Avaya Solution & Interoperability Test Lab

Configuring Alcatel OmniPCX Enterprise with Avaya

Aura® Communication Manager 6.0.1 and Avaya
Aura® Session Manager 6.1 – Issue 1.0

Abstract

These Application Notes present a sample configuration for a network consisting of an

Avaya Aura® Communication Manager and Alcatel OmniPCX Enterprise. These two
systems are connected via a common Avaya Aura® Session Manager.

Testing was conducted via the Internal Interoperability Program at the Avaya Solution
and Interoperability Test Lab.

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 1. Introduction
The purpose of this interoperability application note is to validate Alcatel OmniPCX
Enterprise (OXE) with Avaya Aura® Communication Manager (CM) which are both
connected to an Avaya Aura® Session Manager via a separate SIP Trunk. Voicemail
integration between Alcatel OmniPCX Enterprise and Avaya Aura® Messaging was not
included in the scope of this Application Notes. The sample network is shown in Figure
1, where the Alcatel OmniPCX Enterprise supports the Alcatel ipTouch 4028 / 4038 /
4068 IP Telephones. SIP trunks are used to connect Avaya Aura® Communication
Manager and Alcatel OmniPCX Enterprise to Avaya Aura® Session Manager. All inter-
system calls are carried over these SIP trunks. Avaya Aura® Session Manager can support
flexible inter-system call routing based on dialed number, calling number and system
location, and can also provide protocol adaptation to allow for multi-vendor systems to
interoperate. Avaya Aura® Session Manager is managed by a separate Avaya Aura®
System Manager, which can manage multiple Avaya Aura® Session Managers.

Figure 1: Connection of Alcatel OmniPCX Enterprise and Avaya Aura®

Communication Manager via Avaya Aura® Session Manager using SIP Trunks

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 Alcatel phones are registered to Alcatel OmniPCX Enterprise. Alcatel OmniPCX
Enterprise registered stations use extensions 3600x. One SIP trunk is provisioned to the
Avaya Aura® Session Manager to manage calls to/from Alcatel OmniPCX Enterprise.
One SIP trunk is provisioned to the Avaya Aura® Session Manager to manage calls
to/from Avaya Aura® Communication Manager.

2. Equipment and Software Validated

The following equipment and software/firmware were used for the sample configuration:

Hardware Component Software Version

Avaya Aura® Communication Manager
Avaya S8800 Media Servers with G650
6.0.1
Media Gateway
(R016x.00.1.510.1)
Avaya S8510 Server Avaya Aura® Session Manager 6.1 SP0
Avaya S8800 Server Avaya Aura® System Manager 6.1 SP0
Avaya A175 Desktop Video Device 1.0.1
Avaya 1140 IP Telephone (SIP) 04.00.04.00
Avaya 96x1 IP Telephone (SIP) 6.1 SP2
Avaya 96x1 IP Telephone (H.323) 6.0 SP2
Avaya 2420 Digital Telephone -
Alcatel OmniPCX Enterprise 9.1 (I1.605-16-c)
Alcatel ipTouch NOE Telephone 4.20.71

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 3. Configure Avaya Aura® Communication Manager
This section shows the configuration in Communication Manager. All configurations in
this section are administered using the System Access Terminal (SAT). These
Application Notes assumed that the basic configuration has already been administered.
For further information on Communication Manager, please consult with reference
Error! Reference source not found.. The procedures include the following areas:
 Verify Avaya Aura® Communication Manager License
 Administer System Parameters Features
 Administer IP Node Names
 Administer IP Network Region and Codec Set
 Administer SIP Signaling Group and Trunk Group
 Administer Route Pattern
 Administer Private Numbering
 Administer Locations
 Administer Dial Plan and AAR analysis
 Save Changes

3.1. Verify Avaya Aura® Communication Manager License

Use the display system-parameter customer options command to compare the
Maximum Administered SIP Trunks field value with the corresponding value in the
USED column. The difference between the two values needs to be greater than or equal
to the desired number of simultaneous SIP trunk connections.

Note: The license file installed on the system controls the maximum features permitted.
If there is insufficient capacity or a required feature is not enabled, contact an authorized
Avaya sales representative to make the appropriate changes.
display system-parameters customer-options Page 2 of 11
OPTIONAL FEATURES

IP PORT CAPACITIES USED

Maximum Administered H.323 Trunks: 30 0
Maximum Concurrently Registered IP Stations: 18000 9
Maximum Administered Remote Office Trunks: 0 0
Maximum Concurrently Registered Remote Office Stations: 0 0
Maximum Concurrently Registered IP eCons: 0 0
Max Concur Registered Unauthenticated H.323 Stations: 0 0
Maximum Video Capable Stations: 10 1
Maximum Video Capable IP Softphones: 10 4
Maximum Administered SIP Trunks: 100 55

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 3.2. Administer System Parameters Features
Use the change system-parameters features command to allow for trunk-to-trunk
transfers. This feature is needed to allow for transferring an incoming/outgoing call
from/to a remote switch back out to the same or different switch. For simplicity, the
Trunk-to-Trunk Transfer field was set to all to enable all trunk-to-trunk transfers on a
system wide basis.

Note: This feature poses significant security risk and must be used with caution. As an
alternative, the trunk-to-trunk feature can be implemented using Class Of Restriction or
Class Of Service levels.
change system-parameters features Page 1 of 18
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? y
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? n
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
AAR/ARS Dial Tone Required? y
Music/Tone on Hold: none
Music (or Silence) on Transferred Trunk Calls? no
DID/Tie/ISDN/SIP Intercept Treatment: attd
Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred
Automatic Circuit Assurance (ACA) Enabled? n

3.3. Administer IP Node Names

Use the change node-names ip command to add entries for Communication Manager
and Session Manager that will be used for connectivity. In the sample network, clan and
10.10.9.45 are entered as Name and IP Address for the CLAN card in Communication
Manager running on the Avaya S8800 Server. In addition, sm100 and 10.10.9.34 are
entered for Session Manager.
change node-names ip Page 1 of 2
IP NODE NAMES
Name IP Address
sm100 10.10.9.34
clan 10.10.9.45
default 0.0.0.0
gateway 10.10.9.1
medpro 10.10.9.46
procr 10.10.9.42
procr6 ::

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 3.4. Administer IP Network Region and Codec Set
Use the change ip-network-region n command, where n is the network region number
to configure the network region being used. In the sample network, ip-network-region 1
is used. For the Authoritative Domain field, enter the SIP domain name configured for
this enterprise and a descriptive Name for this ip-network-region. Set Intra-region IP-
IP Direct Audio and Inter-region IP-IP Direct Audio to yes to allow for direct media
between endpoints. Set the Codec Set to 1 to use ip-codec-set 1.
change ip-network-region 1 Page 1 of 20
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: mmsil.local
Name: To ASM61
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 46
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5

Use the change ip-codec-set n command, where n is the existing codec set number to
configure the desired audio codec.
change ip-codec-set 1 Page 1 of 2

IP Codec Set

Codec Set: 1

Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)
1: G.711MU n 2 20
2: G.729 n 2 20

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 3.5. Administer SIP Signaling Group and Trunk Group
3.5.1. SIP Signaling Group
In the test configuration, Communication Manager acts as an Evolution Server. An IMS
enabled SIP trunk is not required. Use signal group 150 along with trunk group 150 to
reach the Session Manager. Use the add signaling-group n command, where n is the
signaling-group number being added to the system. Use the values defined in Section 3.3
and 3.4 for Near-end Node Name, Far-End Node-Name and Far-End Network
Region. The Far-end Domain is left blank so that the signaling group accepts any
authoritative domain. Set IMS Enabled to n and Peer Detection Enabled to y.
add signaling-group 150 Page 1 of 1
SIGNALING GROUP

Group Number: 150 Group Type: sip

IMS Enabled? n Transport Method: tcp
Q-SIP? n SIP Enabled LSP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: SM

Near-end Node Name: clan Far-end Node Name: sm100

Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 1

Far-end Domain:
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

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 3.5.2. SIP Trunk Group
Use the add trunk-group n command, where n is the new trunk group number being
added to the system. The following screens show the settings used for trunk group 150.
Enter the following:
 Group Type sip
 TAC a numeric value i.e. 150
 Service Type tie
 Signaling Group the signaling group defined in Section 3.5.1, i.e. 150
 Number of Members set to a numeric value, i.e. 10
add trunk-group 150 Page 1 of 21
TRUNK GROUP

Group Number: 150 Group Type: sip CDR Reports: y

Group Name: Trunk 150 COR: 1 TN: 1 TAC: 150
Direction: two-way Outgoing Display? y
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Member Assignment Method: auto
Signaling Group: 150
Number of Members: 10

Navigate to Page 3 and enter private for Numbering Format.

add trunk-group 150 Page 3 of 21
TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y

Numbering Format: private

UUI Treatment: service-provider

Replace Restricted Numbers? n

Replace Unavailable Numbers? n

Modify Tandem Calling Number: no

Show ANSWERED BY on Display? y
Navigate to Page 4 and enter 97 for Telephone Event Payload Type and From for
Identity for Calling Party Display.
add trunk-group 150 Page 4 of 21
PROTOCOL VARIATIONS

Mark Users as Phone? n

Prepend '+' to Calling Number? n
Send Transferring Party Information? y
Network Call Redirection? n
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 97

Convert 180 to 183 for Early Media? n

Always Use re-INVITE for Display Updates? n
Identity for Calling Party Display: From
Enable Q-SIP? n

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 3.6. Administer Route Pattern
Configure a route pattern to correspond to the newly added SIP trunk group. Use the
change route-pattern n command, where n is the route pattern number specified in
Section 3.9. Configure this route pattern to route calls to trunk group number 150
configured in Section 3.5.2. Assign the lowest FRL (facility restriction level) to allow all
callers to use this route pattern. Assign 0 to No. Del Dgts.
change route-pattern 150 Page 1 of 3
Pattern Number: 150 Pattern Name: To ASM
SCCAN? n Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC
No Mrk Lmt List Del Digits QSIG
Dgts Intw
1: 150 0 0 n user
2: n user
3: n user
4: n user

BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n unre none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none

3.7. Administer Private Numbering

Use the change private-numbering command to define the calling party number to be
sent out through the SIP trunk. In the sample network configuration below, all calls
originating from a 5-digit extension beginning with 40 and 41 will result in a 5-digit
calling number. The calling party number will be in the SIP “From” header.
change private-numbering 0 Page 1 of 2
NUMBERING - PRIVATE FORMAT

Ext Ext Trk Private Total

Len Code Grp(s) Prefix Len
5 40 5 Total Administered: 4
5 41 5 Maximum Entries: 540

3.8. Administer Locations

Use the change locations command to define the proxy route to use for outgoing calls. In
the sample network the proxy route will be the trunk group defined in Section 3.5.2.
change locations Page 1 of 1
LOCATIONS

ARS Prefix 1 Required For 10-Digit NANP Calls? y

Loc Name Timezone Rule NPA Proxy Sel

No Offset Rte Pat
1: Main + 00:00 0 150

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 3.9. Administer Dial Plan and AAR analysis
Configure the dial plan for dialing 5-digit extensions beginning with 36 to stations
registered with Alcatel OXE. Use the change dialplan analysis command to define
Dialed String 36 as an aar Call Type.
change dialplan analysis Page 1 of 12
DIAL PLAN ANALYSIS TABLE
Location: all Percent Full: 2

Dialed Total Call Dialed Total Call Dialed Total Call

String Length Type String Length Type String Length Type
1 3 dac
36 5 aar
39990 5 ext
39995 5 aar
40 5 ext
41 5 ext
6 3 fac
7 5 ext
* 2 fac
# 2 fac

Use the change aar analysis 0 command to configure an aar entry for Dialed String 36
to use Route Pattern 150. Add an entry for the SIP phone extensions which begin with
41. Use unku for call type.
change aar analysis 0 Page 1 of 2
AAR DIGIT ANALYSIS TABLE
Location: all Percent Full: 1

Dialed Total Route Call Node ANI

String Min Max Pattern Type Num Reqd
36 5 5 150 unku n
3999 5 5 150 unku n
41 5 5 150 unku n

3.10. Save Changes

Use the save translation command to save all changes.
save translation

SAVE TRANSLATION

Command Completion Status Error Code

Success 0

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 4. Configuring Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. For further
information on Session Manager, please consult with references [1], [2], and [3]. The
procedures include the following areas:
 Log in to Avaya Aura® Session Manager
 Administer SIP domain
 Administer Locations
 Administer Adaptations
 Administer SIP Entities
 Administer Entity Links
 Administer Time Ranges
 Administer Routing Policies
 Administer Dial Patterns
 Administer Avaya Aura® Session Manager
 Add Avaya Aura® Communication Manager as an Evolution Server
 Administer SIP users

4.1. Log in to Avaya Aura® Session Manager

Access the System Manager using a Web Browser and entering http://<ip-
address>/SMGR, where <ip-address> is the IP address of System Manager. Log in using
appropriate credentials.

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 In the next screen under Elements column select Routing.

In the main panel, a short procedure for configuring Network Routing Policy is shown.

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 4.2. Administer SIP Domain
Add the SIP domain, for which the communications infrastructure will be authoritative,
by selecting Domains on the left panel menu and clicking the New button (not shown) to
create a new SIP domain entry. Complete the following options:
 Name The authoritative domain name (e.g., mmsil.local)
 Type SIP
 Notes Description for the domain (optional)

Click Commit to save changes.

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 4.3. Administer Locations
Session Manager uses the origination location to determine which dial patterns to look at
when routing the call if there are dial patterns administered for specific locations.
Locations are also used to limit the number of calls coming out of or going to a physical
location. This is useful for those locations where the network bandwidth is limited. To
add a Location, select Locations on the left panel menu and then click the New button
(not shown). Enter the following for each Location:
Under General:
 Name An informative name (e.g. Dublin)
Under Location Pattern:
 IP Address Pattern An IP address pattern for this location

Select Add to add more locations. Click Commit to save changes.

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 4.4. Administer Adaptations
Create an adaptation entry for a call to Alcatel OXE. For the Alcatel OXE adaptation,
enter the following information:
 Name An informative name for the adaptation
 Adaptation Module Enter a DigitConversionAdapter to ensure the request
URI domain on outgoing calls to Alcatel OXE is
node1.mmsil.local (the Alcatel FQDN). See Section 5.2
 Digit Conversion for incoming Calls to SM
Matching Pattern 360 with a minimum and maximum of
5 digits long, which is the dial pattern for a station
registered with Alcatel OXE. Delete Digits has value 0 to
indicate no digits are to be deleted.

Click Commit to save changes.

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 4.5. Administer SIP Entities
A SIP Entity must be added for Session Manager and for each SIP-based telephony
system supported by a SIP Trunk. To add a SIP Entity, select SIP Entities on the left
panel menu and then click on the New button (not shown). Enter the following for each
SIP Entity:
Under General:
 Name An informative name (e.g., SessionManager)
 FQDN or IP Address IP address of the signaling interface on the Session
Manager, CM or OXE.
 Type Session Manager for Session Manager, CM for CM and
SIP Trunk for OXE.
 Location Dublin
 Time Zone Time zone for this location

The following screen shows the SIP Entity for Session Manager.

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 Under Port, click Add, and then edit the fields in the resulting new row.
 Port Port number on which the system listens for SIP requests
 Protocol Transport protocol to be used to send SIP requests

The following screen shows the Port definitions for the Session Manager SIP Entity.
Click Commit to save changes.

The following screen shows the SIP Entity for CM.

The following screen shows the SIP Entity for OXE.

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 4.6. Administer Entity Links
A SIP trunk between a Session Manager and a telephony system is described by an Entity
Link. To add an Entity Link, select Entity Links on the left panel menu and click on the
New button (not shown). Fill in the following fields in the new row that is displayed.
 Name An informative name
 SIP Entity 1 Select SessionManager
 Port Port number to which the other system sends its SIP requests
 SIP Entity 2 The other SIP Entity for this link, created in Section 4.5
 Port Port number to which the other system expects to receive SIP
requests
 Trusted Whether to trust the other system
 Protocol Transport protocol to be used to send SIP requests

Click Commit to save changes. The following screen shows the Entity Links used in the
sample network.

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 4.7. Administer Time Ranges
Before adding routing policies (see next step), time ranges must be defined during which
the policies will be active. To add this time range, select Time Ranges from the left panel
menu and then click New on the right. Fill in the following fields.
 Name An informative name (e.g. always)
 Mo through Su Check the box under each day of the week for inclusion
 Start Time Enter start time (e.g. 00:00 for start of day)
 End Time Enter end time (e.g. 23:59 for end of day)

In Session Manager, a default policy (24/7) is available that would allow routing to occur
at anytime. This time range was used in the sample network.

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 4.8. Administer Routing Policies
Create routing policies to direct how calls will be routed to a system. Two routing
policies must be added, one for Communication Manager (H.323 and Digital phones) and
one for Alcatel OXE. To add a routing policy, select Routing Policies on the left panel
menu and then click on the New button (not shown).
Under General:
 Enter an informative Name
 Under SIP Entity as Destination:
 Click Select, and then select the appropriate SIP entity to which this routing
policy applies
Under Time of Day:
 Click Add, and then select the time range configured in the Section 4.7.

Click Commit to save changes. The following screen shows the Routing Policy Details
for calls to Communication Manager.

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 The following is screen shows the Routing Policy Details for Alcatel OXE.

4.9. Administer Dial Patterns

A dial pattern must be defined that will direct calls to the appropriate telephony system.
In the sample network, 5-digit extensions beginning with 360 reside on Alcatel OXE and
5-digit extensions beginning with 40 (H.323 and Digital phones) reside on CM. To
configure the Alcatel OXE Dial Pattern select Dial Patterns on the left panel menu and
then click on the New button (not shown).
Under General:
 Pattern Dialed number or prefix
 Min Minimum length of dialed number
 Max Maximum length of dialed number
 Notes Comment on purpose of dial pattern
 SIP Domain Select ALL

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 Navigate to Originating Locations and Routing Policy List and select Add (not
shown). Under Originating Location select Apply The Selected Routing Policies to
All Originating Locations and under Routing Policies select Alcatel PBX. Click Select
button to confirm the chosen options and then be returned to the Dial Pattern screen
(shown previously), select Commit button to save.

A dial pattern must be defined that will direct calls to CM (H.323 and Digital phones). To
configure the CM Dial Pattern select Dial Patterns on the left panel menu and then click
on the New button (not shown).
Under General:
 Pattern Dialed number or prefix
 Min Minimum length of dialed number
 Max Maximum length of dialed number
 Notes Comment on purpose of dial pattern
 SIP Domain Select ALL

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 Navigate to Originating Locations and Routing Policy List and select Add (not
shown). Under Originating Location select all locations by checking the box next to
Apply The Selected Routing Policies to All Originating Locations and under Routing
Policies select the Routing Policy created for CM in Section 4.8. Click Select button to
confirm the chosen options and then be returned to the Dial Pattern screen (shown
above), select Commit button to save.

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 4.10. Administer Avaya Aura® Session Manager
To complete the configuration, adding the Session Manager will provide the linkage
between System Manager and Session Manager. On the SMGR management screen
under the Elements column select Session Manager.

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 In the left panel on the next screen, select Session Manager Administration and in the
right panel under Session Manager Instances select New (not shown). Fill in the fields
as described below and shown in the following screen:
Under General:
 SIP Entity Name Select the name of the SIP Entity added for Session Manager
 Description Descriptive comment (optional)
 Management Access Point Host Name/IP
Enter the IP address of the Session Manager management
interface
Under Security Module:
 Network Mask Enter the network mask corresponding to the IP address of
Session Manager
 Default Gateway Enter the IP address of the default gateway for Session
Manager

Use default values for the remaining fields. Click Commit to add this Session Manager.

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 4.11. Add Avaya Aura® Communication Manager as an Evolution
Server
In order for Communication Manager to provide configuration and Evolution Server
support to SIP phones when they register to Session Manager, Communication Manager
must be added as an application.

4.11.1.Create a CM Instance
On the SMGR management screen under the Elements column select Inventory.

Select Manage Elements on the left. Click on New.

On the next screen (not shown), for Type select CM.

Click on the Applications tab and enter the following fields. Use defaults for the
remaining fields:
 Name A descriptive name
 Description A description of the CM instance

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  Node Enter the IP address for CM SAT access

Click on the Attributes tab and enter the following:

 Login Login used for SAT access
 Password Password used for SAT access
 Confirm Password Password used for SAT access

Click on Commit to save.

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 4.11.2.Create an Evolution Server Application
On the SMGR management screen under the Elements column select Session Manager.

Select Application Configuration  Applications on the left menu. Click on New (not
shown). Enter following fields and use defaults for the remaining fields. Click on
Commit to save.
 Name A descriptive name
 SIP Entity Select the CM SIP Entity defined in Section 4.5
 CM System for SIP Entity Select the CM instance created in Section 4.11.1

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 4.11.3.Create an Evolution Server Application Sequence
Select Application Configuration  Application Sequences on the left menu. Click on
New (not shown). Enter a descriptive Name. Click on the + sign next to the appropriate
Available Applications and they will move up to the Applications in this Sequence
section. Click on Commit to save.

4.11.4.Synchronize Avaya Aura® Communication Manager Data

On the SMGR management screen under the Elements column select Inventory.

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 Select Synchronization  Communication System on the left. Select the appropriate
Element Name. Select Initialize data for selected devices. Then click on Now. This
may take some time.

4.12. Administer SIP Users

SIP users must be added via Session Manager and the details will be updated on the CM.
On the SMGR management screen under the Users column select User Management.

Select Manage Users on the left. Then click on New.

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 Click on the Identity tab and enter the following and use defaults for other fields:
 Last Name A desired last name
 First Name A desired first name
 Login Name The desired phone extension [email protected] where
domain was defined in Section 4.2
 Password Password for user to log into System Manager (SMGR)

Click on the Communication Profile tab. Enter the following and defaults for the
remaining fields:
 Shared Communication
Profile Password Password to be entered by the user when logging
into the phone
 Type Select Avaya SIP
 Fully Qualified Address Enter the extension number and select the domain

Click on Add.

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 Navigate to the Session Manager Profile and Endpoint Profile sections. Select the
appropriate Session Manager server for Primary Session Manager. For Origination
Application Sequence and Termination Application Sequence select the application
sequence created in Section 4.11.3. Choose the Home Location created in Section 4.3.
Click on Endpoint Profile to expand that section. Enter the following fields and use
defaults for the remaining fields:
 System Select the CM Entity
 Extension Enter a desired extension number
 Template Select a telephone type template
 Port Select IP

Click on Commit to save (not shown).

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 5. Configure Alcatel OmniPCX Enterprise
This section shows the configuration in Alcatel OmniPCX Enterprise. All configurations
in this section are administered using the Command Line Interface. These Application
Notes assumed that the basic configuration has already been administered. For further
information on Alcatel OmniPCX Enterprise, please consult with reference Error!
Reference source not found.. The procedures include the following areas:
 Verify Alcatel OXE Licences
 Access the Alcatel OXE Manager
 Administer IP Domain
 Administer SIP Trunk Group
 Administer SIP Gateway
 Administer SIP Proxy
 Administer SIP External Gateway
 Administer Network Routing Table
 Administer Prefix Plan
 Administer Codec on SIP Trunk Group

Note: All configuration is completed using the OXE manager menu. To enter the menu,
type mgr at the CLI prompt.

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 5.1. Verify Alcatel OXE Licenses
From the CLI prompt, use the spadmin command and from the menu shown, select
option 2 Display active file. This will show the license files installed on the system.
Display current counters ........................... 1
Display active file ................................ 2
Check active file coherency ........................ 3
Install a new file ................................. 4
Read the system CPUID .............................. 5
CPU-Ids management ................................. 6
Display active and new file ........................ 7
Display OPS limits ................................. 8
Display ACK code ................................... 9
Exit ............................................... 0

5.2. Access the Alcatel OXE Manager

Establish a Telnet connection to the CS board of the OXE. At the CLI prompt type mgr
and a menu is then presented.
+-Select an object-----------------+
¦ ¦
¦ -> Shelf ¦
¦ Media Gateway ¦
¦ PWT/DECT System ¦
¦ System ¦
¦ Translator ¦
¦ Classes of Service ¦
¦ Attendant ¦
¦ Users ¦
¦ Users by profile ¦
¦ Set Profile ¦
¦ Groups ¦
¦ Speed Dialing ¦
¦ Phone Book ¦
¦ Entities ¦
¦ Trunk Groups ¦
¦ External Services ¦
¦ Inter-Node Links ¦
¦ X25 ¦
¦ DATA ¦
¦ Applications ¦
¦ Specific Telephone Services ¦
¦ ATM ¦
¦ Events Routing Discriminator ¦
¦ Security and Access Control ¦
¦ IP ¦
¦ SIP ¦
¦ DHCP Configuration ¦
¦ Alcatel-Lucent 8&9 Series ¦
¦ SIP Extension ¦
¦ Encryption ¦
¦ Passive Com. Server ¦
¦ SNMP Configuration ¦
¦ ¦
+----------------------------------+

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 5.3. Administer IP Domain
To create an IP domain select IP  IP domain. Complete the following option:
 IP Domain Name node1.mmsil.local, this is the domain name the OXE
expects in the from header for incoming SIP Invites

Click ctrl+v to complete.

+-Create: IP domain---------------------------------------------------------------------+
¦ ¦
¦ Node Number (reserved) : 1 ¦
¦ Instance (reserved) : 1 ¦
¦ IP Domain Number : 0 ¦
¦ ¦
¦ IP Domain Name : node1.mmsil.local ¦
¦ Country + Default ¦
¦ Intra-domain Coding Algorithm + Default ¦
¦ Extra-domain Coding Algorithm + Default ¦
¦ FAX/MODEM Intra domain call transp + NO ¦
¦ FAX/MODEM Extra domain call transp + NO ¦
¦ G722 allowed in Intra-domain + NO ¦
¦ G722 allowed in Extra-domain + NO ¦
¦ Tandem Primary Domain : -1 ¦
¦ Domain Max Voice Connection : -1 ¦
¦ IP Quality of service : 0 ¦
¦ Contact Number : ------------------------------ ¦
¦ Backup IP address : ----------------------------------------------- ¦
¦ Trunk Group ID : 10 ¦
¦ IP recording quality of service : 0 ¦
¦ Time Zone Name + System Default ¦
¦ Calling Identifier : ------------------------------ ¦
¦ Supplement. Calling Identifier : ------------------------------ ¦
¦ SIP Survivability Mode + NO ¦
¦ ¦
+---------------------------------------------------------------------------------------+

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 5.4. Administer SIP Trunk Group
To add a SIP Trunk Group select Trunk Groups  Create. Complete the following
options:
 Trunk Group ID A desired ID number
 Trunk Group Type T2
 Trunk Group Name A desired name

Click ctrl+v to continue.

+-Create: Trunk Groups--------------------------------------------------------------+
¦ ¦
¦ Node Number (reserved) : 1 ¦
¦ Trunk Group ID : 10 ¦
¦ ¦
¦ Trunk Group Type + T2 ¦
¦ Trunk Group Name : To sm100 ¦
¦ UTF-8 Trunk Group Name : ------------------------------------------- ¦
¦ Number Compatible With : -1 ¦
¦ Remote Network : 255 ¦
¦ Shared Trunk Group + False ¦
¦ Special Services + Nothing ¦
¦ ¦
+-----------------------------------------------------------------------------------+

On the next screen complete the following options:

 Q931 Signal Variant ABC-F
 T2 Specification SIP
Click ctrl+v to complete configuration.
+-Create: Trunk Groups-------------------------------------------------------+
¦ ¦
¦ Node number : 1 ¦
¦ Transcom Trunk Group + False ¦
¦ Auto.reserv.by Attendant + False ¦
¦ Overflow trunk group No. : -1 ¦
¦ Tone on seizure + False ¦
¦ Private Trunk Group + False ¦
¦ Q931 Signal variant + ABC-F ¦
¦ SS7 Signal variant + No variant ¦
¦ Number Of Digits To Send : 0 ¦
¦ Channel selection type + Quantified ¦
¦ Auto.DTMF dialing on outgoing call + NO ¦
¦ T2 Specification + SIP ¦
¦ Homogenous network for direct RTP + NO ¦
¦ Public Network COS : 0 ¦
¦ DID transcoding + False ¦
¦ Can support UUS in SETUP + True ¦
¦ ¦
¦ Implicit Priority ¦
¦ ¦
¦ Activation mode : 0 ¦
¦ Priority Level : 0 ¦
¦ ¦
¦ Preempter + NO ¦
¦ Incoming calls Restriction COS : 10 ¦
¦ Outgoing calls Restriction COS : 10 ¦
¦ Callee number mpt1343 + NO ¦
¦ Overlap dialing + YES ¦
¦ Call diversion in ISDN + NO ¦
¦ ¦
+----------------------------------------------------------------------------+

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 5.5. Administer SIP Gateway
To configure a SIP Gateway select SIP  SIP Gateway. Complete the following
options:
 SIP Trunk Group SIP trunk group number defined in Section 5.4
 DNS Local Domain Name Enter domain name for the OXE

Click ctrl+v to complete.

+-Review/Modify: SIP Gateway------------------------------------------------------------+
¦
¦ Node Number (reserved) : 1
¦ Instance (reserved) : 1
¦ Instance (reserved) : 1
¦
¦ SIP Subnetwork : 9
¦ SIP Trunk Group : 10
¦ IP Address : 10.10.9.111
¦ Machine name - Host : node1
¦ SIP Proxy Port Number : 5060
¦ SIP Subscribe Min Duration : 1800
¦ SIP Subscribe Max Duration : 86400
¦ Session Timer : 1800
¦ Min Session Timer : 1800
¦ Session Timer Method + RE_INVITE
¦ DNS local domain name : mmsil.local
¦ DNS type + DNS A
¦ SIP DNS1 IP Address : -----------------------------------------------
¦ SIP DNS2 IP Address : -----------------------------------------------
¦ SDP in 18x + False
¦ Cac SIP-SIP + False
¦ INFO method for remote extension + True
¦ Dynamic Payload type for DTMF : 97
+---------------------------------------------------------------------------------------+

5.6. Administer SIP Proxy

To configure a SIP Proxy select SIP  SIP Proxy. Complete the following options:
 Minimal authentication method SIP None

Click ctrl+v to complete.

+-Review/Modify: SIP Proxy--------------------------------------------------------------+
¦
¦ Node Number (reserved) : 1
¦ Instance (reserved) : 1
¦ Instance (reserved) : 1
¦ SIP initial time-out : 500
¦ SIP timer T2 : 4000
¦ Dns Timer overflow : 5000
¦ Recursive search + False
¦ Minimal authentication method + SIP None
¦
¦ Authentication realm : --------------------------------------------------
¦ Only authenticated incoming calls + False
¦ Framework Period : 3
¦ Framework Nb Message By Period : 25
¦ Framework Quarantine Period : 1800
¦ TCP when long messages + True
¦
+---------------------------------------------------------------------------------------+

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 5.7. Administer SIP External Gateway
Configure a SIP connection to the Session Manager by creating a SIP External Gateway.
Select SIP  SIP Ext Gateway  Create. Complete the following options:
 SIP External Gateway ID A desired ID number
 Gateway Name A desired name
 SIP Remote domain Enter sm100 ip address from Section 3.3
 SIP Port Number 5060
 SIP Transport Type TCP
 Trunk Group Number The trunk group number defined in Section 5.4
 SDP in 18x This must be set to False for Avaya Digital to
Alcatel ipTouch calls to work
 Minimal authentication
method SIP None

Click ctrl+v to complete.

+-Create: SIP Ext Gateway---------------------------------------------------------------+
¦
¦ Node Number (reserved) : 1
¦ Instance (reserved) : 1
¦ SIP External Gateway ID : 0
¦
¦ Gateway Name : Session Manager
¦ SIP Remote domain : 10.10.9.34
¦ PCS IP Address : -----------------------------------------------
¦ SIP Port Number : 5060
¦ SIP Transport Type + TCP
¦ RFC3262 Forced use + True
¦ Belonging Domain : --------------------------------------------------
¦ Registration ID : --------------------------------------------------
¦ Registration ID P_Asserted + False
¦ Registration timer : 0
¦ SIP Outbound Proxy : --------------------------------------------------
¦ Supervision timer : 0
¦ Trunk group number : 10
¦ Pool Number : -1
¦ Outgoing realm : --------------------------------------------------
¦ Outgoing username : --------------------------------------------------
¦
¦ Outgoing Password : ----------
¦ Confirm : ----------
¦
¦ Incoming username : --------------------------------------------------
¦ Incoming Password : ----------
¦ Confirm : ----------
¦
¦ RFC 3325 supported by the distant + True
¦ DNS type + DNS A
¦ SIP DNS1 IP Address : -----------------------------------------------
¦ SIP DNS2 IP Address : -----------------------------------------------
¦ SDP in 18x + False
¦ Minimal authentication method + SIP None
¦ INFO method for remote extension + False
¦ Send only trunk group algo + False
¦ To EMS + False
¦ Routing Application + False
¦ Dynamic Payload type for DTMF : 97
+---------------------------------------------------------------------------------------+

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 5.8. Administer Network Routing Table
In the sample configuration, network number 15 was used. To administer the routing
table for network number 15, select Translator  Network Routing Table and then
select 15. Complete the following options:
 Associated Ext SIP gateway Use the SIP External Gateway ID defined in
Section 5.7

Click ctrl+v to complete.

+-Review/Modify: Network Routing Table-------------------------------+
¦ ¦
¦ Node Number (reserved) : 1 ¦
¦ Instance (reserved) : 1 ¦
¦ Network Number : 15 ¦
¦ ¦
¦ Rank of First Digit to be Sent : 1 ¦
¦ Incoming identification prefix : -------- ¦
¦ Protocol Type + ABC_F ¦
¦ Numbering Plan Descriptor ID : 11 ¦
¦ ARS Route list : 0 ¦
¦ Schedule number : -1 ¦
¦ ATM Address ID : -1 ¦
¦ Network call prefix : -------- ¦
¦ City/Town Name : -------------------- ¦
¦ Send City/Town Name + False ¦
¦ Associated Ext SIP gateway : 0 ¦
¦ Enable UTF8 name sending + True ¦
¦ ¦
+--------------------------------------------------------------------+

5.9. Administer Prefix Plan

In the sample configuration, Avaya SIP phones are 5 digits in length and begin with 41.
To administer the prefix plan for dialing Avaya SIP phones from OXE, select Translator
 Prefix Plan  Create. Complete the following options:
 Number 41
 Prefix Meaning Routing No

Click ctrl+v to continue.

+-Create: Prefix Plan-----------------------------------------------------+
¦ ¦
¦ Node Number (reserved) : 1 ¦
¦ Instance (reserved) : 1 ¦
¦ Number : 41 ¦
¦ ¦
¦ Prefix Meaning + Routing No. ¦
¦ ¦
+-------------------------------------------------------------------------+

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 On the next screen complete the following options:
 Network Number Use network number administered in Section 5.8
 Node Number/ABC-F Trunk Group
Use the trunk group number administered in Section 5.4
 Number of Digits 5

Click ctrl+v to complete.

+-Create: Prefix Plan--------------------------------+
¦ ¦
¦ Network Number : 15 ¦
¦ Node Number/ABC-F Trunk Group : 10 ¦
¦ Number of Digits : 5 ¦
¦ Number With Subaddress (ISDN) + NO ¦
¦ Default X25 ID.pref. + NO ¦
¦ ¦
+----------------------------------------------------+

5.10. Administer Codec on SIP Trunk Group

To create a codec on the SIP Trunk Group select Trunk Groups  Trunk Group. The
parameter IP Compression Type has two possible values, G711 and Default. If the
parameter Default is chosen then this value is determined by the parameter Compression
Type administered in System  Other System Param.  Compression Parameters.
Compression type is either G.729 or G.723.
+-Review/Modify: Compression Parameters----------------------------------+
¦ ¦
¦ Node Number (reserved) : 1 ¦
¦ Instance (reserved) : 1 ¦
¦ Instance (reserved) : 1 ¦
¦ System Option + Compression Type ¦
¦ ¦
¦ Compression Type + G 729 ¦
¦ ¦
+------------------------------------------------------------------------+

For the above values to hold true, all other options for compression in the Alcatel OXE
must be set to non-compressed options. Ensure the following parameters are set
accordingly:
Navigate to IP  IP Domain
 Intra-Domain Coding Algorithm = default
 Extra-Domain Coding Algorithm = default
Navigate to IP  TSC/IP
 Default Voice Coding Algorithm = without compression
Navigate to IP  INT/IP
 Default Voice Coding Algorithm = without compression

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 6. Verification
This section provides the verification tests that can be performed on Session Manager,
Communication Manager and Alcatel OmniPCX Enterprise to verify their proper
configuration.

6.1. Verify Avaya Aura® Session Manager

On the SMGR management screen under the Elements column select Session Manager.
On the left menu, select System Status  SIP Entity Monitoring. Verify as shown
below that none of the SIP entities for Alcatel or CM links are down, indicating that they
are all reachable for routing.

Click on the SIP Entity Names Alcatel PBX and CM-ES, shown in the previous screen,
and verify that the connection status is Up, as shown in the following screenshots. Alcatel
connection status is show below:

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 Communication Manager connection status is show below:

6.2. Verify Avaya Aura® Communication Manager

Verify the status of the SIP trunk group by using the status trunk n command, where n
is the trunk group number being investigated. Verify that all trunks are in the in-
service/idle state as shown below.
status trunk 150

TRUNK GROUP STATUS

Member Port Service State Mtce Connected Ports

Busy

0150/001 T00001 in-service/idle no

0150/002 T00002 in-service/idle no
0150/003 T00003 in-service/idle no
0150/004 T00004 in-service/idle no
0150/005 T00005 in-service/idle no
0150/006 T00006 in-service/idle no
0150/007 T00007 in-service/idle no
0150/008 T00008 in-service/idle no
0150/009 T00009 in-service/idle no
0150/010 T00010 in-service/idle no

Verify the status of the SIP signaling-group by using the status signaling-group n
command, where n is the signaling group number being investigated. Verify that the
signaling group is in the in-service state as shown below.
status signaling-group 150
STATUS SIGNALING GROUP

Group ID: 150

Group Type: sip

Group State: in-service

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 6.3. Verify Alcatel OmniPCX Enterprise
Verify the status of the SIP trunk group by using the trkstat n command, where n is the
trunk group number being investigated. Verify that all trunks are in the Free state as
shown below.
trkstat 10

+==============================================================================+
| S I P T R U N K S T A T E Trunk group number : 10 |
| Trunk group name : To ASM60 |
| Number of Trunks : 62 |
+------------------------------------------------------------------------------+
| Index : 1 2 3 4 5 6 7 8 9 10 11 12 13 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 14 15 16 17 18 19 20 21 22 23 24 25 26 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 27 28 29 30 31 32 33 34 35 36 37 38 39 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 40 41 42 43 44 45 46 47 48 49 50 51 52 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 53 54 55 56 57 58 59 60 61 62 |
| State : F F F F F F F F F F |
+------------------------------------------------------------------------------+
| F: Free | B: Busy | Ct: busy Comp trunk | Cl: busy Comp link |
| WB: Busy Without B Channel| Cr: busy Comp trunk for RLIO inter-ACT link |
| WBD: Data Transparency without chan.| WBM: Modem transparency without chan. |
| D: Data Transparency | M: Modem transparency |
+------------------------------------------------------------------------------+

6.4. Verified Scenarios

The following scenarios have been verified for the configuration described in these
Application Notes.
 Basic calls between various telephones on Communication Manager and Alcatel
OXE can be made in both directions using G.711MU/A-law and G.729A.
 Proper display of the calling and called party name and number information was
verified for all telephones with the basic call scenario.
 Supplementary calling features were verified. The feature scenarios involved
additional endpoints on the respective systems, such as performing an unattended
transfer of the SIP trunk call to a local endpoint on the same site, and then
repeating the scenario to transfer the SIP trunk call to a remote endpoint on the
other site. The supplementary calling features verified are shown below.
o Unattended transfer
o Attended transfer
o Hold/Unhold
o Consultative hold
o Call forwarding
o Conference
o Calling number block
o DTMF tone sending

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 7. Conclusion
As illustrated in these Application Notes, Alcatel OmniPCX Enterprise can interoperate
via Avaya Aura® Session Manager with Avaya Aura® Communication Manager using
SIP trunks. The following is a list of interoperability items observed:
 In the case where an Avaya Digital phone dials an Alcatel phone, there is no
audio path. To resolve this issue, set “SDP in 18x” to false in the SIP Ext
Gateway configuration in the Alcatel OmniPCX Enterprise.
 In the case where an Avaya phone dials an Alcatel phone and then the Alcatel
phone performs an unattended transfer to another Avaya phone, an issue was seen
whereby the Alcatel OmniPCX Enterprise tears down the completed call after 20
seconds. To prevent this from happening, disable shuffling on the Avaya Aura®
Communication Manager.
 In the case where an Avaya phone dials an Alcatel phone and then the Alcatel
phone performs an unattended transfer to another Alcatel phone, an issue was
seen whereby the completed call was torn down after 5 seconds and the SIP trunk
on the Alcatel OmniPCX Enterprise side was blocked. To prevent this happening,
do not assign a DNS ip address in the SIP External Gateway configuration in the
Alcatel OmniPCX Enterprise.
 For conference calls and attended/unattended transfers, phone displays were not
updated correctly for both username and number. This is an Alcatel OmniPCX
Enterprise issue as SIP 180 & 200 messages sent by Alcatel OmniPCX Enterprise
do not contain a user part in the contact header.

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 8. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com
[1] Avaya Aura® Session Manager Overview, Doc # 03603323, Issue 1 Release 6.1
[2] Administering Avaya Aura® Session Manager, Doc # 03603324, Issue 1 Release 6.1
[3] Maintaining and Troubleshooting Avaya Aura® Session Manager, Doc # 03603325,
Issue 1 Release 6.1
[4] Administering Avaya AuraTM Communication Manager, Doc # 03-300509, Issue 6.0

Product documentation for Alcatel products may be found at

http://enterprise.alcatel-lucent.com/?dept=ResourceLibrary&page=Landing
[5] http://enterprise.alcatel-lucent.com/?product=OmniPCXEnterprise&page=overview

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 ©2011 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ®
and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other
trademarks are the property of their respective owners. The information provided in
these Application Notes is subject to change without notice. The configurations,
technical data, and recommendations provided in these Application Notes are believed to
be accurate and dependable, but are presented without express or implied warranty.
Users are responsible for their application of any products specified in these Application
Notes.

Please e-mail any questions or comments pertaining to these Application Notes along
with the full title name and filename, located in the lower right corner, directly to the
Avaya Solution & Interoperability Test Lab at [email protected]

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