On, Voicemail Off, Voicemail Node.
Release: 1.0+.
Example: Retrieve Messages from Specific Mailbox
This short code allows a user to retrieve messages from the mailbox of the hunt group 'Sales'.
This usage is not supported on Voicemail Pro running in Intuity emulation mode unless a custom
call flow has been created for the hunt group, refer to the Voicemail Pro help.
Short Code: *89
Telephone Number: "?Sales"
Feature: VoicemailCollect
Example: Record Message to Specific Mailbox
To allow users to deposit a message directly to Extn201's Voicemail box. This short code is useful
when you know the person is not at her/his desk and you want to immediately leave a message
rather than call the person and wait to be redirected to voicemail.
Short Code: *201
Telephone Number: "#Extn201"
Feature: VoicemailCollect
Example: Accessing a Specific Voicemail Pro Module
This short code can be used in instances where you have a conference bridge set up on the
system and a module has been created via Voicemail Pro to access this conference bridge. A

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 Voicemail Collect

short code can be created for internal access to the module. In the sample short code below, the
telephone number field contains the name of the module. In this example, if a short burst of ringing
is required before connecting the module, "#conferenc" would be used as the telephone number.
Short Code: *100
Telephone Number: "conferenc"
Feature: VoicemailCollect
Example: Record Voicemail Pro Messages for Outbound Contact Express
Short Code: *99
Telephone Number: "edit_messages"
Feature: VoicemailCollect
This short code allows users to record Voicemail Pro messages used by the Outbound Contact
Express solution. For example:
• Queuing messages.
• A message intended for an answering machine.
• Messages an agent can play for a customer.
• On hold messages. (Always specify message number “0” for the hold treatment message.)
• Messages played by a Virtual Agent.
Recorded message files are stored in the folder /opt/vmpro/Wavs/Modules/CPAPrompts.
When invoked, the user is prompted to enter a number to associate with the message. The
Outbound Contact Express Proactive Contact component ships with the following default English
messages:
• 0: Hold message
• 1: First outbound queue message – Female
• 2: Second outbound queue message - Female
• 3: Third outbound queue message - Female
• 4: Fourth outbound queue message – Female
• 9: First outbound queue message – Male
• 10: Second outbound queue message - Male
• 11: Third outbound queue message - Male
• 12: Fourth outbound queue message - Male
• 17: Message to play to an answering machine or message to play by a virtual agent - Female
• 18: Message to play to an answering machine or message to play by a virtual agent – Male
• 19: Message played when default F6 agent key is pressed (Release the Line, completion
Code 20)
Related links
Short Code Features on page 809

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Short Code Features

Voicemail Node
Similar to Voicemail Collect but used for calls being directed to a Voicemail Pro Short Codes start
point. Useful if you have set up a short code start point with Voicemail Pro and want to give direct
internal access to it.
Details
Telephone Number: Voicemail Pro Short Code start point name without quotation marks.
Default Short Code:
Programmable Button Control:
See also: Voicemail Collect.
Release: 2.0+.
Example
Having created a short codes start point call flow called Sales, the following system short code
can be used to route calls to that call flow:
• Short Code: *96
• Telephone Number: Sales
• Feature: VoicemailNode
Related links
Short Code Features on page 809

Voicemail On
This feature enables the user's voicemail mailbox to answer calls which ring unanswered or arrive
when the user is busy.
Details
Telephone Number: None.
Default Short Code: *18
Programmable Button Control: VMOn
See also: Voicemail Off.
Release: 1.0+.
Example
This short code can be used to toggle the feature on.
Short Code: *18

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 Voicemail Off

Feature: VoicemailOn
Related links
Short Code Features on page 809

Voicemail Off
This feature disables the user's voicemail mailbox box from being used to answer calls. It does not
disable the voicemail mailbox being used as the target for other functions such as call recording or
messages forwarded from other mailboxes.
Details
Telephone Number: None.
Default Short Code: *19
Programmable Button Control: VMOff
See also: Voicemail On.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *19
Feature: VoicemailOff
Related links
Short Code Features on page 809

Voicemail Ringback On
This feature enables voicemail ringback to the user's extension. Voicemail ringback is used to call
the user when they have new voicemail messages. The ringback takes place each time the
extension is used. This feature is useful for users who do not have voicemail light/button indicators
on their telephone.
If the user has been configured to receive message waiting indication for any hunt groups, a
separate voicemail ringback will occur for each such group and for the users own mailbox.
Details
Telephone Number:
Default Short Code: *48

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Short Code Features

Programmable Button Control: VMRB+

See also: Voicemail Ringback Off.
Release: 1.0+. For Release 3.2, the Voicemail On and Voicemail Ringback On short code features
toggled. For Release 4.0 and higher, they no longer toggle.
Example
This short code can be used to turn the feature on.
Short Code: *48
Feature: VoicemailRingbackOn
Related links
Short Code Features on page 809

Voicemail Ringback Off

This feature disables voicemail ringback to the user's extension.
Details
Telephone Number:
Default Short Code: *49
Programmable Button Control: VMRB-
See also: Voicemail Ringback On.
Release: 1.0+.
Example
Below is a sample of the short code setup.
Short Code: *49
Feature: VoicemailRingbackOff
Related links
Short Code Features on page 809

Whisper Page
This feature allows you to intrude on another user and be heard by them without being able to
hear the user's existing call which is not interrupted. For example: User A is on a call with user B.

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 Whisper Page

When user C intrudes on user A, they can be heard by user A but not by user B who can still hear
user A. Whisper page can be used to talk to a user who has enabled private call.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
The system support a range of other call intrusion methods in addition to this feature.
Details
Telephone Number: Target extension number.
Default Short Code:
Programmable Button Control: Whisp.
See also: Call Intrude, Call Listen, Coaching Intrusion, Dial Inclusion.
Release: 8.0+.
Related links
Short Code Features on page 809

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Chapter 19: Button Programming Overview

This section provides an overview of system actions that can be assigned to programmable buttons
on Avaya phones.
Button assignment can be done through the system configuration using Manager and for some
functions using the phone itself. Using Manager, if only button programming changes are required,
the configuration changes can be merged back to the system without requiring a reboot.
Appearance Functions The functions Call Appearance, Bridged Appearance, Coverage and
Line Appearance are collectively known as "appearance functions". For full details of their
operation and usage refer to the Appearance Button Operation section. The following restrictions
must be observed for the correct operation of phones.
Phone Support Note that not all functions are supported on all phones with programmable buttons.
Where possible exceptions, have been indicated. Those buttons will typically play an error tone
when used on that phone. Programming of those features however is not restricted as users may
hot desk between different types of phones, including some where the feature is supported.
Actions that use status feedback are only supported on buttons that provide that feedback through
lamps or icons.
Related links
Programming Buttons with Manager on page 902
Programming Button via the Menu Key on page 904
Programming Button via an Admin Button on page 906
BST Button Programming on page 908
T3 Self-Administration on page 910
Interactive Button Menus on page 912
Label Templates on page 912

Programming Buttons with Manager

About this task
Procedure
1.
Select the User required to display their configuration details.
2. Select Button Programming.

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 Programming Buttons with Manager

The number of button displayed is based on the phone associated with the user when the
configuration was loaded. This can be overridden by selecting Display All Buttons. This
may be necessary for users who switch between different phones using hot desking or
have an expansion unit attached to their phone.
3. For the required button, either select the button and then click Editor double-click the
button.
4. Edit the settings as required.
Use the ... button to display the menu for selecting the required button action. Select the
action and set the action data, then click OK.

5. Click OK.
Repeat for any other buttons.
6. Click OK.
Result
An alternate method for the above programming is to right-click on the various fields. To do this
start with the Action field, then Action Data and then Label if required.
Related links
Button Programming Overview on page 902

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Button Programming Overview

Programming Button via the Menu Key

On 4412D+, 4424D+, 4612IP, 4624IP, 6408D, 6416D, 6424D phones the Menu button can be
used to program some functions against other buttons. This programming also includes
programmable buttons on any associated add-on units associated with the phone. Buttons already
programmed as appearance buttons cannot be altered using these methods.
A Self-Administer button can also be added to allow the phone user to program the functions on
their other buttons, see Self-Administer.
Related links
Button Programming Overview on page 902

Setting a Button to Dial a Number

About this task
This process sets the selected programmable button to the Dial function in the system
configuration.
Procedure
1. With the phone idle and on-hook, press MENU .
2. Press and select PROG.
3. Enter the number required.
The left-most display button can be used to backspace and the right-most display button
can be used to Clear the whole number.
4. Press the programmable button against which the number should be set.
5. If the button is already programmed, options to replace (Repla), keep (Keep) or delete
(Delet) the buttons existing programming appear.
Select the option required.
6. The message BUTTON PROGRAMMED! indicates that the button is now programmed.

Select Cont and then press Exit .

Setting a Button to a Switch Function

About this task
This process allows users to program there own Group, User, and Park slot monitor buttons. It
also allows the programming of Dial and Flash Hook buttons.

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 Programming Button via the Menu Key

Procedure
1. With the phone idle and on-hook, press Menu twice.
2. Press and select ProgA.
3. Press and select DSS.
4. Use the and buttons to display the function required. Press the display button below the
function to select it.
5. If the function requires a telephone number value set, enter the number.
The left-most display button can be used to backspace and the right-most display button
can be used to Clear the whole number.
6. Press the programmable button against which the number should be set.
7. If the button is already programmed, options to replace (Repla), keep (Keep) or delete
(Delet) the buttons existing programming appear.
Select the option required.
8. The message BUTTON PROGRAMMED! indicates that the button is now programmed.

Select Cont and then press Exit .

Setting Buttons to Admin Function

About this task
Phones with a Menu key can program a range of self-administer functions onto their
programmable buttons. These are:
Dir - Directory.
Drop - Drop.
HFAns - Internal Auto-Answer.
Timer - Timer.
AutCB - Automatic Callback.
Prog - Abbreviated Dial Program.
CFrwd - Call Forwarding All.
CPark - Call Park.
SAC - Send All Calls.
TmDay - Time of Day.
Admin - Self-Administer.
Acct - Account Code Entry.
AD - Abbreviated Dial.

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Button Programming Overview

Call Park
GrpPg - Group Paging.
CPkUp - Call Pickup.
DPkUp - Directed Call Pickup.
RngOf - Ringer Off.
Spres - AD Suppress.
HdSet - Headset Toggle.
HGNS+ - Set Hunt Group Night Service.
This is the same set of functions that can be programmed by users with a button set to Self-
Administer (see Self-Administer).
Procedure
1. With the phone idle and on-hook, press Menu .
2. Press twice and select Admin.
3. Use the and keys to display the function required and then select it by pressing the
display button below the feature.
Selecting Expl? changes the display from short name mode to long name mode. In this
mode the full names of the features are displayed. Select SHORTMODE to return to that
mode.
4. If the function requires a telephone number value set, enter the number.
The left-most display button can be used to backspace and the right-most display button
can be used to Clear the whole number.
5. Press the programmable button against which the number should be set.
6. If the button is already programmed, options to replace (Repla), keep (Keep) or delete
(Delet) the buttons existing programming appear.
Select the option required.
7. The message BUTTON PROGRAMMED! indicates that the button is now programmed.

Select Cont and then press Exit .

Programming Button via an Admin Button

The Admin (also called Self-Administer) function can be assigned to a programmable button on a
user's phone. That button then allows the user to program functions against other programmable
buttons on their phone, except those already set as appearance buttons.

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 Programming Button via an Admin Button

Admin buttons are only supported on 2410, 2420, 4406D+, 4412D+, 4424D+, 4606IP, 4612IP,
4624IP, 5410, 5420, 6408D, 6416D and 6424D.
On 4412D+, 4424D+, 4612IP, 4624IP, 6408D, 6416D, 6424D phones:
• Admin can be permanently accessed via Menu , , , Admin.
• Admin1 can be permanently accessed via Menu , Menu , , ProgA, , , DSS.
Related links
Button Programming Overview on page 902

Using an Admin Button

About this task
Procedure
1. With the phone idle and on-hook, press the button programmed to Admin or Admin1.
The list of available functions is shown.
2. Use the and buttons to move through the list.
Selecting Expl? changes the display from short name mode to long name mode. In this
mode the full names of the features are displayed. Select SHORTMODE to return to that
mode.
3. Select the function required.
4. If the function requires a telephone number value set, enter the number.
The left-most display button can be used to backspace and the right-most display button
can be used to Clear the whole number.
5. Press the programmable button against which the number should be set.
On phones with multiple pages of buttons use the and button to select the required
page before pressing the button to program.
6. If the button is already programmed, options to replace, keep or delete the button's existing
programming appear.
Select the option required.
7. The message BUTTON PROGRAMMED! indicates that the button is now programmed.
8. Select Cont. and then press Exitor lift the handset to go off-hook.

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Button Programming Overview

BST Button Programming

About this task
The process below can be used to assign functions to programmable buttons on T-Series and M-
Series phones. Existing button can be overwritten except buttons set to appearance functions.
Procedure
1. Press Feature *3.
If a security code is requested, enter your phone login code and press #.
2. Use one of the processes below.
Press * to switch between processes (or More if displayed). On T7000 phones, only the
first process is supported.
3. Select Button and then Function
a. Press the button to program.
b. Enter the feature code of the function required (the only * function supported is *7 for
contrast).
c. If the button has an existing function it is displayed and the option to replace the
button or return to function selection.
4. Select Function and then Button.
Enter the number for the feature required or use the volume buttons to move through the
list of functions.
01. Speed dial
02. Ring Again
03. Conference
04. Call Forward All
05. Last Number Redial
06. Page Group
07. Voicemail
08. Automatic Intercom
09. Priority Call
10. Transfer
11. Call Park
12. Group Pickup
13. Direct Pickup
14. Timer

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 BST Button Programming

15. Do Not Disturb On

16. Contrast
17. Group Listen On
18. Time of Day
17. Call Log
18. Self-Administer
19. Account Code
20. Forward on Busy
21. Forward on No Answer
22. Pickup
23. Directory
24. Flash Hook
25. Internal Auto Answer
26. Set Hunt Group Night Service
27. Twinning
28. Ringer Off
a. Press Holdto select a currently displayed function.
b. Press the button to which the function should be assigned.
c. If the button has an existing function it is displayed and the option to replace the
button or return to function selection.
5. When Default Buttons is displayed, press Hold(or the Prog softkey if displayed).
The phone buttons are defaulted to those appropriate to the phone type. Note that only
buttons that have a default function on the phone type are defaulted. It does not affect the
functions assigned to any buttons that do not have default functions.
Default Buttons
For T-Series and M-Series phones, default button functions are assigned to buttons when
a phone is first connected to the extension port. The functions assigned depend on the
particular phone model.
The default functions for the phone model are also assigned when Feature *3 is used to
default the phone's buttons. Buttons without a default function are not overwritten when the
buttons are defaulted.
Related links
Button Programming Overview on page 902

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Button Programming Overview

T3 Self-Administration
Note:
IP Office R11 does not support T3 and T3 IP Phones.
Release 4.2+ supports functions for T3 phone users to be able to program their own buttons. This
is similar to the existing Self-Administer button supported on other phones but is configured and
accessed via different methods.
The user accesses button programming through Menu | Settings | Button programming. This
function is not available by default, instead it must be configured as available for the user using
the method detailed below.
Once enabled, the user is able to configure the following functions on buttons:
Function Description
empty Returns the button to it normal default function.
Account Code Allows the user to enter an account code before or
during a call. The account code can be preset or
entered after the button press. See the Account
Code Entry function.
Callback Set a callback from the currently dialed extension
number. See the Automatic Callback function.
Call list Displays a list of calls received. See the Call List
function.
Call Tracing Activate malicious call tracing. See the MCID
Activate function and Malicious Call Tracing (MCID).
Dial Dial a preset number or partial number that can be
completed after the button press. See the Dial
function.
Dial Intercom Make a page call to the selected target if it supports
handsfree answer. See Dial Intercom.
Directory Display the system directory. See the Directory
function.
Do not disturb Toggle the phone between do not distrub on and off.
See the Send All Calls function.
Follow me here Activate/cancel follow me here. See the Follow Me
Here function.
Forward unconditional Activate/cancel forward all calls. See the Forward
Unconditional On function.
Group Paging Page a group of phones. See the Group Paging
function.
Group Membership Enable/disable the user membership of a group or
all groups. See the Hunt Group Enable function.
Table continues…

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 T3 Self-Administration

Group State Change a hunt group's out of service status. See

the Set Hunt Group Out of Service function.
Headset Switch between handset and headset modes. See
the Headset Toggle function.
Internal Auto-Answer Auto connect internal calls after a single tone. See
the Internal Auto-answer function.
Login Access the menu for phone log in. See the Extn
Login function.
Logout Log out from the phone. See the Extn Logout
function.
Night Service Change a hunt group's night service status. See the
Set Hunt Group Night Service function.
Paging Page an extension or group. See the Dial Paging
function.
Pickup Answer a call alerting on the system. See the Call
Pickup function.
Pickup Member Answer a call alerting the hunt group of which the
user is a member. See the Call Pickup Members
function.
Twinning Switch mobile twinning on/off and set the twinning
destination. Also used to pull a call answered at the
twinned number back to the users primary
extension. See the Twinning function.
User Monitor the status of a user. Also used to call them
or to pickup calls alerting them. See the User
function.
Visual Voice Create a visual voice access button. See Visual
Voice.
Voicemail Equivalent to the Voicemail Collect function.
Voicemail on/off Switch the use of the user's mailbox to answer
unanswered calls calls on/off. See the Voicemail On
function.

The user will need to be made aware of which physical buttons can be programmed as this varies
between the different T3 phones. See T3 Compact, T3 Classic and T3 Comfort.
Configuring a T3 User for Button Programming
1. Using Manager, receive the configuration from the system.
2. Select the T3 user and then select Menu Programming.
3. Set the action for one of the menus to Self-Administer.
4. Send the configuration back to the system.
5. The user will now be able to access button programming from their phone via Menu |
Settings | Button programming.

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Button Programming Overview

Related links
Button Programming Overview on page 902

Interactive Button Menus

For certain functions, on display phones where a button has been configured without a specific
number, a menu for number entry is displayed. The menu includes a Dir option for selecting a
number from the directories held by the system.
Functions that use the interactive menu are:
Feature Directory lists... Feature Directory lists...
Automatic Intercom Users Follow Me Here Cancel Users
Acquire Call/Call Steal Users Follow Me Here Users
Call Forwarding All Users Follow Me To Users
Call Intrude Users Forward Number Users/Groups
Call Park To Other Users Forward Busy Number Users/Groups
Extension
Dial Inclusion Users Group Paging Users/Groups
Dial Intercom Users Leave Word Calling Users/Groups
Directed Call Pickup Users/Groups Priority Calling Users/Groups

User and Group buttons can be used to indicate the required user or hunt group only if those
buttons are on an associated button module. User and Group buttons on the users extension are
not accessible while the interactive button menu is being displayed.
For functions supported across a multi-site network, the directory will include remote users and
advertised hunt groups.
For M-Series and T-Series phone, the volume buttons are used to scroll through the list of
matching names. If this is done during a call or while a call is alerting, this will also adjust the call
or ring volume.
Related links
Button Programming Overview on page 902

Label Templates
The attached zip file below contains Word document templates for the paper programmable key
labels on various phones supported by the system. Two templates are provided, one for A4 size
paper, the other for US Letter sized paper.

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 Label Templates

DSS Key Label Template File (Microsoft Word .dot Files)

For ETR, M-Series, T-Series, 1400 and 1600 phones, a number of tools and perforated printable
labels are available. For further details visit http://support.avaya.com and search for information on
DESI. Alternatively visit http://www.desi.com.
Manager is able to pass user button information to a DESI application on the same PC. This
allows printing of labels using the label text set within the system configuration. Currently only
ETR, M-Series, T-Series, 1400 and 1600 phonesphones are supported by DESI.
Related links
Button Programming Overview on page 902

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Chapter 20: Button Programming Actions

The following sections provide details for each of the button actions supported by system. Note that
this does not include buttons on phones on a system running in Partner Edition mode.
For each the following details are listed:
• Action Indicates the selection path to the action from within the list of actions displayed in
Manager.
• Action Data Indicates the type of data required by the action. For some actions no data is
required while for others action data may be optional. The option to enter the data after
pressing the button is not available for all phones, see Interactive Button Menus.
• Default Label This is the default text label displayed on phones which provide a display area
next to programmable buttons. Alternate labels can be specified in the system configuration or
entered by the phone user (refer to the telephone user guide). Note that for buttons with action
data set, the action data may also be displayed as part of the default label. Depending on the
display capacity of the particular phone, either a short or long label is displayed.
• Toggles Indicates whether the action toggles between two states, typically on or off.
• Status Indication Indicates whether the button provides status indication relevant to the
feature if the button has status lamps or display. If the Status Indication is listed as Required
it indicates that the button action is only supported on programmable buttons that can provide
status indication.
• User Admin This item indicates that users with a Self-Administer button can assign the action
to other buttons themselves.
• Phone Support This is only a general indication of support or otherwise of an action by phones
within particular series. On phones with 3 or less programmable buttons those button can only
be used for the Call Appearance action. In addition some actions are only supported on
phones where the programmable buttons provide status indication and or a display for data
entry once the feature is invoked.
Table of Button Programming Actions
The following tables list the actions available for programmable buttons on system.

Login Code Required Some function may require the user to enter their log in code. This
typically applies when the action data is left blank for entry when the button is pressed.

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 General
Action Action Data Default Label
Dial Any number. Dial
Group "Group name" in quote marks. <Group name>
User "User name" in quote marks. <User name>

Appearance
Action Action Data Default Label
Appearance None. a=
Bridged Appearance User name and call appearance <user name><appearance label>
button number.
Coverage Appearance User name. <user name>
Line Appearance Line appearance ID. Line

Emulation
Action Action Data Short Label Long Label
Abbreviated Dial Any number. AD Abbreviate Dial
Abbreviated Dial Pause None. Pause –
Abbreviated Dial None. Prog –
Program
Abbreviated Dial Stop None. Stop –
Account Code Entry Account code or blank for Acct Account Code
entry when pressed.
ACD Agent Statistics None. Stats –
ACD Stroke Count None. Count –
AD Special Function None. Mark –
Mark
AD Special Function Wait None. Wait –
AD Special Functions None. Sfunc –
AD Suppress None. Spres Suppress Digits
Automatic Callback None. AutCB Auto Callback
Automatic Intercom User number or name. Iauto Auto Intercom
Call Forwarding All Any number or blank for CFrwd Call Forward All
entry when pressed.
Call Park Park slot ID CPark Call Park
(alphanumeric) or blank
for menu of slots in use.
Call Park To Other User number. RPark Call Park to Other
Extension
Table continues…

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Button Programming Actions

Action Action Data Short Label Long Label

Call Pickup None. CpkUp Call Pickup Any
Cancel Leave Word None. CnLWC –
Calling
Consult None. Cnslt –
Dial Intercom User number or name or Idial Auto Intercom
blank for entry when
pressed.
Directed Call Pickup User number or name or DpkUp Call Pickup
group number or name or
or blank for entry when
pressed..
Directory None. Dir –
Drop None. Drop Drop Call
Group Paging User or group number or GrpPg Page
name or blank for entry
when pressed.
Headset Toggle None or FF HdSet –
Inspect None. Inspt –
Internal Auto-Answer None. HFAns Auto Answer
Leave Word Calling None. LWC –
Manual Exclude None. Excl –
Priority Calling None. Pcall –
Ringer Off None. RngOf Ringer Off
Self-Administer Blank or 1 or 2 Admin Self Administer
Send All Calls None. SAC Send All Calls
Stored Number View None. BtnVu –
Time of Day None. TmDay –
Timer None. Timer –
Twinning None. Twinning Twinning
Visual Voice None. Voice Voice

Advanced
Action Action Data Category Short Label Long Label
Acquire Call User number or Call Acquir Acquire
blank for last call
transferred.
Table continues…

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 Action Action Data Category Short Label Long Label
Break Out System name or IP Dial BkOut Breakout
address or blank for
selection when
pressed.
Busy None. Busy Busy –
Busy On Held 0 (off) or 1 (on). Busy BusyH –
Call Intrude User number or Call Intru Call Intrude
blank for entry
when pressed.
Call List None. Call LIST –
Call Listen User number. Call Listn Listen
Call Log None. Call Call Log
Call Pickup Any None. Call PickA Pickup Any
Call Pickup Group Group number or Call PickG Pickup Group
name.
Call Pickup Group number or Call PickM Pickup Members
Members name.
Call Queue User number. Call Queue Queue
Call Record None. Call Recor Record
Call Screening None. Call CallScreen Call Screening
Call Steal User number or Call Steal –
blank for last call
transferred.
Call Waiting Off None. Call CWOff –
Call Waiting On None. Call CWOn –
Call Waiting None. Call CWSus –
Suspend
Cancel All None. Call FwdOf Call Forward Off
Forwarding
Cancel Ring Back None. Miscellaneous RBak- –
When Free
Channel Monitor Channel number. Call ChMon –
Clear Call None. Call Clear Clear
Clear CW None. Call ClrCW –
Clear Hunt Group Group number. Call HGNS- –
Night Service
Clear Hunt Group Group number. Call HNOS- –
Out Of Service
Table continues…

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Button Programming Actions

Action Action Data Category Short Label Long Label

Clear Quota "Service name" Call Quota –
within quote marks
or "" for all services.
Coaching Intrusion User number or Call Coach Coaching
name or blank for Intrusion
entry when
pressed.
Conference Invoke the Call Conf –
conference
process. (M and T-
Series phones only)
Conference Add None. Call Conf+ Conference Add
Conference Meet Conference name Call CnfMM Conf. Meet Me
Me or number.
Dial 3K1 Any number. Dial D3K1 Dial 3K1
Dial 56K Any number. Dial D56K Dial 56K
Dial 64K Any number. Dial D64K Dial 64K
Dial CW User number. Dial DCW Dial Call Waiting
Dial Direct User number or Dial Dirct Auto Intercom
name or blank for
entry when
pressed.
Dial Emergency Any number. Dial Emrgy Dial Emergency
Dial Inclusion User number or Dial Inclu Dial Inclusion
name or blank for
entry when
pressed.
Dial Paging User or group Dial Page Page
number or name or
blank for entry
when pressed.
Dial Physical Extn Extension port Dial PhyEx Dial Physical Extn
by Number Base Extension
number.
Dial Physical Extn Extension port ID Dial DialP Dial Extn by Id
by Id number. (Release
1.4+)
Dial Speech Any number. Dial DSpch Dial Speech
Dial V110 Any number. Dial DV110 Dial V110
Dial V120 Any number. Dial DV120 Dial V120
Dial Video Any number. Dial Dvide Dial Video
Table continues…

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 Action Action Data Category Short Label Long Label
Display Msg Command string. Dial Displ –
Do Not Disturb None Do Not Disturb NoAI No Auto Int Calls
Auto-Intercom Deny
Do Not Disturb Any number. Do Not Disturb DNDX+ –
Exception Add
Do Not Disturb Any number. Do Not Disturb DNDX- –
Exception Delete
Do Not Disturb Off None. Do Not Disturb DNDOf –
Do Not Disturb On None. Do Not Disturb DNDOn Do Not Disturb
Extn Login None. Extension Login Login
Extn Logout None. Extension Logof Logout
Flash Hook None. Miscellaneous Flash Flash Hook
Follow Me Here User number. Follow Me Here+ Follow Me Here
Follow Me Here User number or Follow Me Here- Follow Me Here-
Cancel blank for entry
when pressed.
Follow Me To User name or user Follow Me FolTo Follow Me To
number or blank for
entry when
pressed.
Forward Hunt None. Forward FwdH+ –
Group Calls On
Forward Hunt None. Forward FwdH- Fwd HG Calls
Group Calls Off
Forward Number Any number or Forward FwdNo Fwd Number
blank for entry
when pressed.
Forward On Busy Any number or Forward FwBNo Fwd Busy Number
Number blank for entry
when pressed.
Forward On Busy None. Forward FwBOf –
Off
Forward On Busy None. Forward FwBOn Fwd Busy
On
Forward On No None. Forward FwNOf –
Answer Off
Forward On No None. Forward FwNOn Fwd No Answer
Answer On
Forward None. Forward FwUOf –
Unconditional Off
Table continues…

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Button Programming Actions

Action Action Data Category Short Label Long Label

Forward None. Forward FwUOn Fwd
Unconditional On Unconditional
Group Listen On None. Extension GroupListenOn –
Hold Call ISDN Exchange Hold Hold –
slot number.
Hold CW None. Hold HoldCW –
Hold Music None. Hold Music Hold Music
Hunt Group Disable Group number or Hunt Group HGDis
name or blank for
all groups.
Hunt Group Enable Group number or Hunt Group HGEna HG Enable
name or blank for
all groups.
Last Number Redial Redial the last Call Again –
number dialed. (M
and T-Series
phones only)
MCID Activate None. Miscellaneous MCID Malicious Call
Monitor Analogue Line appearance Voicemail TrkMW Trunk MWI
Trunk MWI ID.
Off Hook Station None. Miscellaneous OHStn –
Pause Recording None. Call PauseRec Pause Recording
Priority Call User number or Call PCall Priority Call
name.
Private Call None. (Release Call PrivC Private Call
4.0+)
Relay Off 1 or 2. Relay Rely- –
Relay On 1 or 2. Relay Rely+ Relay On
Relay Pulse 1 or 2. Relay Relay Relay Pulse
Resume Call ISDN Exchange Call Resum –
slot number.
Retrieve Call ISDN Exchange Call Retriv –
slot number.
Ring Back When None. Miscellaneous RBak+ Auto Callback
Free
Set Absent Text String for selected Set Absnt Absence Text
message and
custom text.
Table continues…

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 Action Action Data Category Short Label Long Label
Set Account Code Blank or valid Set Acct Account Code
account code.
(Release 2.1+)
Set Hunt Group Group number. Set HGNS+ HG Night Service
Night Service
Set Hunt Group Out Group number. Set HGOS+ HG Out of Service
Of Service
Set Inside Call Seq Value 0 to 10. Set ICSeq –
Set Night Service Group number. Set SetNSG HG NS Group
Group (Release 4.2+)
Set No Answer Time in seconds Set NATim No Answer Time
Time (range 6 to 99999).
Set Outside Call Value 0 to 10. Set OCSeq –
Seq
Set Out of Service Group number. Set SetOOSG HG OS Group
Group (Release 4.2+)
Set Ringback Seq Value 0 to 10. Set RBSeq –
Set Wrap Up Time Time in seconds Set WUTim Wrap-up Time
(range 0 to 99999).
Speed Dial Initiate the speed Dial SpdDial –
dial selection
process. (M and T-
Series phones only)
Stamp Log None. Miscellaneous StmpL Stamp Log
Suspend Call ISDN Exchange Suspend Suspe –
slot number.
Suspend CW ISDN Exchange Suspend SusCW –
slot number.
Toggle Calls None. Call Toggl –
Transfer Initiate the call Call Xfer –
transfer process. (M
and T-Series
phones only)
Unpark Call Park slot ID Call Ride –
(alphanumeric).
Voicemail Collect See notes. Voicemail VMCol VMail Collect
Voicemail Off None. Voicemail VMOff –
Voicemail On None. Voicemail VMOn VMail On
Voicemail Ringback None. Voicemail VMRB- –
Off
Table continues…

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Button Programming Actions

Action Action Data Category Short Label Long Label

Voicemail Ringback None. Voicemail VMRB+ VMail Ringback
On
Whisper Page User number or Call Whisp Whisper Page
name or blank for
entry when
pressed.

Abbreviated Dial
This function allows quick dialing of a stored number.
Action: Emulation | Abbreviated Dial.
Action Data:
• Full Number The number is dialled.
• Partial Number The partial number is dialled and the user can then complete dialing the full
number.
Default Label: AD or Abbreviate Dial.
Toggles: No.
Status Indication: No.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1] [1]
1600 Series: Yes 3810: No 5600 Series: Yes 9600: No
[1] [1]

[1] Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602 models.

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 Abbreviated Dial Pause

Abbreviated Dial Pause

Not supported. Provided for CTI emulation only. Allows a user to enter a pause character when
programming an abbreviated dial.
Action: Emulation | Abbreviated Dial Pause.
Action Data: None.
Default Label: Pause.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models if detailed below.

Abbreviated Dial Program

Not supported. Provided for CTI emulation only. Allows a user to program abbreviated dialing
numbers against other programmable buttons. This function cannot be used to overwrite call
appearance buttons.
Action: Emulation | Abbreviated Dial Program.
Action Data: None.
Default Label: Prog.
Toggles: No.
Status Indication: No.

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Button Programming Actions

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series 4100 Series: No 6400 Series: No D100: No
1100 Series: No 2400 Series: No 4400 Series: No 7400 Series: No M-Series: No
1200 Series: No 3600 Series: No 4600 Series: No 9040: No T-Series: No
1400 Series: No 3700 Series: No 5400 Series: No 9500 Series: No T3/T3 IP Series:
No
1600 Series: No 3810: No 5600 Series: No 9600 Series: No

Abbreviated Dial Stop

Not supported. Provided for CTI emulation only. Allows a user to enter a stop character when
programming an abbreviated dial.
Action: Emulation | Abbreviated Dial Stop.
Action Data: None.
Default Label: Stop.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models if detailed below.

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 Account Code Entry

Account Code Entry

Enter an account code for a call. This button can be used before dialing a number or during a call.
Action: Emulation | Account Code Entry.
Action Data: Optional. If an code is set it must match an account code set in the account codes
list. If no account code is set, the phone display will request entry of a valid code. This option is
not supported on XX02 phones and the T7000 phone.
Default Label: Acct or Account Code.
Toggles: No.
Status Indication: No.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0 DT software.

ACD Agent Statistics

Not supported. Provided for CTI emulation only.
Action: Emulation | ACD Agent Statistics.
Action Data: None.
Default Label: Stats.
Toggles: No.
Status Indication: No.
User Admin: No.

February 2019 Administering Avaya IP Office™ Platform with Manager 925

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Button Programming Actions

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

ACD Stroke Count

Not supported. Provided for CTI emulation only.
Action: Emulation | ACD Stroke Count.
Action Data: None.
Default Label: Count.
Toggles: No.
Status Indication: No.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0 DT software.

February 2019 Administering Avaya IP Office™ Platform with Manager 926

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 Acquire Call

Acquire Call
See Call Steal.

AD Special Functions
Not supported. Provided for CTI emulation only. Allows a user to enter a special character (mark,
pause suppress, wait) when entering an abbreviated dial.
Action: Emulation | AD Special Functions.
Action Data: None.
Default Label: Sfunc.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

AD Special Function Mark

Not supported. Provided for CTI emulation only. Allows a user to enter a mark character when
programming abbreviated dial.
Action: Emulation | AD Special Function Mark.
Action Data: None.

February 2019 Administering Avaya IP Office™ Platform with Manager 927

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Button Programming Actions

Default Label: Mark.

Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

AD Special Function Wait

Not supported. Provided for CTI emulation only. Allows a user to enter a Wait for Dial Tone
character when programming an abbreviated dial.
Action: Emulation | AD Special Function Wait.
Action Data: None.
Default Label: Wait.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 928

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 AD Suppress

1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes

[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

No
1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

AD Suppress
Suppresses the display of dialed digits on the telephone display. Dialed digits are replaced with an
s character.
Action: Emulation | AD Suppress.
Action Data: None.
Default Label: Spres or Suppress Digits.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 929

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Button Programming Actions

1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

After Call Work

This button is used by users configured as a Customer Call Reporter (CCR) Agent (User |
Telephony | Supervisor Settings) and working with the CCR application. It shows the CCR
agent their current After Call Work (ACW) status and allow them to manually change status. While
in ACW state, the agent will not receive hunt group calls.
CCR Agents can be automatically put into and taken out of ACW by the system if the user is
configured for Automatic After Call Work (User | Telephony | Supervisor Settings). Those users
must have an After Call Work button.
Action: Advanced | Miscellaneous | After Call Work
Action Data: None.
Default Label: ACWrk or After Call Work.
Toggles: Yes.
Status Indication: Yes. Required.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: No 6400 Series: No D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: No 7400 Series: No M-Series: Yes
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 930

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 Appearance

1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: No T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1] [1]
1600 Series: Yes 3810: No 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602.

Appearance
Creates a call appearance button. This can be used to answer and make calls. Users with multiple
call appearance buttons can handle multiple calls.
Call appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Appearance buttons can be set with a ring delay if required or to not ring. This does not affect the
visual alerting displayed next to the button. The delay uses the user's Ring Delay (User |
Telephony | Multi-line Options) setting.
Details
Action: Appearance | Appearance.
Action Data: Optional text label.
Default Label: a=.
Toggles: No.
Status Indication: Yes, required. See “Call Appearance Button Indication” in Administering IP
Office with Manager.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: Yes 6400 Series: Yes D100: No
[1]
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: Yes M-Series: Yes [1]
[1]
1200 Series: No 3600 Series: Yes 5400 Series: Yes 9040: Yes T-Series: Yes [1]
1400 Series: Yes 3700 Series: No 4600 Series: Yes 9500 Series: Yes
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 931

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Button Programming Actions

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes

1. 4100 Series and 7400 Series phones support virtual call appearance button operation.
This also applies to T7000, T7100, M7100 and M7100N phones and the Audio
Conferencing Unit (ACU).
Virtual Call Appearances
4100 Series and 7400 Series phones support virtual call appearance button operation. This also
applies to T7000, T7100, M7100 and M7100N phones and the Audio Conferencing Unit (ACU).
Virtual call appearance operation is similar to an analog phone with call waiting enabled. However,
it does not use the call waiting on/off settings, instead it uses call appearance buttons.
The number of virtual call appearances is set by the call appearance buttons programmed in the
user's settings. These must be programmed as a single block start from button 1. It is
recommended that only a maximum of 3 call appearances are used, however the user must have
at least 1 call appearance programmed in order to make and receive calls.
Virtual Call Appearance Usability
If the user goes off-hook, they are connected to the alerting call if any, else to dial tone in order to
make an outgoing call. This uses one of their virtual call appearance buttons.
With a call connected:
• If another call arrives on another virtual call appearance, the user will hear a call waiting tone
on the set. The display, if the phone has one, will switch between details of the current and
the waiting caller.
• If the user presses Hold, the connected call is placed on hold and:
If there are any available virtual call appearances, dial tone is heard. This allows the user to make
a call or to use short codes that may affect the held or waiting calls. The following are some of the
default short codes that can be used:
• *26: Clear CW Drop the previous call and answer the waiting call.
• *52: Clear Call Drop the previous call.
• *47: Conference Add Start a conference between the user and any held calls.
• Else, if there is a call waiting, that call is answered.
• Else, if there is a call on hold, that call is reconnected.
If the user presses Release or Drop or goes on-hook during a call, the current call is ended and
the user's phone returns to idle. If there is a waiting call, it starts ringing. The user can answer the
call by going off hook or pressing Hold.
With the phone idle:
If the user goes off hook:
• The first alerting call appearance is answered if any.
• Else, the first idle call appearance is seized and the user hears dial tone.
• The user can press Holdto switch between virtual call appearances. This will answer or
retrieve any call on next virtual call appearance or else hear dial tone to make a call.
With the phone idle but a call alerting:

February 2019 Administering Avaya IP Office™ Platform with Manager 932

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 Automatic Callback

Going off-hook or pressing Hold will answer the call.

When all the users virtual call appearances are in use, they are busy to any further calls. Calls will
follow forward on busy if set, else go to voicemail is available or else get busy indication.
The only other appearance button controls applied and supported are
Reserve Last CA This setting can be enabled for the extension user. When selected, the last
available call appearance is reserved for outgoing calls only. For example, for a user with 3 call
appearances, they return busy to any further calls when 2 virtual appearances are in use. The
extension user can press hold to get dial tone on the reserved call appearance. An available call
appearance is also required when using Feature 70 to initiate a call transfer.
Coverage Appearances Other users can have Coverage Appearance buttons set to provided
coverage to the virtual call appearance user. The virtual appearance users Individual Coverage
Time setting is applied.

Automatic Callback
Sets a ringback on the extension being called. When the target extension ends its current call, the
ringback user is rung (for their set No Answer Time) and if they answer, a new call is made to the
target extension.
Ringback can also be cleared using the Cancel Ring Back When Free function.
Action: Emulation | Automatic Callback.
Action Data: None.
Default Label: AutCB or Auto Callback.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 933

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Button Programming Actions

1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Auto-Intercom Deny
Use the Auto-Intercom Deny function to block automatic intercom calls.
Action: Advanced | Do Not Disturb | Auto Intercom Deny.
Action Data: Blank.
Default Label: NoAI or No Auto Int Calls.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 934

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 Automatic Intercom

1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Automatic Intercom
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
This feature can be used as part of handsfree announced transfers.
Action: Emulation | Automatic Intercom.
Action Data: User number or name.
This field can be left blank for number entry when pressed.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: Iauto or Auto Intercom.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 935

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Button Programming Actions

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Break Out
This feature is usable within a system multi-site network. It allows a user on one system in the
network to specify that the following dialing be processed by another system on the network as if
the user dialed it locally on that other system.
On phones with a multi-line display, if the target system is not specified in the button settings, a
menu of the available systems in the network is displayed from which a selection can be made.
Action: Advanced | Dial | Break Out.
Action Data: Optional. The system name or IP address of the required system can be specified. If
no system name or IP address is set, on display phones a list of systems within the network is
displayed when the button is pressed.
Default Label: BkOut or Breakout.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes M-Series: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No T-Series: No
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1] [1]
1600 Series: Yes 3810: No 5600 Series: Yes D100: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602 models.

February 2019 Administering Avaya IP Office™ Platform with Manager 936

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 Bridged Appearance

Bridged Appearance
Creates an appearance button that follows the state of another user's call appearance button. The
bridged appearance can be used to make and answer calls on behalf of the call appearance user.
The bridged appearance button user must also have at least one call appearance button
programmed.
Bridged appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Appearance buttons can be set with a ring delay if required or to not ring. This does not affect the
visual alerting displayed next to the button. The delay uses the user's Ring Delay (User |
Telephony | Multi-line Options) setting.
Action: Appearance | Bridged Appearance.
Action Data: User name and call appearance button number.
Default Label: <user name><call appearance label>.
Toggles: No.
Status Indication: Yes. Required. See Bridge Appearance Button Indication.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes [1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes [1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes

1. Not supported on T7000, T7100, M7100, M7100N and the Audio Conferencing Unit (ACU).

Busy
Not used.

February 2019 Administering Avaya IP Office™ Platform with Manager 937

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Button Programming Actions

Busy On Held
When on, busy on held returns busy to new calls while the user has an existing call on hold. While
this feature can be used by users with appearance keys, it is not recommended as this overrides
the basic call handling intent of appearance keys.
Action: Advanced | Busy | Busy on Held.
Action Data: 1 for on, 0 for off.
Default Label: BusyH.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Call Forwarding All

Switches forward unconditional on and sets the forward number to the number specified or
prompts the user to enter a number if none is specified.
Action: Emulation | Call Forwarding All.
Action Data: Telephone number or blank for entry when pressed.
If blank, user's with a log in code will be prompted to enter that code to use this function.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: CFrwd or Call Forward All.

February 2019 Administering Avaya IP Office™ Platform with Manager 938

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 Call Intrude

Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Call Intrude
This feature allows you to intrude on the existing connected call of the specified target user. All call
parties are put into a conference and can talk to and hear each other. A Call Intrude attempt to a
user who is idle becomes a Priority Call.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
The system support a range of other call intrusion methods in addition to this feature.
Action: Advanced | Call | Call Intrude.
Action Data: User number or blank for entry when pressed.

February 2019 Administering Avaya IP Office™ Platform with Manager 939

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Button Programming Actions

On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: Intru or Intrude.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Call Listen
This feature allows you to monitor another user's call without being heard. Monitoring can be
accompanied by a tone heard by all parties. Use of the tone is controlled by the Beep on Listen
setting on the System | Telephony | Tones & Music tab. The default for this setting is on. If
enabled, this is the only indication of monitoring given to the monitored user. There is no phone
display indication of monitoring.

Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
The use of call listen is dependant on:
The target being a member of the group set as the user's Monitor Group (User | Telephony |
Supervisor Settings). The user does not have to be a member of the group.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded

February 2019 Administering Avaya IP Office™ Platform with Manager 940

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 Call Listen

(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
This feature uses system conference resources. If insufficient conference resource are available it
will not be possible to use this feature.
A number of features are supported for call listening:
• Users can be given privacy features that allow them to indicate that a call cannot be
monitored. See Private Calls.
• IP extensions can be monitored including those using direct media. Previously the monitoring
of IP extensions could not be guaranteed.
• The monitoring call can be initiated even if the target user is not currently on a call and
remains active until the monitoring user clears the monitoring call.
• The user who initiated the call listen can also record the call.
Intruding onto an a user doing silent monitoring (Call Listen) is turned into a silent monitoring call.
1400, 1600, 9500 and 9600 Series phones with a user button can initiate listening using that
button if the target user meets the criteria for listening.
The system support a range of other call intrusion methods in addition to this feature.
Action: Advanced | Call | Call Listen.
Action Data: User number.
Default Label: Listn or Listen.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

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Button Programming Actions

Call Log
This function provides access to a list of received calls.
Action: Advanced | Call | Call Log.
Action Data: None.
Default Label: Call Log.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support This function is only supported on M-Series and T-Series phones with a display.
Not supported on phones on a DECT systems.

Call Park
Allows the user to park and unpark calls. The button can be used in two ways, either associated
with a specified park slot number or unspecified.
When associated with a specific park slot number, the button will park and unpark calls from that
park slot and indicate when a call is parked in that park slot. Similarly the Park buttons within
application (for example SoftConsole and one-X Portal) can be used to park, retrieve and indicate
parked calls.
When not associated with a specific park slot number, the button will park calls by assigning them
a park slot number based on the users extension number. For example, for extension XXX, the
first parked call is assigned to park slot XXX0, the next to XXX1 and so on up to XXX9. The button
will indicate when there are parked calls in any of those slots. On the T7000 phone, only a single
automatic part slot XXX0 is supported.
• With a call connected, pressing the button will park that call using a park slot number
assigned by the system based on the extension number.
• With no call connected, pressing the button will display details of any calls parked by the
extension and allow their retrieval.
Action: Emulation | Call Park.
Action Data: Optional. Either blank or a specific park slot number. Name ca
Park slot IDs can be up to 15 digits in length. Names can also be used for application park slots.
Default Label: CPark or Call Park.
Toggles: .
Status Indication: .

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 Call Park and Page

Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- Calls CPark CPark Green Red flash Green Blue Slow
parked by flash flash flash
extension
- Call CPark CPark Red flash Red on Red flash Green Slow
Parked by flash
other
extension
- No CPark CPark Off Off Off Grey Off
parked
calls

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes [2]
[1]
1200 Series: No 3600 Series: Yes 5400 Series: Yes 9040: Yes T-Series: Yes [2]
[1]
1400 Series: Yes 3700 Series: No 4600 Series: Yes 9500 Series: Yes
[1] [1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602.
2. M-Series/T-Series: The button is equivalent to Feature 74.

Call Park and Page

Parks the user's current call into the park slot number specified on the System | Telephony | Park
& Page tab, in the Central Park Range field.
On M/T-series phones, 14xx/16xx phones, and the 9504 phone, the user is presented with up to
three Page Target Groups. On other 95xx/96xx phones, the Page action displays a scrolling list of
possible Page Target Groups. The user may also directly enter a Page target number, or use the
system Directory to find a Page target.
A call Parked within the Central Park Range (regardless of the origin of the Park action) can be
retrieved by directly dialing the desired Central Park Range slot on which that call is Parked.

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Button Programming Actions

Action: Emulation | Call Park and Page.

Action Data: None.
Default Label: ParkPage
Toggles: No.
Status Indication: No.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: No 6400 Series: No D100: Yes
1100 Series: Yes 2400 Series: No 4400 Series: No 7400 Series: No M-Series: Yes [1]
1200 Series: Yes 3600 Series: No 5400 Series: No 9040: Yes T-Series: Yes [1]
1400 Series: Yes 3700 Series: No 4600 Series: No 9500 Series: Yes
1600 Series: Yes 3810: No 5600 Series: No 9600 Series: Yes

1. M-Series/T-Series: Feature 74 is equivalent to this button when a Central Park Range is

defined. On an M7000 phone, if this feature is invoked, the call always attempts to Park on
the highest defined Central Park Range slot. See the Call Park and Page short code
description for details.

Call Park To Other Extension

Allows the user to park their current call against another user's extension. The parked call
indication on that extension is then activated according to the telephone type.
If the target extension has a Call Park button with no specific park slot number, the parked call will
be indicated by that button and can be unparked from the list of parked calls shown when that
button is pressed.
The park slot number assigned to the parked call is based on the number of the extension parking
the call. For example, calls parked by extension 201 are assigned the park slot ID 2010, 2011 and
so on up to 2019 depending on the number of calls parked.
Action: Emulation | Call Park To Other Extension.
Action Data: User number. This field can be left blank for number entry when pressed.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: RPark or Call Park to Other.
Toggles: Yes .
Status Indication: Yes. This is the status indication on the extension parking the call.

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 Call Pickup

Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- Parked RPark RPark Green Red flash Green Blue Slow
call flash flash flash
- No RPark RPark Off Off Off Grey Off
parked
call

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 5400 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 4600 Series: Yes 9500 Series: Yes
[1] [1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602.

Call Pickup
Answer an alerting call on the system.
Action: Emulation | Call Pickup.
Action Data: None.
Default Label: CpkUp or Call Pickup Any.
Toggles: No.
Status Indication: No.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.

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Button Programming Actions

Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No

1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Call Pickup Any

Pick up the first available ringing call on the system.
Action: Advanced | Call | Call Pickup Any.
Action Data: None.
Default Label: PickA or Pickup Any.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. T3 Phones: Displays a list of call ringing from which the user can select a call to answer.
• Classic/Comfort icon: Displays .

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 Call Pickup Group

• DSS Link LED: None.

Call Pickup Group

Pick up a call ringing any hunt group of which the user is a member or set to pick up calls from a
specific group.
The user can use this feature even if their membership of the group is currently set as disabled.
Action: Advanced | Call | Call Pickup Group.
Action Data: Optional. To pick up calls from a specific group, use the group number or name.
Default Label: PickG or Pickup Group.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models as detailed below.
T3 Phones: Displays a list of calls ringing the hunt group from which the user can select which call
to answer.
• Classic/Comfort icon: Displays followed by group name.
• DSS Link LED: None.
M-Series/T-Series: The button is equivalent to Feature 75.

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Button Programming Actions

Call Pickup Members

This feature can be used to pick up any call to an extension that is a member of the hunt group
specified. The call picked up does not have to be a hunt group call. The call picked up does not
have to be a hunt group call. The function includes group members even if their membership of
the group is currently disabled.
Action: Advanced | Call | Call Pickup Members.
Action Data: Group number or name.
Default Label: PickM or Pickup Members.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models as detailed below.
T3 Phones: Displays a list of calls ringing the hunt group from which the user can select which call
to answer.
• Classic/Comfort icon: Displays followed by group name.
• DSS Link LED: None.

Call Queue
Transfer the call to the target extension if free or busy. If busy, the call is queued to wait for the
phone to become free. This is similar to transfer except it allows you to transfer calls to a busy
phone.

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 Call Record

Action: Advanced | Call | Call Queue.

Action Data: User number.
Default Label: Queue.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Call Record
This feature allows you to record a conversation and requires Voicemail Pro to be installed. An
advice of recording warning will be given if configured on the voicemail system. The recording is
placed in the mailbox specified by the user's Manual Recording Mailbox setting. Call recording
also requires available conference resources similar to a three-party conference.
Action: Advanced | Call | Call Record.
Action Data: None.
Default Label: Recor or Record.
Toggles: Yes.
Status Indication: Yes.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.

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Button Programming Actions

Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Call Screening
This function is used to enable or disable call screening. While enabled, when a caller is
presented to the user's voicemail mailbox, if the user's phone is idle they will hear through the
phone's handsfree speaker the caller leaving the message and can select to answer or ignore the
call.
This feature is supported on 1408, 1416, 1608, 1616, 9500 Series, 9600 Series, M7310, M7310N,
M7208, M7208N, M7324, M7324N, T7208, T7316 and T7316E phones. It can be used with both
Embedded Voicemail and Voicemail Pro.
Call screening is only applied as follows:
• It is only applied to calls that have audible alerted at the user's extension before going to
voicemail. This requires the user to have both voicemail coverage and call screening enabled
and the phone's ringer not set to silent. However it is not applied if the user transfers the call
to voicemail.
• It is only applied if the user's phone is idle, ie. not on a call or with a call held pending transfer
or conference.
• Calls that ring the user, are then rerouted (for example follow a forward on busy setting) and
then return to the user's mailbox are screened.
While a call is being screened, the phone can be used to either answer or ignore the screened
call. Auto answer options are ignored.
Answering a screened call
A screened call can be answered by pressing the Answer soft key (if displayed) or lifting the
handset. Pressing the call appearance or line button on which the call is indicated will also answer
the call.
When answered:
• The phone's microphone is unmuted and a normal call between the user and caller now
exists.

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 Call Screening

• The voicemail recording stops but that portion of the call already recorded is left as a new
message in the user's mailbox.
Ignoring a screened call
A screened call can be ignored by pressing the Ignore soft key if displayed. On 1400, 1600, 9500
and 9600 Series phones, pressing the SPEAKER button will ignore the call. On M-Series and T-
Series phones, pressing the Release key will ignore the call.
When ignored:
• The call continues to be recorded until the caller hangs up or transfers out of the mailbox.
• The user's phone returns to idle with call screening still enabled. However any other call that
has already gone to voicemail is not screened.
Screened call operation
While a call is being screened:
• The mailbox greeting played and the caller can be heard on the phone's speakerphone. The
caller cannot hear the user.
• The user is regarded as being active on a call. They will not be presented with hunt group
calls and additional personal calls use abbreviated ringing.
• 1400/1600/9500/9600 Series phones: If the phone's default audio path is set to headset or
the phone is idle on headset, then the screened call is heard through the headset.
• Any additional calls that go to the user's mailbox when they are already screening a call,
remain at the mailbox and are not screened even if the existing call being screened is ended.
• Making or answering another call while listening to a screened call is treated as ignoring the
screened call. For users with Answer Pre-Select enabled (User | Telephony | Multi-line
Options), pressing an appearance button to display details of a call is also treated as ignoring
the screened call.
• Other users cannot access a call that is being screened. For example they cannot use call
pickup, bridged appearance or line appearance buttons, call intrude or call acquire functions.
• Phone based administration cannot be accessed and the hold, transfer and conference
buttons are ignored.
• The screened caller using DTMF breakout ends the call screening.
Enabling do not disturb overrides call screening except for calls from numbers in the user's do not
disturb exceptions list.
Locking the phone overrides call screening.
Manual call recording cannot be applied to a call being screened.
While a call is being screened, it uses one of the available voicemail channels. If no voicemail
channels are available, call screening does not occur.
Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.

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Button Programming Actions

Details
Action: Advanced | Call | Call Screening.
Action Data: None.
Default Label: CallScreen or Call Screening.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: No 6400 Series: No D100: No
1100 Series: No 2400 Series: No 4400 Series: No 7400 Series: No M-Series: Yes
1200 Series: No 3600 Series: No 4600 Series: No 9040: No T-Series: Yes [1]
1400 Series: Yes 3700 Series: No 5400 Series: No 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: No 5600 Series: No 9600 Series: Yes
[1]

1. Not 1403, 1603, T7406E.

Call Steal
This function can be used with or without a specified user target.
If the target has alerting calls, the function will connect to the longest waiting call.
If the target has no alerting calls but does have a connected call, the function will take over the
connected call, disconnecting the original user. This usage is subject to the Can Intrude setting of
the Call Steal user and the Cannot Be Intruded setting of the target.
If no target is specified, the function attempts to reclaim the user's last ringing or transferred call if
it has not been answered or has been answered by voicemail.
Action: Advanced | Call | Call Steal.

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 Call Waiting Off

Action Data: User number or blank for last call transferred.

Default Label: Aquir or Aquire.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models if detailed below.

Call Waiting Off

Switches call waiting off for the user. This button function is obsolete. The Call Waiting On button
function toggles on/off and indicates current status.
Action: Advanced | Call | Call Waiting Off.
Action Data: None.
Default Label: CWOff.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
Table continues…

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Button Programming Actions

1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models if detailed below.

Call Waiting On
Enables call waiting on the user's extension. When the user is on a call and another call arrives,
they will hear a call waiting tone.
Note:
Call waiting does not operate for user's with call appearance buttons. See Call Waiting.
Details
Action: Advanced | Call | Call Waiting On.
Action Data: None.
Default Label: CWOn or Call Waiting On.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.

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 Call Waiting Suspend

Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No

1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Call Waiting Suspend

Disables call waiting, if on, for the duration of the extension's next call.
Action: Advanced | Call | Call Waiting Suspend.
Action Data: None.
Default Label: CWSus.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

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Button Programming Actions

Cancel All Forwarding

Cancels forward unconditional, forward on busy, forward on no answer, follow me and do not
disturb if any of those are active on the user's extension.
Action: Advanced | Call | Cancel All Forwarding.
Action Data: None.
Default Label: FwdOf or Call Forward Off.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Cancel Leave Word Calling

Not supported. Provided for CTI emulation only. Cancels the last Leave Word Calling message
originated by the user.
Action: Emulation | Cancel Leave Word Calling.
Action Data: None.
Default Label: CnLWC.
Toggles: No.
Status Indication: No.
User Admin: No.

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 Cancel Ring Back When Free

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Cancel Ring Back When Free

Cancels any existing ring back set by the user, see Ring Back When Free. Note that the Ring
Back When Free button toggles to set or cancel ring back when free and also indicates the current
status.
Action: Advanced | Miscellaneous | Cancel Ring Back When Free.
Action Data: None.
Default Label: RBak-.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No [2]
Table continues…

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Button Programming Actions

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models if detailed below.
M-Series/T-Series: The button is equivalent to Feature #2.

Clear Call
This feature can be used to end the last call put on hold. This can be used in scenarios where a
first call is already on hold and simply ending the second call will cause an unsupervised transfer
of the first call.
Action: Advanced | Call | Clear Call.
Action Data: None.
Default Label: Clear.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

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 Clear CW

Clear CW
End the user's current call and answer any call waiting. Requires the user to also have call waiting
indication on. This function does not work for users with multiple call appearance buttons.
Action: Advanced | Call | Clear CW.
Action Data: None.
Default Label: ClrCW.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Clear Hunt Group Night Service

Changes the specified hunt group from Night Service mode to 'In Service' mode. This button
function is obsolete. The Set Hunt Group Night Service function can be used to toggle a group
in/out of service and provides lamp status indication.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Action: Advanced | Call | Clear Hunt Group Night Service.
Action Data: Group number.

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Button Programming Actions

If left blank, the button will affect all hunt groups of which the user is a member.
The Set Hunt Group Night Service and Clear Hunt Group Night Service short code and button
features can be used to switch an SSL VPN service off or on respectively. The service is indicated
by setting the service name as the telephone number or action data. Do not use quotation marks.
Default Label: HGNS-.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Clear Hunt Group Out Of Service

Changes the specified hunt groups status from Out of Service mode to 'In Service' mode. This
button function is obsolete. The Set Hunt Group Out Of Service function can be used to toggle a
group in/out of service and provides lamp status indication.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Action: Advanced | Call | Clear Hunt Group Out of Service.
Action Data: Group number.
If left blank, the button will affect all hunt groups of which the user is a member.
Default Label: HGOS-.
Toggles: No.
Status Indication: No.

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 Clear Quota

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Clear Quota
Quotas can be assigned on outgoing calls to data services such as internet connections. The
quota defines the number of minutes available for the service within a time frame set within the
service, for example each day, each week or each month.
The Clear Quota function can be used to reset the quota for a specific service or for all services.
Action: Advanced | Call | Clear Quota.
Action Data: Service name" or "" (all services).
Default Label: Quota.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
Table continues…

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Button Programming Actions

1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Coaching Intrusion
This feature allows the you to intrude on another user's call and to talk to them without being
heard by the other call parties to which they can still talk. For example: User A is on a call with
user B. When user C intrudes on user A, they can hear users A and B but can only be heard by
user A.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.

Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
The system support a range of other call intrusion methods in addition to this feature.
Action: Advanced | Call | Coaching Intrusion.
Action Data: User number or name or blank for entry when pressed.
Default Label: Coach or Coaching Intrusion.
Toggles: No.
Status Indication: No.
User Admin: No feedback provided..
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No. 20 Series: No. 4100 Series: No 6400 Series: No D100: No
1100 Series: No. 2400 Series: No 4400 Series: No 7400 Series: No M-Series: No
1200 Series: No. 3600 Series: No 4600 Series: No 9040: No T-Series: No
Table continues…

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 Conference

1400 Series: Yes 3700 Series: No 5400 Series: No 9500 Series: Yes T3/T3 IP Series:
[1][2] [2] No
1600 Series: Yes 3810: No 5600 Series: No 9600 Series: Yes
[1]

1. Not 1403, 1603.

2. Not supported on non-IP telephones when using a headset.

Conference
This function is intend for use with Avaya M-Series and T-Series phones only. When pressed, the
button invokes the same conference process as dialing Feature 3.
Action: Advanced | Call | Conference.
Action Data: None.
Default Label: Conf or Conference Add.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support This function is only supported on Avaya M-Series and T-Series phones.
M-Series/T-Series: The button is equivalent to Feature 3.

Conference Add
Conference add controls can be used to place the user, their current call and any calls they have
on hold into a conference. When used to start a new conference, the system automatically assigns
a conference ID to the call. This is termed ad-hoc (impromptu) conferencing.
If the call on hold is an existing conference, the user and any current call are added to that
conference. This can be used to add additional calls to an ad-hoc conference or to a meet-me
conference. Conference add can be used to connect two parties together. After creating the
conference, the user can drop from the conference and the two incoming calls remain connected.
For R11.0 and higher, the button has additional features:
• When pressed during a normal two-party call, that call is turned into a two-party conference
call. This then provides access to the phone’s other conference control, such as to add other
parties, without interrupting the call.
• During an existing conference, pressing the button (on 1400, 1600, 9500, 9600 and J100
Series phones) provides a menu to enter the number of an additional party to add to the

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Button Programming Actions

conference without put the conference on hold. The other parties in the conference can hear
the call progress and if answered the other party is immediately in the conference.
For further details refer to the Conferencing section.
Action: Advanced | Call | Conference Add.
Action Data: None.
Default Label: Conf+ or Conference Add.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Conference Meet Me
Conference meet-me refers to features that allow a user or caller to join a specific conference by
using the conference's ID number (either pre-set in the control or entered at the time of joining the
conference).
Note:
Conference Meet Me features can create conferences that include only one or two parties.
These are still conferences that are using resources from the host system's conference
capacity.
Conference ID Numbers
By default, ad hoc conferences are assigned numbers starting from 100 for the first conference in
progress. Therefore, for conference Meet Me features specify a number away from this range
ensure that the conference joined is not an ad hoc conference started by other users. It is not

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 Conference Meet Me

possible to join a conference using conference Meet Me features when the conference ID is in use
by an ad-hoc conference.
User Personal Conference Number Each user's own extension number is treated as their own
personal conference number. Only that user is able to start a conference using that number as the
conference ID. Any one else attempting to start a conference with that number will find themselves
in a conference but on hold until the owner also joins. Personal conferences are always hosted on
the owner's system.
Note:
When a user calls from their mobile twinned number, the personal conference feature will only
work if they access the conference using an FNE 18 service.
Multi-Site Network Conferencing
Meet Me conference IDs are now shared across a multi-site network. For example, if a conference
with the ID 500 is started on one system, anyone else joining conference 500 on any system will
join the same conference. Each conference still uses the conference resources of the system on
which it was started and is limited by the available conference capacity of that system.
Previously separate conferences, each with the same conference ID, could be started on each
system in a multi-site network.
Other Features
Transfer to a Conference ButtonA currently connected caller can be transferred into the
conference by pressing TRANSFER, then the Conference Meet Me button and TRANSFER again
to complete the transfer. This allows the user to place callers into the conference specified by the
button without being part of the conference call themselves. This option is only support on Avaya
phones with a fixed TRANSFER button (excluding T3 and T3 IP phones).
Conference Button Status Indication When the conference is active, any buttons associated
with the conference ID indicate the active state.
Details
Action: Advanced | Call | Conference Meet Me.
Action Data: Conference number. This can be an alphanumeric value up to 15 characters.
User Personal Conference Number Each user's own extension number is treated as their own
personal conference number. Only that user is able to start a conference using that number as the
conference ID. Any one else attempting to start a conference with that number will find themselves
in a conference but on hold until the owner also joins. Personal conferences are always hosted on
the owner's system.
Note:
When a user calls from their mobile twinned number, the personal conference feature will only
work if they access the conference using an FNE18 service.
Default Label: CnfMM <conference number> or Conf. Meet Me <conference number>.
Toggles: No.
Status Indication: Yes

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Button Programming Actions

Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J169, J179
6400, 9500
Series
- CnfMM CnfMM Green on Red on Green on Green On
Conferenc
e In Use
- CnfMM CnfMM Off Off Off Grey Off
Conferenc
e Idle

For a Conference Meet Me configured to the user's own extension number, the indicator flashes
red when the conference is in use but the user has not joined. There is also an abbreviated ring
when the indicator changes to flashing red. It changes to solid red when the user joins.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series:Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Consult
Not supported. Provided for CTI emulation only.
Action: Emulation | Consult.
Action Data: None.
Default Label: Cnslt.
Toggles: No.
Status Indication: No.
User Admin: No.

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 Coverage Appearance

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Coverage Appearance
Creates a button that alerts when a call to the specified covered user is unanswered after that
users Individual Coverage Timer expires.
The call coverage appearance button user must also have at least one call appearance button
programmed. The covered user does not need to be using call appearance buttons.
Coverage appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
IP Office: Appearance buttons can be set with a ring delay if required or to not ring. This does not
affect the visual alerting displayed next to the button. The delay uses the user's Ring Delay (User |
Telephony | Multi-line Options) setting.
Action: Appearance | Coverage Appearance.
Action Data: User name.
Default Label: <user name>.
Toggles: No.
Status Indication: Yes. See Coverage Button Indication.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes [1]
Table continues…

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Button Programming Actions

1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes [1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes

1. Not supported on T7000, T7100, M7100, M7100N and the Audio Conferencing Unit (ACU).

Dial
This action is used to dial the number contained in the Telephone Number field. A partial number
can be enter for the user to complete. On buttons with a text label area, Dial followed by the
number is shown.
Action Data: Telephone number or partial telephone number.
Default Label: Dial.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. T3 Phones
• Classic/Comfort icon: Displays the telephone number set.
• DSS Link LED: None.

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 Dial 3K1

Dial 3K1
The call is presented to local exchange as a "3K1 Speech Call". Useful in some where voice calls
cost less than data calls.
Action: Advanced | Dial | Dial 3K1.
Action Data: Telephone number.
Default Label: D3K1 or Dial 3K1.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial 56K
The call presented to local exchange as a "Data Call".
Action: Advanced | Dial | Dial 56K.
Action Data: Telephone number.
Default Label: D56K or Dial 56K.
Toggles: No.
Status Indication: No.
User Admin: No.

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Button Programming Actions

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial 64K
The call is presented to local exchange as a "Data Call".
Action: Advanced | Dial | Dial 64K.
Action Data: Telephone number.
Default Label: D64K or Dial 64K.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

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 Dial CW

Dial CW
Call the specified extension number and force call waiting indication on if the extension is already
on a call. The call waiting indication will not work if the extension called has multiple call
appearance buttons in use.
Action: Advanced | Dial | Dial CW.
Action Data: User number.
Default Label: DCW or Dial Call Waiting.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial Direct
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
This feature can be used as part of handsfree announced transfers.
Action: Advanced | Dial | Dial Direct.
Action Data: User number or name or blank for entry when pressed.
If left blank, the Dial Direct button can be used with User buttons to specify the target.

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Button Programming Actions

Default Label: Dirct or Auto Intercom.

Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1]
1100 Series: No 2400 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
[1]
1200 Series: No 3600 Series: No 4400 Series: Yes 7400 Series: No M-Series: Yes
1400 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1] [1]
1600 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial Emergency
Dials the number specified regardless of any outgoing call barring applicable to the user.
Action: Advanced | Dial | Dial Emergency.
Action Data: Telephone number.
Default Label: Emrgy or Dial Emergency.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 972

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 Dial Inclusion

1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial Inclusion
This feature allows you to intrude on another user's call to talk to them. Their current call is put on
hold while you talk and automatically reconnected when you end the intrusion. The intruder and
the target extension can then talk but cannot be heard by the other party. This can include
intruding into a conference call, where the conference will continue without the intrusion target.
During the intrusion all parties hear a repeated intrusion tone. When the intruder hangs-up the
original call parties are reconnected. Attempting to hold a dial inclusion call simply ends the
intrusion. The inclusion cannot be parked.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
The system support a range of other call intrusion methods in addition to this feature.
Action: Advanced | Dial | Dial Inclusion.
Action Data: User number or name or blank for user selection when pressed.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: Inclu or Dial Inclusion.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 973

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Button Programming Actions

1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial Intercom
Automatic intercom functions allow you to call an extension and have the call automatically
answered on speaker phone after 3 beeps. The extension called must support a handsfree
speaker. If the extension does not have a handsfree microphone then the user must use the
handset if they want to talk. If the extension is not free when called, the call is presented as a
normal call on a call appearance button if available.
This feature can be used as part of handsfree announced transfers.
Action: Emulation | Dial Intercom.
Action Data: User number or name or blank for number entry when pressed.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: Idial or Auto Intercom.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 974

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 Dial Paging

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays followed by the set number.
• DSS Link LED: None.
M-Series/T-Series: The button is equivalent to Feature 66 <number>.

Dial Paging
Makes a paging call to an extension or group specified. If no number is specified, this can be
dialed after pressing the button. The target extension or group members must be free and must
support handsfree auto-answer in order to hear the page.
On Avaya phones with a CONFERENCE button, a paged user can convert the page call into a
normal call by pressing that button.
Action: Advanced | Dial | Dial Paging.
Action Data: User number or name or group number or name or blank for number entry when
pressed.
Default Label: Page.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 975

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Button Programming Actions

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays followed by target number if set.
• DSS Link LED: None.

Dial Physical Extn by Number

Call the specified extension using its Base Extension number setting. This is regardless of the
current user logged in at that extension and any forwarding, follow me or do not disturb settings
applied by the extension user. This function requires the extension to be assigned a default
extension number in the system configuration. If the extension does not have a default extension
number, Dial Physical Extn by Id should be used.
Action: Advanced | Dial | Dial Physical Extn by Number.
Action Data: Extension port base extension number.
Default Label: PhyEx or Dial Physical Extn.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

February 2019 Administering Avaya IP Office™ Platform with Manager 976

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 Dial Physical Number by ID

Dial Physical Number by ID

Call the specified extension, if free, regardless of the current user logged in at that extension and
any forwarding, follow me or do not disturb settings applied by the extension user. This function
uses the port ID shown in the system configuration.
Action: Advanced | Dial | Dial Physical Extn by Id.
Action Data: Extension port ID number.
Default Label: DialP or Dial Extn by Id.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial Speech
This feature allows a short code to be created to force the outgoing call to use the Speech bearer
capability.
Action: Advanced | Dial | Dial Speech.
Action Data: Telephone number.
Default Label: DSpch or Dial Speech.
Toggles: No.
Status Indication: No.
User Admin: No.

February 2019 Administering Avaya IP Office™ Platform with Manager 977

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Button Programming Actions

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Dial V110
The call is presented to local exchange as a "Data Call".
Action: Advanced | Dial | Dial V110.
Action Data: Telephone number.
Default Label: DV110 or Dial V110.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

February 2019 Administering Avaya IP Office™ Platform with Manager 978

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 Dial V120

Dial V120
The call is presented to local exchange as a "Data Call".
Action: Advanced | Dial | Dial V120.
Action Data: Telephone number.
Default Label: DV120 or Dial V120.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Display Msg
Allows the sending of text messages to digital phones on the local system.
Action: Advanced | Dial | Display Msg.
Action Data: The telephone number takes the format N";T" where:
• N is the target extension.
• T is the text message. Note that the "; before the text and the " after the text are required.
Default Label: Displ.
Toggles: No.
Status Indication: No.
User Admin: No.

February 2019 Administering Avaya IP Office™ Platform with Manager 979

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Button Programming Actions

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Dial Video
The call is presented to the local exchange as a "Video Call".
Action: Advanced | Dial | Dial Video.
Action Data: Telephone number.
Default Label: Dvide or Dial Video.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

February 2019 Administering Avaya IP Office™ Platform with Manager 980

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 Directed Call Pickup

Directed Call Pickup

Pickup a call ringing at a specific extension or hunt group.
Action: Emulation | Directed Pickup.
Action Data: User number or name or group number or name or blank for number entry when
pressed.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: DpkUp or Call Pickup.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
M-Series/T-Series: The button is equivalent to Feature 76.

Directory
A Dir button provides access to various directories and allows telephone number selection by
dialed name matching. The directories available for searching depend on the phone type, see
User Directory Access. Once they user has selected a directory, dialing on the dial pad letter keys
is used to display matching names, with controls for scrolling through the matching names and for
calling the currently displayed name.

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Button Programming Actions

The method of name matching is controlled by the Dial by Name (System | Telephony | Telephony)
setting in the system configuration:
• With Dial By Name on Matching is done against all the dial keys pressed. For example,
dialing 527 matches names starting with JAS (for example "Jason") and KAR (for example
"Karl"). Only the first 50 matches are displayed.
• With Dial By Name off Matching is done against the first letter only. For example pressing 5
displays names beginning with J. Press 5 again displays names beginning with K. Only the
first 50 matches are displayed. This mode is not supported by Release 5.0+.
Name dialing functions on the system assume that the phone is using the standard ITU keypad as
follows:

Dialing Spaces
To enter a name with a space, the 0 key is used for the space. For example "John S..." is dialed as
564607.
Details
Action: Emulation | Directory.
Action Data: None.
Default Label: Dir.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 3600 Series: No 5400 Series: Yes 9040: No T3/T3 IP Series:
[1] Yes
1400 Series: No 3700 Series: No 4600 Series: Yes 9500 Series: No
[1]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]
20 Series: Yes 4100 Series: No 6400 Series: Yes M-Series: Yes
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 982

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 Do Not Disturb Exception Add

2400 Series: Yes 4400 Series: Yes 7400 Series: No T-Series: Yes [1]
[1]

1. Not 1603, 2402, 4601, 4602, 5402, 5601, 5602 and T7100 models.
T3 Phones:
• Classic/Comfort icon: Displays .
• DSS Link LED: None.

Do Not Disturb Exception Add

Adds a number to the user's "Do Not Disturb Exception List". This can be the number of an
internal user or a number to match the CLI of a particular external caller.Calls from that number,
except hunt group calls, will ignore the user's Do Not Disturb setting. For further details see Do
Not Disturb (DND).
Action: Advanced | Do Not Disturb | Do Not Disturb Exception Add.
Action Data: Telephone number or CLI. Up to 31 characters. For CLI numbers any prefix added
by the system must also be included.
Default Label: DNDX+.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

February 2019 Administering Avaya IP Office™ Platform with Manager 983

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Button Programming Actions

Do Not Disturb Exception Delete

Removes a number from the user's "Do Not Disturb Exception List". This can be the number of an
internal user or a number to match the CLI of a particular external caller.
Action: Advanced | Do Not Disturb | Do Not Disturb Exception Delete.
Action Data: Telephone number or CLI.
Default Label: DNDX-.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Do Not Disturb Off

Cancels the user's 'do not disturb' mode if set. This button function is obsolete as the do not
disturb on function toggles on/off and indicates the button status.
Action: Advanced | Do Not Disturb | Do Not Disturb Off.
Action Data: None.
Default Label: DNDOf.
Toggles: No.
Status Indication: No.
User Admin: No.

February 2019 Administering Avaya IP Office™ Platform with Manager 984

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 Do Not Disturb On

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
M-Series/T-Series: The button is equivalent to Feature #85.

Do Not Disturb On
Enables the user's 'do not disturb' mode.
For CCR Agents, using this function button on the following phones will be requested the user to
select a reason code - 1400, 1600, 2400, 4600, 5400, 5600, 9500 and 9600 Series phones with
available programmable buttons.
Action: Advanced | Do Not Disturb | Do Not Disturb On.
Action Data: None.
Default Label: DNDOn or Do Not Disturb.
Toggles: Yes.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 985

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Button Programming Actions

1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
M-Series/T-Series: The button is equivalent to Feature 85.

Drop
This action is supported on phones which do not have a permanent Drop button.
For a currently connected call, pressing Drop disconnects the call. When drop is used to end a
call, silence is returned to the user rather than dial tone. This is intended operation, reflecting that
Drop is mainly intended for use by call center headset users.
If the user has no currently connected call, pressing Drop will redirect a ringing call using the
user's Forward on No Answer setting if set or otherwise to voicemail if available.
For a conference call, on phones with a suitable display, Drop can be used to display the
conference parties and allow selection of which party to drop from the conference.
Action: Emulation | Drop.
Action Data: None.
Default Label: Drop or Drop Call.
Toggles: No.
Status Indication: No.
User Admin: .
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: No D100: No
1100 Series: No 2400 Series: No 4400 Series: Yes 7400 Series: No M-Series: No
1200 Series: No 3600 Series: Yes 5400 Series: Yes 9040: Yes T-Series: No
1400 Series: No 3700 Series: No 4600 Series: Yes 9500 Series: Yes T3/T3 IP Series:
No
1600 Series: No 3810: Yes 5600 Series: Yes 9600 Series: Yes

February 2019 Administering Avaya IP Office™ Platform with Manager 986

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 Extn Login

Extn Login
Extn Login allows a user who has been configured with a Login Code (User | Telephony |
Supervisor Settings) to take over ownership of any extension. That user's extension number
becomes the extension number of the extension while they are logged. This is also called ‘hot
desking’.
Hot desking is not supported for H175, E129 and J129 telephones.
When used, the user is prompted to enter their extension number and then their log in code. Login
codes of up to 15 digits are supported with Extn Login buttons. Login codes of up to 31 digits are
supported with Extn Login short codes.
When a user logs in, as many of their user settings as possible are applied to the extension. The
range of settings applied depends on the phone type and on the system configuration.
By default, on 1400 Series, 1600 Series, 9500 Series and 9600 Series phones, the user's call log
and personal directory are accessible while they are logged in. This also applied to M-Series and
T-Series telephones.
On other types of phone, those items such as call logs and speed dials are typically stored locally
by the phone and will not change when users log in and log out.
If the user logging in was already logged in or associated with another phone, they will be
automatically logged out that phone.
Action: Advanced | Extension | Extn Login.
Action Data: None.
Default Label: Login.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
Table continues…

February 2019 Administering Avaya IP Office™ Platform with Manager 987

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Button Programming Actions

1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Extn Logout
Logs out a user from the phone. The phone will return to its normal default user, if an extension
number is set against the physical extension settings in the configuration. Otherwise it takes the
setting of the NoUser user. This action is obsolete as Extn Login can be used to log out an
existing logged in user.
If the user who logged out was the default user for an extension, dialing *36 will associate the
extension with the user unless they are set to forced log in.
This feature cannot be used by a user who does not have a log in code.
Action: Advanced | Extension | Extn Logout.
Action Data: None.
Default Label: Logof or Logout.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
Table continues…

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 Flash Hook

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. May have limited support on some specific T3 phone models if detailed below.

Flash Hook
Sends a hook flash signal to the currently connected line if that line is an analog line.
Action: Advanced | Miscellaneous | Flash Hook.
Action Data: Optional. Normally this field is left blank. It can contain the destination number for a
Centrex Transfer for external calls on a line from a Centrex service provider.
Default Label: Flash or Flash Hook.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Follow Me Here
Causes calls to the extension number specified, to be redirected to this user's extension. User's
with a log in code will be prompted to enter that code when using this function.

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Button Programming Actions

Action: Advanced | Follow Me | Follow Me Here.

Action Data: User name or user number.
If a user name or user number has been entered in the Action Data field, when the interactive
menu opens, press Enterto activate Follow Me Here for the number displayed on the screen.
This field can be left blank for number entry when pressed.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: Here+ or Follow Me Here.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays followed by the user name.
• DSS Link LED: On when active.

Follow Me Here Cancel

Cancels any 'Follow Me Here' set on the specified extension. Only works if entered at the
extension to which the extension's calls are being sent by the follow me action.
Action: Advanced | Follow Me | Follow Me Here Cancel.
Action Data: User number or blank for number entry when pressed.

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 Follow Me To

If a user name or user number has been entered in the Action Data field, when the interactive
menu opens, press Enterto deactivate Follow Me Here for the number displayed on the screen.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: Here- or Follow Me Here-.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Follow Me To
Leaving the extension blank prompts the user to enter the extension to which their calls should be
redirected. User's with a log in code will be prompted to enter that code when using this function.
Action: Advanced | Follow Me | Follow Me To.
Action Data: User name or user number or blank for number entry when pressed.
If a user name or user number has been entered in the Action Data field, when the interactive
menu opens, press Enterto activate Follow Me To for the number displayed on the screen.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: FolTo or Follow Me To.
Toggles: Yes.

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Button Programming Actions

Status Indication: Yes. On/off status indication is provided if the button is programmed with a
user name or number.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Forward Hunt Group Calls Off

Cancels the forwarding of the user's hunt group calls. This function is obsolete since the button
function Forward Hunt Group Calls On toggles on/off and indicates status.
Action: Advanced | Forward | Forward Hunt Group Calls Off.
Action Data: None.
Default Label: FwdH-.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.

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 Forward Hunt Group Calls On

Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No

1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Forward Hunt Group Calls On

Forward the user's hunt group calls (internal and external). This function only works when forward
unconditional is also on and uses the same forwarding number as forward unconditional.
This option is only applied for calls to Sequential and Rotary type hunt groups. Calls from other
hunt group types are not presented to the user when they have Forward Unconditional active.
Note also that hunt group calls cannot be forwarded to another hunt group.
Action: Advanced | Forward | Forward Hunt Group Calls On.
Action Data: None.
Default Label: FwdH+ or Fwd HG Calls.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
Table continues…

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Button Programming Actions

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Forward Number
Sets the number to which calls are forwarded when the user has forwarding on. Used for all
forwarding options unless a separate Forward On Busy Number is also set. Forwarding to an
external number is blocked if Inhibit Off-Switch Transfers is selected within the system
configuration.
Action: Advanced | Forward | Forward Number.
Action Data: Telephone number.
The field to be left blank to prompt the user for entry when the button is pressed. If blank, users
with a log in code will be prompted to enter that code.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: FwdNo or Fwd Number.
Toggles: No.
Status Indication: Yes. For a button with a prefixed number, status indication will indicate when
that number matches the users current set number. For a button with a no number, status
indication will show when a number has been set.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
Table continues…

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 Forward On Busy Number

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Forward On Busy Number

Sets the number to which calls are forwarded when using 'Forward on Busy' and/or 'Forward on
No Answer'. Forwarding to an external number is blocked if Inhibit Off-Switch Transfers is
selected within the system configuration.
Action: Advanced | Forward | Forward on Busy Number.
Action Data: Telephone number.
The field to be left blank to prompt the user for entry when the button is pressed. If blank, users
with a log in code will be prompted to enter that code.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: FwBNo or Fwd Busy Number.
Toggles: No.
Status Indication: Yes. For a button with a prefixed number, status indication will indicate when
that number matches the users current set number. For a button with a no number, status
indication will show when a number has been set.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
Table continues…

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Button Programming Actions

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Forward On Busy Off

Switches forward on busy off. This button function is obsolete, as Forward On Busy On can be
used to switch forward on busy on/off and provides status indication.
Action: Advanced | Forward | Forward on Busy Off.
Action Data: None.
Default Label: FwBOf.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Forward On Busy On
Enables forwarding when the user's extension is busy. For users with call appearance buttons,
they will only return busy when all call appearance buttons are in use. Uses the Forward Number
as its destination unless a separate Forward on Busy Number is set.

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 Forward On No Answer Off

Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Action: Advanced | Forward | Forward on Busy On.
Action Data: None.
Default Label: FwBOn or Fwd Busy.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, J69, M-Series
4400, J179
6400, 9500
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Forward On No Answer Off

Switches forward on no answer off. This button function is obsolete, as Forward On No Answer
On can be used to switch forward on no answer on/off and provides status indication.
Action: Advanced | Forward | Forward on No Answer Off.
Action Data: None.
Default Label: FwNOf.
Toggles: No.
Status Indication: No.

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Button Programming Actions

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Forward On No Answer On
Switches forward on no answer on/off. The time used to determine the call as unanswered is the
user's no answer time. Uses the Forward Number as its destination unless a separate Forward
on Busy Number is set.
Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
Action: Advanced | Forward | Forward on No Answer On.
Action Data: None.
Default Label: FwNOn or Fwd No Answer.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
Table continues…

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 Forward Unconditional Off

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Forward Unconditional Off

Switch 'forward all calls' off. This does not affect 'Forward on No Answer' and/or 'Forward on Busy'
if also on. This function is obsolete as a button set to Forward Unconditional On toggles on/off and
indicates when on.
Action: Advanced | Forward | Forward Unconditional Off.
Action Data: None.
Default Label: FwUOf.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Forward Unconditional On
This function is also known as 'divert all' and 'forward all'. It forwards all calls, except hunt group
and page calls, to the forward number set for the user's extension. To also forward hunt group
calls to the same number 'Forward Hunt Group Calls On' must also be used.

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Button Programming Actions

Forward Internal (User | Forwarding) can also be used to control whether internal calls are
forwarded.
In addition to the lamp indication shown below, most phones display D when forward unconditional
is on.
Action: Advanced | Forward | Forward Unconditional On.
Action Data: None.
Default Label: FwUOn or Fwd Unconditional.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays followed by the user name.
• DSS Link LED: On when active.
M-Series/T-Series: The button is equivalent to Feature 4 <number>.

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 Group

Group
Monitors the status of a hunt group queue. This option is only supported for hunt groups with
queuing enabled. The user does not have to be a member of the group.
Depending on the users button type, indication is given for when the group has alerting calls and
queued calls (queued in this case is defined as more calls waiting than there are available group
members).
Pressing a Group button answers the longest waiting call.
The definition of queued calls include group calls that are ringing. However, for operation of the
Group button, ringing calls are separate from other queued calls.
Action: Group.
Action Data: Group name enclosed in " " double-quotes or group number.
Default Label: <group name>.
Toggles: No.
Status Indication: Yes, Required.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- No calls Main Main Off Off Off Grey Off
- Call Main Main Green Red flash Green Blue Slow
alerting flash flash flash
- Calls Main Main Red flash Red on Red flash Green Slow
queued flash

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No
Table continues…

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Button Programming Actions

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Group Listen On
Using group listen allows callers to be heard through the phone's handsfree speaker but to only
hear the phone's handset microphone. This enables listeners at the user’s phone to hear the
connected party whilst limiting the connected party to hear only what is communicated via the
phone handset
When group listen is enabled, it modifies the handsfree functionality of the user’s phone in the
following manner
When the user’s phone is placed in handsfree/speaker mode, the speech path from the connected
party is broadcast on the phone speaker but the phone's base microphone is disabled.
The connected party can only hear speech delivered via the phone's handset microphone.
Group listen is not supported for IP phones or when using a phone's HEADSET button.
For T-Series and M- Series phones, this option can be turned on or off during a call. For other
phones, currently connected calls are not affected by changes to this setting, instead group listen
must be selected before the call is connected.
Group listen is automatically turned off when the call is ended.
Action: Advanced | Extension | Group Listen On.
Action Data: None.
Default Label: Group Listen On.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

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 Group Paging

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: 96x1 Series:
1100 Series: 2400 Series: 4400 Series: 7400 Series: D100:
1200 Series: 3600 Series: 4600 Series: 9040: M-Series: [2]
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T-Series: [2]
1600 Series: 3810: 5600 Series: 9600 Series: T3/T3 IP Series:

1. Not 1403.
2. M-Series/T-Series: The button is equivalent to Feature 802 (On) and Feature #802 (Off).

Group Paging
Makes a paging call to an extension or group specified. If no number is specified, this can be
dialed after pressing the button. The target extension or group members must be free and must
support handsfree auto-answer in order to hear the page.
On Avaya phones, a paged user can convert the page call into a normal call by pressing the
Conference button.
Action: Emulation | Group Paging.
Action Data: User number or name or group number or name.
On large display phones, if configured without a preset target, this type of button will display an
interactive button menu for target selection.
Default Label: GrpPg.
Toggles: No.
Status Indication: Yes.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: Yes
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: T3/T3 IP Series:
[1] No [2]
Table continues…

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Button Programming Actions

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays followed by target number if set.
• DSS Link LED: None.
M-Series/T-Series: The button is equivalent to Feature 60 <number>.

Headset Toggle
This function is intended for use with Avaya phones that have separate handset and headset
sockets but do not provide a dedicated Headset button, for example older style 4400 Series and
4600 Series phones. On phones without a headset socket or with a dedicated headset button this
control will have no effect.
Action: Miscellaneous | Headset Toggle.
Action Data: None.
Default Label: HdSet.
Toggles: Yes.
Status Indication: Yes.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 3600 Series: 4600 Series: [1] 9040: T3/T3 IP Series:
[2]
1400 Series: 3700 Series: 5400 Series: 9500 Series:
1600 Series: 3810: 5600 Series: 9600 Series:
20 Series: 4100 Series: 6400 Series: M-Series:
2400 Series: 4400 Series: 7400 Series: T-Series:

1. 4606, 4612 and 4624 only.

2. T3 Phones
• Classic/Comfort icon: Displays HdSet.

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 Hold Call

• DSS Link LED: On when active.

:

Hold Call
This uses the Q.931 Hold facility, and "holds" the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The Hold Call feature "holds" the current call to a slot. The current call is
always automatically placed into slot 0 if it has not been placed in a specified slot. Only available if
supported by the ISDN exchange.
Action: Advanced | Hold | Hold Call.
Action Data: ISDN Exchange hold slot number or blank (slot 0).
Default Label: Hold.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Hold CW
Place the user's current call on hold and answers the waiting call. This function is not supported
on phones which have multiple call appearance buttons set.
Action: Advanced | Hold | Hold CW.
Action Data: None.
Default Label: HoldCW.
Toggles: No.

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Button Programming Actions

Status Indication: No.

User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Hold Music
This feature allows the user to listen to the system's music on hold. See Music On Hold for more
information.
Action: Advanced | Hold | Hold Music.
Action Data: Optional. Systems can support multiple hold music sources. However only the
system source is supported for Hold Music buttons.
Default Label: Music or Hold Music.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

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 Hunt Group Enable

Hunt Group Enable

An individual users membership of any particular hunt groups is programmed through the system
configuration. This control allows the user to enable or disable that membership. While enabled,
the user can receive hunt group calls when logged in.
In addition to the lamp indication below, phones display G when any group membership is
enabled.
Action: Advanced | Hunt Group | Hunt Group Enable.
Action Data: Group number or name or blank for all groups of which the user is a member.
Default Label: HGEna or HG Enable.
Toggles: Yes.
Status Indication: Yes. Required.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays followed by the group number or * for all if
programmed with no specific group number.
• DSS Link LED: On when active.

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Button Programming Actions

Hunt Group Disable

This function is obsolete, the Hunt Group Enable function being able to toggle membership
between enabled and disabled and providing lamp indication of when membership is enabled.
An individual user's membership of any particular hunt groups is programmed through the system
configuration. This control allows the user to disable that membership. They will no longer receive
calls to that hunt group until their membership is enabled again.
Action: Advanced | Hunt Group | Hunt Group Disable.
Action Data: Group number or blank for all groups of which the user is a member.
Default Label: HGDis.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Inspect
Not supported. Provided for CTI emulation only. Allows users on display phones to determine the
identification of held calls. Allows users on an active call to display the identification of incoming
calls.
Action: Emulation | Inspect.
Action Data: None.
Default Label: Inspt.
Toggles: No.
Status Indication: No.
User Admin: No.

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 Internal Auto-Answer

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Internal Auto-Answer
This function is also known as handsfree auto-answer. It sets the user's extension to automatically
connect internal calls after a single tone. This function should only be used on phones that support
handsfree operation.
Action: Emulation | Internal Auto-Answer.
Action Data: Optional.
• If left blank this function acts as described above for internal auto-answer.
• FF can be entered. In that case the button will enable/disable headset force feed operation
for external calls. In this mode, when headset mode is selected but the phone is idle, an
incoming external call will cause a single tone and then be automatically connected. This
operation is only supported on Avaya phones with a fixed HEADSET button. Ring delay is
applied if set on the appearance button receiving the call before the call is auto-connected.
Default Label: HFAns or Auto Answer.
Toggles: Yes.
Status Indication: Yes. Required.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.

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Button Programming Actions

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays HFAns.
• DSS Link LED: On when active.

Last Number Redial

This function is intend for use with Avaya M-Series and T-Series phones only. When pressed, the
button invokes the same last number redial process as dialing Feature 5.
Action: Advanced | Call | Last Number Redial.
Action Data: None.
Default Label: Again.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support This function is only supported on Avaya M-Series and T-Series phones.
M-Series/T-Series: The button is equivalent to Feature 5.

Leave Word Calling

Not supported. Provided for CTI emulation only. Leaves a message for the user associated with
the last number dialed to call the originator.
Action: Emulation | Leave Word Calling.
Action Data: None.
Default Label: LWC.

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 Line Appearance

Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Line Appearance
Creates an line appearance button linked to the activity of a specified line appearance ID number.
The button can then be used to answer and make calls on that line.
The line appearance button user must also have at least one call appearance button programmed
before line appearance buttons can be programmed.
Line appearance functions, assigned to buttons that do not have status lamps or icons, are
automatically disabled until the user logs in at a phone with suitable buttons.
Release 3.2+: Appearance buttons can be set with a ring delay if required or to not ring. This does
not affect the visual alerting displayed next to the button. The delay uses the user's Ring Delay
(User | Telephony | Multi-line Options) setting.
Release 4.2+: Line appearances are supported on T3 and T3 IP phones. These phones do not
require (or support) call appearance buttons in order to use line appearances.
Action: Appearance | Line Appearance.
Action Data: Line ID number.
Default Label: Line <Line ID number>.
Toggles: No.
Status Indication: Yes. See Line Appearance Button Indication.
User Admin: No.
Phone Support: The following table indicates phones which support the programmable button:

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Button Programming Actions

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series: [1]
1200 Series: 3600 Series: 4600 Series: 9040: T-Series: [1]
1400 Series: 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: 3810: 5600 Series: 9600 Series:

1. Not supported on T7000, T7100, M7100, M7100N and the Audio Conferencing Unit (ACU).

MADN Call Appearance

Multiple Appearance Directory Number (MADN) emulates an Avaya Communication Server 1000
key and lamp style feature.
When using normal appearance buttons to answer or make calls, the information (name and
number) presented to the other end of the call is that of the button user (subject to any other line
and short code settings). When using a MADN call appearance button, the information presented
is that of the user to which the button is associated rather than that of the button user.
The user associated with a MADN button does not need to have a license or an active extension.
However, they must have an extension number. The system considers the user’s records when
the user makes a call using the MADN buttons. You can have up to 30 MADN buttons associated
with the same user.
MADN can operate in two modes:
• MADN Single Call Appearance (SCA)
The button is configured with the user name of the associated user and one of their call
appearances. This provides the following behaviors:
- Incoming extension calls: The button acts like a Bridged Appearance button to the
associated user.
- Incoming group calls: The button alerts if associated user is a member of the hunt group
and alerting.
- Outgoing calls: The button acts like a Call Appearance. It presents the call as originating
from the button user but with the number and name of the associated user in the calling
party information.
• MADN Multiple Call Appearance (MCA)
The button is configured with just the user name of the associated user. This provides the
following behaviors:
- Incoming extension calls: The button acts like a Coverage Appearance to the associated
user.
- Incoming groups calls: The button does not alert.

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 Manual Exclude

- Outgoing calls: The button acts like a Call Appearance. It presents the call as originating
from the button user but with the number of the associated user in the calling party
information.
Action Either:
• Appearance | MADN Single Call Appearance
• Appearance | MADN Multiple Call Appearance
Action Data:
• MADN Single Call Appearance: User Name, Call Appearance button number and Ring Delay.
• MADN Multiple Call Appearance: User Name and Ring Delay.
Default Label:
• MADN SCA: <MADN number S=>
• MADN MCA: <MADN number M=>
Toggles: No.
Status Indication:
• MADN SCA: Yes. See Bridge Appearance Button Indication.
• MADN MCA: Yes. See Coverage Button Indication.
User Admin: No.
Phone Support: The following table indicates phones which support the programmable button:
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: 9040: T-Series:
1400 Series: 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: 3810: 5600 Series: 9600 Series:

Manual Exclude
Not supported. Provided for CTI emulation only.
Action: Emulation | Manual Exclude
Action Data: None.
Default Label: Excl.
Toggles: No.
Status Indication: No.
User Admin: No.

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Button Programming Actions

Phone Support: Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 64000: Yes D100: No
[1]
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
1200 Series: No 3600 Series: No 4600 Series: Yes 9040 Series: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810 Series: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

MCID Activate
This action is used with ISDN Malicious Caller ID call tracing. It is used to trigger a call trace at the
ISDN exchange. The call trace information is then provided to the appropriate legal authorities.
This option requires the line to the ISDN to have MCID enabled at both the ISDN exchange and
on the system. The user must also be configured with Can Trace Calls (User | Telephony |
Supervisor Settings) enabled.
Note:
Currently, in Server Edition network, MCID is only supported for users using an MCID button and
registered on the same IP500 V2 Expansion system as the MCID trunks.
Action: Advanced | Miscellaneous | MCID Activate.
Action Data: None.
Default Label: MCID or Malicious Call.
Toggles: No.
Status Indication: Yes.
User Admin: No.
Phone Support: Note that support for particular phone models is also dependant on the system
software level.
Analog: No 3600 Series: No 4600 Series: Yes 9040: Yes T3/T3 IP Series:
[1] No
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes
Table continues…

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 Monitor Analogue Trunk MWI

1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]
20 Series: No 4100 Series: No 6400 Series: No M-Series: Yes
2400 Series: Yes 4400 Series: Yes 7400 Series: No T-Series: Yes

1. Not 1603, 4601, 4602, 5601 and 5602.

Monitor Analogue Trunk MWI

Enables a user to receive message waiting indicator (MWI) signals from analog trunks terminating
on the ATM4U-V2 card. MWI is a telephone feature that turns on a visual indicator on a telephone
when there are recorded messages.
Action: Advanced | Voicemail | Monitor Analogue Trunk MWI.
Action Data: The line appearance ID of the analog line for which MWI will be received.
Default Label: Trunk MWI.
Toggles: No.
Status Indication: No.
User Admin: No.

Off Hook Station

Enables the user's extension to be controlled by an application, for example SoftConsole. Calls
can then be answered and cleared through the application without having to manually go off or on
hook. Requires the phone to support full handsfree operation.
Action: Advanced | Miscellaneous | Off Hook Station.
Action Data: None.
Default Label: OHStn.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
Table continues…

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Button Programming Actions

- On. <Label> <Label> Green on Red on Green on Green On

- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 3600 Series: 4600 Series: [1] 9040: T3/T3 IP Series:
1400 Series: 3700 Series: 5400 Series: [1] 9500 Series:
1600 Series: 3810: 5600 Series: [1] 9600 Series:
20 Series: 4100 Series: 6400 Series: M-Series:
2400 Series: [1] 4400 Series: 7400 Series: T-Series:

1. Not 2402, 4601, 4602, 5402, 5601 and 5602 models.

Pause Recording
This feature can be used to pause any call recording. It can be used during a call that is being
recorded to omit sensitive information such as customer credit card information. This feature can
be used with calls that are recorded both manually or calls that are recorded automatically.
The button status indicates when call recording has been paused. The button can be used to
restart call recording. The system Auto Restart Paused Recording (System | Voicemail) setting
can be used to set a delay after which recording is automatically resumed.
If the voicemail system is configured to provide advice of call recording warnings, then pausing the
recording will trigger a "Recording paused" prompt and a repeat of the advice of call recording
warning when recording is resumed.
Action: Advanced | Call | Pause Recording.
Action Data: None.
Default Label: PauseRec or Pause Recording.
Toggles: Yes.
Status Indication: Yes.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series:
Table continues…

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 Priority Call

1200 Series: 3600 Series: 4600 Series: 9040: T-Series:

1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: 9600 Series:

1. Not 1403, 1603.

Priority Call
This feature allows the user to call another user even if they are set to 'do not disturb'. A priority
call will follow forward and follow me settings but will not go to voicemail.
Action: Advanced | Call | Priority Call.
Action Data: User number or name.
Default Label: PCall or Priority Call.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support: The following table indicates phones which support the programmable button:
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Priority Calling
Not supported. Provided for CTI emulation only.
Action: Emulation | Priority Calling.
Action Data: None.
Default Label: Pcall.
Toggles: No.

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Button Programming Actions

Status Indication: No.

Phone Support: The following table indicates phones which support the programmable button:

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Private Call
When on, any subsequent calls cannot be intruded on until the user's private call status is
switched off. The exception is Whisper Page which can be used to talk to a user on a private call.
Note that use of private calls is separate from the user's intrusion settings. If the user's Cannot be
Intruded (User | Telephony | Supervisor Settings) setting is enabled, switching private calls off
does not affect that status. To allow private calls to be used to fully control the user status, Cannot
be Intruded (User | Telephony | Supervisor Settings) should be disabled for the user.
If enabled during a call, any current recording, intrusion or monitoring is ended.
Action: Advanced | Call | Private Call.
Action Data: None.
Default Label: PrivC or Private Call.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.

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 Relay Off

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: 3810: 5600 Series: [1] 9600 Series:

1. Not 1603, 4601, 4602, 5601 and 5602.

Relay Off
Opens the specified switch in the system's external output port (EXT O/P).
This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Action: Advanced | Relay | Relay Off.
Action Data: Switch number (1 or 2).
Default Label: Rely-.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Relay On
Closes the specified switch in the system's external output port (EXT O/P).

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Button Programming Actions

This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Action: Advanced | Relay | Relay On.
Action Data: Switch number (1 or 2).
Default Label: Rely+ or Relay On.
Toggles: Yes.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Relay Pulse
Closes the specified switch in the system's external output port (EXT O/P) for 5 seconds and then
opens the switch.
This feature is not supported on Linux based systems. For Server Edition, this option is only
supported on Expansion System (V2) units.
Action: Advanced | Relay | Relay Pulse.
Action Data: Switch number (1 or 2).
Default Label: Relay or Relay Pulse.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.

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 Resume Call

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[12]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays S1 or S2 dependant on switch number.
• DSS Link LED: None.

Resume Call
Resume a call previously suspended to the specified ISDN exchange slot. The suspended call
may be resumed from another phone/ISDN Control Unit on the same line.
Action: Advanced | Call | Resume Call.
Action Data: ISDN Exchange suspend slot number.
Default Label: Resum.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

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Button Programming Actions

Request Coaching Intrusion

This feature allows a user to request that another user intrude on a call and talk to them without
being heard by the other call parties to which they can still talk.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.

Warning:
The use of features to listen to a call without the other call parties being aware of that
monitoring may be subject to local laws and regulations. Before enabling the feature you must
ensure that you have complied with all applicable local laws and regulations. Failure to do so
may result in severe penalties.
The system support a range of other call intrusion methods in addition to this feature.
Action: Advanced | Call | Request Coaching Intrusion.
Action Data: None.
Default Label: Request Coach or Request Coaching Intrusion.
Toggles: Yes.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: 9600 Series:

1. Not 1403, 1603.

The Request Coaching Intrusion feature exhibits the following behavior:
• A coaching request can be sent to a user or a group.
• While the request is pending, the user can cancel the request by pressing the Request
Coach button again.
• Once a coaching session is established, the user that initiated the request can include the
coach in the call, transfer the call to the coach, or drop the coach from the call.
• Once a coaching session is established, the coach can join the call or steal the call. The
coach cannot transfer or conference the call.

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 Retrieve Call

• Once the primary call ends, the coaching call continues.

Retrieve Call
Retrieves a call previously held to a specific ISDN exchange slot. Only available when supported
by the ISDN exchange.
Action: Advanced | Call | Retrieve Call.
Action Data: Exchange hold slot number.
Default Label: Retriv.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: Yes 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: Yes 3600 Series: No 4600 Series: Yes 9040: T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: No T3/T3 IP Series:
[1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: No
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Ring Back When Free

Sets a ringback on the extension being called. When the target extension ends its current call, the
ringback users is rung (for their set No Answer Time) and if they answer, a new call is made to the
target extension.
Ringback can be cleared using the Cancel Ring Back When Free function.
Action: Advanced | Miscellaneous | Ring Back When Free.
Action Data: None.

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Button Programming Actions

Default Label: AutCB or Auto Callback.

Toggles: No.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
M-Series/T-Series: The button is equivalent to Feature 2.

Ringer Off
Switches the phone's call alerting ring on/off.
Action: Emulation | Ringer Off.
Action Data: None.
Default Label: RngOf or Ringer Off.
Toggles: Yes.
Status Indication: Yes Required.

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 Self-Administer

Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Self-Administer
Allows a user to program features against other programmable buttons themselves.
Appearance can no longer be used to create call appearance buttons. Similarly, existing call
appearance button cannot be overwritten using any of the other Admin button functions.
User's with a log in code will be prompted to enter that code when they use this button action.
T3 phone users can access a similar set of functions for button programming, see T3 Phone Self-
Administration.
On 4412D+, 4424D+, 4612IP, 4624IP, 6408D, 6416D, 6424D phones:
• Admin can be permanently accessed via Menu , , , Admin. See Using a Menu Key.
• Admin1 can be permanently accessed via Menu , Menu , , ProgA, , , DSS.
Action: Emulation | Self-Administer.
Action Data: See below.

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Button Programming Actions

Value T-Series and M-Series phones Other Phones

None The Feature *3 process is started with If no value is set, the button allows user
an alternate set of possible functions. programming of the following emulation
actions:
• Abbreviated Dial
• Abbreviated Dial Program
• Account Code Entry
• AD Suppress
• Automatic Callback
• Break Out
• Call Forwarding All
• Call Park
• Call Park and Page
• Call Park To Other Extension
• Call Pickup
• Call Pickup Any
• Conference Meet Me
• Dial Paging
• Directed Call Pickup
• Directory
• Drop
• Group Paging
• Headset Toggle
• Hook Flash
• Internal Auto-Answer
• Ringer Off
• Self-Administer
• Send All Calls
• Set Absent Text
• Set Hunt Group Night Service
• Time of Day
• Timer
• Twinning
Table continues…

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 Self-Administer

Value T-Series and M-Series phones Other Phones

1 The Feature *1 process is started for If 1 is entered as the telephone number,
assigning Abbreviated Dial button. allows user programming of the
following system functions.
• Abbreviated Dial
• Group
• CPark
• User
• Flash Hook
2 The Feature *6 process is started for If 2 is entered, the button can be used
setting the ring type. for viewing details of the control unit
type and its software version. This
option is available. If the user has a log
in code set, they will be prompted to
enter that code. System phone users
(see System Phone Features on
page 715) can also use the button to
manually set the system's date and
time.
3 The option 3 is used with M-Series and Not used.
T-Series sets to enable display contrast
control.

Default Label: Admin or Self Administer.

Toggles: No.
Status Indication: No.
User Admin: Yes.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 3600 Series: 4600 Series: [1] 9040: T3/T3 IP Series:
[2]
1400 Series: 3700 Series: 5400 Series: [1] 9500 Series:
1600 Series: 3810: 5600 Series: [1] 9600 Series:
20 Series: 4100 Series: 6400 Series: M-Series:
2400 Series: 4400 Series: 7400 Series: T-Series:

1. Not 1403, 1603, 2402, 5402, 4601, 4602, 5601 and 5602.
2. See T3 Phone Self-Administration.

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Button Programming Actions

Send All Calls

Sets the user's extension into 'Do Not Disturb' mode. Callers, other than those on the user's do not
disturb exception list, receive busy or are diverted to the users voicemail mailbox. Note that with a
call already connected and other calls already alerting, enabling Do Not Disturb will not affect
those calls already existing. For full details of see Do Not Disturb.
When on, most phones display an N on the display. This function and the Do Not Disturb On
function work in parallel, ie. setting one sets the other.
Action: Emulation | Send All Call.
Action Data: None.
Default Label: SAC or Send All Calls.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays .
• DSS Link LED: On when active.

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 Set Absent Text

Set Absent Text

This feature can be used to select the user's current absence text. This text is then displayed to
internal callers who have suitable display phones or applications. It doesn't changes the users
status. The absence text message is limited to 128 characters. Note however that the amount
displayed will depend on the caller's device or application.
The text is displayed to callers even if the user has forwarded their calls or is using follow me.
Absence text is supported across a multi-site network.
Note:
The user still has to select Set or Clear on their phone to display or hide the text.
Action: Advanced | Set | Set Absent Text.
Action Data: Optional.
Note:
On certain phones (1400, 1600, 9500 and 9600), if the button is set without any Action Data,
the user is prompted to select their absence text and switch it on/off through a menu shown on
the phone display.
The telephone number should take the format "y,n,text" where:
• y = 0 or 1 to turn this feature off or on repesctively.
• n = the number of the absent statement to use:
0 = None. 4 = Meeting until. 8 = With cust. til.
1 = On vacation until. 5 = Please call. 9 = Back soon.
2 = Will be back. 6 = Don't disturb until. 10 = Back tomorrow.
3 = At lunch until. 7 = With visitors until. 11 = Custom.

text = any text to follow the absent statement..

Default Label: Absnt or Absence Text.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
Table continues…

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Button Programming Actions

1200 Series: No 3600 Series: No 5400 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 4600 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] [1] No
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602.

Set Account Code

Dials an account code and then returns dial tone for the user to dial a number. Can also be used
to enter an account code after a call has been connected.
Action: Advanced | Set | Set Account Code..
Action Data: Account code or blank. If blank, the user is prompted to dial an account code after
pressing the button. This option is not supported on XX02 phone modules.
Default Label: Acct or Account Code.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays 1234.
• DSS Link LED: None.

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 Set Hunt Group Night Service

Set Hunt Group Night Service

Puts the specified hunt group into Night Service mode. Calls to a group set to night service,
receive busy or are diverted to voicemail if available or are diverted to the group's night service
fallback group if set.
Setting and clearing hunt group night service can be done using either manual controls or using a
system time profile. The use of both methods to control the night service status of a particular hunt
group is not supported.
This function is not supported between systems in a multi-site network. It can only be used by a
user currently logged onto the same system as hosting the hunt group.
Action: Advanced | Set | Set Hunt Group Night Service.
Action Data: Hunt group extension number.
Release 4.0+: If left blank, the button will affect all hunt groups of which the user is a member.
The Set Hunt Group Night Service and Clear Hunt Group Night Service short code and button
features can be used to switch an SSL VPN service off or on respectively. The service is indicated
by setting the service name as the telephone number or action data. Do not use quotation marks.
Default Label: HGNS+ or HG Night Service.
Toggles: Yes.
Status Indication: Yes Required. If the button is blank (no specific hunt group) it will indicate on if
any one of the hunt groups of which the user is a member is set to night service. If the button is
set for multiple hunt groups it will indicate on if any one of those groups is set to night service.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
Table continues…

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Button Programming Actions

1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones: Supported on Avaya T3 Classic, T3 Comfort phones and DSS Link units only.
• T3 Classic/T3 Comfort icon: Displays followed by the group number. The
background uses the same settings as the LED below.
• DSS Link LED: On when all related groups are in night service. Slow flash if related
hunt groups are in mixed states.

Set Hunt Group Out Of Service

Puts the specified hunt group into Out of Service mode. Calls to a group set to out of service
receive busy or are diverted to voicemail if available or are diverted to the group's out of service
fallback group if set.
This function can be used to used to override hunt groups already set to night service mode by an
associated time profile.
Action: Advanced | Set | Set Hunt Group Out of Service.
Action Data: Hunt group extension number.
If left blank, the button will affect all hunt groups of which the user is a member.
Default Label: HGOS+ or HG Out of Service.
Toggles: Yes.
Status Indication: Yes Required. If the button is blank (no specific hunt group) it will indicate on if
any one of the hunt groups of which the user is a member is set out of service. If the button is set
for multiple hunt groups it will indicate on if any one of those groups is set out of service.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.

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 Set Inside Call Seq

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays – followed by the group number. The background uses
the same settings as the LED below.
• DSS Link LED: On when set. On when all related groups are out of service. Slow flash
if related hunt groups are in mixed states.

Set Inside Call Seq

This feature allows the user to select the ringing used on their analog extension for internal calls.
The number entered corresponds to the ring pattern required. This is 0 for Default Ring, 1 for
RingNormal, 2 for RingType1, etc. For more information on selectable ringing patterns, see Ring
Tones. Use of this short code function is applicable to analog phone users only. The distinctive
ringing pattern used for other phones is set by the phone type.

Set Night Service Destination

This button allows the user to change the Night Service target of a hunt group. The button user
does not have to be a member of the hunt group. In a multi-site network this function can be used
for hunt groups on remote systems.
Changing the destination does not affect calls already ringing at the hunt groups previous night
service destination.
Action: Advanced | Set | Set Night Service Group.
Action Data: Hunt group extension number. This is the group for which the night service
destination is being set.
Default Label: SetNSG or HG NS Group.
Toggles: No.

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Button Programming Actions

Status Indication: No.

User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
On T3 phones this option is accessible through the phone's menus.

Set No Answer Time

Allows the user to change their no answer time setting. This is the time calls ring before going to
voicemail or following the user's divert on no answer setting if set on.
In situations where call coverage is also being used, the user's no answer time must be greater
than their individual coverage time for coverage to occur.
Action: Advanced | Set | Set No Answer Time.
Action Data: Time in seconds.
Default Label: NATim or No Answer Time.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
Table continues…

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 Set Out of Service Destination

1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Set Out of Service Destination

This button allows the user to change the Out of Service target of a hunt group. The button user
does not have to be a member of the hunt group. In a multi-site network this function can be used
for hunt groups on remote systems.
Changing the destination does not affect calls already ringing at the hunt groups previous Out of
Service destination.
Action: Advanced | Set | Set Out of Service Group.
Action Data: Hunt group extension number. This is the group for which the night service
destination is being set.
Default Label: SetOOSG or HG OS Group.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
On T3 phones this option is accessible through the phone's menus.

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Button Programming Actions

Set Outside Call Seq

This feature allows the user to select the ringing used on their analog extension for external calls.
The number entered corresponds to the ring pattern required. This is 0 for Default Ring, 1 for
RingNormal, 2 for RingType1, etc. For more information on selectable ringing patterns, see Ring
Tones. Use of this short code function is applicable to analog phone users only. The distinctive
ringing pattern used for other phones is set by the phone type.

Set Ringback Seq

This feature allows the user to select the ringing used on their analog extension for ringback calls.
The number entered corresponds to the ring pattern required. This is 0 for Default Ring, 1 for
RingNormal, 2 for RingType1, etc. For more information on selectable ringing patterns, see Ring
Tones. Use of this short code function is applicable to analog phone users only. The distinctive
ringing pattern used for other phones is set by the phone type.

Set Wrap Up Time

Allows users to change their Wrap-up Time (User | Telephony | Call Settings) setting.
Other phones or applications monitoring the user's status will indicate the user as still being busy
(on a call).
Hunt group calls will not be presented to the user.
If the user is using a single line set, direct calls also receive busy treatment. If the user is using a
mutli-line set (multiple call appearances), direct calls to them will ring as normal.
It is recommended that this option is not set to less than the default of 2 seconds. 0 is used to
allow immediate ringing.
For users set as an CCR Agent, the After Call Work Time (User | Telephony | Supervisor Settings)
setting should be used.
Action: Advanced | Set | Set Wrap Up Time.
Action Data: Time in seconds. Range 0 to 99999 seconds.
Default Label: WUTim or Wrap-up Time.
Toggles: No.
Status Indication: No.
User Admin: No.

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 Speed Dial

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Speed Dial
When pressed, the button invokes the same process as dialing Feature 0.
If Feature 0 is followed by a 3-digit index number in the range 000 to 999, the system directory
entry with the matching index number is dialed.
If Feature 0 is followed by * and a 2-digit index number in the range 00 to 99, the personal
directory entry with the matching index number is dialed. Note: Release 10.0 allows users to have
up to 250 personal directory entries. However, only 100 of those can be assigned index numbers.
Action: Advanced | Dial | Speed Dial.
Action Data: None.
Default Label: SpdDial.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series 4100 Series: No 6400 Series: No D100: No
1100 Series: Yes 2400 Series: No 4400 Series: No 7400 Series: No M-Series: Yes
1200 Series: Yes 3600 Series: No 4600 Series: No 9040: No T-Series: Yes
1400 Series: No 3700 Series: No 5400 Series: No 9500 Series: No T3/T3 IP Series:
No
1600 Series: No 3810: No 5600 Series: No 9600 Series: No

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Button Programming Actions

Stamp Log
The stamp log function is used to insert a line into any System Monitor trace that is running. The
line in the trace indicates the date, time, user name and extension plus additional information. The
line is prefixed with LSTMP: Log Stamped and a log stamp number. When invoked from a Avaya
phone with a display, Log Stamped# is also briefly displayed on the phone. This allows users to
indicate when they have experienced a particular problem that the system maintainer want them
to report and allows the maintainer to more easily locate the relevant section in the monitor trace.
The log stamp number is set to 000 when the system is restarted. The number is then
incremented after each time the function is used in a cycle between 000 and 999. Alternately if
required, a specific stamp number can be assigned to the button or short code being used for the
feature.
Action: Advanced | Miscellaneous | Stamp Log.
Action Data: Optional. Blank or any 3 digit number.
Default Label: Stamp Log.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series: [1]
1200 Series: 3600 Series: 4600 Series: 9040: T-Series: [1]
1400 Series: 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: 3810: 5600 Series: 9600 Series:

1. Not supported on T7000, T7100, M7100, M7100N and the Audio Conferencing Unit (ACU).

Stored Number View

Not supported. Provided for CTI emulation only. Allows a user to view the contents of any
programmed feature button.
Action: Emulation | Stored Number View.
Action Data: None.
Default Label: BtnVu.
Toggles: No.

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 Suspend Call

Status Indication: No.

User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Suspend Call
Uses the Q.931 Suspend facility. Suspends the incoming call at the ISDN exchange, freeing up
the ISDN B channel. The call is placed in exchange slot 0 if a slot number is not specified. Only
available when supported by the ISDN exchange.
Action: Advanced | Suspend | Suspend.
Action Data: Exchange slot number or blank (slot 0).
Default Label: Suspe.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

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Button Programming Actions

Suspend CW
Uses the Q.931 Suspend facility. Suspends the incoming call at the ISDN exchange and answer
the call waiting. The call is placed in exchange slot 0 if a slot number is not specified. Only
available when supported by the ISDN exchange.
Action: Advanced | Suspend | Suspend CW.
Action Data: Exchange slot number or blank (slot 0).
Default Label: SusCW.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Swap CLID Name/Number

Allows the user to toggle between Caller Name and Caller ID.
Action: Emulation | Swap CLID Name/Number
Action Data: None.
Default Label:
Toggles: Yes.
Status Indication: No.
User Admin: No.
Phone Support: Note that support for particular phone models is also dependant on the system
software level.

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 Time of Day

Analog: No 20 Series: No 4100 Series: No 64000: No D100: No

[1]
1100 Series: No 2400 Series: No 4400 Series: No 7400 Series: No M-Series: Yes
1200 Series: No 3600 Series: No 4600 Series: No 9040 Series: No T-Series: Yes
1400 Series: No 3700 Series: No 5400 Series: No 9500 Series: No T3/T3 IP Series:
No
1600 Series: No 3810 Series: No 5600 Series: No 9600 Series: No

Time of Day
Displays the time and date on the user's telephone. This function is ignored on those Avaya
phones that display the date/time by default.
Action: Emulation | Time of Day.
Action Data: None.
Default Label: TmDay.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: : No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] [1] No
Table continues…

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Button Programming Actions

1600 Series: Yes 3810: No 5600 Series: Yes 9600 Series: No

[1] [1]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602 models.

Time Profile
You can manually override a time profile. The override settings allow you to mix timed and manual
settings.
The button indicator will show the Time Profile state and pressing the button will present a menu
with five options and an indication of the current state. The menu options are listed below.
Menu Option Description
Timed Operation No override. The time profile operates as configured.
Active Until Next Timed Inactive Use for time profiles with multiple intervals. Select to
make the current timed interval active until the next
inactive interval.
Inactive Until Next Timed Active Use for time profiles with multiple intervals. Select to
make the current active timed interval inactive until the
next active interval.
Latch Active Set the time profile to active. Timed inactive periods are
overridden and remain active.
Latch Inactive Set the time profile to inactive. Timed active periods are
overridden and remain inactive.

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 Time Profile

Active
Inactive
7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Active until next Inactive

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Inactive untill next Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Active

7 AM 9 AM 11 AM 1 PM 3 PM 5 PM 7 PM

Time Profile
Override

Set Latch Inactive

Action: Emulation | Time Profile

Action Data: Time profile name.
Default Label: TP or Time Profile
Toggles: No.
Status Indication:
Status 1400, 1600, 9600 Series 9608, 9611, J139, 9621, 9641
J169, J179
On Green Red on Green On Green
Off Off Off Off Grey

User Admin: No
Phone Support: Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: No 4100 Series: No 6400 Series: No D100: No
Table continues…

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Button Programming Actions

1100 Series: No 2400 Series: No 4400 Series: No 7400 Series: No M-Series: No

1200 Series: No 3600 Series: No 4600 Series: No 9040: No T-Series: No
1400 Series: Yes 3700 Series: No 5400 Series: No 9500 Series: No T3/T3 IP Series:
No
1600 Series: Yes 3810: No 5600 Series: No 9600 Series: Yes

Timer
Starts a timer running on the display of the user's extension. The timer disappears when the user
ends a call.
This function can be used on Avaya phones (except 9600 Series) that display a call timer next to
each call appearance. The button will temporarily turn the call timer on or off for the currently
selected call appearance. The change only applies for the duration of the current call.
Action: Emulation | Timer.
Action Data: None.
Default Label: Timer.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 4600, 4400, 1400, 9600 9608, 9621, T-Series,
5400 5600 6400 1600, Series 9611, 9641 M-Series
Series Series Series 9500 J139,
Series J169,
J179
- On. <Label> <Label> Green on Off – – – On

- Off. <Label> <Label> Off Off – – – Off

User Admin: Yes.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] [1] No
Table continues…

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 Transfer

1600 Series: Yes 3810: No 5600 Series: Yes 9600 Series: Yes
[1] [1] [2]

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602 models.
2. Supported on 96x1, but not 96x0.

Transfer
This function is intend for use with Avaya M-Series and T-Series phones only. When pressed, the
button invokes the same transfer process as dialing Feature 70.
Action: Advanced | Call | Transfer.
Action Data: None.
Default Label: Xfer.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support This function is only supported on Avaya M-Series and T-Series phones.

Toggle Calls
Cycle between the user's current call and any held calls.
Action: Advanced | Call | Toggle Calls..
Action Data: None.
Default Label: Toggl.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
Table continues…

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Button Programming Actions

1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Twinning
This action can be used by user's setup for mobile twinning. This action is not used for internal
twinning.
While the phone is idle, the button allows the user to set and change the destination for their
twinned calls. It can also be used to switch mobile twinning on/off and indicates the status of that
setting.
When a call has been routed by the system to the user's twinned destination, the Twinning button
can be used to retrieve the call at the user's primary extension.
In configurations where the call arrives over an IP trunk and the outbound call is on an IP trunk,
multi-site network may optimise the routing and in this case the button may not be usable to
retrieve the call.
For user's setup for one-X Mobile Client, changes to their Mobile Twinning status made through
the system configuration or using a Twinning button are not reflected in the status of the
Extension to Cellular icon on their mobile client. However, changes to the Extension to Cellular
status made from the mobile client are reflected by the Mobile Twinning field in the system
configuration. Therefore, for one-X Mobile Client users, it is recommended that they control their
Mobile Twinning status through the one-X Mobile Client rather than through a Twinning button.
Mobile Twinning Handover When on a call on the primary extension, pressing the Twinning
button will make an unassisted transfer to the twinning destination. This feature can be used even
if the user's Mobile Twinning setting was not enabled.
During the transfer process the button will wink.
Pressing the twinning button again will halt the transfer attempt and reconnect the call at the
primary extension.
The transfer may return if it cannot connect to the twinning destination or is unanswered within the
user's configured Transfer Return Time (if the user has no Transfer Return Time configured, a
enforced time of 15 seconds is used).
Action: Emulation | Twinning.
Action Data: None.
Default Label: Twinning.
Toggles: Yes.

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 Unpark Call

Status Indication: Yes.

Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. Twinning Twinning Green on Red on Green on Green On
- Off. Twinning Twinning Off Off Off Grey Off
- Twinned Twinning Twinning Red on Red flash Red on Blue On
call at
secondary

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: M-Series:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: T-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T3/T3 IP Series:
1400 Series: [1] 3700 Series: 5400 Series: [1] 9500 Series:
1600 Series: [1] 3810: 5600 Series: [1] D100:

1. Not 1403, 1603, 2402, 4601, 4602, 5402, 5601 and 5602 models.

Unpark Call
This function is obsolete, since the Call Park function can be used to both park and retrieve calls
and provides visual indication of when calls are parked. Retrieve a parked call from a specified
system park slot.
Action: Advanced | Call | Unpark Call.
Action Data: System park slot number. This must match a park slot ID used to park the call.
Default Label: UnPark.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.

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Button Programming Actions

Analog: 20 Series: 4100 Series: 6400 Series: D100:

1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
M-Series/T-Series: The button is equivalent to Feature #74 <park slot number>.

User
Monitors whether another user's phone is idle or in use. The Telephone Number field should
contain the users name enclosed in double quotes. The button can be used to make calls to the
user or pickup their longest waiting call when ringing. On buttons with a text label, the user name
is shown.
The actions performed when the button is pressed will depend on the state of the target user and
the type of phone being used. It also depend on whether the user is local or on a remote multi-site
network system.
Phone Large display 1400, 1600, 2400, 4600, 5400, 5600, Other Phones or across a
9500, 9600, M-Series and T-Series Phones multi-site network
Idle Call the user. Whilst ringing the phone displays options to Callback (set an
automatic callback ) and Drop (end the call attempt).
Ringing • Call Pickup: Pickup the ringing call. Picks up the call.
• Call: Make a call to the user.
On a Call The following options are displayed (name lengths No action.
may vary depending on the phone display):
For 1400, 1600, 9500 and
• Call: Make a call to the user. 9600 Series phones, the
Call, Voicemail and
• Message: Cause a single burst of ringing on the
Callback options are
target phone. On some phones, when they end
supported.
their current call their phone will then display
PLEASE CALL and your extension number.
• Voicemail: Call the user's voicemail mailbox.
• Callback: Set an automatic callback.
Table continues…

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 User

For 1400, 1600, 9500 and 9600 Series phones the

following additional options are displayed:
• Drop Disconnect the user's current call.
• Acquire: Shown if able to intrude on the user. Take
control of the call.
• Intrude: Shown if able to intrude on the user.
Intrude into the call, turning it into a 3-way
conference.
• Listen: Shown if configured to be able to listen to
(monitor) the user. Start silent monitoring of the
user's call.

A User button can be used in conjunction with other buttons to indicate the target user when those
buttons have been configured with no pre-set user target. In cases where the other button uses
the phone display for target selection this is only possible using User buttons on an associate
button module.
The following changes have been made to the indication of user status via BLF (busy lamp field)
indicators such as a User button:
The status shown for a logged out user without mobile twinning will depend on whether they have
Forward Unconditional enabled.
• If they have Forward Unconditional enabled the user is shown as idle.
• If they do not have Forward Unconditional enabled they will show as if on DND.
The status shown for a logged out user with mobile twinning will be as follows:
• If there are any calls alerting or in progress through the system to the twinned destination, the
user status is shown as alerting or in-use as appropriate. This includes the user showing as
busy/in-use if they have such a call on hold and they have Busy on Held enabled.
• If the user enables DND through Mobile Call Control or one-X Mobile client, their status will
show as DND.
• Calls from the system direct to the user's twinned destination number rather than redirected
by twinning will not change the user's status.
Action: User.
Action Data: User name enclosed in "double-quotes".
Default Label: <the user name>.
Toggles: No.
Status Indication: Yes.

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Button Programming Actions

Status 2400 4600 4400 1400 9600 9608, 9621, T-Series,

Series, Series, Series, Series, Series 9611, 9641 M-Series
5400 5600 6400 1600 J139,
Series Series Series Series, J169,
9500 J179
Series
- Idle. Extn221 Extn221 Off Off Off Off Grey Off
- Extn221 Extn221 Green Red flash Red flash Red flash Blue Slow
Alerting. flash flash
- In Use/ Extn221 Extn221 Green on Red wink Red wink Red wink Blue Fast
Busy. flash
- DND Extn221 Extn221 Green on Red on Red on Red on Green On

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: Yes 4100 Series: No 6400 Series: Yes D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: Yes
[1]
1200 Series: No 3600 Series: Yes 4600 Series: Yes 9040: Yes T-Series: Yes
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] No [2]
1600 Series: Yes 3810: Yes 5600 Series: Yes 9600 Series: Yes
[1] [1]

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
T3 Phones:
• Classic/Comfort icon: Displays the user name.
• DSS Link LED: On when busy, flashing when call alerting user.

Visual Voice
This action provides the user with a menu for access to voicemail mailboxes. The menu provides
the user with options for listening to messages, leaving messages and managing the mailbox. If
no action data is specified, then it is the user's mailbox. Action Data can be used to specify the
mailbox of another user or group.

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 Visual Voice

Note:
You can also use the “H” and “U” user source numbers to add another mailbox to your Visual
Voice menu. See User | Source Numbers
If the Action Data has been configured, pressing the button for an incoming call or while a call is
connected sends the call to the user mailbox specified in the action data. If no Action Data is
configured, the user is prompted to enter a mailbox.
On phones that have a display but do not support full visual voice operation as indicated below,
use of the button for user mailbox access using voice prompts and for direct to voicemail transfer
during a call is supported (does not include T3 and T3 IP phones).
Access to Visual Voice on supported phones can be triggered by the phone's MESSAGES button
rather than requiring a separate Visual Voice programmable button. This is done using the option
System | Voicemail | Messages button goes to Visual Voice.
Action: Emulation | Visual Voice.
Action Data: All local users and groups and all users and groups on systems in the network,
except for the user on which the button is being programmed.
Default Label: Voice.
Toggles: No.
Status Indication: When action data is configured, the status lamp provides a message waiting
indicator for the monitored mailbox.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: No 20 Series: No 4100 Series: No 6400 Series: No D100: No
1100 Series: No 2400 Series: Yes 4400 Series: Yes 7400 Series: No M-Series: No
[1]
1200 Series: No 3600 Series: No 4600 Series: Yes 9040: Yes T-Series: No
[1]
1400 Series: Yes 3700 Series: No 5400 Series: Yes 9500 Series: Yes T3/T3 IP Series:
[1] Yes [2]
1600 Series: Yes 3810: No 5600 Series: Yes 9600 Series: No
[1]

1. Not 1403, 1603, 2402, 5402, 4601, 4602, 5601 and 5602.
2. Takes the user direct to the listen part of Visual Voice. For the full Visual Voice menu
options the user should use Menu | Settings | Voicemail Settings.
Visual Voice Controls
The arrangement of options on the screen will vary depending on the phone type and display size.
Listen Access your own voicemail mailbox. When pressed the screen will show the number of
New, Old and Saved messages. Select one of those options to start playback of messages in that

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Button Programming Actions

category. Use the up arrow and arrow keys to move through the message. Use the options
below.
Listen Play the message.
Pause Pause the message playback.
Delete Delete the message.
Save Mark the message as a saved message.
Call Call the message sender if a caller ID is available.
Copy Copy the message to another mailbox. When pressed a number of additional options are
displayed.
Message Record and send a voicemail message to another mailbox or mailboxes.
Greeting Change the main greeting used for callers to your mailbox. If no greeting has been
recorded then the default system mailbox greeting is used.
Mailbox Name Record a mailbox name. This feature is only available on systems using
Embedded Voicemail.
Email This option is only shown if you have been configured with an email address for voicemail
email usage in the system configuration. This control allows you to see and change the current
voicemail email mode being used for new messages received by your voicemail mailbox. Use
Change to change the selected mode. Press Donewhen the required mode is displayed. Possible
modes are:
Password Change the voicemail mailbox password. To do this requires entry of the existing
password.
Voicemail Switch voicemail coverage on/off.

Voicemail Collect
Connects to the voicemail server. The telephone number must indicate the name of the Voicemail
box to be accessed, eg. "?Extn201" or "#Extn201". The ? indicates "collect Voicemail" and the #
indicates "deposit Voicemail". This action is not supported by voicemail using Intuity emulation
mode.
When used with Voicemail Pro, names of specific call flow start points can also be used to directly
access those start points via a short code. In these cases ? is not used and # is only used if
ringing is required before the start points call flow begins.
Action: Advanced | Voicemail | Voicemail Collect.
Action Data: See above.
Default Label: VMCol or VMail Collect.
Toggles: No.
Status Indication: No.

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 Voicemail Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.
M-Series/T-Series: For access to the users own mailbox, this button is equivalent to
Feature 65 and Feature 981.

Voicemail Off
Disables the user's voicemail box from answering calls that ring unanswered at the users
extension. This does not disable the user's mailbox and other methods of placing messages into
their mailbox.
This button function is obsolete as the Voicemail On function toggles on/off.
Action: Advanced | Voicemail | Voicemail Off.
Action Data: None.
Default Label: VMOff.
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
Table continues…

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Button Programming Actions

1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Voicemail On
Enables the user's voicemail mailbox to answer calls which ring unanswered or arrive when the
user is busy.
Action: Advanced | Voicemail | Voicemail On.
Action Data: None.
Default Label: VMOn or VMail On.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.
2. Limited support on some specific T3 phone models as detailed below.

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 Voicemail Ringback Off

T3 Phones: Supported on Avaya T3 Classic, Comfort and Compact phones for Release
4.2+.
• Classic/Comfort icon: Displays . The background uses the same settings as the
LED below.
• DSS Link LED: On when set.

Voicemail Ringback Off

Disables voicemail ringback to the user's extension. This button function is obsolete as the
Voicemail Ringback On function toggles on/off.
Action: Advanced | Voicemail | Voicemail Ringback Off.
Action Data: None.
Default Label: VMRB-
Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 2402, 4601, 4602, 5601 and 5602 except where 4602 is supported on
Release 2.1 and 3.0DT software.

Voicemail Ringback On
Enables voicemail ringback to the user's extension. Voicemail ringback is used to call the user
when they have new voicemail messages in their own mailbox or a hunt group mailbox for which
they have been configured with message waiting indication.
The ringback takes place when the user's phone returns to idle after any call is ended.
Action: Advanced | Voicemail | Voicemail Ringback On.

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Button Programming Actions

Action Data: None.

Default Label: VMRB+ or VMail Ringback.
Toggles: Yes.
Status Indication: Yes.
Status 2400, 5400 4600, 5600 1400, 9600 9608, 9621, 9641 T-Series,
Series Series 1600, Series 9611, M-Series
4400, J139,
6400, 9500 J169, J179
Series
- On. <Label> <Label> Green on Red on Green on Green On
- Off. <Label> <Label> Off Off Off Grey Off

User Admin: No.

Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: [1] 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: [1] 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: T3/T3 IP Series:
1600 Series: [1] 3810: 5600 Series: [1] 9600 Series:

1. Not 1403, 1603, 4601, 4602, 5601 and 5602 except where 4602 is supported on Release
2.1 and 3.0DT software.

Whisper Page
This feature allows you to intrude on another user and be heard by them without being able to
hear the user's existing call which is not interrupted. For example: User A is on a call with user B.
When user C intrudes on user A, they can be heard by user A but not by user B who can still hear
user A. Whisper page can be used to talk to a user who has enabled private call.
The ability to intrude and be intruded is controlled by two configuration settings, the Can Intrude
(User | Telephony | Supervisor Settings) setting of the user intruding and the Cannot Be Intruded
(User | Telephony | Supervisor Settings) setting of target being intruded on. The setting of any
other internal party is ignored. By default, no users can intrude and all users are set to cannot be
intruded.
The system support a range of other call intrusion methods in addition to this feature.
Action: Advanced | Call | Whisper Page.
Action Data: User number or name or blank for entry when pressed.

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 Whisper Page

Default Label: Whisp or Whisper Page.

Toggles: No.
Status Indication: No.
User Admin: No.
Phone Support Note that support for particular phone models is also dependant on the system
software level.
Analog: 20 Series: 4100 Series: 6400 Series: D100:
1100 Series: 2400 Series: 4400 Series: 7400 Series: M-Series:
1200 Series: 3600 Series: 4600 Series: 9040: T-Series:
1400 Series: [1] 3700 Series: 5400 Series: 9500 Series: [2] T3/T3 IP Series:
[2]
1600 Series: [1] 3810: 5600 Series: 9600 Series:

1. Not 1403, 1603.

2. Not supported on non-IP telephones when using a headset.

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Chapter 21: Appearance Button Operation

Many Avaya phones supported on system have a programmable keys or buttons (the terms 'key'
and 'button' mean the same thing in this context). Various actions can be assigned to each of these
keys, allowing the phone user to access that action.
Many of the phones also have indicator lamps next to the programmable buttons. These lamps are
used to indicate the status of the button, for example 'on' or 'off'. On other phones the programmable
buttons use an adjacent area of the phones display to show status icons and text labels for the
buttons.
Example The example below shows the display and programmable buttons on an Avaya 5421
phone where a number of programmable features have been assigned to the user.

This type of phone displays text labels for the programmed features. On other phones a paper label
may have to be updated to indicate the programmed feature.
The system supports the following 'appearance' actions - Call Appearance, Bridged Appearance,
Line Appearance and Call Coverage Appearance. These actions can be assigned to the
programmable buttons on a user's phone. Those 'appearance' buttons can then be used to answer,
share, switch between and in some case make calls. This type of call handling is often called 'key
and lamp mode'.
This document covers the programming and operation of phones using the appearance functions.
Details of the other actions that can be assigned to programmable keys are covered in Button
Programming.

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 Appearance Button Features

Note:
For all the examples within this documentation, it is assumed that Auto Hold is on and Answer
Pre-Select is off unless otherwise stated.
The text shown on phone displays are typical and may vary between phone types, locales and
system software releases.
Related links
Appearance Button Features on page 1059
Call Appearance Buttons on page 1060
Bridged Appearance Buttons on page 1065
Call Coverage Buttons on page 1070
Line Appearance Buttons on page 1074
Selected Button Indication on page 1080
Idle Line Preference on page 1081
Ringing Line Preference on page 1083
Answer Pre-Select on page 1085
Auto Hold on page 1087
Ring Delay on page 1087
Delayed Ring Preference on page 1089
Collapsing Appearances on page 1091
Joining Calls on page 1092
Multiple Alerting Appearance Buttons on page 1094
Twinning on page 1095
Busy on Held on page 1096
Reserving a Call Appearance Button on page 1096
Logging Off and Hot Desking on page 1096
Applications on page 1097
Programming Appearance Buttons on page 1097

Appearance Button Features

Appearance functions are only supported on Avaya phones which have programmable buttons
and also support multiple calls. Appearance functions are also only supported on those buttons
that have suitable adjacent indicator lamps or a display area. Appearance buttons are not
supported across a multi-site network.
Related links
Appearance Button Operation on page 1058

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Appearance Button Operation

Call Appearance Buttons

Call appearance buttons are used to display alerts for incoming calls directed to a user's extension
number or to a hunt group of which they are a member. Call appearance buttons are also used to
make outgoing calls.
By having several call appearance buttons, a user is able to be alerted about several calls, select
which call to answer, switch between calls and take other actions.

When all the user's call appearance buttons are in use or alerting, any further calls to their
extension number receive busy treatment. Instead of busy tone, the user's forward on busy is
used if enabled or otherwise voicemail if available.
Call appearance buttons are the primary feature of key and lamp operation. None of the other
appearance button features can be used until a user has some call appearance button
programmed[1].
There are also addition requirements to programming call appearance buttons:
Call appearance buttons must be the first button programmed for the user.
Programming a single call appearance button for a user is not supported. The normal default is 3
call appearances per user except on phones where only two physical buttons are available.
[1] For Release 4.2+, T3 phones support the use of Line Appearance buttons. These can be
programmed against buttons on T3 phones without requiring call appearance buttons. See T3
Phone Line Appearances.
Related links
Appearance Button Operation on page 1058
Call Appearance Example 1 on page 1060
Call Appearance Example 2 on page 1061
How are Call Appearance Buttons Treated? on page 1062
Call Appearance Button Indication on page 1063

Call Appearance Example 1

In this example, the user has multiple call appearance buttons.

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 Call Appearance Buttons

Phone Idle The phone is currently idle.

First Call Alerts A call arrives. It alerts against the

first available call appearance button. Pressing that
button will answer the call.

Call Answered The call is now connected.

Second Call Alerts A second call arrives whilst the

first is still connected. It alerts against the next
available call appearance button. As the user has a
call in progress, the alert gives just a single ring and
briefly display details of the caller.

Pressing the Second Call Appearance Pressing

the second call appearance button will hold the first
call and answer the second.

Related links
Call Appearance Buttons on page 1060

Call Appearance Example 2

In this example, the user will use their call appearances to make two calls and start a conference
between those calls.
Initial Call The user has a call in progress, shown
on their first call appearance button. It is decided to
conference another user into the call.

Table continues…

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Appearance Button Operation

Make Conference Enquiry Pressing the

CONFERENCE button on the users phone
automatically places the current call on hold and
takes the phone off hook on the next available call
appearance.

Enquiry in Progress The other extension has been

dialed and invited to join a conference call. The user
presses the CONFERENCE button on their
phone again.

Conference Starts The conference call has

started. The separate call appearances have
collapsed to a single appearance that represents
the conference.

Related links
Call Appearance Buttons on page 1060

How are Call Appearance Buttons Treated?

For incoming calls
Call Waiting settings are ignored except for hunt group call waiting where the call waiting tone is
replaced by an alert on a call appearance button if available.
Follow Me, Forward Unconditional and Forward Hunt Group Calls are used when set.
If Do Not Disturb is set, only calls from numbers in the user's Do Not Disturb Exception list will
alert if a call appearance is available.
Busy status
For calls direct to the user's extension number The user is busy when all their available call
appearances are in use. Instead of busy tone, the user's forward on busy is used if enabled or
otherwise voicemail if available.
For calls to a hunt group of which the user is a member The user is busy to further hunt group
calls when they have any appearance button in use on their phone. The only exception is calls to
a collective hunt group with call waiting.
In both cases above, even when busy, the user may still receive alerts on other appearance
buttons.
For outgoing calls
Outgoing calls are treated exactly the same as calls made by non-appearance button users.

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 Call Appearance Buttons

External Calls made on a call appearance, which route out on a line for which the user also has a
line appearance, will remain on the call appearance. The line appearance will indicate 'in use
elsewhere'.
For call appearance buttons matched by a bridged appearance button
If the bridged appearance is used to make or answer calls, the state of the call appearance will
match that of the bridged appearance.
If the call is put on hold by the bridged appearance user, the call appearance will show 'on hold
elsewhere'.
Other
Held/Parked Call Timeout If the user has parked a call, the parked call timer only starts running
when the user is idle rather than on another call.
Incoming calls routed directly to the user as the incoming call routes destination on a line for which
the user also has a line appearance, will only alert on the line appearance. These calls do not
follow any forwarding set but can be covered.
Related links
Call Appearance Buttons on page 1060

Call Appearance Button Indication

On phones with a text display area next to the button, by default a=, b= and so on is displayed.
This can be replaced by another label if required.
When the user is not connected to a call, the button indicated as selected is the button that will be
used if the user goes off hook without pressing an appearance button. When a user is connected
to a call, that call is the selected button.
The following table shows how the different states of call appearance buttons (alerting, held, etc)
are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay.
Icon Button Dual LED Button Call Appearance Button State
Idle The call appearance is not in
use and is not currently selected.
Red off,
Green off.
Idle + Selected The call
appearance is not in use but is
Red on, the current selected button that
Green off. will be used if the user goes off
hook.
Table continues…

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Appearance Button Operation

Alerting The matching call

appearance is alerting for an
Flashing icon. Red off, incoming call. This is
Green steady flash. accompanied by ringing. If the
user is already on a call, only a
single ring is given.
Alerting + Selected As above
but Ringing Line Preference has
Flashing icon. Red on, made this the user's current
Green steady flash. selected button.
In Use Here The user has a call
connected on the call appearance
Red on, or is dialing.
Green on.
In Use Elsewhere The call
appearance button is in use on a
Red off, bridged appearance.
Green on.
On Hold Here The call has been
put on hold by this user.
Red off,
Green fast flash.
On Hold Pending Transfer
Applies to 1400, 1600, 9500 and
Red fast flash, 9600 Series phones in Release
Green fast flash 8.1 and higher.
On Hold Elsewhere A call on a
bridged appearance button
Red off, matched to the call appearance
Green intermittent flash. has been put on hold. Calls on a
call appearance that are put on
hold by another user will continue
to show connected lamp status,
though the phone display will
indicate a held call.
Inaccessible The button pressed
is not accessible. The call is still
Icon Red off, dialing, ringing or cannot be
flashes off. Green broken flash. bridged into.

Related links
Call Appearance Buttons on page 1060

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 Bridged Appearance Buttons

Bridged Appearance Buttons

A bridged appearance button shows the state of one of another user's call appearance buttons. It
can be used to answer or join calls on that user's call appearance button. It can also be used to
make a call that the call appearance user can then join or retrieve from hold.

When the user's call appearance button alerts, any associated bridged appearance buttons on
other user's phones also alert. The bridged appearance buttons can be used to answer the call on
the call appearance button user's behalf.
When the call appearance button user answers or makes a call, any associated bridged
appearance buttons on other users' phones show the status of the call, ie. active, on hold, etc.
The bridged appearance button can be used to retrieve the call if on hold or to join the call if active
(subject to intrusion permissions).
Note Bridged appearance buttons are different from the action of bridging into a call (joining a
call). See Joining Other Calls (Bridging).
Bridged appearance buttons are not supported between users on different systems in a multi-site
network.
Related links
Appearance Button Operation on page 1058
Bridged Appearance Example 1 on page 1065
Bridged Appearance Example 2 on page 1066
Bridged Appearance Example 3 on page 1067
How are Bridged Appearances Treated? on page 1068
Bridged Appearance Button Indication on page 1068

Bridged Appearance Example 1

In this example, one user is able to see the status of the other user's call appearances, and when
necessary answer calls for the other user. Both users have Ringing Line Preference and Auto
Hold on.

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Appearance Button Operation

Call Appearance User Bridged Appearance User Both Phone Idle Our user has
bridged appearance buttons that
match a colleague's call
appearances buttons.

First Call The colleague has a

call alerting on their first call
appearance button. It also alerts
on our user's first bridged
appearance button.

Call Answered The colleague

has answered the call. The
bridged appearance indicates 'in
use elsewhere'.

Second Call Another call alerts

at the colleagues phone and
again is mirrored on our user's
second bridged call appearance
button.

Call Answered Our user has

gone off hook and answered the
incoming call alerting on the
bridged call appearance.

Related links
Bridged Appearance Buttons on page 1065

Bridged Appearance Example 2

In this example, the bridged appearance user makes a call on behalf of the call appearance user.
Once the call is connected, they put it on hold. The call appearance user is able to take the call off
hold using their call appearance button. Both users have Ringing Line Preference and Auto
Hold on.
Call Appearance User Bridge Appearance User Both Phones Idle Our user has
bridged appearance buttons that
match a colleague's call
appearances buttons.

Table continues…

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 Bridged Appearance Buttons

Bridged User Makes Call Our

user has pressed a bridged
appearance and made a call on it.
The matching call appearance
shows 'in use elsewhere'.

Call Put on Hold Having made

the call, the bridged user puts it
on hold. The matching call
appearance indicates 'on hold
elsewhere'.

Call Taken Off Hold By pressing

the call appearance, the first user
has answered the held call. The
bridged appearance user returns
to idle.

Related links
Bridged Appearance Buttons on page 1065

Bridged Appearance Example 3

In this example, a call is passed from the call appearance user to the bridged appearance user.
Both users have Ringing Line Preference and Auto Hold on.
Bridged Appearance User Call on Colleague's Phone The call appearance
user has answered a call on one of their call
appearances. The bridged appearance user's
matching bridged appearance shows 'in use
elsewhere'.

Call Held by Colleague The call appearance user

has put the call on hold and called the bridged
appearance user. The first bridged call appearance
shows a call 'on hold elsewhere' whilst the second
matches the call between users.

Enquiry Call Between Colleagues By going off

hook, the bridged appearance user has answered
the call from the call appearance user. They are
asked to pickup the call on the colleagues first call
appearance.

Table continues…

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Appearance Button Operation

Call Taken Off Hold Pressing the first bridged

appearance button takes that call off hold and
connects it to the bridged appearance user.
In this example, Auto Hold is not set for the
system, so pressing the bridged appearance button
disconnected the call from the colleague.
If Auto Hold had been set, the colleague's call
would have been put on hold until they hung up.

Related links
Bridged Appearance Buttons on page 1065

How are Bridged Appearances Treated?

Bridged appearance buttons operate in parallel with their matching call appearance button.
Whose user settings control the call? Until answered on a bridged appearance button, calls
alerting on a bridged appearance button follow the settings of the user or hunt group to which the
call was originally directed.
If the call appearance is in use, any matching bridged appearance will indicate the same.
If a bridged appearance is in use, the call appearance it matches will indicate the same.
The bridge appearance will only alert if the call appearance is alerting. For example, direct
intercom and paging call to the call appearance will show on the bridged appearance but will not
give any audible alert.
If the bridged appearance user put the call on hold, the call appearance indicates 'on hold
elsewhere'.
Bridged appearances to a user who has logged out, or has logged into a phone without
appearance buttons will not operate.
If the bridged appearance user has 'do not disturb' (DND) enabled, the bridge appearance button
icon or lamps operates but alerting and ringing line preference selection is not applied unless the
caller is in the user’s DND exception list.
Bridged appearance buttons are not supported between users on different systems in a multi-site
network.
Related links
Bridged Appearance Buttons on page 1065

Bridged Appearance Button Indication

On phones with a text display area next to the button, the name of the bridged user and the label
from the bridged user's call appearance key are displayed.

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 Bridged Appearance Buttons

The following table shows how the different states of bridged appearance buttons (alerting, held,
etc) are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay.
Icon Button Dual LED Button Bridge Appearance Button State
Idle The bridged appearance is
not in use.
Red off,
Green off.
Alerting The matching call
appearance is alerting for an
Flashing icon. Red off, incoming call. This is
Green steady flash. accompanied by ringing. If the
user is already on a call, only a
single ring is given.
Alerting + Selected As above
but Ringing Line Preference has
Flashing icon. Red on, made this the user's current
Green steady flash. selected button.
In Use Elsewhere The matching
call appearance button is in use.
Red off,
Green on.
In Use Here The user has made
a call or answered a call on the
Red on, bridged appearance, or bridged
Green on. into it.
On Hold Here The call has been
put on hold by this user.
Red off,
Green fast flash.
On Hold Elsewhere The call on
that call appearance has been put
Red off, on hold by another user.
Green intermittent flash.
Icon flashes off. Inaccessible The button pressed
is not usable. The call is still
Red off, dialing, ringing or cannot be
Green broken flash. bridged into.

Related links
Bridged Appearance Buttons on page 1065

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Appearance Button Operation

Call Coverage Buttons

Call coverage allows a user to be alerted when another user has an unanswered call.

The user being covered does not necessarily have to be a key and lamp user or have any
programmed appearance buttons. Their Individual Coverage Time setting (default 10 seconds)
sets how long calls will alert at their extension before also alerting on call coverage buttons set to
that user.
The user doing the covering must have appearance buttons including a call coverage appearance
button programmed to the covered users name.
Call coverage appearance buttons are not supported between users on different systems in a
multi-site network.
Related links
Appearance Button Operation on page 1058
Call Coverage Example 1 on page 1070
Call Coverage Example 2 on page 1071
How is Call Coverage Treated? on page 1072
Call Coverage Button Indication on page 1073

Call Coverage Example 1

In this example, the covering user is able to answer their colleagues call when it rings
unanswered. Both users have Ringing Line Preference and Auto Hold on.
Covered User Covering User Both Phones Idle Our user has a
call coverage button to cover their
colleague.

Table continues…

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 Call Coverage Buttons

Call to Covered User A call

arrives for the covered user.

Call Alerts to Coverage After

ringing for the covered user's
Individual Coverage Time, the call
also begins alerting on the call
coverage button .

Covering User Answers By

going off hook or pressing the
alerting button, the covering user
has answered the call.

Related links
Call Coverage Buttons on page 1070

Call Coverage Example 2

In this example, the covered user has calls on all their available call appearances. Both users
have Ringing Line Preference and Auto Hold on.
Covered User Covering User Calls in Progress The covered
user already has a number of
calls in progress on all their call
appearance keys.

Call Alerts to Coverage The

covered user is treated as busy,
so their next call goes
immediately to call coverage.

Covering User Answers The

covering user has answered the
call.

Related links
Call Coverage Buttons on page 1070

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Appearance Button Operation

How is Call Coverage Treated?

Whose user settings control the call ?
Until answered, calls alerting on a call coverage button follow the settings of the user to
which the call was originally directed.
Once answered, the call follows the user settings of the user who answered it.
Coverage is applied to :
• Internal calls dialed to the covered user's extension number.
• External calls routed to the covered user by a incoming call route.
• Calls forwarded internally by the covered user or on follow me from the covered user.
Coverage is not applied to :
• Hunt group calls to a hunt group of which the covered user is a member.
• Calls forwarded to the covered user using forward or follow me functions.
• Calls alerting on the covered user's bridged appearance and call coverage buttons.
• Coverage is only applied to calls alerting on a line appearance if the call was also routed to
that user by an incoming call route.
• Page and intercom calls.
• Parked, transferred and held calls ringing back to the user.
• Automatic callback calls set by the covered user.
• Voicemail ringback calls.
• Call coverage appearance buttons are not supported between users on different systems in a
multi-site network.
Coverage is applied :
• If the covered user's phone is available, call coverage is applied only after the covered user's
Individual Coverage Time has expired.
• If the covered user's phone is busy, call coverage is applied immediately.
• If the covered user is using follow me or forward all to an internal number to divert their calls,
call coverage is still applied.
• If the covered user has 'do not disturb' on, call coverage is applied immediately except for
calls from numbers in the covered user's do not disturb exceptions list.
Other items :
If the call is not answered after the covered user's No Answer Time it will go to the covered user's
voicemail if available or follow their forward on no answer settings.
If the covered user has several alerting calls, the call answered by the call coverage button is the
covered user's longest ringing call.

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 Call Coverage Buttons

Calls will not alert at a covering user who has 'do not disturb' enabled, except when the calling
number is in the covering user's do not disturb exception list.
Related links
Call Coverage Buttons on page 1070

Call Coverage Button Indication

On phones with a text display area next to the button, the name of the covered user is displayed
followed by the word Cover.
When the user is not connected to a call, the button indicated as selected is the button that will be
used if the user goes off hook without pressing an appearance button. When a user is connected
to a call, that call is the selected button.
The following table shows how the different states of call coverage appearance buttons (alerting,
held, etc) are indicated. This is a general table, not all phone button types are covered. The ring
that accompanies the visual indication can be delayed or switched off. See Ring Delay.
Icon Button Dual LED Button Call Coverage Button State
Idle The button is not in use.

Red off,
Green off.
Alerting The call coverage is
alerting for an unanswered call at
Flashing icon. Red off, the covered user's phone. This is
Green steady flash. accompanied by ringing. If the
user is already on a call, only a
single ring is given.
Alerting + Selected As above
but Ringing Line Preference has
Flashing icon. Red on, made this the user's current
Green steady flash. selected button.
In Use Here The user has
answered the call requiring
Red on, coverage.
Green on.
On Hold Here The covered call
has been put on hold by the call
Red off, coverage button user.
Green fast flash.

Related links
Call Coverage Buttons on page 1070

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Appearance Button Operation

Line Appearance Buttons

Line appearance buttons allow specific individual line to be used when making calls or answered
when they have an incoming call. It also allows users to bridge into calls on a particular line.

Incoming call routing is still used to determine the destination of all incoming calls. Line
appearance buttons allow a call on a specific line to alert the button user as well as the intended
call destination. When these are one and the same, the call will only alert on the line appearance
but can still receive call coverage.
When alerting on suitable phones, details of the caller and the call destination are shown during
the initial alert.
Individual line appearance ID numbers to be assigned to selected lines on a system. Line
appearance buttons are only supported for analog, E1 PRI, T1, T1 PRI, and BRI PSTN trunks;
they are not supported for other trunks including E1R2, QSIG and IP trunks.
Line appearance buttons are not supported for lines on remote systems in a multi-site network.
Using Line Appearances for Outgoing Calls In order to use a line appearance to make
outgoing calls, changes to the normal external dialing short codes are required. For full details see
Outgoing Line Programming.
Private Lines Special behaviour is applied to calls where the user has both a line appearance for
the line involved and is also the Incoming Call Route destination of that call. Such calls will alert
only on the Line Appearance button and not on any other buttons. These calls will also not follow
any forwarding.
T3 Phone Line Appearances Line appearances are supported on T3 and T3 IP phones, see T3
Phone Line Appearances.
Related links
Appearance Button Operation on page 1058
Line Appearance Example 1 on page 1075
Line Appearance Example 2 on page 1075
How are Line Appearances Treated? on page 1076
Line Appearance Button Indication on page 1077
T3 Phone Line Appearances on page 1078

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 Line Appearance Buttons

Line Appearance Example 1

In this example, the user is able to answer a call alerting on a particular line.
Line Goes Active A call is active on the line with
line ID number 601. This is indicated as 'in use
elsewhere'.
For an incoming call, the line will show active but
will not alert until call routing has been determined.
On analog ICLID lines, alerting is delayed until the
ICLID that might be used to do the call routing has
been received.
Line Appearance Alerting The routing of the call
has been complete and it is ringing against its
destination. On our user's phone the line
appearance also alerts and ringing line preference
has made it the current selected button.

Answer Call By going off hook or pressing the line

appearance, our user has answered the call on that
line.

Related links
Line Appearance Buttons on page 1074

Line Appearance Example 2

In this example, two users exchange a call using line appearance buttons set to the same line.
Note that this requires that the user who first answers the call to have Cannot be Intruded off.
Both users have Ringing Line Preference and Auto Hold on.
Idle The two users has line
appearances for the same line.

Call Alerts A call arrives. Either

user can answer it by pressing
the alerting line appearance

Table continues…

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Appearance Button Operation

Call Answered The first user has

answer the call.

Line Held The first user has put

the call on hold.

Line Retrieved The second user

has retrieved the held call by
pressing the line appearance.

Related links
Line Appearance Buttons on page 1074

How are Line Appearances Treated?

Incoming Calls
Until answered using a line appearance button, incoming calls alerting on a line
appearance, follow the settings of the incoming call route's destination group or user. They
do not follow the settings of any line appearance user.
If an incoming calls destination is voicemail, or once the incoming call has passed from its
destination to voicemail, it cannot be answered or bridged into using a line appearance button.
If the line appearance user is also the incoming call route destination for the call, the call will alert
on their line appearance only. In this case:
• It will alert on the line appearance even if all call appearances are in use.
• The call will not follow any of the user's forwarding settings .
• The call will receive call coverage from other user's with call coverage buttons set to the line
appearance user.
• The ring delay used is that of the first free call appearance.
For analog lines set to ICLID, any line appearances show active while the system waits for ICLID
information. During this time the line has not been routed and cannot be answered using a line
appearance button.
Calls alerting on a line appearance can also alert on a call coverage appearance on the same
phone. If Ringing Line Preference is set, the current selected button will change from the line
appearance to the call coverage appearance.

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 Line Appearance Buttons

If the line appearance user has do not disturb (DND) enabled, the line appearance button icon or
lamps will still operate but alerting and ringing line preference selection are not applied unless the
caller is in their DND exception list.
Outgoing Calls
In order to be used for making outgoing calls, some additional system programming may be
required. See Outgoing Line Programming.
Calls made on a call appearance, which are routed out on a line for which the user also has a line
appearance, will remain on the call appearance. The line appearance will indicate 'in use
elsewhere'.
Additional Notes
Calls alerting on a line appearance do not receive call coverage or go to a users voicemail unless
the user was the call's original incoming call route destination.
If a call indicated by a line appearance is parked, it cannot be joined or unparked by using another
line appearance.
Where a line appearance button is used to answer a call for which automatic call recording is
invoked, the recording will go to the automatic recording mailbox setting of the original call
destination.
Line appearance buttons are not supported for lines on remote systems in a multi-site network.
Related links
Line Appearance Buttons on page 1074

Line Appearance Button Indication

On phones with a text display area next to the button, the label Line and the line number are
displayed.
When the user is not connected to a call, the button indicated as selected is the button that will be
used if the user goes off hook without pressing an appearance button. When a user is connected
to a call, that call is the selected button.
The following table shows how the different states of line appearance buttons (alerting, held, etc)
are indicated. This is a general table, not all phone button types are covered. The ring that
accompanies the visual indication can be delayed or switched off. See Ring Delay.
Icon Button Dual LED Button Line Appearance Button State
Idle The associated line is not in
use.
All off.
Idle + Selected The associated
line is not in use but the button is
the user currently selected button.
Table continues…

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Appearance Button Operation

Flashing icon. Red off, Green steady Alerting The line is ringing at its
flash. incoming call route destination.
This is accompanied by ringing. If
the user is already on a call, only
a single ring is given.
Flashing icon. Red on, Green steady Alerting + Selected As above
flash. but Ringing Line Preference has
made this the user's current
selected button.
Red off, Green on. In Use Elsewhere The line is in
use.
Red on, Green on. In Use Here The user has
answered the line, made a call on
it or bridged into the call on the
line.
Red off, Green fast flash. On Hold Here The call on the line
has been put on hold by this user.
Red off, Green On Hold Elsewhere The call on
intermittent flash. the line has been put on hold by
another appearance button user.
Icon flashes off. Red off, Green broken Inaccessible The button pressed
flash. is not accessible. The call is still
dialing, ringing, routing or cannot
be bridged into.

Related links
Line Appearance Buttons on page 1074

T3 Phone Line Appearances

Note:
IP Office R11 does not support T3 and T3 IP Phones.
Release 4.2+: Line appearances are supported on T3 and T3 IP phones. As these phones do not
support call appearance, bridge appearance or call coverage appearance buttons the user can be
programmed with just line appearance buttons.
Soft Key LED Key Line Appearance Button State
L601 Off Idle The associated line is not in
use.
601 Off Idle + Selected The associated
line is not in use but the button is
the user currently selected button.
Table continues…

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 Line Appearance Buttons

L601 alternating with bell symbol. Fast flashing Alerting The line is ringing at it
incoming call route destination.
This is accompanied by ringing. If
the user is already on a call, only
a single ring is given.
L601 alternating with bell symbol. Fast flashing Alerting + Selected As above
but Ringing Line Preference has
made this the user's current
selected button.
L601 On In Use Elsewhere The line is in
use.
601 On In Use Here The user has
answered the line, made a call on
it or bridged into the call on the
line.
L601 Slow flash Slow flash On Hold Here The call on the line
has been put on hold by this user.
L601 Slow flash Slow flash On Hold Elsewhere The call on
the line has been put on hold by
another appearance button user.
-601 Off Inaccessible The button pressed
is not accessible. The call is still
dialing, ringing, routing or cannot
be bridged into. A single tone is
also given.

Notes
Hot Desking The following applies to appearance button programmed for a user on a system with
T3 phones.
• From a T3 Phone If a T3 user with programmed line appearances but no programmed call
appearances hot desks onto a phone type that requires call appearances, the phone will not
operate correctly. This configuration is not supported by Avaya.
• To a T3 Phone If appearance buttons other than line appearance are programmed for a user,
when that user in on a T3 phone those other appearance buttons will be treated as blank.
Depending on the button and type of T3 phone the button may assume its default T3 phone
function. See T3 Compact, T3 Classic and T3 Comfort.
Call Waiting Line appearances will ignore the T3 phones user selected call waiting setting. So
with a call connected and call waiting off, calls can still alert on line appearances.
Multiple Calls T3 phones are limited to a maximum of 6 associated calls at any time, including
calls connected, on hold and alerting.
Delayed Ringing The only Ring Delay options supported are Immediate or No Ring. Any other
delayed
Preference Idle line preference is always used, however T3 phones will never default to using a
line appearance for an outbound call.

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Appearance Button Operation

Joining/Bridging Joining a call active on a line appearance is supported. This is subject to the
intrusion settings of the users involved. The call then becomes a conference call.
Related links
Line Appearance Buttons on page 1074

Selected Button Indication

During appearance button usage, one of the user's appearance buttons may be indicated as the
user's current selected button. This is the appearance button already in use, or if idle, the
appearance button that will be used if the user goes off hook by lifting the handset.
On phones with a display area next to each button, the current selected button is indicated by
either an _ underscore of the button label, a * star or a shaded background.

On phones with twin LED lamps, the current selected button is indicated by the red lamp being on

.
On Transtalk 9040 phones, the current selected button is indicated by a icon.
The system sets which appearance button is the current selected button using the following
methods:
• Idle Line Preference This feature can be set on or off for each individual user, the default is
on. When on, it sets the current selected button as the first available idle call/line appearance
button. See Idle Line Preference.
• Ringing Line Preference This feature can be set on or off for each individual user, the default
is on. When on, it sets the current selected button as the button which has been alerting at
the users phone for the longest. Ringing Line Preference overrides Idle Line Preference. See
Ringing Line Preference.
• Delayed Ring Preference: This setting is used in conjunction with ringing line preference and
appearance buttons set to delayed or no ring. It sets whether ringing line preference should
observe or ignore the delayed ring applied to the user's appearance buttons when
determining which button should have current selected button status.
• User Selection The phone user can override both Idle Line Preference and Ringing Line
Preference by pressing the appearance button they want to use or answer. That button will
then remain the current selected button whilst active.

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 Idle Line Preference

If the user currently has a call connected, pressing another appearance button will either hold
or disconnect that call. The action is determined by the system's Auto Hold setting.
Answer Pre-Select: Normally when a user has multiple alerting calls, only the details of the call on
current selected button are shown. Pressing any of the alerting buttons will answer the call on that
button, going off-hook will answer the current selected button. Enabling the user telephony setting
Answer Pre-Select allows the user to press any alerting button to make it the current selected
button and displaying its call details without answering that call. To answer a call when the user
has Answer Pre-Select enabled, the user must press the alerting button to display the call details
and then either press the button again or go off-hook.
Related links
Appearance Button Operation on page 1058

Idle Line Preference

Idle Line Preference determines the user's currently selected button as the first available idle call/
line appearance button. Selected button indication is applied to that button and if the user goes
off-hook, for example by lifting their handset, an outgoing call is started on that button.
Idle Line Preference is overridden by Ringing Line Preference if also on for the user.
By default Idle Line Preference is on for all users.
For appearance button users with Idle Line Preference off, going off-hook (lifting the handset or
pressing SPEAKER, HEADSET, etc) will have no effect until an appearance button is pressed.
If all the available call/line appearance buttons are in use, no current selected button choice is
made by Idle Line Preference. In this case, going off hook will have no effect.
?Why Would I Use Just Idle Line Preference In environments that are focused on making
outgoing calls, for example telemarketing, incoming calls are infrequent and user's go off-hook
expecting to make a call. Using Idle Line Preference without Ringing Line Preference ensures
that the user doesn't inadvertently answer a call when expecting to make a call.
Idle Line Preference Example 1
In this example, only Idle Line Preference has been programmed for the user. Ringing Line
Preference has not been programmed.
Phone Idle The phone is idle. The current selected
button determined by Idle Line Preference is the
first available idle call appearance button. This is
shown by the _ underscore of the button text.

Table continues…

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Appearance Button Operation

First Call to User A call for the user arrives. It

alerts on the first available call appearance button.
Idle Line Preference has changed the current
selected button to the next available idle call
appearance.

1. User Goes Off Hook With the call still alerting,

if the user goes off hook, it will be interpreted
as making a call using the currently selected
button, not as answering the alerting button.
2. To answer the alerting call, the user should
press the alerting button.

Idle Line Preference Example 2

In this example, only Idle Line Preference has been programmed for the user. Ringing Line
Preference has not been not programmed.
Two Calls Alerting The users has two incoming
calls alerting. Idle Line Preference has set the
currently selected button to their third call
appearance.

First Caller Abandons If the first incoming caller

disconnects, the currently selected button changes
to the first call appearance as this is now the first
available idle call appearance button.

Idle Line Preference Example 3

In this example, both Idle Line Preference and Ringing Line Preference are set for the user.
Phone Idle The phone is idle and Idle Line
Preference has assigned current selected button to
the first call appearance.

Call Alerting A call has arrived and Ringing Line

Preference keeps the current selected button at the
first call appearance.

Table continues…

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 Ringing Line Preference

Call Answered With the call answered it retains

current selected button status.

Call Held When the call is put on hold, Idle Line

Preference assigns current selected button status
to the next available call appearance button.

Idle Line Preference Example 4

In this example, only Idle Line Preference has been programmed for the user. Ringing Line
Preference has not been programmed.

All Call Appearances Alerting In this case, all the users call appearance buttons are alerting
incoming calls. Idle Line Preference has changed the currently selected button to the first available
line appearance.
Related links
Appearance Button Operation on page 1058

Ringing Line Preference

Ringing Line Preference determines the user's currently selected button as the button which has
been alerting the longest. Selected button indication is applied to that button and if the user goes
off-hook, for example by lifting their handset, the alerting call on that button is answered.
Ringing Line Preference includes calls alerting on call appearance, line appearance, bridged
appearance and call coverage buttons.
Ringing Line Preference overrides Idle Line Preference.
By default Ringing Line Preference is on for all users.
Ringing Line Preference Order When a user's longest waiting call alerts on several of the user's
appearance buttons and Ringing Line Preference is set for the user, the order used for current
selected button assignment is;
Call appearance.

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Appearance Button Operation

Bridged appearance.
Call coverage.
Line appearance.
Example: A user has a call to a covered user alerting initially on a line appearance button.
Ringing Line Preference assigns current selected button status to the line appearance. When the
same call also begins to alert on the call coverage appearance button, current selected button
status switches to the call coverage appearance button.
Ring Delay and Ringing Line Preference Appearance buttons can be set to Delayed Ring or
No Ring. These buttons still alert visually but do not give an audible ring or tone. Ringing line
preference is still applied to alerting buttons even if set to Delayed Ring or No Ring.
Delayed Ring Preference For users with Ringing Line Preference selected, their Delayed Ring
Preference setting sets whether ringing line preference is used or ignores buttons that are visually
alerting but have Delayed Ring or No Ring set. The default is off, ie. ignore ring delay.
Ringing Line Preference Example 1
In this example, both Ring Line Preference and Idle Line Preference have been set for the user.
They also have Ringing Line Preference on and Auto Hold is on. Answer Pre-Select is off.
Phone Idle The phone is idle. The current selected
button has been determined by Idle Line Preference
as the first available idle call appearance button.
This is shown by the _ underscore next to that
button.

First Call Alerting A call for the user arrives. It

alerts on the first available call appearance button.
Ringing Line Preference uses this as the currently
selected button as it is the only alerting call.

Second Call Alerting Another call for the user

arrives. It alerts on the next available call
appearance button. As the first call has been
alerting longer, under Ringing Line Preference it
retains the currently select button status.

The First Call Abandons The first caller

disconnects. Ringing Line Preference changes the
currently selected button status to the second call
appearance button.

Table continues…

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 Answer Pre-Select

Another Call Arrives Another call arrives. It alerts

as the first free call appearance button. However
the call at the second call appearance has been
alerting longer and so under Ringing Line
Preference retain the currently selected button
status.

Ringing Line Preference Example 2

In this example, the user has both Ring Line Preference and Idle Line Preference programmed.
They also have Ringing Line Preference on and Auto Hold is on. Answer Pre-Select is off.
First Call to User A call for the user arrives. It
alerts on the first available call appearance button.
Ringing Line Preference uses this as the currently
selected button as it is the only alerting call.

Call on Line 601 The user's Line Appearance is

alerting due to an incoming call on the associated
line. Details of the call and its destination are
shown. Ringing Line Preference keeps the currently
selected button status on the call appearance
button as this has been alerting longest.
Second Call to User A second call to the user
arrives and alerts on the second call appearance
button. Ringing Line Preference keeps the currently
selected button status on the call appearance
button as this has been alerting longest.

The First Caller Abandons The first call to the

user disconnects. Ringing Line Preference passes
the currently selected button status to the Line
Appearance button as this has been alerting
longest.

Related links
Appearance Button Operation on page 1058

Answer Pre-Select
On some phones, only the details of the call alerting or connected on the current selected button
are shown. The details of calls alerting on other buttons are not shown or only shown briefly when
they are first presented and are then replaced again by the details of the call on the current
selected button.

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Appearance Button Operation

By default, pressing any of the other alerting buttons will answer the call on that button. Answer
pre-select allows a user to press alerting buttons other than the current selected button without
actually answering them. Instead the button pressed becomes the current selected button and its
call details are displayed.
Note that using answer pre-select with a currently connected call will still either hold or end that
call in accordance with the system's Auto Hold setting.
Answer Pre-Select Example 1
Phone Idle The phone is idle. The current selected
button has been determined by Idle Line Preference
as the first available idle call appearance button.
This is shown by the _ underscore next to that
button.

First Call Alerting A call for the user arrives. It

alerts on the first available call appearance button.
Ringing Line Preference uses this as the currently
selected button as it is the only alerting call.

Second Call Alerting Another call for the user

arrives. It alerts on the next available call
appearance button. As the first call has been
alerting longer, under Ringing Line Preference it
retains the currently select button status.

The User Presses the Second Call Appearance

Pressing the second call appearance overrides
ringing line preference and assigns current selected
button status to the button without actually
answering the call. The details of the caller are
displayed.
The User Answers the Call The user can press
the button again to answer the call or just go off-
hook to answer as it is now the currently selected
button.

Related links
Appearance Button Operation on page 1058

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 Auto Hold

Auto Hold
Auto Hold is a system wide feature that affects all appearance button users. This feature
determines what happens when a user, who is already on a call, presses another appearance
button. The options are:
• If Auto Hold is off, the current call is disconnected.
• If Auto Hold is on, the current call is placed on hold.
On Release 4.0 and higher systems Auto Hold is on by default. On previous levels of system
software the default for US was off.
Auto Hold Example 1
In this example, the user has two calls currently shown on call appearance buttons. Answer Pre-
Select is off.
1. This user has three call appearance buttons.
They have answer one call and are still
connected to it, shown by the icon. A second
call is now alerting on their second call
appearance button, shown by the icon.
2. What happens when the user presses the
second call appearance key is determined by
the system's Auto Hold setting:
Auto Hold On When the second call appearance
key is pressed, that call is answered and the first
call is put on hold, shown by the icon. The user
can switch between calls using the call appearance
buttons and make/receive other calls if they have
additional call appearance buttons
Auto Hold Off When the second call appearance
key is pressed, that call is answered and the first
call is disconnected.

Related links
Appearance Button Operation on page 1058

Ring Delay
Ring delay can be applied to appearance buttons. This option can be used with all types of
appearance buttons and can be selected separately for each appearance button a user has. Using

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Appearance Button Operation

ring delay does not affect the buttons visual alerting through the display and display icons or
button lamps.
Ring delay is typically used with line appearance buttons for lines which a user wants to monitor
but does not normally answer. However ring delay can be applied to any type of appearance
button.
The selectable ring delay options for an appearance button are listed below. The option is selected
as part of the normal button programming process.
Immediate Provide audible alerting as per normal system operation.
Delayed Ring Only provide audible alerting after the system ring delay or, if set, the individual
user's ring delay.
No Ring Do not provide any audible alerting.
There are two possible sources for the delay used when delayed ringing is selected for a button.
System | Telephony | Telephony | Ring Delay: Default = 5 seconds, Range 1 to 98 seconds. This is
the setting used for all users unless a specific value is set for an individual user.
User | Telephony | Multi-line Options | Ring Delay: Default = Blank (Use system setting), Range 1
to 98 seconds. This setting can be used to override the system setting. It allows a different ring
delay to be set for each user.
Notes
Calls That Ignore Ring Delay Ring delay is not applied to hold recall calls, park recall calls,
transfer return calls, voicemail ringback calls and automatic callback calls. For phones using
Internal Twinning, ring delay settings are not applied to calls alerting at a secondary twinned
extension (except appearance buttons set to No Ring which are not twinned).
Auto Connect Calls Ring delay is applied to these calls before auto-connection. This does not
apply to page calls.
Multiple Alerting Buttons Where a call is presented on more than one button on a user's phone,
see Multiple Alerting Buttons, the shortest delay will be applied for all the alerting buttons. For
example, if one of the alerting buttons is set to Immediate, that will override any alerting button set
to Delayed Ring. Similarly if one of the alerting buttons is set to No Ring, it will be overridden if
the other alerting button is set to Immediate or Delayed Ring.
Line Appearance Buttons Calls routed to a user that could potentially be presented on both a
call appearance button and a line appearance button are only presented on the line appearance
button. In this scenario, the ring delay settings used is that of the first free call appearance button.
Delay on Analog Lines Analog lines set to Loop Start ICLID already delay ringing whilst the
system waits for the full ICLID in order to resolve incoming call routing. In this scenario the ring
delay operates in parallel to the routing delay.
Ring Delay and Ringing Line Preference Appearance buttons can be set to Delayed Ring or
No Ring. However ringing line preference is still applied to alerting buttons even if set to Delayed
Ring or No Ring.

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 Delayed Ring Preference

The user's Delayed Ring Preference setting is used to determine whether ringing line preference
is used with or ignores buttons that are alerting but have Delayed Ring or No Ring set.
Ring Delay Example 1
In this example, the user has a line appearance button set but configured to no ring.
Phone Idle The phone is idle. The current selected
button has been determined by Idle Line Preference
as the first available call appearance button. This is
shown by the _ underscore next to that button.

Incoming Call Alerting on the Line An incoming

call arrives on the line and begin to alert
somewhere on the system. The user's line
appearance button shows this visually but doesn't
ring audibly. Ringing line preference would makes
the line appearance the user's currently selected
button and therefore they would answer the line if
they went off-hook.

Related links
Appearance Button Operation on page 1058

Delayed Ring Preference

When a call is alerting at an idle phone, by default Ringing Line Preference sets the call as the
currently selected button and if the user then goes off-hook they will answer that call.
In most situations this is acceptable as the user hears ringing which informs them that there is a
call waiting to be answered. If the user wants to make a call instead, they can press another call
appearance button to go off-hook on that other button.
When ring delay is being used there can potentially be a problem if the user lifts the handset to
make a call without looking at the display. If they do this while the a call is alerting silently on a
button with ring delay, the user will actually answer the waiting call rather than get dial tone to
make a call.
Once the call alerting on a button has currently selected call status, it retains that status even if a
prior call on a button with ring delay applied comes out of its ring delay period.
Delayed Ring Preference Example 1
In this example the user has a line appearance button for a line they monitor. This line appearance
button has been set to no ring as the user occasionally need to use that line but does not normally
answer calls on that line.

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Appearance Button Operation

Phone Idle The phone is idle. The current selected

button has been determined by Idle Line Preference
as the first available call appearance button. This is
shown by the _ underscore next to that button.

Incoming Call Alerting on the Line An incoming

call arrives on the line and begin to alert
somewhere on the system. The user's line
appearance button shows this visually but doesn't
ring audibly.
Normally ringing line preference would make the
line appearance the user's currently selected button
and therefore they would answer the line if they
went off-hook expecting to make a call.
However, because Delayed Ring Preference is on
for the user, ringing line preference is not applied
and idle line preference makes their current
selected button the first call appearance. If the user
were to go off-hook they would be making a call on
that call appearance.
Call Alerting for the User A call for the user
arrives. It alerts on the first available call
appearance button. Ringing line preference is
applied and makes that the users currently selected
button. If the user goes off-hook now that will
answer the call on the call appearance and not the
line appearance.

Delayed Ring Preference Example 2

This is similar to the previous example except that the user and the line has been configured for a
15 second ring delay. This informs the users that the line has not been answered for some reason
and allows them to answer it by just going off-hook.
Phone Idle The phone is idle. The current selected
button has been determined by Idle Line Preference
as the first available call appearance button. This is
shown by the _ underscore next to that button.

Table continues…

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 Collapsing Appearances

Incoming Call Alerting on the Line An incoming

call arrives on the line and begin to alert
somewhere on the system. The user's line
appearance button shows this visually but doesn't
ring audibly. Because Delayed Ring Preference is
on for the user, ringing line preference is not applied
and idle line preference makes their current
selected button the first call appearance. If the user
were to go off-hook they would be making a call on
that call appearance.
Call Continues Alerting When the ring delay for
the line appearance expires, if no other call has
taken ringing line preference it becomes the current
selected call and will be answered if the user goes
off-hook.

Related links
Appearance Button Operation on page 1058

Collapsing Appearances
This topic covers what happens when a user with several calls on different appearance buttons,
creates a conference between those calls. In this scenario, the call indication will collapse to a
single appearance button and other appearance buttons will return to idle. The exception is any
line appearance buttons involved which will show 'in use elsewhere'.
Collapsing Appearances Example 1
In this example, the user will setup a simple conference. Ringing Line Preference and Idle Line
Preference are set for the user. Auto Hold for the system is on. Answer Pre-Select is off.
Initial Call The user has a call in progress, shown
on their first call appearance button. It is decided to
conference another user into the call.

Make Conference Enquiry Pressing the

CONFERENCE button on the users phone
automatically places the current call on hold and
takes the phone off hook on the next available call
appearance.

Table continues…

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Appearance Button Operation

Enquiry in Progress The other extension has

answered and is invited to join a conference call.
The user presses the CONFERENCE button on
their phone again.

Conference Starts/Call Appearances Collapse

The conference call has started. The call
appearances have collapsed to a single
appearance.

Related links
Appearance Button Operation on page 1058

Joining Calls
Appearance buttons can be used to "join" existing calls and create a conference call. A user can
join calls that are shown on their phone as 'in use elsewhere'.
This feature is often referred to as 'bridging into a call'. However this causes confusion with
Bridged appearance buttons and so the term should be avoided.
The ability to join calls is controlled by the following feature which can be set for each user:
• Cannot be Intruded: Default = On If this option is set on for the user who has been in the
call the longest, no other user can join the call. If that user leaves the call, the status is taken
from the next internal user who has been in the call the longest. The exceptions are:
- Voicemail calls are treated as Cannot be Intruded at all times.
- When an external call is routed off switch by a user who then leaves the call, the Cannot
be Intruded status used is that of the user who forwarded the call off switch.
- Any call that does not involve an internal user at any stage is treated as Cannot be
Intruded on. For example:
• When an external call is routed off switch automatically using a short code in the
incoming call route.
• multi-site network calls from other systems that are routed off-switch.
• VoIP calls from a device not registered on the system.
• The Can Intrude setting is not used for joining calls using appearance buttons.
The following also apply:

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 Joining Calls

Inaccessible In addition to the use of the Cannot be Intruded setting above, a call is
inaccessible if:
• The call is still being dialed, ringing or routed.
• It is a ringback call, for example a call timing out from hold or park.
• If all the internal parties, if two or more, involved in the call have placed it on hold.
• Conferencing Resources The ability to bridge depends on the available conferencing
resource of the system. Those resources are limited and will vary with the number of existing
parties in bridged calls and conferences. The possible amount of conferencing resource
depends on the system type and whether Conferencing Center is also installed.
• Conference Tone When a call is joined, all parties in the call hear the system conferencing
tones. By default this is a single tone when a party joins the call and a double-tone when a
party leaves the call. This is a system setting.
• Holding a Bridged Call If a user puts a call they joined on hold, it is their connection to the
joined call (conference) that is put on hold. The other parties within the call remain connected
and can continue talking. This will be reflected by the button status indicators. The user who
pressed hold will show 'on hold here' on the button they used to join the call. All other
appearance users will still show 'in use here'.
• Maximum Two Analog Trunks Only a maximum of two analog trunks can be included in a
conference call.
• Parked Calls A Line Appearance button may indicate that a call is in progress on that line.
Such calls to be unparked using a line appearance.
Joining Example 1: Joining with a Bridged Appearance
In this example, the user joins a call using a bridged appearance button. Answer Pre-Select is
off.
User with Bridged Appearance Buttons The user
has bridged appearance buttons that match their
colleagues call appearance buttons.

Call on Bridged Appearance The colleague has

a call in progress on their first call appearance. This
is matched on the first bridged appearance button.

User Joins the Call Pressing the bridged

appearance button will take our user off hook and
join them into their colleagues call, creating a
conference call.

Joining Example 2: Joining with a Line Appearance

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Appearance Button Operation

In this example, the user joins a call by pressing a line appearance button. Answer Pre-Select is
off.
Line Goes Active A call is active on the line with
line ID number 601.
If this is an incoming call, it will show active but will
not alert until its call routing has been determined.
On ICLID analog lines, alerting is delayed until the
ICLID that might be used to do that routing has
been received.
Line Appearance Alerting The call routing is
completed and the call is now ringing against its
target. The line appearance also begins alerting and
Ringing Line Preference has made it the current
selected button.

Call Answered Alerting on the line appearance has

stopped but the line is still active. This indicates that
the call has probably been answered. As our user's
phone is idle, Idle Line Preference has returned the
current select button to the first available call
appearance button.
User Joins the Call Our extension user has been
asked by their colleague to join the call just
answered on line 601. By pressing the line
appearance button they have joined the call on that
line and created a conference call.

Related links
Appearance Button Operation on page 1058

Multiple Alerting Appearance Buttons

In some scenarios, it is may be potentially possible for the same call to alert on several
appearance buttons. In this case the following apply:
• Line appearance buttons override call and bridged appearance buttons In cases where
a call on a line goes directly to the user as the incoming call route's destination, the call will
only alert on the line appearance. In this scenario the ring delay settings used is that of the
first free call appearance button.
• A call can alert both call appearance, line appearance and bridged appearance buttons
The most common example of this will be hunt group calls where the hunt group members
also have bridged call appearances to each other. In this case the button used to answer the
call will remain active whilst the other button will return to idle.

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 Twinning

• Calls on a line/bridged appearance buttons can also alert on call coverage button In
this case alerting on the call coverage button may be delayed until the covered user's
Individual Coverage Time has expired.
• Ringing Line Preference Order When a call alerts on several of the user's appearance
buttons and Ringing Line Preference is set for the user, the order used for current selected
button assignment is:
1. Call appearance.
2. Bridged appearance.
3. Call coverage.
4. Line appearance.
Example A user has a call to a covered user alerting initially on a line appearance button.
Ringing Line Preference will assign current selected button status to the line appearance. When
the same call also begins to alert on the call coverage appearance button, current selected button
status switches to the call coverage appearance button.
Ring Delay Where ring delays are being used, the shortest delay will be applied for all the alerting
buttons. For example, if one of the alerting buttons is set to Immediate, that will override any
alerting button set to Delayed Ring. Similarly if one of the alerting buttons is set to No Ring, it will
be overridden if the other alerting button is set to Immediate or Delayed Ring.
Related links
Appearance Button Operation on page 1058

Twinning
Twinning is a mechanism that allows an user to have their calls alert at two phones. The user's
normal phone is referred to as the primary, the twinned phone as the secondary.
By default only calls alerting on the primary phone's call appearance buttons are twinned. For
internal twinning, the system supports options to allow calls alerting on other types of appearance
buttons to also alert at the secondary phone. These options are set through the User | Twinning
section of the system configuration and are Twin Bridge Appearances, Twin Coverage
Appearances and Twin Line Appearances. In all cases they are subject to the secondary having
the ability to indicate additional alerting calls.
Call alerting at the secondary phone ignoring any Ring Delay settings of the appearance button
being used at the primary phone. The only exception is buttons set to No Ring, in which case calls
are not twinned.
Related links
Appearance Button Operation on page 1058

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Appearance Button Operation

Busy on Held
For a user who has Busy on Held selected, when they have a call on hold, the system treats
them as busy to any further calls. This feature is intended primarily for analog phone extension
users. Within Manager, selecting Busy on Held for a user who also has call appearance keys will
cause a prompt offering to remove the Busy on Held selection.
Related links
Appearance Button Operation on page 1058

Reserving a Call Appearance Button

Functions such as transferring calls using a Transfer key require the user to have at least one
available call appearance button in order to complete the outgoing call part of the process.
However, by default all call appearance button are available to receive incoming calls at all times.
Through the system configuration it is possible to reserve the user's last call appearance button
for making outgoing calls only.
1400, 1600, 9500 and 9600 Series telephone users can put a call on hold pending transfer if they
already have held calls even if they have no free call appearance button available. See Context
Sensitive Transfer.
Reserving a Call Appearance
The method for doing this depends on the system software level.
Pre-4.0 IP Office: On the User | Source Numbers tab, enter the line RESERVE_LAST_CA= .
Release 4.0+: On the User | Telephony | Multi-line Options tab, select the option Reserve Last
CA.
Related links
Appearance Button Operation on page 1058

Logging Off and Hot Desking

Users can be setup to log in and log out at different phones, this is called 'hot desking'. All the
users settings, including their extension number, are transferred to the phone at which the user is
logged in. This includes their key and lamp settings and appearance buttons.
This type of activity has the following effect on appearance buttons:
If logged out, or logged in at a phone that doesn't support appearance button functions:
• Bridged appearances set to the user will be inactive.

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 Applications

• Call coverage set to the user will still operate.

If logged in at a phone with fewer buttons than programmed for the user:
• Those buttons which are inaccessible on the logged in phone will be inactive.
• Any bridged appearances to those button from other users will be inactive.
Remote Hot Desking
Release 4.0+ supports, through the addition of license keys, users hot desking between systems
within a multi-site network. However, the use of appearance buttons (call coverage, bridged
appearance and line appearance) within a multi-site network is not supported. Therefore when a
user logs in to a remote system, any such button that they have will no longer operate. Similarly
any button that other users have with the remote user as the target will not operate.
Related links
Appearance Button Operation on page 1058

Applications
A number of system applications can be used to make, answer and monitor calls. These
applications treat calls handled using key and lamp operation follows:
SoftConsole These applications are able to display multiple calls to or from a user and allow
those calls to be handled through their graphical interface.
• All calls alerting on call appearance buttons are displayed.
• Calls on line, call coverage and bridged appearance buttons are not displayed until
connected using the appropriate appearance button
• Connected and calls held here on all appearance button types are displayed.
Related links
Appearance Button Operation on page 1058

Programming Appearance Buttons

About this task
This section covers the programming of appearance buttons for users into existing system
configurations.
Appearance Functions The functions Call Appearance, Bridged Appearance, Coverage and
Line Appearance are collectively known as "appearance functions". For full details of their
operation and usage refer to the Appearance Button Operation section. The following restrictions
must be observed for the correct operation of phones.

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Appearance Button Operation

Appearance functions programmed to buttons without suitable status lamps or icons are treated
as disabled. These buttons are enabled when the user logs in on a phone with suitable buttons in
those positions.
Line appearance buttons require line ID numbers to have been assigned, see Programming Line
Appearance Numbers. The use of line appearances to lines where incoming calls are routed using
DID (DDI) is not recommended.
How many buttons are allowed? The recommended limits depend on the type of system. The
are 10 for IP500 V2 systems, 20 for Server Edition and 40 for Server Edition Select. The limits are
applied as follows:
• Number of bridged appearances to the same call appearance.
• Number of line appearances to the same line.
• Number of call coverage appearances of the same covered user.
Programming Appearance Buttons Using Manager
If only button programming changes are required, the configuration changes can be merged back
to the system without requiring a reboot.
Procedure
1. Start Manager and load the current configuration from the system.
2. Locate and select the user for whom appearance buttons are required.
3. Select Button Programming.

The number of buttons displayed is based on the phone associated with the user when the
configuration was loaded from the system. This can be overridden by selecting Display all
buttons.
4. For the required button, click the button number and then click Edit.
5. Click the ... button.

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 Programming Appearance Buttons

6. From the list of options that appears, click Appearance.

7. Select the type of appearance button required.
8. Use the Action Data drop-down fields to select the required settings.
Click OK.
9. Repeat for any additional call appearance buttons required.
Click OK.
10. Repeat for any other users requiring appearance buttons.
Related links
Appearance Button Operation on page 1058
Appearance Function System Settings on page 1099
Appearance Function User Settings on page 1100
Programming Line Appearance ID Numbers on page 1101
Outgoing Line Programming on page 1103

Appearance Function System Settings

System settings are applied to all users and calls. The system settings that affect appearance
operation are found on the System | Telephony tabs and are:
• Auto Hold
• Conferencing Tone
• Ring Delay
• Visually Differentiate External Call
Related links
Programming Appearance Buttons on page 1097

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Appearance Button Operation

Appearance Function User Settings

User settings are applied separately to each individual user. In addition to button programming,
the following user settings are applicable to appearance button operation:
Cannot be Intruded: Default = On. This feature controls whether other users can use their
appearance buttons to join the users call. It applies when the user is the longest present internal
party already within the call.
• Individual Coverage Time (secs): Default = 10 seconds, Range 1 to 99999 seconds.
This function sets how long the phone will ring at your extension before also alerting at any
call coverage users. This time setting should not be equal to or greater than the No Answer
Time applicable for the user.
• Ring Delay: Default = Blank (Use system setting). Range = 0 (use system setting) to 98
seconds. This setting is used when any of the user's programmed appearance buttons is set
to Delayed ringing. Calls received on that button will initially only alert visually. Audible
alerting will only occur after the ring delay has expired.
• Coverage Ring: Default = Ring. This field selects the type of ringing that should be used for
calls alerting on any the user's call coverage and bridged appearance buttons. Ring selects
normal ringing. Abbreviated Ring selects a single non-repeated ring. No Ring disables
audible ringing. Note that each button's own ring settings (Immediate, Delayed Ring or No
Ring) are still applied.
The ring used for a call alerting on a call coverage or bridged appearance button will vary
according to whether the user is currently connected to a call or not.
• If not currently on a call, the Coverage Ring setting is used.
• If currently on a call, the quieter of the Coverage Ring and Attention Ring settings is used.
Attention Ring Setting Coverage Ring Setting
Ring Abbreviated Off
Ring Ring Abbreviated Off
Abbreviated Abbreviated Abbreviated Off

• Attention Ring: Default = Abbreviated Ring. This field selects the type of ringing that should
be used for calls alerting on appearance buttons when the user already has a connected call
on one of their appearance buttons. Ring selects normal ringing. Abbreviated Ring selects
a single ring. Note that each button's own ring settings (Immediate, Delayed Ring or No
Ring) are still applied.
• Ringing Line Preference: Default = On. For users with multiple appearance buttons. When
the user is free and has several calls alerting, ringing line preference assigns currently
selected button status to the appearance button of the longest waiting call. Ringing line
preference overrides idle line preference.
• Idle Line Preference: Default = On. For users with multiple appearance buttons. When the
user is free and has no alerting calls, idle line preference assigns the currently selected
button status to the first available appearance button.

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 Programming Appearance Buttons

• Delayed Ring Preference: Default = Off. This setting is used in conjunction with
appearance buttons set to delayed or no ring. It sets whether ringing line preference should
use or ignore the delayed ring settings applied to the user's appearance buttons.
When on, ringing line preference is only applied to alerting buttons on which the ring delay has
expired.
When off, ringing line preference can be applied to an alerting button even if it has delayed ring
applied.
• Answer Pre-Select: Default = Off. Normally when a user has multiple alerting calls, only the
details and functions for the call on currently selected button are shown. Pressing any of the
alerting buttons will answer the call on that button, going off-hook will answer the currently
selected button. Enabling Answer Pre-Select allows the user to press any alerting button to
make it the current selected button and displaying its call details without answering that call
until the user either presses that button again or goes off-hook. Note that when both Answer
Pre-Select and Ringing Line Preference are enabled, once current selected status is
assigned to a button through ringing line preference it is not automatically moved to any other
button.
• Reserve Last CA: Default = Off. Used for users with multiple call appearance buttons. When
selected, this option stops the user's last call appearance button from being used to receive
incoming calls. This ensures that the user always has a call appearance button available to
make an outgoing call and to initiate actions such as transfers and conferences.
1400, 1600, 9500 and 9600 Series telephone users can put a call on hold pending transfer if they
already have held calls even if they have no free call appearance button available. See Context
Sensitive Transfer.
Abbreviated Ring: This option has been replaced by the Attention Ring setting above.
Related links
Programming Appearance Buttons on page 1097

Programming Line Appearance ID Numbers

Line appearances are supported for analog, E1 PRI, T1, T1 PRI, and BRI PSTN trunks. They are
not supported for E1R2, QSIG and IP trunks.
Note that setting and changing line settings including line appearance ID numbers requires the
system to be rebooted.
Related links
Programming Appearance Buttons on page 1097

Automatic Renumbering
About this task
Procedure
1. Select Tools | Line Renumber.

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Appearance Button Operation

2. Select the starting number required for line numbering and click OK.
3. All lines that support Line Appearance ID will be numbered in sequence.

Manual Renumbering
About this task
Procedure
1. Start Manager and load the current configuration from the system.
2.
Select Line.
3. Select the line required.
The tab through which line appearance ID numbers are set will vary depending on the type
of line. A couple of examples are shown below.
a. Analog Line
On the Line Settings tab select Line Appearance ID and enter the ID required.

b. Basic/Primary Rate Trunks

On the Channels tab select the individual channel and click Edit. Select Line
Appearance ID and enter the required ID, then click OK. Repeat for all the channels
required.

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 Programming Appearance Buttons

4. Click OKand repeat for any other lines.

Outgoing Line Programming

Assigning line ID numbers to lines and associating line appearance buttons to those lines is
sufficient for answering incoming calls on those lines. However, to use line appearance buttons for
outgoing calls may require further programming.
Short Codes and Outgoing Line Appearance Calls Once a line has been seized using a line
appearance button, short code matching is still applied to the number dialed. That can include
user, system and ARS or LCR short codes.
The short codes matching must resolve to an off-switch number suitable to be passed direct to the
line.
The final short code applied must specify a 'dial' feature. This allows call barring of specific
matching numbers to be applied using short codes set to features such as 'Busy'.
Related links
Programming Appearance Buttons on page 1097

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Chapter 22: Overview of Data Routing

The system is a network router. In this role it can connect users on its LAN to remote services by
using WAN links and telephone trunk connections. It can also allow users to dial-in and then act as if
they were using a PC on the LAN.
As well as being a network router, the system is a telephone system. These dual roles allow it to
support a range of functions that involve traffic between the network and telephony interfaces.
These functions use internal data channels. The number of internal data channels that can be
connected from the system's LAN interface to its telephony interface at any time is restricted.
An internal data channel is a connection between the system's telephony and LAN interfaces. For
example a Voicemail connection, an internet connection or a RAS user.
Calls using a VCM channel do not use a data channel.
The number of data channels in use does not necessarily match the number of users:
• Several LAN network users, browsing the internet using the same service to an ISP would be a
single data channel.
• Several dial-in network users would each have a separate data channel.
The maximum number of data channels that can be simultaneously in use for voicemail is restricted.
These channels also require entry of an appropriate license.
The restriction depends on the type of control unit being used.
System Control Unit Internal Data Channels Maximum Data Channels for
Voicemail
Small Office Edition 18 10
IP403 18 10
IP406 V1 24 20
IP406 V2 40 20
IP412 100 30
IP500 V2 48[1] 40

1. Reduced to 44 when an IP500 V2 4-Port Expansion card is installed.

Related links
Network Address Translation (NAT) on page 1105
Dynamic Host Configuration Protocol (DHCP) on page 1105
Simple ISDN Internet Connection on page 1106

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 Network Address Translation (NAT)

ISDN Link Between IP Offices on page 1106

Using a Dedicated T1/PRI ISP Link on page 1107
Remote Access on page 1110
Creating a VoIP Link via the WAN Port Using PPP on page 1113

Network Address Translation (NAT)

NAT allows the addresses used within your LAN to be replaced by a different address when
connecting to an external service.
Typically a service provider will allocate you a single IP address to be used when connecting to
their service. NAT allows all your user's traffic to appears to be coming from that single address
without having to change any of your user's real addresses. This is useful as internally most
networks use addresses that have been reserved for public use within networks but are not valid
for routing across the internet (since the same addresses may be being used on other networks).
Also as stated it allows multiple users to use the same service simultaneously.
The use of NAT is automatically enabled if the system Service being used includes an IP address
that is not in the same domain as the its LAN1 IP address.
An exception to the above applies for systems with two LAN's, LAN1 and LAN2. For these units,
on each LAN, Enable NAT can be selected and then applied to traffic between the two LAN's.
Related links
Overview of Data Routing on page 1104

Dynamic Host Configuration Protocol (DHCP)

The system can act as a simple DHCP server. When switched on with a defaulted configuration,
the Control Unit request IP address information from a DHCP server. If it gets no response it
assumes the role of DHCP server for the LAN.
In DHCP Server mode, by default the Control Unit issues itself the address 192.168.42.1. It
allocates 200 addresses for DHCP clients, 192.168.42.1 to 19.168.42.200. This leaves
192.168.42.201 to 192.168.42.254 available for any computers that need to be allocated a fixed or
static IP address. 192.168.42.255 is not used as this is a broadcast address for the LAN.
Related links
Overview of Data Routing on page 1104

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Overview of Data Routing

Simple ISDN Internet Connection

In this example, we want all non-local data traffic to be routed to the Internet. The Internet Service
Provider (ISP) has provided the account details required. Using the system's Network Address
Translation (NAT), a single account can be used for all users.

Select Service and add a normal service. Change the following settings and click OK.
Name: Internet
Account Name: As provided by the ISP.
Password: As provided by the ISP.
Telephone Number: As provided by the ISP.
Check Request DNS.

Select IP Route and add a new route. Change the following settings and click OK.
1. Leave the IP Address and IP Mask blank. This will then match any data traffic that isn't
matched by any other IP Route record.
2. Select the service created above as the Destination.
Alternate In the example above, a default IP Route was created which then routed all traffic to the
required Service. An alternate method to do this with system is to select Default Route within the
Service settings.
Related links
Overview of Data Routing on page 1104

ISDN Link Between IP Offices

To create a data link between two sites via ISDN configure the Control Unit as per the following
example:
At Site A on IP address 192.168.43.1
1. Create a Normal Service: The Service name can be any text and is used to identify this
particular Service. The Account Name and password are presented to the remote end,
therefore must match the User name and password configured at Site B. The Telephone
Number is the number of the remote end.
2. Create an IP Route: In the IP Address field enter the network address of the remote end,
not the IP address of the Control Unit. Under Destination select the Service created above.
3. Create a User: Under the Dial In tab tick Dial In On. This User account is used to
authenticate the connection from the Site B. Note that as the Service and User have the
same names, these two configuration forms are automatically linked and become an

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 Using a Dedicated T1/PRI ISP Link

Intranet Service. The User password is displayed at the bottom of the Service tab as the
Incoming Password.
4. Setup RAS: Check the default RAS settings "Dial In" are available, otherwise create a new
one. If the RAS settings are given the same name as the Service and User they are
automatically linked and become a WAN Service. Ensure that the Encrypted Password
option is not checked when using a WAN Service.
5. Setup an Incoming Call Route: Check the default Incoming Call Route is available,
otherwise create a new one. If the Incoming Number is left blank, the Incoming Call Route
accepts data calls on any number. Under Destination select the RAS service created
above. The Bearer Capability should be AnyData.
At Site B on IP address 192.168.45.1
Repeat the above process but altering the details to create an route from Site B to Site A.
Related links
Overview of Data Routing on page 1104

Using a Dedicated T1/PRI ISP Link

This section shows an example of a dedicated WAN PPP link to an Internet Service Provider (ISP)
over a set of T1 or T1 PRI line channels. The ISP must support this mode of connection and will
need to provide details of the required settings. If multiple channels are to be used, then the ISP
must support Multilink PPP.
Related links
Overview of Data Routing on page 1104

Tasks for Using a Dedicated T1/PRI ISP Link

About this task
Procedure
1. Create a New WAN Service
2. Create the Virtual WAN Port
3. Create an IP Route
4. T1 PRI Trunk

Create a New WAN Service

About this task
A service is used to define connection settings such as name, password, bandwidth, etc.

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Overview of Data Routing

Procedure
1.
Select Service to display the existing services.
2. Click on and select WAN Service.
3. Select the Service tab.
4. In the Name field enter an appropriate name, such as “Internet”.
Note that the system will also automatically create User record and a RAS record with the
same name.
5. Enter the Account Name, Password and Telephone Number details provided by the ISP.
6. For the Firewall Profile select the firewall created previously.
7. Click the Bandwidth tab.
a. Set the Maximum No. of Channels to the maximum number of channels that the
service should use.
In this example, 12 channels were used.
b. Leave all the other records at their default values.
c. If the ISP has allocated IP address details these are entered through the IP tab.
If the IP Address and IP Mask define a different domain from the system LAN, then
NAT is automatically applied.
8. Click the IP tab.
a. In the IP Address field enter the IP address specified by the ISP.
b. In the IP Mask field enter the IP Mask specified by the ISP.
c. The settings shown are typical.
The actual settings must match those required by the ISP. For example, if Cisco
routers are being used then IPHC needs to be ticked.
9. Click the PPP tab.
Ensure that the following options are selected. Leave all other options at their default
settings.
• Multilink.
• Compression Mode: Disable.
• Callback Mode: Disable.
• Access Mode: Digital64
10. Click OK.

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 Using a Dedicated T1/PRI ISP Link

Create the Virtual WAN Port

About this task
In this stage, a WAN port is defined that actually uses T1 or T1 ISDN trunk channels.
Procedure
1.
Select WAN Port to display existing ports.
2. Click on and select WAN Port.
3. In the Name field, enter either LINEx.y where:
• LINE must be in uppercase.
• x is the line number. For a trunk card in Slot A, this will be 1. For a trunk card in Slot B,
this will be 5.
• y is the lowest numbered channel number to be used by the WAN link minus 1. For
example, if the lowest channel to be used is channel 1 then y = 1 - 1 = 0.
4. In the Speed field, enter the total combined speed of the maximum number of channels
sets in the Service.
In this example, 12 channels x 64000 bits = 76800.
5. Set the Mode to SyncPPP.
6. In the RAS Name field, select the name used for the Service.
7. Click OK.

Create an IP Route
About this task
By creating an IP route with blank IP address details, it becomes the default route for outgoing IP
traffic.
Procedure
1.
Select IP Route to display existing routes.
2. Click on and select IP Route.
3. Leave the IP Address and IP Mask fields blank.
4. In the Destination field, select the WAN service.
5. Leave the Metric at default value of 1.
6. Click OK.

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Overview of Data Routing

7. Configure the Line Channels This stage of the process differs according to the type of
trunk being used.
8. T1 Trunk Use the following for a T1 trunk.
9.
Click Line to display the existing lines.
10. Double-click on the line previously entered in the WAN Port settings.
11. Check that the Channel Allocation order matches that required by the ISP.
Cisco routers typically use 1|24.
12. Select the channels to be used in the WAN PPP link and change their Channel Type to
“Clear Channel 64k”.

13. Click OK.

14. Click OKagain.
15. Send the configuration to the system and reboot.

T1 PRI Trunk
About this task
Use the following for a T1 PRI trunk.
Procedure
1.
Click on Line to display the list of existing lines.
2. Double-click on the line previously entered in the WAN Port settings.
3. Check that the Channel Allocation order matches that required by the ISP.
Cisco routers typically use 1|23.
4. Select the channels to be used in the WAN PPP link and change their Admin to “Out of
Service”.
5. Click OK.
6. Click OKagain.
7. Send the configuration to the system and reboot.

Remote Access
The system support remote access for incoming data calls on trunks.

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 Remote Access

To do remote access, an incoming call is passed through the following elements of the system
configuration.

Incoming Call Route A Incoming Call Route is used to match incoming remote access calls
and pass them to a RAS service as the destination.

RAS Service The RAS service defines settings relating to the data traffic methods usable with
the call.

User The user defines the name and password required for the RAS service. The user must
have Dial In On enabled.
An R setting on the user's Source Numbers tab can be used to define the ICLID from which RAS
calls are accepted.

Time Profile The user settings can specify a time profile. The time profile then controls when
remote access is allowed.

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Overview of Data Routing

Firewall Profile The user settings can specify a firewall profile. The firewall profile then
controls what traffic is allowed through the remote access connection.
Static NAT The system supports the use of Static NAT records in firewall profiles. These are
used to translate external IP addresses to internal IP addresses.

System | LAN The system can provide DHCP support for remote access connections when it
is set to Server or Dial in modes. Alternatively the remote access client can use a static IP
address on the system's subnet.

IP Route If the remote access client uses a IP address that is from a different subnet from the
system, then a IP route record is required for returning data. The RAS service is set as the
destination.
ISDN Remote Access Example

Create a User The required details are:

• In the User tab: Enter a Name and Password. The system is case sensitive. Remember to
take care with passwords as this is a remote access link into your network.
• In the Dial In tab: Ensure that Dial In On is ticked. The Firewall Profile and Time Profile are
optional.

Create a RAS Record

In the RAS tab: Enter the same name as the user that you created earlier. Again, remember this
is case sensitive.

Create an Incoming Call Route

• Set the Bearer Capability to Any Data.
• In the Destination drop-down list, select the RAS record created above.
• The values that you enter for any of the other fields will depend on whether the remote user
will be calling in on a particular line, number or from a set ICLID.
1. Is a Return IP Route Needed ? Go to Step 5.
2.
Create a IP Route (Optional) If the remote user has an IP address that is not in the
same domain as the system, then an IP Route is needed for return data. This is not

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 Creating a VoIP Link via the WAN Port Using PPP

necessary if the remote user's dial-up connection method is set to 'Obtain an IP Address
Automatically' and the system's DHCP mode is set to Server or Dial In.
• Enter the IP Address and IP Mask of the remote system.
• In the Destination drop-down list select the RAS record created above.
Analog Remote Access Example

Configuration for a connection from an analog modem call is very similar to the ISDN example.
However the system must be able to answer modem calls. This can be done in the following ways:
Analog Trunk Modem Mode On systems with an analog trunk card in the control unit, the first
analog trunk can be set to answer V.32 modem calls. This is done by checking the Modem
Enabled option on the analog line settings or using the default short code *9000* to toggle this
service on or off.
IP500 ATM4 Uni Trunk Card Modem Support It is not required to switch the card's modem port
on/off. The trunk card's V32 modem function can be accessed simply by routing a modem call to
the RAS service's extension number. The modem call does not have to use the first analog trunk,
instead the port remains available for voice calls.
When using an analog modem, the Bearer Capability of the incoming call route used should be
Any Voice.
Related links
Overview of Data Routing on page 1104

Creating a VoIP Link via the WAN Port Using PPP

A VoIP link across a leased line requires the Control Unit at both ends to have a Voice
Compression Module installed. These provide for a fixed number of channels to use VoIP at any
time. They are used to compress voice down to either 6k3 (G.723) or 8k (G.729) and provide echo
cancellation.
Both ends must using the same version of software and configured to use the same speed and
compression.
For example, consider this VoIP link:
• Site A on IP address 192.168.42.1
• Site B on IP address 192.168.45.1

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Overview of Data Routing

Perform the following steps, once for Site A and once for Site B.
1. Create a Normal Service: The Account Name and password is presented to the remote
end, therefore must match the User name and password configured at Site B. The
Encrypted Password option can only be used if the remote end also supports CHAP.
2. Create a User: Under the Dial In tab tick Dial In On. This User account is used to
authenticate the connection from the Site B. As the Service and User have the same name
these two configuration forms are automatically linked and become an Intranet Service.
The User password is displayed at the bottom of the Service tab as the Incoming
Password.
3. Name: SiteB
4. Dial In | Dial In On: Enabled.
5. Create a RAS service: If CHAP is to be used on this link, then the Encrypted Password
option must be checked in the Service and in the RAS service. The name of the RAS
service must match the name of the Service at Site B. If the RAS service is given the same
name as the Service and User, they are automatically linked and become a WAN Service.
Ensure that the Encrypted Password option is not checked when using a WAN Service.
6. Edit the WANPort: Note - do not create a new WANPort, this is automatically detected. If
a WANPort is not displayed, connect the WAN cable, reboot the Control Unit and receive
the configuration. The WANPort configuration form should now be added.
RAS Name: SiteA
7. Create an IP Route: The IP Address is the network address of the remote end. Under
Destination select the Service created above.
8. Create a new Line: The Line Number and Line Group ID must be unique, in other words,
not used by any other line. The Gateway IP Address is the IP Address of the Control Unit
at the remote end. The Compression Mode used is dependent on the Voice Compression
Card the Control Unit is running and the speed of the link.
9. Create a Short Code: To route all calls where the number dialed starts with 8 via Line
Group ID 1, therefore via the VPN Line created above.
10. Short Code: 8N
11. Telephone Number: N
12. Line Group ID: 1
13. Feature: Dial
Related links
Overview of Data Routing on page 1104

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Chapter 23: Appendix: SMDR

The control unit is able to send SMDR (Station Message Detail Reporting) records to a specified IP
address and port.
Typically an SMDR record is output for each call between two parties (internal and or external) that
is handled by the system. In some scenarios, for examples transfers, where a call involves multiple
parties then multiple SMDR records may be output for each part of the call.
Each SMDR record contains call information in a comma-separated format (CSV), that is variable-
width fields with each field separated by commas.
The recommended limit for authorization codes is 1000 entries
The IP500 V2 control units can store any buffered SMDR records during any controlled system
power downs or reboots.
Note:
Outbound Contact Express
The Outbound Contact Express solution does not generate SMDR records.
Enabling SMDR
1. Receive the configuration from the system.
2. Select System and then select the CDR/SMDR tab.
3. Use the Output drop down box to select SMDR only.
4. In the SMDR settings, enter the required IP Address and TCP Port.
Overview of SMDR Records
An SMDR record is generated for each call between two devices on the system. Devices include
extensions, trunk lines (or channels on a trunk), voicemail channels, conference channels and
system tones.
Calls which are not presented to another device do not generate an SMDR record. For example
internal users dialing short code that simply changes a configuration setting.
The SMDR record is generated when the call ends, therefore the order of the SMDR records output
does not match the call start times.
Each record contains a call ID which is increased by 1 for each subsequent call.
When a call moves from one device to another, an SMDR record is output for the first part of the call
and an additional SMDR record will be generated for the subsequent part of the call.
Each of these records will have the same Call ID.

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Appendix: SMDR

Each record for a call indicates in the Continuation field if there will be further records for the same
call.
Note:
The SMDR record length is not fixed. New fields may be added to the end as required.
Call Times
Each SMDR record can include values for ringing time, connected time, held time and parked time.
The total duration of an SMDR record is the sum of those values.
The time when a call is not in any one of the states above, for example when one party to the call
has disconnected, is not measured and included in SMDR records.
Where announcements are being used, the connected time for a call begins either when the call is
answered or the first announcement begins.
All times are rounded up to the nearest second.
Each SMDR record has a Call Start time taken from the system clock time. For calls being
transferred or subject to call splitting, each of the multiple SMDR records will have the same Call
Start time as the original call.
Related links
SMDR Fields on page 1116
SMDR Examples on page 1121

SMDR Fields
The SMDR output contains the following fields. Note that time values are rounded up to the
nearest second.
1. Call Start
Call start time in the format YYYY/MM/DD HH:MM:SS. For all transferred call segment this
is the time the call was initiated, so each segment of the call has the same call start time.
2. Connected Time
Duration of the connected part of the call in HH:MM:SS format. This does not include
ringing, held and parked time. A lost or failed call will have a duration of 00:00:00. The total
duration of a record is calculated as Connected Time + Ring Time + Hold Time + Park
Time.
3. Ring Time
Duration of the ring part of the call in seconds.
• For inbound calls this represents the interval between the call arriving at the switch and
it being answered, not the time it rang at an individual extension.

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 SMDR Fields

• For outbound calls, this indicates the interval between the call being initiated and being
answered at the remote end if supported by the trunk type. Analog trunks are not able to
detect remote answer and therefore cannot provide a ring duration for outbound calls.
4. Caller
The callers' number. If the call was originated at an extension, this will be that extension
number. If the call originated externally, this will be the CLI of the caller if available,
otherwise blank.
For SIP trunks, the field can contain the number plus IP address. For example
[email protected].
5. Direction
Direction of the call – I for Inbound, O for outbound. Internal calls are represented as O for
outbound. This field can be used in conjunction with Is_Internal below to determine if the
call is internal, external outbound or external inbound.
6. Called Number
This is the number called by the system. For a call that is transferred, this field shows the
original called number, not the number of the party who transferred the call.
• Internal calls: The extension, group or short code called.
• Inbound calls: The target extension number for the call.
• Outbound calls: The dialed digits.
• Voice Mail: Calls to a user's own voicemail mailbox.
7. Dialled Number
For internal calls and outbound calls, this number dialed by the user. This may differ from
the Called Number due to the effect of short codes and other features. For inbound calls,
this is the DDI of the incoming caller.
8. Account
The last account code attached to the call.
Note:
System account codes may contain alphanumeric characters.
9. Is Internal
0 or 1, denoting whether both parties on the call are internal or external (1 being an internal
call). Calls to destinations on other switches in a network are indicated as internal.
Direction Is Internal Call Type
I 0 Incoming external call.
O 1 Internal call.
O 0 Outgoing external call.

10. Call ID

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Appendix: SMDR

This is a number starting from 1,000,000 and incremented by 1 for each unique call. If the
call has generates several SMDR records, each record will have the same Call ID. Note
that the Call ID used is restarted from 1,000,000 if the system is restarted.
11. Continuation
1 if there is a further record for this call id, 0 otherwise.
12. Party1Device
The device 1 number. This is usually the call initiator though in some scenarios such as
conferences this may vary. If an extension/hunt group is involved in the call its details will
have priority over a trunk. That includes remote network destinations.
Type Party Device Party Name
Internal Number E<extension number> <name>
Voicemail V<9500 + channel number> VM Channel <channel
number>
Conference V<1><conference number> CO Channel <conference
+<channel number> number.channel number>
Line T<9000+line number> Line <line number>.<channel if
applicable>
Other V<8000+device number> U<device class> <device
number>.<device channel>
Unknown/Tone V8000 U1 0.0

13. Party1Name
The name of the device – for an extension or agent, this is the user name.
14. Party2Device
The other party for the SMDR record of this call segment. See Party1Device above.
For barred calls, this field is populated with “Barred”.
15. Party2Name
The other party for the SMDR record of this call segment. See Party1Name above.
For barred calls, this field is populated with “Barred”.
16. Hold Time
The amount of time in seconds the call has been held during this call segment.
17. Park Time
The amount of time in seconds the call has been parked during this call segment.
18. AuthValid
This field is used for authorization codes. This field shows 1 for valid authorization or 0 for
invalid authorization.
19. AuthCode

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 SMDR Fields

For security, this field shows n/a regardless of whether an authorization code was used.
20. User Charged
This and the following fields are used for ISDN Advice of Charge (AoC). The user to which
the call charge has been assigned. This is not necessarily the user involved in the call.
21. Call Charge
The total call charge calculated using the line cost per unit and user markup.
22. Currency
The currency. This is a system wide setting set in the system configuration.
23. Amount at Last User Change
The current AoC amount at user change.
24. Call Units
The total call units.
25. Units at Last User Change
The current AoC units at user change.
26. Cost per Unit
This value is set in the system configuration against each line on which Advice of Charge
signalling is set. The values are 1/10,000th of a currency unit. For example if the call cost
per unit is £1.07, a value of 10700 should be set on the line.
27. Mark Up
Indicates the mark up value set in the system configuration for the user to which the call is
being charged. The field is in units of 1/100th, for example an entry of 100 is a markup
factor of 1 .
28. External Targeting Cause
This field indicates who or what caused the external call and a reason code. For example
U FU indicates that the external call was caused by the Forward Unconditional setting of a
User.
Targeted by Reason Code
HG Hunt Group. fb Forward on Busy.
U User. fu Forward unconditional.
LINE Line. fnr Forward on No
Response.
AA Auto Attendant. fdnd Forward on DND.
ICR Incoming Call Route. CfP Conference proposal
(consultation) call.
Table continues…

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Appendix: SMDR

Targeted by Reason Code

RAS Remote Access Cfd Conferenced.
Service.
? Other. MT Mobile Twinning.
TW Teleworker.
XfP Transfer proposal
(consultation) call.
Xfd Transferred call.

29. External Targeter Id

The associated name of the targeter indicated in the External Targeting Cause field. For
hunt groups and users this will be their name in the system configuration. For an Incoming
Call Route this will be the Tag if set, otherwise ICR.
30. External Targeted Number
This field is used for forwarded, Incoming Call Route targeted and mobile twin calls to an
external line. It shows the external number called by the system as a result of the off switch
targeting where as other called fields give the original number dialled.
31. Server IP address of the caller extension
Fields 31 to 34 are used to uniquely identify a call made in an IP Office Server Edition
solution. This IP address identifies the server where the call was initiated. If the field does
not contain an IP address, then the call originated outside the IP Office network.
32. Unique call id for the caller extension
Numerical value that is a unique identifier of the call on the server where the call was
initiated.
33. Server IP address of the called extension
This IP address identifies the server where the called extension is logged in. If the field
does not contain an IP address, then the call is to a trunk outside the IP Office network.
34. Unique call id for the called extension
Numerical value that is a unique identifier of the call on the server where the called
extension is logged in.
35. UTC time
Coordinated Universal Time (UTC) using the format YYYY/MM/DD HH:MM:SS.
Related links
Appendix: SMDR on page 1115

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 SMDR Examples

SMDR Examples
The following are examples of system SMDR records for common call scenarios.
Lost incoming Call
In this record, the Call duration is zero and the Continuation field is 0, indicating that the call was
never connected. The Ring Time shows that it rang for 9 seconds before ending.
2014/06/28 09:28:41,00:00:00,9,8004206,I,4324,4324,,0,1000014155,0,E4324,Joe
Bloggs,T9161,LINE 5.1,0,0,,,,,,,,,,,,,

Call Answered by Voicemail

In this example, 215 has made a call to 211. However the Party2Device and Party2Name show
that the call was answered by voicemail.
2014/10/20 06:43:58,00:00:10,21,215,O,211,211,,I,28,0,E215,Extn215,V9051,VM Channel
1,0,0,,,,,,,,,,,,,

Call Transferred to Voicemail

In this example, the Continuation field in the first record tells us that it wasn't the end of the call.
The matching Call ID identifies the second record as part of the same call. The change in Party 1
details between the two records show that the call was transferred to voicemail.
2014/06/28 09:30:57,00:00:13,7,01707392200,I,299999,299999,,0,1000014160,1,E4750,John
Smith,T9002,LINE 1.2,11,0,,,,,,,,,,,,, 2014/06/28 09:30:57,00:00:21,0,01707392200,I,
299999,299999,,0,1000014160,0,V9502,VM Channel 2,T9002,LINE 1.2,0,0,,,,,,,,,,,,,

External Call
The Is Internal field being 0 shows this to be a external call. The Direction field as I shows that it
was an incoming call. The Ring Time was 7 seconds and the total Connected Time was 5
seconds.
2014/08/01 15:14:19,00:00:05,7,01707299900,I,
403,390664,,0,1000013,0,E403,Extn403,T9001,Line 1.2,0,0,,,,,,,,,,,,,,

Internal Call
The Is Internal field being 1 shows this to be a internal call. The Ring Time was 4 seconds and the
total Connected Time was 44 seconds.
2014/06/26 10:27:44,00:00:44,4,4688,O,4207,4207,,1,1000013898,0,E4688,Joe
Bloggs,E4207,John Smith,0,0,,,,,,,,,,,,,

Outgoing Call
The combination of the Direction field being outbound and the Is Internal field be 0 show that this
was a outgoing external call. The line (and in this case channel) used are indicated by the Party2
Name and being a digital channel the Ring Time before the call was answered is also shown.
2014/06/28 08:55:02,00:08:51,9,4797,O,08000123456,08000123456,,0,1000014129,0,E4797,Joe
Bloggs,T9001,LINE 1.1,0,0,,,,,,,,,,,,,

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Appendix: SMDR

Voicemail Call
The two records below show calls to voicemail. The first shows the Dialed Number as*17, the
default short code for voicemail access. The second shows the Dialed Number as VoiceMail,
indicating some other method such as the Message key on a phone was used to initiate the call.
2014/06/28 09:06:03,00:00:19,0,4966,O,*17,*17[1],,1,1000014131,0,E4966,John
Smith,V9501,VM Channel 1,0,0,,,,,,,,,,,,, 2014/06/28
09:06:03,00:00:19,0,4966,O,VoiceMail,VoiceMail,,1,1000014134,0,E4966,John
Smith,V9501,VM Channel 1,0,0,,,,,,,,,,,,,

Parked Call
In this example the first record has a Park Time showing that the call was parked. The
Continuation field indicates that the call did not end this way and there are further records. The
second record has the same Call ID and shows a change in the Party2Name [4], indicating that
party unparked the call. Note also that both records share the same call start time.
2014/10/20 07:18:31,00:00:12,3,215,O,
210,210,,1,38,1,E215,Extn215,E210,Extn210,0,7,,,,,,,,,,,,, 2014/10/20
07:18:31,00:00:10,0,215,O,210,210,,1,38,0,E215,Extn215,E211,Extn211,0,0,,,,,,,,,,,,,

Incoming call with Account Code

In this example, at some stage as the call was made or during the call, an Account Code has been
entered.
2014/06/28 11:29:12,00:00:02,2,5002,I,1924,1924,Support,
0,1000014169,0,E1924,Extn1924,T9620,LINE 8.20,0,0,,,,,,,,,,,,,

Conference Using Conference Add Short Code

In this example 2101 has made a call and put put it on hold (record 2), then made another call and
put it on hold (record 1) and then dialled the default short code *47 to conference all their held
calls (record 3). The records for the first two calls have the Continuation field set as 1 indicating
that the calls continued in further records.
Record 3 shows 2101 making a new call in which they dial *47, which places them and their held
calls into a conference. This is shown by the Party Device and Party Name details as being a
conference (100) and the conference channel used for each.
For both the Continuation fields show that the calls do not end but rather have subsequent
records.
2014/07/09 17:55,00:00:03,3,2101,O,8262623#,8262623#,,
0,1000024,1,E2101,Extn2101,T9002,Line 2.1,8,0,,,,,,,,,,,,,
2014/07/09 17:54,00:00:29,7,2101,O,
2121,2121,,1,1000023,1,E2101,Extn2101,E2121,Extn2121,23,0,,,,,,,,,,,,,
2014/07/09 17:55,00:00:46,0,2101,O,*47,*47,,1,1000026,0,E2101,Extn2101,V11001,CO
Channel 100.1,0,0,,,,,,,,,,,,,
2014/07/09 17:54,00:00:49,0,,O,
71234567890,71234567890,,1,1000023,0,E2121,Extn2121,V11003,CO Channel
100.3,0,0,,,,,,,,,,,,,
2014/07/09 17:55,00:00:49,0,,O,8262623#,8262623#,,0,1000024,0,V11002,CO Channel
100.2,T9002,Line 2.1,0,0,,,,,,,,,,,,,

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 SMDR Examples

Conference Using Conference Button

In this example, an extension user answers a call and then brings in another user by using the
Conference button on their phone. Again we see records for the initial call, the conference
proposal call and then for the 3 parties in the conference that is created.
2014/07/09 15:05:41,00:00:04,3,203,O,
201,201,,1,1000009,1,E203,Extn203,E201,Extn201,0,0,,,,,,,,,,,,,
2014/07/09 15:05:26,00:00:09,3,207,O,
203,203,,1,1000008,1,E207,Extn207,E203,Extn203,10,0,,,,,,,,,,,,,
2014/07/09 15:05:41,00:00:08,0,,O,,,,1,1000009,0,E201,Extn201,V11001,CO Channel
100.1,0,0,,,,,,,,,,,,,
2014/07/09 15:05:50,00:00:10,0,203,O,201,201,,1,1000010,0,E203,Extn203,V11002,CO
Channel 100.2,0,0,,,,,,,,,,,,,
2014/07/09 15:05:26,00:00:10,0,207,O,203,203,,1,1000008,0,E207,Extn207,V11003,CO
Channel 100.3,0,0,,,,,,,,,,,,,

Adding a Party to a Conference

This example is a variant on that above. Having started a conference, extension 203 adds another
party.
2014/07/09 15:08:31,00:00:03,3,203,O,
201,201,,1,1000014,1,E203,Extn203,E201,Extn201,0,0,,,,,,,,,,,,,
2014/07/09 15:08:02,00:00:22,6,207,O,
203,203,,1,1000013,1,E207,Extn207,E203,Extn203,9,0,,,,,,,,,,,,,
2014/07/09 15:08:45,00:00:02,4,203,O,403,403,,0,1000016,1,E203,Extn203,E403,Libby
Franks,0,0,,,,,,,,,,,,,
2014/07/09 15:08:02,00:00:24,0,207,O,203,203,,1,1000013,0,E207,Extn207,V11003,CO
Channel 100.3,0,0,,,,,,,,,,,,,
2014/07/09 15:08:39,00:00:17,0,203,O,201,201,,1,1000015,0,E203,Extn203,V11002,CO
Channel 100.2,8,0,,,,,,,,,,,,,
2014/07/09 15:08:31,00:00:26,0,,O,,,,1,1000014,0,E201,Extn201,V11001,CO Channel
100.1,0,0,,,,,,,,,,,,,
2014/07/09 15:08:45,00:00:12,0,,O,403,403,,0,1000016,0,E403,Libby Franks,V11004,CO
Channel 100.4,0,0,,,,,,,,,,,,,

Transfer
In this example 2126 has called 2102. The record (1) for this has the Continuation set a 1
indicating that it has further records. In the following record (3) with the same Call ID it can be
seen that the Party 2 Device and Party 2 Name fields have changed, indicating that the call is now
connected to a different device, in this example 2121. We can infer the blind transfer from the
intermediate record (2) which shows a call of zero Connected Time between the original call
destination 2102 and the final destination 2121.
2014/07/09 17:51,00:00:38,18,2126,O,
2102,2102,,1,1000019,1,E2126,Extn2126,E2102,Extn2102,19,0,,,,,,,,,,,,,
2014/07/09 17:52,00:00:00,7,2102,O,
2121,2121,,1,1000020,0,E2102,Extn2102,E2121,Extn2121,0,0,,,,,,,,,,,,,
2014/07/09 17:51,00:00:39,16,2126,O,
2102,2102,,1,1000019,0,E2126,Extn2126,E2121,Extn2121,0,0,,,,,,,,,,,,,

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Appendix: SMDR

In this second example extension 402 answers an external call and then transfers it to extension
403. Again the two legs of the external call have the same time/date stamp and same call ID.
2014/08/01 15:23:37,00:00:04,7,01707299900,I,
4001,390664,,0,1000019,1,E402,Extn402,T9001,Line 1.1,6,0,,,,,,,,,,,,,,
2014/08/01 15:23:46,00:00:00,3,402,O,
403,403,,1,1000020,0,E402,Extn402,E403,Extn403,0,0,,,,,,,,,,,,,,
2014/08/01 15:23:37,00:00:04,4,01707299900,I,
4001,390664,,0,1000019,0,E403,Extn403,T9001,Line 1.1,0,0,,,,,,,,,,,,,,

Busy/Number Unavailable Tone

In this example 2122 calls 2123 who is set to DND without voicemail. This results in 2122
receiving busy tone.
The records shows a call with a Connected Time of 0. The Call Number field shows 2123 as the
call target but the Party 2 Device and Party 2 Name fields show that the connection is to a virtual
device.
2014/07/09 17:59,00:00:00,0,2122,O,2123,2123,,1,1000033,0,E2122,Extn2122,V8000,U1
0.0,0,0,,,,,,,,,,,,,

Call Pickup
The first record shows a call from 2122 to 2124 with a Connected Time of zero but a Ring Time of
8. The Continuation field indicates that the call has further records.
The second record has the same Call ID but the Party 2 Device and Party 2 Name details show
that the call has been answered by 2121.
2014/07/09 18:00,00:00:00,8,2122,O,
2124,2124,,1,1000038,1,E2122,Extn2122,E2124,Extn2124,0,0,,,,,,,,,,,,,
2014/07/09 18:00,00:00:38,1,2122,O,
2124,2124,,1,1000038,0,E2122,Extn2122,E2121,Extn2121,0,0,,,,,,,,,,,,,

Internal Twinning
The records for scenarios such as internal call forwarding or follow me indicate the rerouting in a
single record by having Caller and Called Number details that differ from the final Party 1 and
Party 2 details. Internal twinning differs is showing a call answered at the twin exactly the same as
having been answered at the primary.
203 is internally twinned to 201. Call from 207 to 203 but answer at 201.
2014/07/09 16:25:26,00:00:03,7,207,O,
203,203,,1,1000037,0,E207,Extn207,E203,Extn203,0,0,,,,,,,,,,,,,

Park and Unpark

Parking and unparking of a call at the same extension is simply shown by the Park Time field of
the eventual SMDR record. Similarly calls held and unheld at the same extension are shown by
the Held Time field of the eventual SMDR record for the call. The records below however show a
call parked at one extension and then unparked at another.

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 SMDR Examples

The records show a call from 207 to 203. 203 then parks the call shown by the Park Time. The call
is unparked by 201, hence the first record is indicated as continued in its Continuation field. The
matching Call ID indicates the subsequent record for the call.
2014/07/09 16:39:11,00:00:00,2,207,O,
203,203,,1,1000052,1,E207,Extn207,E203,Extn203,0,4,,,,,,,,,,,,,
2014/07/09 16:39:11,00:00:02,0,207,O,
203,203,,1,1000052,0,E207,Extn207,E201,Extn201,0,0,,,,,,,,,,,,,

Distributed Hunt Group Call

An incoming call to site A is targeted to a distributed hunt group member on site B. They transfer
the call back to a hunt group member on site A.
2014/08/01 15:32:52,00:00:10,19,01707299900,I,
4002,390664,,0,1000024,1,E209,Luther-209,T9001,Line 1.2,0,0,,,,,,,,,,,,,,
2014/08/01 15:33:19,00:00:00,2,209,I,
403,403,,0,1000025,0,E209,Luther-209,E403,Extn403,0,0,,,,,,,,,,,,,,
2014/08/01 15:32:52,00:00:03,3,01707299900,I,
4002,390664,,0,1000024,0,E403,Extn403,T9001,Line 1.2,0,0,,,,,,,,,,,,,,

Voicemail Supervised Transfer

A call is routed to a voicemail module that performs a supervised transfer.
2014/08/01 16:36:04,00:00:09,0,01707299900,I,xfer,390664,,0,1000061,1,T9001,Line
1.1,V9508,VM Channel 8,0,0,,,,,,,,,,,,,,
2014/08/01 16:36:07,00:00:03,4,,I,402,402,,0,1000062,0,E402,Extn402,V8000,U12
0.8,0,0,,,,,,,,,,,,,,
2014/08/01 16:36:04,00:00:09,0,01707299900,I,
402,390664,,0,1000061,0,E402,Extn402,T9001,Line 1.1,0,0,,,,,,,,,,,,,,

Outgoing External Call

The External Targeting Cause indicates that the external call was caused by a user. The lack of
specific reason implies that it was most likely dialed. The External Targeter ID is the user name in
this example
… 16:23:06,00:00:04,5,203,O,9416,9416,,0,1000035,0,E203,Extn203,T9005,Line
5.1,0,0,,,Extn203,,,,,,,,U,Extn203,,

Rerouted External Call

In this example an incoming external call has been rerouted back off switch, shown by the Party 1
fields and the Party 2 fields being external line details. The External Targeter Cause shows that
rerouting of the incoming call was done by an incoming call route (ICR). The External Targeter ID
in this case is the Tag set on the incoming call route. The External Targeted Number is the actual
external number call.
… 08:14:27,00:00:03,5,392200,I,9416,200,,0,1000073,0,T9005,Line 5.1,T9005,Line
5.2,0,0,,,,0000.00,,0000.00,0,0,618,0.01,ICR,Main ICR,416,

External Forward Unconditional

In this example, user 203 has a forward unconditional number set for calls. This is indicated by the
External Targeting Cause showing user and forward unconditional. The External Targeter ID

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Appendix: SMDR

shows the source of the call being forwarded, in this example user 207. The External Targeted
Number shows the actual external number called by the system.
… 16:22:41,00:00:02,5,207,O,203,203,,0,1000034,0,E207,Extn207,T9005,Line
5.1,0,0,,,Extn203,0000.00,,0000.00,0,0,618,1.00,U fu,Extn207,9416,

Transferred Manually
In this example the internal user transfers a call to an external number. The External Targeting
Cause in the first record indicates that this external call is the result of a user (U) transfer proposal
(XfP) call. The Continuation field indicates that another record with the same Call ID will be output.
The additional records are output after the transferred call is completed. The first relates to the
initial call prior. The second is the transferred call with the External Targeting Cause now indicating
user (U) transferred (Xfd).
… 16:33:19,00:00:05,3,203,O,9416,9416,,0,1000044,1,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U XfP,Extn207,,
… 16:33:09,00:00:02,2,207,O,
203,203,,1,1000043,0,E207,Extn207,E203,Extn203,11,0,,,,,,,,,,,,,,
… 16:33:19,00:00:04,0,207,O,9416,9416,,0,1000044,0,E207,Extn207,T9005,Line
5.1,0,0,,,Extn207,,,,,,,,U Xfd,Extn203,,

Mobile Twinned Call Answered Internally

For this example user 203 has mobile twining enabled to the external number 9416 as twin. Their
mobile dial delay is set to 2 seconds. The call is answered at the user's internal extension.
In this scenario the record for the external call part of twinning is output immediately the call is
answered internally. The Call Start for this record differs dues to the user's Mobile Dial Delay
setting. The External Targeting Cause indicates the external call was the result of user (U) mobile
twinning (MT) settings. If the call had been answered before the mobile dial delay expired, no
external call and therefore no record would be produced. When the call is completed the second
record is output.
… 16:17:59,00:00:00,7,,O,9416,9416,,0,1000028,0,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U MT,Extn203,9416,
… 16:17:58,00:00:07,9,207,O,
203,203,,1,1000027,0,E207,Extn207,E203,Extn203,0,0,,,,,,,,,,,,,,

Mobile Twinned Call Answered at the Mobile Twin

This is the same scenario as the example above except that the call is answered at the external
mobile twinning destination. Unlike the previous example the external call record has a non-zero
Call Time showing that the call was also answered externally.
… 16:17:04,00:00:06,9,,O,9416,9416,,0,1000026,0,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U MT,Extn203,9416
… 16:17:02,00:00:06,11,207,O,
203,203,,1,1000025,0,E207,Extn207,E203,Extn203,0,0,,,,,,,,,,,,,,

Mobile Twinned Call Picked Up Using the Twinning Button

This is the same scenario as the example above, however after answering the call on the external
twinned device, the user has picked it up internally by using a twinning button. The first two

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 SMDR Examples

records are for the answered external call and are output when that call is picked up by the
internal extension. The third record is output when the call is ended internally.
… 16:19:18,00:00:05,11,207,O,
203,203,,1,1000029,1,E207,Extn207,E203,Extn203,0,0,,,,,,,,,,,,,,
… 16:19:20,00:00:05,9,,O,9416,9416,,0,1000030,0,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U MT,Extn203,9416
… 16:19:18,00:00:05,0,207,O,
203,203,,1,1000029,0,E207,Extn207,E203,Extn203,0,0,,,,,,,,,,,,,,

External Conference Party

This is similar to internal conferencing (see examples above) but the conference setup and
progress records include External Targeting Cause codes for user (U) conference proposal (CfP)
and user (U) conferenced (Cfd).
… 16:48:58,00:00:02,2,203,O,9416,9416,,0,1000066,1,E203,Extn203,T9005,Line
5.1,0,0,,,,,,,,,,,U CfP,Extn203,,
… 16:48:37,00:00:04,3,203,O,
207,207,,1,1000064,1,E203,Extn203,E207,Extn207,7,0,,,,,,,,,,,,,,
… 16:49:04,00:00:08,0,203,O,9416,9416,,1,1000067,0,E203,Extn203,V11002,CO Channel
100.2,0,0,,,,,,,,,,,,,,
… 16:48:37,00:00:13,0,,O,,,,1,1000064,0,E207,Extn207,V11003,CO Channel
100.3,0,0,,,,,,,,,,,,,,
… 16:48:58,00:00:13,0,,O,9416,9416,,0,1000066,0,V11001,CO Channel 100.1,T9005,Line
5.1,0,0,,,Extn203,,,,,,,,U Cfd,Extn203,

Call Routed by Incoming Call Route

Call from external number 403 rerouted by incoming call route (ICR) for incoming line group 701
back out to 404.
2014/08/01 11:45:36,00:00:01,2,403,I,9404,,,0,1000007,0,T9001,Line 1.0,T9010,Line
10.0,0,0,0,n/a,,,,,,,,,ICR,ICR701,404

Two Outgoing External Calls Transferred Together

This scenario shows an outgoing call which is then transferred to another outgoing call.
2009/02/19 11:13:26,00:00:06,0,203,O,9403,9403,,0,1000012,1,E203,Extn203,T9001,Line
1.0,8,0,0,n/a,,,,,,,,,U,Extn203,,
2009/02/19 11:13:36,00:00:02,0,203,O,8404,8404,,0,1000013,0,E203,Extn203,T9002,Line
2.0,0,0,0,n/a,,,,,,,,,U XfP,Extn203,,
2009/02/19 11:13:26,00:00:11,0,8404,I,404,,,0,1000012,0,T9002,Line 2.0,T9001,Line
1.0,0,0,0,n/a,,,,,,,,,LINE Xfd,0.1038.0 13 Alog Trunk:2,,

Authorization code
In this example, an authorization code was used and the 0 indicates that it is invalid:
2014/02/20 11:04:59,00:00:00,0,319,O,,,,0,1000009,0,E319,Alice,V8000,U1 0.0,0,0,0,n/
a,,,,,,,,,U,Alice,

In this example, the authorization code is valid.

2014/02/20 11:04:59,00:00:00,0,319,O,,,,0,1000009,0,E319,Alice,V8000,U1 0.0,0,0,1,n/
a,,,,,,,,,U,Alice,

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Appendix: SMDR

Server Edition Call ID

In this example, a call is made from extension 1234 on Expansion 1 to extension 4321 on
Expansion 2.
• Expansion 1 IP address: 192:168:42:192
• Call ID on Expansion 1: 1002
• Expansion 2 IP address: 192:168:42:193
• Call ID on Expansion 2: 1004
Primary output:
2014/04/08 16:42:05,00:00:01,3,1234,O,
4321,4321,,1,1000000,0,E1234,Extn1234,E4321,Extn4321,0,0,,,,,,,,,192:168:42:192,1002,192
:168:42:193,1004,

Expansion 1 output:
2014/04/08 16:42:04,00:00:01,3,1234,O,
4321,4321,,1,1000000,0,E1234,Extn1234,E4321,Extn4321,0,0,,,,,,,,,,192:168:42:192,1002,19
2:168:42:193,1004,

Expansion 2 output:
2014/04/08 13:42:05,00:00:01,3,1234,I,
4321,4321,,1,1000000,0,E1234,Extn1234,E4321,Extn4321,0,0,,,,,,,,,,192:168:42:192,1002,19
2:168:42:193,1004,

Related links
Appendix: SMDR on page 1115

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Chapter 24: Documentation resources

For a listing of documentation resources related to IP Office, see Avaya IP Office™ Platform Start
Here First. Download documents from the Avaya Support website at http://support.avaya.com.
IP Office documentation is also available on the IP Office Knowledgebase at http://
marketingtools.avaya.com/knowledgebase/.
Related links
Finding documents on the Avaya Support website on page 1129

Finding documents on the Avaya Support website

Procedure
1. Navigate to http://support.avaya.com/.
2. At the top of the screen, type your username and password and click Login.
3. Click Support by Product > Documents.
4. In Enter your Product Here, type the product name and then select the product from the
list.
5. In Choose Release, select an appropriate release number.
6. In the Content Type filter, click a document type, or click Select All to see a list of all
available documents.
For example, for user guides, click User Guides in the Content Type filter. The list
displays the documents only from the selected category.
7. Click Enter.
Related links
Documentation resources on page 1129

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Chapter 24: Support

Visit the Avaya Support website at http://support.avaya.com for the most up-to-date documentation,
product notices, and knowledge articles. You can also search for release notes, downloads, and
resolutions to issues. Use the online service request system to create a service request. Chat with
live agents to get answers to questions, or request an agent to connect you to a support team if an
issue requires additional expertise.
For questions regarding IP Office documentation, send an email to [email protected].

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Chapter 24: Using the Avaya InSite
Knowledge Base

The Avaya InSite Knowledge Base is a web-based search engine that provides:
• Up-to-date troubleshooting procedures and technical tips
• Information about service packs
• Access to customer and technical documentation
• Information about training and certification programs
• Links to other pertinent information
If you are an authorized Avaya Partner or a current Avaya customer with a support contract, you can
access the Knowledge Base without extra cost. You must have a login account and a valid Sold-To
number.
Use the Avaya InSite Knowledge Base for any potential solutions to problems.
1. Go to http://www.avaya.com/support.
2. Log on to the Avaya website with a valid Avaya user ID and password.
The system displays the Avaya Support page.
3. Click Support by Product > Product Specific Support.
4. In Enter Product Name, enter the product, and press Enter.
5. Select the product from the list, and select a release.
6. Click the Technical Solutions tab to see articles.
7. Select relevant articles.

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Chapter 24: Viewing Avaya Mentor videos

Avaya Mentor videos provide technical content on how to install, configure, and troubleshoot
Avaya products.
About this task
Videos are available on the Avaya Support website, listed under the video document type, and on
the Avaya-run channel on YouTube.
Procedure
• To find videos on the Avaya Support website, go to http://support.avaya.com and perform one
of the following actions:
- In Search, type Avaya Mentor Videos to see a list of the available videos.
- In Search, type the product name. On the Search Results page, select Video in the
Content Type column on the left.
• To find the Avaya Mentor videos on YouTube, go to www.youtube.com/AvayaMentor and
perform one of the following actions:
- Enter a key word or key words in the Search Channel to search for a specific product or
topic.
- Scroll down Playlists, and click the name of a topic to see the available list of videos posted
on the website.
Note:
Videos are not available for all products.

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Chapter 24: Additional IP Office resources

You can find information at the following additional resource websites.

Avaya
http://www.avaya.com is the official Avaya website. The front page also provides access to
individual Avaya websites for different countries.
Avaya Sales & Partner Portal
http://sales.avaya.com is the official website for all Avaya Business Partners. The site requires
registration for a user name and password. Once accessed, the portal can be customized for
specific products and information types that you wish to see and be notified about by email.
Avaya IP Office Knowledge Base
http://marketingtools.avaya.com/knowledgebase provides access to an online, regularly updated
version of the IP Office Knowledge Base.
Avaya maintenance, lifecycle and warranty information
Avaya support services complement standard Avaya maintenance, lifecycle and warranty policies
that are posted on http://support.avaya.com. For more information, send email to
[email protected].
International Avaya User Group
http://www.iaug.org is the official discussion forum for Avaya product users.

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Index
Numerics certificate management (continued)
windows certificate store ............................................ 568
4400/6400 ..........................................................................440 certificates ..........................................................................122
certificate support .............................................................. 571
file import .................................................................... 577
A file naming and format ................................................ 571
account code configuration ................................................672 identity certificate ........................................................572
account code configuration fields ...................................... 514 signing certificate ........................................................576
voice recording ........................................................... 514 trusted certificate store ............................................... 574
advanced ............................................................................. 57 change mode ....................................................................... 61
erase configuration ....................................................... 57 change working directory .....................................................45
reboot ........................................................................... 58 close configuration ...............................................................43
system shutdown ..........................................................58 codec selection .................................................................. 760
upgrade ........................................................................ 59 configuration field
changing the .bin file directory ...............................61 Access Control Lists ................................................... 261
searching for systems ........................................... 61 account code
advanced view ..................................................................... 74 voice recording .................................................... 514
alarms ................................................................................ 246 analog .........................................................................387
analog extension ................................................................387 analog line .................................................................. 267
analog line ......................................................................... 267 analog line options ......................................................269
analog options ............................................................ 269 analog line settings .....................................................268
line settings .................................................................268 ARS ............................................................................ 543
announcements ................................................................. 445 authorization codes .................................................... 533
ARS configuration fields .................................................... 543 auto attendant
audit trail .............................................................................. 62 actions ................................................................. 531
authorization codes configuration fields .............................533 BOOTP ....................................................................... 190
auto attendant configuration fields .............................528, 529 BRI line ............................................................... 276, 281
actions ........................................................................ 531 call details ...................................................................335
autoconnect ....................................................................... 481 call log ........................................................................ 237
auto intercom deny off ....................................................... 813 control unit .................................................................. 382
auto intercom deny on ....................................................... 813 customer call reporter .................................................256
DHCP pools ................................................................213
dialer ...........................................................................261
B directory ......................................................................504
directory services ................................................240, 243
backup/restore ..................................................................... 73
display
bandwidth .......................................................................... 477
server edition mode .............................................188
barred calls ........................................................................ 648
standard mode .................................................... 187
applying ...................................................................... 648
DNS ............................................................................ 215
overriding ....................................................................649
E1 line .........................................................................282
BOOTP record ................................................................... 190
E1 line channels ......................................................... 289
BRI line .............................................................................. 276
E1 R2 Advanced .........................................................295
channels ..................................................................... 281
E1 R2 channels .......................................................... 293
busy on held validation ........................................................ 77
E1 R2 line ................................................................... 291
button programming ...........................................................439
E1 R2 MFC group .......................................................295
extension ............ 383, 384, 387, 390, 394, 398, 400, 401
C firewall profile
custom .................................................................509
call log ........................................................................237, 431 standard .............................................................. 507
call settings ........................................................................ 422 static NAT ............................................................ 511
centralized licensing .......................................................... 554 group
certificate management ..................................................... 566 fallback ................................................................ 461
overview ..................................................................... 567 overflow ............................................................... 459

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 Index

WAN port (continued) WAN port (continued)

group (continued) SIP engineering .......................................................... 359
select members ................................................... 455 SIP line ....................................................................... 326
SIP .......................................................................473 SIP T38 Fax ................................................................352
voicemail ............................................................. 463 SIP transport ...............................................................330
voice recording .................................................... 469 SIP VoIP ............................................................. 347, 394
H.323 line ................................................................... 315 SMDR ......................................................................... 253
H.323 short codes ...................................................... 318 SM line ........................................................................362
H.323 VoIP ................................................................. 390 SM line T38 Fax ......................................................... 370
H.323 VoIP settings .................................................... 318 SM line VoIP ............................................................... 365
incoming call route SMTP ..........................................................................252
standard .............................................................. 493 system ................................................................ 192, 193
voice recording ............................................ 497, 498 system events .............................................................244
IP DECT ..................................................................... 400 system events alarms .................................................246
IP DECT gateway ....................................................... 322 system events configuration ....................................... 245
IP DECT line ...............................................................321 System Manager ........................................................ 235
IP DECT VoIP .............................................................324 T1 channels ................................................................ 299
IP Office line ............................................................... 371 T1 line .........................................................................297
IP Office line short codes ............................................377 T1 PRI line ..................................................................302
IP Office line T38 Fax ................................................. 380 T38 fax ........................................................................398
IP Office line VoIP settings ......................................... 377 telephony .................................................................... 222
IP route time profile .......................................................... 505, 582
RIP dynamic routing ............................................ 513 tones and music ......................................................... 230
LAN1 ...........................................................................199 TUI ..............................................................................238
LAN2 ...........................................................................214 tunnel
LAN settings ............................................................... 199 IP security tunnel .................................................525
LAN VoIP .................................................................... 201 IKE policies .................................................. 526
license IPSec policies ...............................................527
remote server ...................................................... 518 main ............................................................. 525
line .............................................................................. 266 L2TP tunnel ................................................. 523, 524
location user .............................................................401, 402, 414
address ................................................................549 announcements ...................................................445
network topology ........................................................ 209 button programming ............................................ 439
park and page .............................................................229 dial in ................................................................... 436
PRI trunks ...................................................................281 DND .....................................................................413
remote access server forwarding ............................................................433
PPP ..................................................................... 489 hunt group membership ...................................... 445
ring tones ....................................................................234 menu programming ............................................. 440
S0 line .........................................................................312 mobility ................................................................ 441
service personal directory ................................................447
autoconnect .........................................................481 self administration ............................................... 449
bandwidth ............................................................ 477 SIP .......................................................................446
dial in ................................................................... 485 source numbers ...................................................415
fallback ................................................................ 484 telephony .............................................................421
IP ......................................................................... 479 voicemail ............................................................. 408
PPP ..................................................................... 482 voice recording .................................................... 437
quota ................................................................... 481 user rights
SSL VPN ..................................................... 485, 486 forwarding ............................................................542
SSL VPN fallback ................................................ 488 telephony .............................................................536
SSL VPN NAPT ...................................................487 user ..................................................................... 534
SSL VPN session ................................................ 486 user rights membership ...............................540, 541
short code ...................................................................474 voice compression modules ....................................... 254
shorty codes ............................................................... 414 voicemail .....................................................................215
SIP advanced ............................................................. 354 VoIP .................................................................... 257, 390
SIP credentials ........................................................... 353 VoIP security ...............................................................259
SIP DECT base .................................................. 360, 401 WAN port
SIP DECT line ............................................................ 360 advanced .............................................................503
SIP DECT VoIP .......................................................... 361

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Index

WAN port (continued) F

WAN port (continued)
DLCI .................................................................... 501 fallback .......................................................................461, 484
frame relay .......................................................... 501 fax over SIP ....................................................................... 761
connect to ............................................................................ 76 file menu .............................................................................. 42
Contact Center ...................................................................263 change working directory ..............................................45
Contact field .......................................................................752 close configuration ........................................................43
control unit ......................................................................... 382 open configuration ........................................................ 43
create new config .................................................................55 save configuration ........................................................ 43
customer call reporter ........................................................ 256 save configuration as ....................................................45
firewall ................................................................................507
custom ........................................................................ 509
D standard ......................................................................507
destination URI .................................................................. 751 static NAT ....................................................................511
details pane ................................................................... 74, 90 firewall profile configuration fields ......................................507
details toolbar ...................................................................... 87 custom ........................................................................ 509
DHCP pools ....................................................................... 213 standard ......................................................................507
dialer .................................................................................. 261 static NAT ....................................................................511
dial in ................................................................................. 436 format SD card .....................................................................64
directories ............................................................................ 50 forwarding .......................................................................... 433
directory configuration fields ...................................... 240, 504 From field ...........................................................................751
directory services
HTTP .......................................................................... 243 G
LDAP .......................................................................... 240
discovery ..............................................................................50 general security fields ........................................................ 131
DNS ................................................................................... 215 getting started ...................................................................... 35
do not disturb ..................................................................... 413 Gmail integration ................................................................657
DTMF .................................................................................761 group configuration fields ...........................................451, 456
fallback ....................................................................... 461
overflow ...................................................................... 459
E select members .......................................................... 455
E1 line ................................................................................282 SIP ..............................................................................473
channels ..................................................................... 289 voicemail .....................................................................463
E1 R2 line .......................................................................... 291 voice recording ........................................................... 469
Advanced ....................................................................295 group operation ..................................................................694
channels ..................................................................... 293 group pane .....................................................................74, 88
MFC group ..................................................................295 add record .................................................................... 89
embedded file management ................................................ 64 columns displayed ........................................................ 89
embedded file management menus .................................... 82 delete record .................................................................90
erase configuration .............................................................. 57 show in groups ............................................................. 90
error pane ...................................................................... 74, 92 sorting ...........................................................................88
automatic validation settings ........................................ 93 validate record .............................................................. 90
revalidating settings ......................................................93
viewing error ................................................................. 93 H
exit ....................................................................................... 74
export user ...........................................................................76 H.323 extension VoIP ........................................................ 390
extension configuration fields ............................................ 383 H.323 line ...........................................................................315
analog .........................................................................387 short codes ................................................................. 318
extension .................................................................... 384 VoIP setting ................................................................ 318
H.323 VoIP ................................................................. 390 headers ..............................................................................768
IP DECT ..................................................................... 400 hold scenarios ....................................................................761
SIP DECT base .......................................................... 401 HTTP directory services .................................................... 243
SIP VoIP ..................................................................... 394 huntgroup ...........................................................................440
T38 fax ........................................................................398 hunt group membership .....................................................445
VoIP ............................................................................ 390
extension renumber ............................................................. 75

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 Index

I line configuration fields (continued)

analog line settings .....................................................268
icons BRI line ............................................................... 276, 281
changing size ................................................................97 control unit .................................................................. 382
import/export ........................................................................73 E1 line .........................................................................282
import templates .................................................................. 79 E1 line channels ......................................................... 289
incoming call E1 R2 Advanced .........................................................295
call scenarios ..............................................................757 E1 R2 channels .......................................................... 293
message details ..........................................................755 E1 R2 line ................................................................... 291
routing .........................................................................755 E1 R2 MFC group .......................................................295
incoming call route configuration fields ..............................490 H.323 line ................................................................... 315
standard ......................................................................493 H.323 short codes ...................................................... 318
voice recording ................................................... 497, 498 H.323 VoIP setting ......................................................318
incoming calls IP DECT gateway ....................................................... 322
media path connection ............................................... 756 IP DECT line ...............................................................321
initial configuration ............................................................... 68 IP DECT VoIP .............................................................324
InSite Knowledge Base .................................................... 1131 IP Office line ............................................................... 371
IP DECT .............................................................................400 IP Office line short codes ............................................377
IP DECT gateway .............................................................. 322 IP Office line T38 Fax ................................................. 380
IP DECT line ...................................................................... 321 IP Office line VoIP settings ......................................... 377
IP DECT VoIP .................................................................... 324 PRI trunks ...................................................................281
IP Office line .......................................................................371 S0 line .........................................................................312
short codes ................................................................. 377 SIP advanced ............................................................. 354
VoIP settings ...............................................................377 SIP credentials ........................................................... 353
IP Office line T38 Fax ........................................................ 380 SIP DECT base .......................................................... 360
IP route SIP DECT line ............................................................ 360
configuring .................................................................. 642 SIP DECT VoIP .......................................................... 361
IP route configuration fields ........................................511, 512 SIP engineering .......................................................... 359
RIP dynamic routing ................................................... 513 SIP line ....................................................................... 326
IP security tunnel ............................................................... 525 SIP T38 Fax ................................................................352
IKE policies .................................................................526 SIP transport ...............................................................330
IPSec policies ............................................................. 527 SIP URI .......................................................................335
main ............................................................................525 SIP VoIP ..................................................................... 347
SM line ........................................................................362
SM line T38 Fax ......................................................... 370
L SM line VoIP ............................................................... 365
L2TP tunnel ....................................................................... 523 T1 channels ................................................................ 299
L2TP ........................................................................... 524 T1 line .........................................................................297
PPP ............................................................................ 524 T1 PRI line ..................................................................302
LAN1 ..................................................................................199 line renumber .......................................................................75
LAN2 ..................................................................................214 location
LAN settings ...................................................................... 199 address .......................................................................549
LAN VoIP ........................................................................... 201 location configuration fields ............................................... 547
launch Voicemail Pro ........................................................... 67 LVM greeting utility ...............................................................67
LDAP directory services .................................................... 240
license M
configuring .................................................................. 552
license configuration fields .................................................516 madn ................................................................................1012
remote server ............................................................. 518 main toolbar ......................................................................... 86
license file Manager
uploading .................................................................... 561 synchronize passwords .............................................. 727
license migration ..........................................................80, 565 manager modes ...................................................................28
licensing media path connection .......................................................756
enterprise branch ........................................................564 memory card command ....................................................... 66
line configuration fields ...................................................... 266 menu bar commands ........................................................... 42
analog line .................................................................. 267 advanced .......................................................... 57–59, 61
analog line options ......................................................269 audit trail ................................................................62

February 2019 Administering Avaya IP Office™ Platform with Manager 1137

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Index

tools (continued) NoCallerId alarm (continued)

advanced (continued) suppressing ................................................................ 718
change mode .........................................................61 No User ..............................................................................717
embedded file management ..................................64
erase security settings ...........................................63
format SD card ...................................................... 64
O
initial configuration .................................................68 offline ................................................................................... 55
launch Voicemail Pro .............................................67 create new config ......................................................... 55
LVM greeting utility ................................................ 67 open file ........................................................................ 56
memory card command .........................................66 open file set .................................................................. 56
recreate SD card ................................................... 65 receive config ............................................................... 56
security settings .....................................................63 send config ................................................................... 56
System Status ....................................................... 67 on-boarding: configuring SSL VPN ....................................580
VM locales .............................................................72 open configuration ............................................................... 43
backup/restore ..............................................................73 open file ............................................................................... 56
embedded file management ......................................... 82 open file set ......................................................................... 56
exit ................................................................................ 74 opening a configuration ....................................................... 38
file ............................................. 42, 55–59, 61–68, 72–74 login messages .............................................................39
file menu ................................................................. 43, 45 operator ............................................................................. 192
import/export .................................................................73 Outbound Contact Express
offline ...................................................................... 55, 56 dialer ...........................................................................261
preferences .................................................46, 50–52, 55 outgoing call
security mode ............................................................... 81 call scenarios ..............................................................753
tools Contact field ................................................................752
busy on held validation ..........................................77 destination URI ........................................................... 751
connect to ..............................................................76 From field ....................................................................751
export user ............................................................ 76 message details ..........................................................751
extension renumber ...............................................75 P-Asserted Identity field ..............................................752
import templates ....................................................79 To field ........................................................................ 752
license migration ................................................... 80 overflow ............................................................................. 459
line renumber ........................................................ 75 overview ...............................................................................28
MSN configuration .................................................78
print button labels .................................................. 78
SCN service user management ............................ 77 P
view .............................................................................. 74
panes
menu programming ............................................................440
moving the details pane ................................................96
message waiting indication ................................................704
showing or hiding ..........................................................96
migrating ADI licenses ....................................................... 565
park and page ....................................................................229
mobility ...............................................................................441
P-Asserted Identity field .....................................................752
moving borders .................................................................... 95
PC requirements ..................................................................35
MSN configuration ............................................................... 78
personal directory .............................................................. 447
multiple call appearance .................................................. 1012
PLDS licensing .................................................................. 552
mult-line options .................................................................429
preferences ..........................................................................46
music on hold
directories ..................................................................... 50
alternate source ..........................................................607
discovery ...................................................................... 50
system source ............................................................ 607
preferences ...................................................................46
security ......................................................................... 52
N validation ...................................................................... 55
visual preferences ........................................................ 51
navigation pane ............................................................. 74, 87 print button labels ................................................................ 78
navigation toolbar ................................................................ 87 PRI trunks .......................................................................... 281
network topology ................................................................209 E1 line .........................................................................282
new in release E1 line channels ......................................................... 289
one-X resiliency .......................................................... 733 E1 R2 Advanced .........................................................295
resiliency .....................................................................733 E1 R2 channels .......................................................... 293
new in this release ............................................................... 25 E1 R2 line ................................................................... 291
NoCallerId alarm E1 R2 MFC group .......................................................295

February 2019 Administering Avaya IP Office™ Platform with Manager 1138

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 Index

PRI trunks (continued) security settings ................................................................... 63

T1 channels ................................................................ 299 erase .............................................................................63
T1 line .........................................................................297 self administration ..............................................................449
T1 PRI line ..................................................................302 send config .......................................................................... 56
server edition
default settings ........................................................... 101
Q record consolidation ................................................... 102
quota ..................................................................................481 solution view ................................................................. 98
system inventories ......................................................101
user interface ................................................................98
R server edition licenses
distributing .................................................................. 554
reboot ...................................................................................58
service configuration fields ........................................ 475, 476
receive config .......................................................................56
autoconnect ................................................................ 481
record consolidation ...........................................................102
bandwidth ................................................................... 477
recreate SD card ..................................................................65
dial in .......................................................................... 485
remote access server configuration fields ......................... 488
fallback ....................................................................... 484
PPP ............................................................................ 489
IP ................................................................................ 479
remote server .....................................................................518
PPP ............................................................................ 482
request methods ................................................................ 767
quota ...........................................................................481
resiliancy
SSL VPN ............................................................ 485, 486
location based ............................................................ 742
SSL VPN fallback ....................................................... 488
resilience ............................................................................729
SSL VPN NAPT ..........................................................487
resiliency ............................................................................729
SSL VPN session ....................................................... 486
resizing the manager window .............................................. 95
service users ......................................................................105
resource websites ............................................................ 1133
short code configuration fields ........................................... 474
response methods ............................................................. 767
short code feature
RFC ................................................................................... 766
auto intercom deny off ................................................ 813
ring tones ........................................................................... 234
auto intercom deny on ................................................ 813
RIP dynamic routing .......................................................... 513
short codes ........................................................................ 414
simplified view ......................................................................74
S SIP ..................................................................................... 446
SIP advanced .................................................................... 354
S0 line ................................................................................312 SIP credentials ...................................................................353
save configuration ................................................................43 SIP DECT base ................................................................. 401
save configuration as ...........................................................45 SIP DECT line ....................................................................360
SCN service user management ...........................................77 SIP DECT base .......................................................... 360
security field .......................................................................130 SIP DECT VoIP .......................................................... 361
general ........................................................................131 SIP engineering ................................................................. 359
rights groups SIP extension VoIP ............................................................ 394
configuration ........................................................146 SIP line .............................................................................. 326
group details ........................................................146 sip line appearances ..........................................................343
security administration .........................................148 SIP messaging ...................................................................750
services settings ......................................................... 144 SIP prefix ........................................................................... 749
service users .............................................................. 152 SIP REFER ........................................................................763
system SIP T38 Fax .......................................................................352
certificates ........................................................... 139 SIP transport ......................................................................330
system details ......................................................136 SIP trunk
unsecured interfaces ........................................... 138 configuring .................................................................. 745
system status overview ..................................................................... 744
external ................................................................151 SIP trunks
HTTP ................................................................... 149 configuring .................................................................. 744
security administration .........................................148 sip uri ................................................................................. 336
Telephony APIs ................................................... 149 SIP URI ..............................................................................335
web services ........................................................149 SIP VoIP ............................................................................ 347
security mode menus ...........................................................81 SMDR .......................................................................253, 1115
security preferences ............................................................ 52 examples ...................................................................1121

February 2019 Administering Avaya IP Office™ Platform with Manager 1139

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Index

SMDR (continued) system status (continued)

field descriptions ....................................................... 1116 rights groups (continued)
SM line ...............................................................................362 configuration ........................................................146
T38 Fax ...................................................................... 370 group details ........................................................146
VoIP ............................................................................ 365 security administration .........................................148
SMTP .................................................................................252 services settings ......................................................... 144
source numbers ................................................................. 415 service users .............................................................. 152
SRTP ................................................................................. 125 system details .............................................................136
SSL VPN ....................................................................485, 486 system status
SSL VPN fallback .............................................................. 488 external ................................................................151
SSL VPN NAPT ................................................................. 487 HTTP ................................................................... 149
SSL VPN session .............................................................. 486 security administration .........................................148
Station Message Detail Reporting ....................................1115 Telephony APIs ................................................... 149
examples ...................................................................1121 web services ........................................................149
field descriptions ....................................................... 1116 unsecured interfaces .................................................. 138
status bar ............................................................................. 94 system shutdown ................................................................. 58
supervisors settings ........................................................... 425 System Status ......................................................................67
system configuration field
Avaya Cloud Services .................................................265
Contact Center ........................................................... 263
T
system configuration fields ................................................ 192 T1 line ................................................................................ 297
Access Control Lists ................................................... 261 channels ..................................................................... 299
call log ........................................................................ 237 T1 PRI line ......................................................................... 302
customer call reporter .................................................256 T38 fax ...............................................................................398
DHCP pools ................................................................213 tabs
dialer ...........................................................................261 changing display of .......................................................97
DNS ............................................................................ 215 telephony ................................................................... 222, 421
HTTP directory services ............................................. 243 call log ................................................................ 237, 431
LAN1 ...........................................................................199 call settings .................................................................422
LAN2 ...........................................................................214 mult-line options ......................................................... 429
LAN settings ............................................................... 199 park and page .............................................................229
LAN VoIP .................................................................... 201 ring tones ....................................................................234
LDAP directory services ............................................. 240 supervisor settings ......................................................425
network topology ........................................................ 209 System Manager ........................................................ 235
ring tones ....................................................................234 tones and music ......................................................... 230
SMDR ......................................................................... 253 TUI ......................................................................238, 432
SMTP ..........................................................................252 template
system ........................................................................ 193 analog trunk ................................................................588
system events .............................................................244 creating .......................................................................587
system events alarms .................................................246 templates
system events configuration ....................................... 245 creating a new record from a template ....................... 587
System Manager ........................................................ 235 TFTP log .............................................................................. 74
telephony .................................................................... 222 time profile configuration fields .................................. 505, 582
park and page ..................................................... 229 title bar ................................................................................. 85
tones and music ......................................................... 230 To field ............................................................................... 752
TUI ..............................................................................238 tones and music .................................................................230
voice compression modules ....................................... 254 toolbars .......................................................................... 74, 85
voicemail .....................................................................215 details ........................................................................... 87
VoIP ............................................................................ 257 main ..............................................................................86
VoIP security ...............................................................259 moving .......................................................................... 96
system events ............................................................244, 644 navigation ..................................................................... 87
system events alarms ........................................................ 246 showing or hiding ..........................................................95
system events configuration .............................................. 245 tools menu ........................................................................... 75
System Manager ................................................................235 tooltip ................................................................................... 74
system security fields .........................................................136 transport protocols ............................................................. 767
certificates .................................................................. 139 trunk templates .................................................................. 585
rights groups applying a template to an analog trunk .......................589

February 2019 Administering Avaya IP Office™ Platform with Manager 1140

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 Index

trunk templates (continued) toolsbars (continued)

importing .....................................................................586 group pane (continued)
TUI ............................................................................. 238, 432 columns displayed .................................................89
tunnel configuration fields .......................................... 522–527 delete record ......................................................... 90
twinning ......................................................................441, 712 show in groups ...................................................... 90
sorting ....................................................................88
validate record .......................................................90
U navigation pane ............................................................ 87
upgrade ................................................................................59 server edition ........................................................ 98, 101
changing the .bin file directory ......................................61 solution view ................................................................. 98
searching for systems ...................................................61 status bar ......................................................................94
user ....................................................................................402 system inventories ......................................................101
No User .............................................................. 717, 718 title bar ..........................................................................85
suppressing NoCallerId alarm .................................... 718 toolbars .........................................................................85
user configuration fields .....................................................401 toolsbars
announcements .......................................................... 445 details .................................................................... 87
button programming ................................................... 439 main .......................................................................86
dial in .......................................................................... 436 navigation .............................................................. 87
DND ............................................................................413 user management overview ...............................................651
forwarding ...................................................................433 user rights configuration fields ........................................... 534
hunt group membership ..............................................445 forwarding ...................................................................542
menu programming .................................................... 440 telephony .................................................................... 536
4400/6400 ........................................................... 440 call log ................................................................. 539
huntgroup ............................................................ 440 call settings ..........................................................536
mobility ....................................................................... 441 multi-line options ................................................. 539
personal directory ....................................................... 447 supervisor settings .............................................. 537
self administration .......................................................449 user .............................................................................534
short codes ................................................................. 414 user rights membership ...................................... 540, 541
SIP ..............................................................................446
source numbers ..........................................................415 V
telephony .................................................................... 421
call log ................................................................. 431 validation preferences ..........................................................55
call settings ..........................................................422 videos ...............................................................................1132
mult-line options .................................................. 429 visual preferences ................................................................51
supervisor settings .............................................. 425 VM locales ........................................................................... 72
TUI .......................................................................432 voice compression modules .............................................. 254
user .............................................................................402 voicemail ............................................................215, 408, 463
voicemail .....................................................................408 voice recording .................................. 437, 469, 497, 498, 514
voice recording ........................................................... 437 VoIP ........................................................................... 257, 261
user interface ....................................................................... 85 VoIP security ......................................................................259
configuring
icon size ................................................................ 97
moving borders ......................................................95
W
moving the details pane ........................................ 96 WAN port configuration fields ............................................ 500
moving toolbars ..................................................... 96 advanced .................................................................... 503
resizing the manager window ................................95 DLCI ........................................................................... 501
showing or hiding panes ........................................96 frame relay ..................................................................501
showing or hiding toolbars .....................................95 Web License Manager ....................................................... 553
tab display ............................................................. 97 WebLM ...............................................................................553
default settings ........................................................... 101 installing a license file .................................................560
details pane .................................................................. 90 WebLM host ID .................................................................. 560
error pane ..................................................................... 92
automatic validation settings ................................. 93
revalidating settings ...............................................93
viewing error ..........................................................93
group pane ................................................................... 88
add record ............................................................. 89

February 2019 Administering Avaya IP Office™ Platform with Manager 1141

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