UC Series User Manual

OpenVox Communication Co., Ltd

UC Series User Manual

Version 4.0

OpenVox Communication Co., LTD. 1

 UC Series User Manual

OpenVox Communication Co., Ltd

Address: 10/F, Building 6-A, Baoneng Science and Technology Industrial Park,
Longhua New District, Shenzhen, Guangdong, China 518109

Tel: +86-755-66630978, 82535461, 82535362

Business Contact: [email protected]

Technical Support: [email protected]

Business Hours: 09:00-18:00(GMT+8) from Monday to Friday

URL: www.openvox.cn

Thank You for Choosing OpenVox Products!

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Revision History

Issue version Issue date Detail of change

4.0 Nov.18.2020 Apply to the new version 4.2.0 UC series operating system
firmware

3.0 May. 07, 2020 Apply to the new version 4.1.0 UC series operating system
firmware

2.0 Nov. 08, 2017 Apply to the new UI 2.0 version

1.2 May. 25, 2017 Apply to all UC series IPPBX

1.1 Dec. 05, 2016 Release notes for 1.1.3,2.4

1.0 Jun. 30, 2016 Initial

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Copyright

Copyright© 2020 OpenVox Inc. All rights reserved. No part of this document may be
reproduced without prior written permission.

Confidentiality

Information contained herein is of a highly sensitive nature and is confidential and proprietary to
OpenVox Inc. No part may be distributed, reproduced or disclosed orally or in written form to
any party other than the direct recipients without the express written consent of OpenVox Inc.

Disclaimer

OpenVox Inc. reserves the right to modify the design, characteristics, and products at any time
without notification or obligation and shall not be held liable for any error or damage of any kind
resulting from the use of this document.

OpenVox has made every effort to ensure that the information contained in this document is
accurate and complete; however, the contents of this document are subject to revision without
notice. Please contact OpenVox to ensure you have the latest version of this document.

Statement

This document applies to all UC series IPPBX, including UC300/UC500/UC501. Different types
of IPPBXs may have functional differences. For details, please contact OpenVox sales or technical
support.

Trademarks

All other trademarks mentioned in this document are the property of their respective owners.

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Contents
Revision History .................................................................................................................................... 3

1 Overview ........................................................................................................................................... 11

1.1 Introduction ............................................................................................................................ 11

1.2 Series of Products .................................................................................................................. 12

1.2.1UC300/500 .................................................................................................................. 12

1.2.2UC501 ......................................................................................................................... 12

1.3 Model..................................................................................................................................... 13

1.3.1 UC300/500 Series Model ............................................................................................ 13

1.3.2 UC501 Series Modular Collocation ............................................................................ 13

1.4 Specifications ......................................................................................................................... 17

1.5 Features .................................................................................................................................. 18

1.6 Compatible Endpoints ............................................................................................................ 21

1.7 Log in to the Web GUI ........................................................................................................... 22

1.8 Web GUI overview ................................................................................................................ 22

2 System ............................................................................................................................................... 24

2.1 Dashboard .............................................................................................................................. 24

2.2 Network.................................................................................................................................. 26

2.2.1 Network Parameters .................................................................................................... 26

2.2.2 VPN Client .................................................................................................................. 27

2.2.3 Static Routes ............................................................................................................... 32

2.2.4 DDNS Client ............................................................................................................... 32

2.2.5 DHCP .......................................................................................................................... 33

2.3 Security .................................................................................................................................. 37

2.3.1 Audit ............................................................................................................................ 37

2.3.2 Weak Keys .................................................................................................................. 38

2.3.3 Certifications ............................................................................................................... 41

2.3.4 Hot Standby ................................................................................................................ 42

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2.3.5 Firewall....................................................................................................................... 43

2.3.6 Fail2Ban......................................................................................................................48

2.4 User Permission..................................................................................................................... 51

2.5 Storage................................................................................................................................... 54

2.5.1 Storage Devices.......................................................................................................... 54

2.5.2 Auto Clean Up............................................................................................................ 55

2.6 Email......................................................................................................................................56

2.7 LDAP Server..........................................................................................................................58

2.8 Maintenance...........................................................................................................................60

2.8.1 Firmware Update........................................................................................................ 60

2.8.2 Cloud Management.....................................................................................................60

2.8.3 Backup & Restore.......................................................................................................61

2.8.4 Login Settings.............................................................................................................63

2.8.5 Reboot Settings...........................................................................................................64

2.9 Event Center...........................................................................................................................65

2.9.1 Event Settings............................................................................................................. 65

2.9.2 Event Logs.................................................................................................................. 68

2.10 Tool Kit................................................................................................................................69

2.10.1 Network Capture.......................................................................................................69

2.10.2 Port Monitor..............................................................................................................69

2.10.3 IP Ping and Traceroute..............................................................................................70

2.11 Preference............................................................................................................................ 71

2.11.1 Language...................................................................................................................71

2.11.2 Date/Time................................................................................................................. 72

2.11.3 Currency....................................................................................................................73

2.11.4 About.........................................................................................................................74

2.11.5 Develop Mode...........................................................................................................74

3 PBX...................................................................................................................................................75

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3.1 Extensions..............................................................................................................................75

3.1.1 Extensions...................................................................................................................75

3.1.2 Ring Groups................................................................................................................81

3.1.3 Follow Me...................................................................................................................84

3.2 Trunks.................................................................................................................................... 88

3.3 Call Control............................................................................................................................96

3.3.1 Inbound Routes...........................................................................................................96

3.3.2 Outbound Routes........................................................................................................ 97

3.3.3 Call Restrictions........................................................................................................100

3.3.4 Set CallerID.............................................................................................................. 102

3.3.5 Call Flow Control..................................................................................................... 102

3.3.6 Time Conditions........................................................................................................104

3.3.7 Time Groups............................................................................................................. 104

3.3.8 PIN Sets.................................................................................................................... 105

3.3.9 FXO Channels DIDs.................................................................................................106

3.3.10 AutoCLIP Route..................................................................................................... 107

3.4 Call Features........................................................................................................................ 109

3.4.1 IVR............................................................................................................................109

3.4.2 Queues.......................................................................................................................110

3.4.3 Phonebook.................................................................................................................119

3.4.4 DISA......................................................................................................................... 120

3.4.5 Conference................................................................................................................ 121

3.4.6 Callback.................................................................................................................... 123

3.4.7 Parking Lot................................................................................................................126

3.4.8 Voicemail Blasting....................................................................................................128

3.4.9 Paging and Intercom................................................................................................. 129

3.4.10 Scheduled Broadcast...............................................................................................131

3.4.11 Wakeup Service......................................................................................................132

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3.5 Voice Prompts......................................................................................................................133

3.5.1 Languages................................................................................................................. 133

3.5.2 System Recordings....................................................................................................134

3.5.3 Announcement.......................................................................................................... 135

3.5.4 Route Congestion......................................................................................................136

3.5.5 Music On Hold..........................................................................................................137

3.6 Settings.................................................................................................................................138

3.6.1 Global Settings..........................................................................................................138

3.6.2 Analog Settings.........................................................................................................139

3.6.3 SIP Settings...............................................................................................................145

3.6.4 IAX2 Settings............................................................................................................148

3.6.5 RTP Settings............................................................................................................. 151

3.6.6 Recording Settings....................................................................................................153

3.6.6 Misc Destinations......................................................................................................154

3.6.7 Functions Code......................................................................................................... 155

3.6.8 AMI...........................................................................................................................156

3.7 Recording.............................................................................................................................158

3.7.1 Call Recordings.........................................................................................................158

3.7.2 VoiceMails................................................................................................................158

3.7.3 VoiceMails Admin....................................................................................................159

3.8 Tools.................................................................................................................................... 160

3.8.1 Asterisk-Cli...............................................................................................................160

3.8.2 AI TTS...................................................................................................................... 162

3.8.3 API............................................................................................................................162

3.9 Auto Provision..................................................................................................................... 164

3.10 PBX Monitor......................................................................................................................167

3.11 Conference Panel...............................................................................................................168

3.11.1 Conference List.......................................................................................................168

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3.11.2 Conference Contacts............................................................................................... 170

4 Fax...................................................................................................................................................171

4.1 Virtual Fax........................................................................................................................... 171

4.1.1 Virtual Fax List.........................................................................................................171

4.1.2 Send Fax....................................................................................................................172

4.1.3 Fax Queue................................................................................................................. 173

4.2 Fax Master........................................................................................................................... 174

4.3 Fax Clients........................................................................................................................... 175

4.4 Fax Viewer...........................................................................................................................176

5 Reports............................................................................................................................................177

5.1 CDR Report......................................................................................................................... 177

5.2 Channels Usage....................................................................................................................178

5.3 Graphic Report.....................................................................................................................179

5.4 Summary..............................................................................................................................180

5.5 Missed Calls.........................................................................................................................181

5.6 Downloads........................................................................................................................... 182

6 Extras.............................................................................................................................................. 183

6.1 Calling Cards....................................................................................................................... 183

6.2 Video Conference................................................................................................................ 184

6.3 Hotel.....................................................................................................................................185

6.3.1 Information............................................................................................................... 185

6.3.2 Service.......................................................................................................................185

6.3.3 Configuration............................................................................................................193

6.3.4 Report........................................................................................................................202

7 Logs.................................................................................................................................................205

7.1 Logs Settings........................................................................................................................205

7.2 System Logs.........................................................................................................................206

7.3 Asterisk Logs....................................................................................................................... 207

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7.4 DAHDI Logs........................................................................................................................208

7.5 FXO Monitor Logs.............................................................................................................. 209

7.6 VPN Logs.............................................................................................................................210

8 Me Bar............................................................................................................................................ 211

8.1 Extension Number............................................................................................................... 212

8.2 Call and Recordings.............................................................................................................214

8.3 Voicemail.............................................................................................................................215

8.4 Password Settings................................................................................................................ 216

8.5 Black List.............................................................................................................................217

8.6 White List.............................................................................................................................218

8.7 Downloads........................................................................................................................... 219

9 Web Phone......................................................................................................................................220

9.1 Web Dialing.........................................................................................................................221

9.2 Contacts................................................................................................................................222

9.3 Settings.................................................................................................................................224

9.4 CDR..................................................................................................................................... 225

9.5 VoiceMails...........................................................................................................................226

9.6 Password Settings................................................................................................................ 227

9.7 Functions Code.................................................................................................................... 228

9.8 Downloads........................................................................................................................... 229

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1 Overview

1.1 Introduction
The UC series IPPBX delivers a multi-functional business office telephony system designed for
small to medium enterprises. The series integrates functions such as IP phone, fax, and voice
recording, and is compatible with multiple service platforms such as Cisco CallManager, BroadSoft,
Huawei IMS and Asterisk, and terminals. The products are highly reliable, easy to install and
deploy, and offer a brand-new experience in mobile offices and communications.

In addition, UC series IPPBX supports a wide selection of codecs and signaling protocols, including
G711 (alaw/ulaw), G722, OPUS, AMR-NB/WB, SILK, G723.1 G726, G729, GSM, ADPCM, iLBC,
H263, H263P, H264, VP8. Taking full advantage of open-source platform, the UC Series appliances
support industry standard SIP trunks, IAX2 trunks, analog PSTN trunks, and analog station trunks.

The UC series delivers a full-featured IP Telephony solution. By supporting intelligent

communication functions such as mobile phone extensions, instant multi-party conferences,
call history, click-to-dial, and customer information management, it not only facilitates
seamless communication between enterprise employees and customers, but also provides a
solid basis for enterprises to analyze core business data.

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1.2 Series of Products

1.2.1 UC300/500

The UC 300/500 series is made of aluminum and the fanless exquisite enclosed design provides
important dust and moth protection. It can be operated in harsh industrial environments with ease.

The UC 300/500 series offers 2-8 analog port with up to 300 simultaneous calls in one single device,
supports up to 86,000 minutes of recording and voicemail (.gsm), and supports failover in
combination with FXS and FXO modules.

Figure 1-2-1 UC300

1.2.2 UC501

UC501 IPPBX is an upgraded version of UC500. It can be pre-installed with OpenVox IPPBX
system or other open-source communication system chosen by customers. It has built-in
Uninterruptible Power Supply (UPS) and full PBX functions to meet different usage scenarios.

The UC501 is equipped with up to 8 analog ports and 2 Ethernet interfaces for seamlessly
integrating VoIP trunks and your existing PSTN lines. In addition, the UC501 is modular in design,
equipped with 1FXO/1FXS/4FXO/4FXS modules, and with a detachable chassis, users can easily
change the port type or expand the system.

Figure 1-2-2 UC501

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1.3 Model

1.3.1 UC300/500 Series Model

UC300/500 series supports multiple models with varying amounts of FXO ports and FXS ports, as
shown in the Table 1-1-2.

Table 1-1-2 Product Models

Mode Network port FXS Ports FXO Ports USB SD UPS

UC300-A11EM1 1 x 10/100M Ethernet 1 1 1 1 NO

UC300-A14EM1 1 x 10/100M Ethernet 1 4 1 1 NO

UC300-A02EM1 1 x 10/100M Ethernet 0 2 1 1 NO

UC300-A11EM1 2 x 10/100M Ethernet 1 1 1 1 YES

UC300-A14EM2 2 x 10/100M Ethernet 1 4 1 1 YES

UC300-A02EM2 2 x 10/100M Ethernet 0 2 1 1 YES

UC500-A22EM2 2 x 10/100M Ethernet 2 2 1 1 YES

UC500-A44EM2 2 x 10/100M Ethernet 4 4 1 1 YES

UC500-A08EM2 2 x 10/100M Ethernet 0 8 1 1 YES

1.3.2 UC501 Series Modular Collocation

The UC501 series of products adopt modular design and are divided into new and old modules.

Below is the top view of the inside of the chassis, and the right side is the module
installation area.There are four areas where you can install modules.
（1）For new module：users can choose any combination of the following three modules.
①FXS-200、② FXO-200、③ FXOS-200

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Remove the screw and install the module.

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（2）For old module：The upper area is for the FXO-100/FXS-100 module and the lower area is
for the FXO-400/FXS-400 module. It should be noted that the module cannot be installed on the left
or right side at the same time, only supports ①+②, ③+④, ①+④, ②+③.Users can choose two
module accessories to customize.

• FXO-100+FXO-100
• FXO-100+FXS-100
• FXO-100+FXO-400
• FXO-100+FXS-400
• FXS-100+FXS-100
• FXS-100+FXO-400
• FXS-100+FXS-400
• FXO-400+FXO-400
• FXO-400+FXS-400

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Remove the screw and install the module（②+③）.

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1.4 Specifications
Table 1-4-1 UC Series Product Specification
Item UC300/500/501/501P
Up to 800 extension registers
System Capacity 100 concurrent calls with G.729 codec

300 concurrent calls with G.711 codec

Max Network Interface 2×10/100M port

Max FXS/FXO Interface 8
USB Port 1×USB 2.0 for external storage or disaster recovery system
External Storage 1×SD slot, support up to 128G
Telephony Interface FXS/FXO interface, Optional
RAM DDR3 1GB
Storage 16GB Onboard Flash
Power Consumption 16W Maximum

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1.5 Features

General
Up to 8 FXS/FXO (PSTN/POTS) Analog Port

Support SIP & IAX2

Support Codecs: G.711(a-law &μ-law), G.722, OPUS, AMR-NB/WB, SILK, G723.1,

G726, G729, GSM, ADPCM, iLBC, H263, H263P, H264, VP8, etc.

Abundant HD voice codecs: OPUS, AMR-NB/WB, G.722, SILK

Abundant HD video codecs: H261, H263, H263P, H264, VP8

HD Video Calls

Echo Canceller

System
Simple and Convenient Configuration via Web GUI

User Portal

Extension User Privileges

System Administrators Monitor

Event Notification

Support Backup/Restore

Remote Management

Hot Standby

System resource monitoring

Network
Network configuration

Support VLAN

Support Static Route

Support Fail2ban

Secure SIP calling (TLS encryption)

Support Multiple VPN protocols including OpenVPN, L2TP, N2N, SSTP

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PBX
Import/Export Extensions

Support SIP Forking

Call Transfer

Follow-Me/Ring Group/Queue

Quickly Auto Provision IP Phones

Support IMS VoLTE

Flexible Inbound/Outbound Route

Blacklist

Support international call restrictions

AutoCLIP

Time Condition

PIN List

Automated Attendant (IVR)

Phonebook

LDAP Service

Wakeup Service

DISA (Direct Inward System Access)

Conference

Call Back

Call Parking

Paging and Intercom

Speed Dial

Call Recording

Music On Hold

Support Open API Protocol (based on Asterisk)

Click2call

WebPhone

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Access Control Interface based on ACL

AI TTS

SIP Instant Messaging

Email
Voicemail

Missed Calls Notification

Remote SMTP Email Server

Antispam support

Support Mail Relay

Fax to Email

Report
Call Detail Records (CDR)

Billing Report

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1.6 Compatible Endpoints

Any SIP compatible IP Phone (Desktop Phones and Soft Phones for Windows, Linux, iOS
and also Android platforms).

Desktop phone examples include: OpenVox C Series, CooFone Series IP Phones provided
by ZYCOO, and also Cisco, Grandstream, Yealink, Polycom, Snom, Akuvox, Escene,
Favil, HTek etc.

Soft Phone examples include 3CX, CooCall, Linphone, X-Lite, Zoiper etc.

IAX compatible endpoints, for example, CooFone IP Phones provided by OpenVox and
also Zoiper softphone.

Analog Phones and Fax Machines

Web Extensions (WebPhone)

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1.7 Log in to the Web GUI

Step 1

Use a CAT5 cable to connect the device to the local network where the PC is connected,
or connect the device directly to the PC.

Step 2

Dial “**89” to obtain device IP address by an analog telephone, the device defaults to a fixed IP
address: 172.16.101.1

Step 3

Make sure that the PC and the device are on the same network segment.

Step 4

Enter the device IP address in the browser address bar (e.g. 192.168.2.218);

Step 5

You can enter the login interface for device configuration by selecting your role and entering a
password on the login interface. The default administrator username and password are admin.

Figure 1-7-1 Login interface

1.8 Web GUI overview

The web management interface of the UC series includes three areas: System button area, Menu bar,

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and Configuration area.

Figure 1-8-1 Web GUI layout

Table 1-8-1 Web Management Interface Layout

Item Description
(1) System button area Contains buttons such as Change Password, Reboot, Logout, Skins,
Language, etc., and the event notification bar, displays the current
login user.
(2) Menu bar Displays submenus for your selection when the mouse pointer is moved
onto a menu. The selection result is displayed in the configuration area.
(3) Configuration area View or modify configurations.

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2 System

2.1 Dashboard
The option Dashboard of menu System in UC series is a visualization tool that shows a general
view of the system and gives a faster access to administrative actions in order to allow the user
an easy administration of the server such as "System Resources", "Processes Status", "Hard
Drives". Below a short description of each one.

System Resources: Here shows general information about the system where UC series is running. It
allows to check out the history of CPU and Memory usage over the time.

Figure 2-1-1 System Resource

Processes Status: It shows the enabled and disabled processes. Here you can start, stop and
restart these processes.

Figure 2-1-2 Processes Status

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Hard Drives: Hard Drives shows the free and used space of the hard drives installed on your server.

Figure 2-1-3 Hard Drives

Communication Activity: This applet shows the number of extensions, trunks and calls currently
on sip server.

Figure 2-1-4 Communication Activity

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2.2 Network

2.2.1 Network Parameters

The option Network Parameters of the Menu Network in UC series lets us view and configure the
network parameters of the server.

Navigate to System > Network > Network Parameters to set network parameters according to the
installed network environment.

Figure 2-2-1 Network Parameters Interface

This corresponds to the general network parameters of the server.

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Table 2-2-1 Description of Edit Network Parameters

Item Description
Basic Settings
Host Server Name, for example: pbx.subdomain.com
Work Mode Optional work modes: Single/Double
Gateway IP Address of the Port of Connection (Default Gateway)
Primary DNS IP Address of the Primary Domain Name Server (DNS)
Secondary DNS IP Address of the Secondary or Alternative Domain Name Server (DNS)
IP Configuration
The type of IP address that the Interface has, which could be STATIC when the
Type IP address is fixed or DHCP when the IP address is obtained automatically
from a DHCP server.
IP Address IP Address assigned to the Interface
Mask The Network Mask assigned to the Interface
MAC Physical Address of the network Interface
Status Shows the physical status of the Interface, if it’s connected or not
Default Route Mainly used in Double work mode to determine the default exit for network
traffic
IP Address 2 The second IP assigned to the Interface
Mask 2 The network mask for the second IP

2.2.2 VPN Client

The VPN Client module of the menu Network lets us connect to the VPN Server.

Navigate to System > Network > VPN Client, chose client type and enter the Server IP Address,
switching the Enable to on and save changes. Then the Server will assign this client an IP address.

The UC series offers four common VPN connections: OpenVPN, N2N, L2TP and SSTP, allowing
users to establish virtual private networks, encrypt communications, and enable remote access.

OpenVPN

You can choose to directly upload the configuration package file (.ovpn format) for the connection.

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Figure 2-2-2 OpenVPN/Upload OpenVPN Package

It is also possible to manually configure the server information and upload files such as certificates
to connect.

Figure 2-2-3 OpenVPN/Manual Configuration

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N2N

Enter the server and user information and click the Save button to connect.

Figure 2-2-4 N2N Interface

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L2TP

Enter the corresponding information and click the Save button to connect.

Figure 2-2-5 L2TP Interface

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SSTP

Enter the corresponding information and click the Save button to connect.

Figure 2-2-6 SSTP Interface

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2.2.3 Static Routes

The Static Routes module of the menu “Network” lets users view and add the static routes.

Figure 2-2-7 Static Routes Interface

Table 2-2-2 Description of Static Routes

Item Description
Destination Identified the destination of IP packet.
Subnet Mask Identified the segment where the destination host or router locates with destination.
Gateway Also named Next Hop Router, defined the next hop server the packets send to.
Metric Used to make routing decisions, contains any number of values that help the router
determine the best route among multiple routes to a destination.
Interface The ethernet LAN/WAN interface, defined the interface used to send packet for the
specific destination.

2.2.4 DDNS Client

Select DDNS server, enter user name, password and other information, then click Save to
make DDNS take effect.

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Figure 2-2-8 DDNS Client Interface

2.2.5 DHCP

DHCP Server

DHCP (Dynamic Host Configuration Protocol) is a standardized network protocol used on

Internet Protocol (IP) networks for dynamically distributing network configuration parameters,
such as IP addresses for interfaces and services.

With DHCP, computers/IP phones request IP addresses and networking parameters

automatically from UC series WAN/LAN port which saves administrators a lot of time when
compared with having to configure these settings manually.

The option "DHCP Server" allows configuring UC series’s DHCP service so it can assign
IP addresses in the network.

Navigate to System > Network > DHCP Server:

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Figure 2-2-9 DHCP Server Interface

Here the description of each field.
Table 2-2-3 Description of Static Routes

Item Description
Enable It indicates if the DHCP service is enabled or disabled
Interface Specify the start IP of the port (Network interface configuration must be static).
Start range of IPs This will be the beginning of the IP range that the server will provide.
End ranges of IPs This will be the ending of the IP range that the server will provide.
Address Lease Duration for DHCP server to lease an address to a new device. When the lease
expires, the DHCP server might assign the IP address to a different device.
time Default value is 7200 seconds.
DNS 1 This address is the Primary DNS that the server will provide.
DNS 2 This address is the Secondary DNS that the server will provide.
WINS It is the IP of the WINS Server that will be given to Windows machines.
Gateway This is the address the server will provide as Gateway.

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TFTP Enter the TFTP server address if required which may be used to auto provision
your IP phones.
Status Display current DHCP status.

To save changes just click on the button .

DHCP Client

This module shows a list of DHCP clients and their status info.

Navigate to System > Network > DHCP Client and you will see a list of all devices receiving
their IP address from the UC series system.

Figure 2-2-10 DHCP Client Interface

To see the leased time of each address, click on "View Details".

Assign IP to Host

With this option you can assign an IP address to a specific device through MAC address.
When the device requests an IP address, the DHCP server will provide it according to the MAC
address. All the associations created by the user are shown in a list.

Navigate to System > Network > Assign IP Address to Host.

Figure 2-2-11 Assign IP Address to Host

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To create a new association, click button. Fill out the required information and click

on button.

Figure 2-2-12 Add Assign IP

Address The following table shows the description of each field:

Table 2-2-4 Description of Assign IP Address

Item Description
Host Name Name that you want to assign to the device
IP Address IP Address you want to use for the device
MAC Address MAC number of the device

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2.3 Security

2.3.1 Audit

The module Audit of the menu Security in UC series shows a list of all the users that have logged
in the system with the date, the username, the source IP address and other details. The results can be
filtered by date and string. The coincidences with the string will be highlighted in the results.

Figure 2-3-1 Audit interface

The results of the search can be downloaded in different formats such as PDF, XML and CSV
by clicking on the Generate button.

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Figure 2-3-2 Generate Audit Content

By clicking on the URL above, you can jump to the Reports > Downloads page. Click
the Download button to download the generated file.

Figure 2-3-3 Download Audit Content

2.3.2 Weak Keys

The module Weak Keys of the menu Security lets us identify the keys that are not enough strength
for the extensions created in the UC series (SIP and IAX2). This module shows all the extensions
but you can filter the results by entering a specific extension number or part of it.

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Figure 2-3-4 Weak keys interface

You can generate the results in different formats such as PDF, XML and CSV by clicking on
the Generate button.

Figure 2-3-5 Generate Weak Keys Content

By clicking on the URL above, you can jump to the Reports > Downloads page. Click
the Download button to download the corresponding file.

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Figure 2-3-6 Download Weak Keys Content

Change Key

Figure 2-3-7 Change Weak Keys

If the extension's registration/user password is not strong enough, you will be prompted in the status bar
and you can change the key by click the red prompt So-So. After clicking this link, you will jump

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to the extension setting page where you can set a new password. The password is at least 8 characters
long and must contain at least 1 digit number, at least 1 uppercase letter and at least 1 lowercase
letter. After setting the new secret, click the Save button to apply the changes.

2.3.3 Certifications

The Certifications module of the Security menu greatly enhances the security of the device.
The UC series supports TLS encrypted calling (SIP), which requires SIP phone support.

Figure 2-3-8 Certifications interface

Clicking Action to generates the Server Key, which will overwrite the original certificate if it already
exists. Click Download to download the Server Key (including the asterisk.pem and ca.crt files).

Note: After regenerating the Server Key, the original Client Keys will be invalid and will need to
be recreated in the Client Key.

Enter the Key Name and IP Address in the Client Key to Create the certificate.

Note that if the device changes its IP, the corresponding client key will need to be generated again.

Download the Client Key (including [Key Name].pem and ca.crt), please import the Client Key into
your SIP phone for encrypted transmission using TLS.

After mutual authentication between the client and the server, the phone can make encrypted calls.
The specific parameters of the Certification module can be set in the column of Transports >
TLS under PBX>Settings>SIP Settings.

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2.3.4 Hot Standby

Hot Standby is a highly reliable application of software and hardware combination. The Hot Standby
system consists of two identical UC devices and control software system. The two devices appear as
a single system in the network, and externally as an independent network IP, and control and
management in the mode of a single system. The system mirrors the data and operational status of
the two devices (including hard disk data and memory data), enables hot backup between the master
and slave devices and seamless switching. Thus, providing stable and reliable services for users and
achieving the high availability solution of dual-unit systems.

Figure 2-3-9 Hot Standby interface

Table 2-3-1 Description of Hot Standby Parameters

Options Description
HA-Mode Peer-Peer hot standby mode
Mode The default is slave mode. The device that turns on the hot standby firstly
is the master server.
Sync NIC The network adapter which is used to heartbeat and synchronous data.
Local Hostname Hostname of the local host

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Peer Hostname Hostname of the peer host

Peer IP Address IP address of the peer host
Encryption key A phrase or password to use for encryption. It has to match on both nodes.
Virtual IP Enter an unused IP address. The extensions would communicate with the
server via the virtual IP address. The two PBX in the hot standby mode
Address
should configure the same Virtual IP address.
Advert Time It sets the interval at which Heartbeat keep-alive packets are sent. The
default is 2s, the default dead time is 3 * advert time.

2.3.5 Firewall

Firewall Rules

UC series system has been preconfigured with a built-in firewall that protects your IP phone
system from unauthorized access, phone calls and other attacks. It allows building Firewall rules to
control the packets that send and receive by the UC devices. To manage the firewall, navigate to
web menu Security->Firewall.

The firewall is off by default and has seven built-in default rules: accept all internal traffic, block
all traffic from outside, and block all ports. After checking the Enable Firewall, click the Save
button and the firewall will be turned on. If you don't want to be pinged by another device, you can
check the Disable Ping.

Once the firewall is enabled, you can create, delete, modify, disable and reorder firewall rules.
Click the Save button after each operation or it will be invalid in the system. Click the Save button
every time a new or edited rule is completed, and then the list will automatically display your
changes, otherwise they are invalid in the system.

Figure 2-3-10 Firewall Rules

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Adding a New Rule

Click add to fill out the form to create a new firewall rule. The form will vary depending on the
parameter selected for Service. You can simply select the Action and Service type, or customize
the Service and set the port range.

In the Source IP Address/Subnet Mask field, you must enter an IP address in the format
x.x.x.x/y, where y is the subnet mask and should be a number between 0 and 32. If you enter the
default IP address (0.0.0.0), the subnet mask will be 0.

Once the rule is created, click the Save button and the new rule will appear in the list. Be sure to save
the changes, otherwise, they will not take effect in the system.

Figure 2-3-11 Add a new rules interface

Table 2-3-2 Description of Firewall Rule Parameters

Options Description
Name Give this rule a descriptive name to help you identify it.
Description A brief description of this rule. For example: accept a specific host to access the
web interface for configuration.
Action Accept: The device will accept access to the specified address.
Deny: The PBX will deny the connection from the specified address and will

send an error message to the other side informing them that the device has
denied the connection.

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Ignore: The device will ignore the connection from the specified address, drop
the data directly, and do not give any feedback.
To improve the security of your IPPBX system, you can use Ignore actions to

avoid malicious attacks to detect the server information of your device.

Service Optional or customizable system services are available. By selecting a service,
the default port for that service is selected. Of course, you can also customize
the firewall service by selecting "Custom" and filling in the "Protocol" and
"Port" options.
MAC Address The MAC address format is: XX:XX:XX:XX:XX:XX:XX:XX:XX:XX.
Type Select the type that matches this rule, either an IP address or a domain name.
Source IP The IP address format is: IP address/subnet mask, subnet mask needs to be
Address/ written in full format, the short format is not supported.
Subnet Mask
For example, 192.168.5.100/255.255.255.255 means that the rule applies to
192.168.5.100;
192.168.5.0/255.255.255.255.0 means that the rule applies to IP between

192.168.5.0 and 192.168.5.255.

Domain Name Appears when "Domain Name" is selected for Type. The firewall rules will
match the domain name filled in here.
Protocol Appears when the service is selected "Custom", selects the protocol that applies
to this rule, selects UDP, TCP and BOTH (UDP and TCP)
Port Appears when the service selects "Custom" to specify the ports for this rule,
which can specify port groups and individual ports.
When specifying a port group, the left side is the start port and the right side is

the end port (included), e.g. "5060:5070" means to specify ports 5060 to 5070
(including 5070).
When specifying a single port, just fill in the same port number on the left as on

the right. For example, "5060:5060" means that port 5060 is specified.

Editing a Rule

To edit an existing rule, click on the icon corresponding to the rule. Here you can modify
parameters of the rule.

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Figure 2-3-12 Edit Firewall rules interface

Deleting a Rule

To delete a rule, just select the corresponding checkbox and click on the button. Be
sure to save the changes or they will not work in the system.
Reordering the Rules
You can modify the order of the rules by clicking on the blue arrows in the column Move. If you

click on the button of a rule, this rule will go up one position and if you click on the button,

it will go down one position. If you click on the arrow, the rule will rise to the highest position

which is the highest priority. Similarly, the arrows move the rule to the lowest position. Make
sure you save the changes, so they will take effect in the system after modifying the position of
the rules.

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Export rules

Firewall rules now support exporting CSV files, just click the button and the browser
will automatically download the exported CSV file. Note that please allow browser pop-ups.

Import rules

The firewall now supports importing CSV files to create rules in bulk, click the button
and a popup will appear as follows

Figure 2-3-13 Import Firewall Rules interface

Click Browse to select the edited CSV file, then click Import to successfully import.
The importation instructions are as follows.

Table 2-3-3 Import Parameters - Firewall Rules

Parameter Importance Restriction Default Value
• The following characters are NOT allowed:
Name Required &"'\<>`| N/A

• The maximum length is 127.

• The following characters are NOT allowed:

Optional system &"'\<>`|
Description services and N/A
custom services • The maximum length is 511.

Optional system Permitted value:

Service services and HTTP HTTPS SSH AMI SIP-UDP SIP- Custom
custom services
TCP SIP-TCP SIP-TLS SIP-

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RTP WEBRTC IAX2 LDAP MYSQL Custom

Permitted value:
Action Required Accept
Accept Reject Drop
Permitted value:
Protocol Required udp
udp tcp both
MAC Optional system
services and MAC address format required. N/A
Address
custom services
Permitted value:
Type Required IP
IP Domain
Source IP Required if Type
Address/Su IP format required. N/A
is IP
bnet Mask
Domain Required if Type Domain format required. N/A
is domain
Port Required The valid port range is 0-65535. N/A

After clicking the link and opening Import Parameters - Firewall Rules page, click and
the browser will automatically download the template of the CSV file.

2.3.6 Fail2Ban

Fail2ban scans log files (e.g. /var/log/apache/error_log) and ban IPs that show the malicious signs --
too many password failures, seeking for exploits, etc. Generally, Fail2Ban is then used to update
firewall rules to reject the IP addresses for a specified amount of time, although any arbitrary other
action (e.g. sending an email) could also be configured. Out of the box, Fail2Ban comes with filters
for various services (apache, courier, ssh, etc).

Fail2Ban is able to reduce the rate of incorrect authentications attempts however it cannot eliminate
the risk that weak authentication presents. Configure services to use only two factors or
public/private authentication mechanisms if you really want to protect services.

The module "Fail2Ban" allows configuring Fail2ban service so it can prevent the UC series from
malicious attacks. Navigate to System > Security > Fail2Ban to configure rules.

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Figure 2-3-14 Fail2Ban interface

Max Retry limits the authentication attempts. Find Time defines the time duration from the first
attempt to the last attempt which reaches the “Max Retry” limitation. Ban Time is the time in
seconds the IPPBX system will block the IP which exceeds max retry. Ban Time don’t take effect
on any whitelisted addresses.

Figure 2-3-15 Fail2Ban add whitelist

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Add whitelist allows you to add a trusted IP addresses or network addresses to the system IP
whitelist. The IPs in the whitelist will always be treated as trusted IP’s and will not be filtered by
the firewall rules.

Figure 2-3-16 Fail2Ban whitelist

If mistakenly disabled, you can log in to that device with another IP and enter the blacklist
to unblock it.

Figure 2-3-17 Fail2Ban blacklist

Jail is generally used for permanent bans, or "top bans", which are disabled by default. When
running Fail2Ban Jail, if an IP has already been banned, and the IP continue to try to access and
reach Max Retry within the set Find Time, then it will be blocked for longer time, this time is set
by Ban Time, if Ban Time is set to -1, then it means permanent blocking.

Figure 2-3-18 Fail2Ban Jail

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2.4 User Permission

System > Users Permission allow the creation and modification of permissions for users accessing the
web interface. An extension that has been granted access can log into the system using the SIP
extension number/login password. It should be noted that by default, the user permissions give the Me
module permissions for all extensions to log in and use some simple features.

Click the button to grant permission to the specified extension, then Click to
save the configuration.

Figure 2-4-1 Create New User

In the User drop-down box, you can select the corresponding extension, and in the Group drop-
down box, you can select Custom/Administrator. If you set the user group as Custom, you can
check the desired function module to give the user web privileges; if you select the user group as
Administrator, all function privileges are enabled by default. Note that if some permissions are
unchecked at this point, they will automatically become the Custom group after saving, in other
words, the Administrators group will have all permissions at all times.

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In addition to Features and Me modules, the other permissions correspond to the function menu
on the left side of the page.

Me Bar provides basic permissions after extension user login and does not
recommend modifications. See 8 Me Bar for details.

Figure 2-4-2 User Permission/Me

The Feature provides enablement of some features associated with the extensions that are also
used in the Me Bar.

Figure 2-4-3 User Permission/Features

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Table 2-4-1 User Permission/Features

Type Option Description

CDR Permission Download CDR Allow the extension to download the CDR associated with
this extension in the Me module
Delete CDR Allows the extension to delete the CDR associated with
this extension in the Me module
Play Recordings Allows extensions to play recordings in the Me module
Recording Download Allows extensions to download recordings in the Me
Permission Recordings module
Delete Recordings Allow extensions to delete recordings in the Me module
Allowed check Allows extensions to view downloads from other extensions in the "Downloads"
and Download section of the Me Bar

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2.5 Storage

2.5.1 Storage Devices

In this module, users can format or mount external storage devices such as TF/SD cards plugged into
UC devices, or add network storage. It should be noted that the system only allows one external
device to be set as the primary storage device, which means that when one external storage device is
mounted, other devices cannot be mounted at the same time. The large files such as audio files
generated by the system will be automatically stored in the mounted external device.

Click System>Storage>Storage Device.

Figure 2-5-1 Storage Devices Interface

Click to format the inserted device. For TF/SD/U disk devices, only EXT4 or FAT file
systems can be mounted. For non-EXT4/FAT file systems, please format them.

Click to mount the device that has been inserted. At that time, large files such as
recordings generated by the system will be automatically stored on the device. The

button will change to gray, and the Unmount button will appear.

Click to unmount the mounted device. Add Network Dive at that time will return to
normal and click is valid.

Click to add network storage, as shown in the following figure.

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Figure 2-5-2 Add Network Drive

Currently network storage only supports CIFS services. Enter the Network Drive information, click
Save, and you can mount it successfully.

2.5.2 Auto Clean Up

The option Auto Clean Up of the menu Storage allows you to configure the clean-up frequency.

Figure 2-5-3 Auto Clean Up Interface

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2.6 Email
The Email is mainly used in conjunction with Event Center, and by setting the remote SMTP
configuration parameters of the mailbox, you can enable the Email service, send event reminder
email and fax email, and provide you with timely and accurate information. It can also be combined
with Voicemail to Email, allowing you to check your voice messages anytime, anywhere.

Note that there is no built-in SMTP server in the UC system, but an external SMTP server is used.

The fields for configuring Email are shown below.

Figure 2-6-1 Email/Basic

Table 2-6-1 Definition of Email

Item Definition
Enable Decide whether to turn on SMTP service
SMTP Server SMTP server type. Multiple server types are built in, associated with
Domain, or can be customized by selecting "other"
Domain SMTP server address. It is automatically filled according to the SMTP
server. When the SMTP Server selects "other", it needs to be filled manually.
Port Port to establish the connection with SMTP Server. Common ports are 25,
465 (SSL), 587 (SSL)
Username Username of email account from SMTP Server.

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Password Password of email account from SMTP Server

To enable certificates of TLS (Transport Layer Security). Generally, this
TLS Enable check is required when using ports that require SSL encryption, such as
465/587. If checked when using a port that does not require encryption, it
will cause the send to fail

After setting Email, if you want to send a test email to check whether the Email function is
enabled correctly, please click Save and then click Test, and a dialog box will pop up for sending.

The Voicemail Template and Faxmail Template options edit the Voicemail and Faxmail Template.
After filling in the template variables in the Subject or Content according to the example shown above,
they will be replaced with the corresponding parameter values when the actual email is sent.

Figure 2-6-2 Email/Voicemail Template

Figure 2-6-3 Email/Faxmail Template

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2.7 LDAP Server

LDAP (Lightweight Directory Access Protocol) is a protocol for accessing directory services. It
is generally used as a phone book on IPPBX. Based on the available LDAP services, it meets the
requirements for fast search of phone directories. You can set up UC IPPBX as a server.

If you want to use LDAP service, just check the Enable LDAP service saving checkbox, and use
the default configuration for the rest of the content. Once LDAP is set up, you can search the LDAP
directory and find contacts on your IP phone.

Figure 2-7-1 LDAP Settings

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The UC has a built-in default phonebook node that contains all extensions on the system, which
cannot be deleted or edited.

Of course, you can also manually add a phone book node, click the button, enter the

phone book name and save. Click to add your contact information.

Figure 2-7-2 Phone Book Nodes

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2.8 Maintenance

2.8.1 Firmware Update

The option Firmware Update of the menu Maintenance allows you to update the firmware version
by uploading firmware file you download from the official website as well as update firmware
online. Note that online upgrades are not recommended if the network is in poor condition.

Figure 2-8-1 Firmware Update

2.8.2 Cloud Management

The UC series has full support for the OpenVox cloud management platform.

Figure 2-8-2 Firmware Update

After the device is connected to the Cloud Management Platform, users can access the gateway's
WEB page or SSH access to the background through the Cloud Management. In addition, it can
monitor whether the device is connected to the Cloud Management, provide functions such as

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password reset, online upgrade, reboot, etc., The Cloud Management Platform can also count your
device model, number, distribution area, monitor your account activity and so on, providing you
with efficient and excellent service and experience.

Table 2-8-1 OpenVox Cloud Management Platform

Options Definition
Enable Yes/No. Indicates that the cloud management function is enable/disable
Account An account or email registered on the cloud management platform.
Password Password for the account registered on the cloud management platform.
Server Three servers are currently supported, including American, China and Europe.
Connect Status Whether or not you are currently connected to a cloud management platform.

2.8.3 Backup & Restore

The Backup & Restore option in the System menu allows you to back up and restore
the configuration of the UC system.

If you have already made a backup before that, you can click Browse to select your backup file,
upload it and select Restore to restore the backup. When you restore a backup, you will be asked if
you want to keep the IP address of your current system. If you choose no, the IP address of your
system will be changed to the IP address of your backup after restoration. You can also click Reset
to restore the factory defaults.

Please note that both the restore backup and reset operations are not reversible.

Figure 2-8-3 Backup & Restore/Basic

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Figure 2-8-4 Backup & Restore/Auto Backup

To enable Auto Backup, navigate to System > Maintenance > Backup & Restore > Auto
Backup, change the disable option to the frequency you want. There are three media you could
select to back up your config file: USB/SD Card, FTP and CIFS.

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2.8.4 Login Settings

Navigate to System > Maintenance > Login Settings to setup the login mode and port.

Figure 2-8-5 Login Settings Interface

The SSH settings page requires Developer Mode to be enabled, see 2.11.5.

After you turn on Developer Mode, you can log in and set up SSH. SSH default port is 13505,
select Enabled-On option, set Name and P. Click Save.

Figure 2-8-6 SSH Settings interface

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2.8.5 Reboot Settings

UC system supports setting timed automatic restart. Navigate to System > Maintenance > Reboot
Settings.

Figure 2-8-7 Reboot Settings Interface

If you want to reboot your system directly, you can click on admin>Reboot in the upper
right corner:

Figure 2-8-8 Reboot

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2.9 Event Center

UC system provides event monitoring and alert function, users can set events that need to be monitored
and notification content, after adding notification contacts, the device will send reminders by sending
emails or calling extensions, so that users can fully grasp the system dynamics.

2.9.1 Event Settings

When the Record column checkbox is checked, the system will record the corresponding event in

the Event Logs, or you can click the at the top to view it; when the checkbox of the
Notification column is checked, you can set the notification by email or phone, but you need to add
the contact information in advance.

Click to edit the Notification Template and personalize the notification.

Figure 2-9-1 Event Settings

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You can set up Notification Contacts to be notified by sending an email or calling when an
event occurs. Click to Add contacts

Figure 2-9-2 Notification Contacts

Once you are done, click Edit to edit the current contact and Delete to delete the contact. Of course,
it is also possible to select multiple contacts for bulk deletion.

Figure 2-9-3 Edit Contact

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Also, you can add a Group for Notification.

Figure 2-9-4 Notification Group

Figure 2-9-5 Edit Group

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2.9.2 Event Logs

You can view logs related to monitored events in both the notification bar in the upper right
corner and the Event Center > Event Logs page.

Figure 2-9-6 Event Logs

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2.10 Tool Kit

2.10.1 Network Capture

The UC series provides network packet capture function for ease of user to analysis, capture
and monitor the network status, RTP streams, protocol and so on.

Figure 2-10-1 Capture interface

2.10.2 Port Monitor

It also provides Port Monitor module for user to monitor and record the port communications.

Figure 2-10-2 Port Monitor interface

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2.10.3 IP Ping and Traceroute

The IP Ping and Traceroute module assist user to check the network connectivity.

Figure 2-10-3 IP Ping and Traceroute interface

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2.11 Preference

2.11.1 Language

Under the Language module in the Preferences menu, you can change the language of the
UC system web interface. Select your desired language from the language list and click Save.

You can also download or upload languages you need.

Figure 2-11-1 Language setting

At the same time, the UC system supports uploading language packs. You can click to
download the current language pack, modify the language pack file based on it, then

and use the new language pack. Note that the language package is cached by default
to ensure system smoothness. When debugging a new language package, you can click

Language Cache, then select Language Debug Yes and save.

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2.11.2 Date/Time

The option Date/Time of the Menu Preferences in UC series lets us configure the Date, Hour and
Timezone for the UC series Web Interface. Select the new date, hour and timezone and click on
the Apply changes button.

Figure 2-11-2 Date/Time Interface

Alternatively, system time can be synchronized automatically with the NTP server/local client.

Figure 2-11-3 Sync time with NTP Server

Figure 2-11-4 Sync time with Client

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2.11.3 Currency

Currency module of menu Preferences allows us change the currency for Reports in UC series.

Figure 2-11-5 Currency Setting interface

Select a currency from the available options and click on the button.

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2.11.4 About

Navigate to System >About, some basic information about the UC System is displayed, you can
see the hardware version, model name, etc.

Figure 2-11-6 About information

2.11.5 Develop Mode

Under About module, five consecutive clicks on the Hardware Version will bring up a
dialog prompt, check it and save it to enter developer mode.

Figure 2-11-7 Developer Mode

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3 PBX
The Menu PBX lets us configure extensions, trunks, routes, dialplan, queues, IVR and so on for
UC series.

In this menu, we can observe that we have different options for configuration.

3.1 Extensions

3.1.1 Extensions

The Extensions Module is used to set up each extension on your system. In the Extensions module,
you will set up the extension number, the name of the extension, the password, voicemail settings
for the extension, and other options.

Normally, each physical phone will be assigned to one extension. If you have a phone that has more
than one "line" button, you would normally make each line button register to the same extension
number, and then use the line buttons to manage multiple calls to and from the same line. However, you
could also create two or more extensions and assign each extension to a different line button.

Figure 3-1-1 Add an Extension interface

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Click one of extensions number and edit it:

Figure 3-1-2 Extension parameter interface

Table 3-1-1 Definition of Extension parameter

Item Description
Basic
User Extension The extension number to dial to reach this user.
Display Name The CallerID name for calls from this user will be set to this name. only enter the
name, NOT the number.
Registration Password configured for the extension to register.
Password
Email Address The email can be used to email notification to the extension user.
Mobile The extension contacts phone number.
Number
User Password Password configured for the extension to login web.
Advanced
Account Code Account code for the device
Max Contacts Maximum number of endpoints that can associate with this device
Web Phone Enable web phone will let user make and receive calls without installing any
plugin in web browser.
Dtmfmode The DTMF signaling mode used by this device, usually rfc2833 for most phone.
Audio Codecs Codecs supported by the device, you can choose the codecs which you want.

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Video Codecs Video codecs supported by the device

Ring Timeout Select the time in seconds.
This sets the allowed transport settings for this device and the default (Primary)
transport for outgoing. The default transport is only used for outbound message
until a registration takes place. During the peer registration the transport type may
change to another supported type if the peer requests so. In most common cases,
Transport this does not have to be changed as most devices register in conjunction with the
host=dynamic setting. If you are using TCP and/or TLS you need to make sure the
general SIP Settings are configured for the system to operate in those modes and
for TLS, proper certificates have been generated and configured. If you are using
websockets (such as WebRTC) then you must select an option that includes WS.
When registering, SIP phones will be sending packets containing the user agent
User Agent string. If the prefix of the user agent does not match the value defined here, the
registration will fail.
Permitted
IP/Subnet Permitted IP/Subnet Mask
Mask
Dictation Allow the device to support dictation service.
Service
Dictation The format of dictation.
Format
Language Code Choose a different language for the user if he/she is not a native speaker than
default system voice prompts.
CID Num Alias The CID Number to use for internal calls, if different from the extension number.
This is used to masquerade as a different user.
If you want to support direct sip dialing of users internally or through anonymous
SIP Alias sip calls, you can supply a friendly name that can be used in addition to the users
extension to call them.
Features
Override the callerid when dialing out a trunk. Any setting here will override the
common outbound callerid set in the trunk admin.
Outbound CID Format: “caller name” <#######>

Leave this filed blank to disable the outbound callerid feature for this user.

Asterisk Dial Cryptic Asterisk Dial Options, check to customize for this extension or un-check to
use system defaults set in Advanced Options. These will not apply to trunk options
Options
which are configured with the trunk.

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Ring Time Number of seconds to ring prior to going to voicemail. Default will use the value
set in Advanced Settings. If no voicemail is configured this will be ignored.
Allow Being Check this option to allow this user to be monitored.
Monitored
Monitor Mode Decide how you will monitor another extension.
Number of seconds to ring during a Call Forward Busy or Call Forward
Call Forward Unavailable call prior to continuing to voicemail or specified destination. Setting
to Always will not return, it will just continue to ring. Default will use the current
Ring Time
Ring Time. If voicemail is disabled and there is not destination specified, it will be
forced into Always mode.
Outbound Maximum number of outbound simultaneous calls that an extension can make.
This is also very useful as a Security Protection against a system that has been
Concurrency
compromised. It will limit the number of simultaneous calls that can be made on
Limit
the compromised extension.
Call Waiting Set the initial/current Call Waiting state for this user’s extension
When set to Intercom, calls to this extension/user from other internal users act as if
they were intercom calls meaning they will be auto-answered if the endpoint
Internal Auto supports this feature and the system is configured to operate in this mode. All the
normal white list and black list settings will be honored if they are set. External
Answer
calls will still ring as normal, as will certain other circumstances such as blind
transfers and when a Follow Me is configured and enabled. If Disabled, the phone
rings as a normal phone.
Call Screening requires external callers to say their name, which will be played
back to the user and allow the user to accept or reject the call. Screening with
memory only verifies a caller for their callerid once. Screening without memory
Call Screening always required a caller to say their name. Either mode will always announce the
caller based on the last introduction saved with that callerID. If any user on the
system uses the memory option, when that user is called, the caller will be required
to re-introduce themselves and all users on the system will have that new
introduction associated with the caller’s CallerID.
Pinless Dialing Enabling Pinless Dialing will allow this extension to bypass any pin codes
normally required on outbound calls.
Emergency This callerid will always be set when dialing out an Outbound Route flagged ad
CID Emergency. The Emergency CID overrides all other CallerID settings.
If this extension is part of a Queue will attempt to use the user’s extension state or
Queue State device state information when determining if this queue member should be called.
In some uncommon situations such as a Follow-Me with no physical device, or
Detection
some virtual extension scenarios, the state information will indicate that this
member is not available when they are. Setting this to ‘Ignore-State’ will make the

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Queue ignore all state information thus always trying to contact this member.
Certain side effects can occur when this route is taken due to the nature of how
Queues handle Local channels, such as subsequent transfers will continue to
show the member as busy until the original call is terminated. In most cases, this
SHOULD BE set to ‘Use State’.
Recording
On Demand Enable or disable the ability to do on demand (one-touch) recording. The overall
calling policy rules still apply and if calls are already being recorded they cannot
Recording
be paused.
Call recording policy priority relative to other extensions when there is a conflict
Record Priority between an extension wanting recording and the other not wanting it. The higher of
Policy the two determines the policy, on a tie the global policy (caller or callee)
determines the policy.
Voicemail
Status Enable or disable the voicemail function.
This is the password used to access the Voicemail system.
Voicemail This password can only contain numbers.

Password
A user can change the password you enter here after logging into the Voicemail
system (*98) with a phone.
Pager Email Page/mobile email address that short Voicemail notifications are sent to.
Address
Email Option to attach Voicemail to email.
Attachment
Play CID Read back caller’s telephone number prior to playing the incoming message, and
just after announcing the date and time the message was left.
Envelope controls whether or not the Voicemail system will play the message
Play Envelope envelope (date/time) before playing the voicemail message. This setting does not
affect the operation of the envelope option in the advanced voicemail menu.
If set to “yes” the message will be delete from the voicemailbox (after having been
Delete emailed). Provides functionality that allows a user to receive their voicemail via
Voicemail email alone, rather than extension handset. CAUTION: must have attach voicemail
to email set to yes otherwise your messages will be lost forever.
Send Voicemail If set to 'yes', the voicemail will be sent by email.
Separate options with pipe( | )
VM Options
Ie: review=yes|maxmessage=60

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This is the voicemail context which is normally set to default. Do not

VM Context
change unless you understand the implications.
Routing
VmX Enable/ disable the VmX locater feature for this user. When enabled all settings are
controlled by the user in the user portal (ARI). Disabling will not delete any
Locater™
existing user settings but will disable access to the feature.
Menu options below are available during your personal voicemail greeting
Use When playback.
Check both to use at all times.

Voicemail Uncheck to play a deep after your personal voicemail greeting.

Instructions
Pressing 0 during your personal voicemail greeting goes to the operator. Uncheck
Press 0 to enter another destination here. This feature can be used while still disabling
VmX to allow an alternative operator extension without requiring the VmX feature
for the user.
The remaining options can have internal extensions, ringgroups, queues and
Press 1 external numbers that may be rung. It is often used to include your cell phone. You
should run a test to make sure that the number is functional any time a change is
made so you don’t leave a caller stranded or receiving invalid number messages.
Use any extensions, ringgroups, queues or external numbers.
Press 2 Remember to re-record your personal voicemail greeting and include instructions.

Run a test to make sure that the number is functional.

Optional destination call is routed to when the call is not answered on an otherwise
No Answer idle phone. If the phone is use and the call is simply ignored, then the busy
destination will be used.
CID Prefix Optional CID prefix to add before sending to this no answer destination.
Optional destination the call is route to when the phone is busy or the call is
Busy rejected the user. This destination is also used on an unanswered call if the phone
is in use and the user choose not pickup the second call.
CID Prefix Optional CID prefix to add before sending to this busy destination.
Not Reachable Optional destination the call is routed when the phone is office, such as a
softphone currently off or a phone unplugged.
CID Prefix Optional CID prefix to add before sending to this not reachable destination.

The extension module allows you create extensions from a CSV file and download a CSV file with
all the extensions that are currently configured in UC series. This makes it easy the migration of data.

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To download a CSV file with all the extensions created in UC series, click on the
button and save the file into your local hard drive.

To upload a CSV with the extensions you want to create, click on button, select the
CSV file and click on "Upload CSV File" button.
Make sure the following indications are taken into account:

• Duplicated extensions are not allowed.

• The first line of the CSV file must contain the headers of the columns.
• The file must have at minimum four columns.
• This type of file can be created and opened with any text editor or spreadsheets such as Open
Office Calc, Excel, etc.
• The separator of the columns is the comma.

3.1.2 Ring Groups

A ring group is a group of extensions that will ring when there is an external incoming call. You can
even put your Mobile Phone number in the ring group if you want to. For the mobile phone to work,
you must have the appropriate route and trunk set up.

You may not want a ring group – it’s entirely up to you. If you don’t require a ring group, you
may ignore this section.

When there is an incoming call to the ring group, the phones nominated in the selected group will
ring. You may select different ring group for each of the incoming trunk or you may nominate
the same group for all the trunks, in which case you will only need to define only one ring group.

The ring group screen is illustrated below:

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Figure 3-1-3 Ring groups interface

Table 3-1-2 Definition of add Ring groups interface

Item Definition
Basic
Ring-Group The number users will dial to ring extensions in this ring group
Number
Group Provide a descriptive title for this Ring Group.
Description
Ring Strategy Ringall : Ring all available channels until one answers (default)
Hunt: Take turns ringing each available extension

Memoryhunt: Ring first extension in the list, then ring the 1 st and 2 nd extension,

then ring 1 st and 2 nd and 3 rd extension in the list…etc.

*-prim: there mode act as described above. However, if the primary extension (first

in list) is occupied, the other extensions will not be rung. If the primary is CF

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unconditional, then all will be rung

First available: ring only the first available channel

Firstnotonphone: ring only the first channel which is not offhook-ignored CW.

Ring Time (max Time in seconds that the phones will ring. For all hunt style ring strategies, this is
300 sec) the time for each iteration of phone(s) that are rung.
Extension List List extensions to ring, one per line, or use the Extension Quick Pick below to insert
them here.
You can include an extension on a remote system, or an external number by

suffixing a number with a ‘#’. Ex:2448089# would dial 2448089 on the appropriate
trunk (see outbound routing)
Extension without a ‘#’ will not ring a user’s Follow-Me. To dial Follow-Me,

Queues and other numbers that are not extensions, put a ‘#’ at the end.
Destination if no If there is no answer, the call will be sent to the destination.
answer
Advanced
Announcement Message to be played to the caller before dialing this group.
To add additional recordings please use the “System Recordings” MENU to the left.

Play Music On If you select a music on hold class to play, instead of ‘Ring’, they will hear that
Hold instead of Ringing while they waiting for someone to pick up.
CID Name You can optionally prefix the callerid name when ringing extensions in this group,
Prefix ie: If you prefix with “Sales:”, a call from John Doe would display as “Sales: John
Doe” on the extensions that ring.
Alert Info ALERT_INFO can be used for distinctive ring with SIP devices.
Ignore CF When checked, agents who attempt to Call Forward will be ignored, this applies to
Settings CF, CFU and CFB. Extensions entered with ‘#’ at the end, for example to access
the extension’s Follow-Me, might not honor this setting.
Enable Call Checking this will allow calls to the ring group to be picked up with the directed
Pickup call pickup feature using the group number. When not checked, individual
extensions that are part of the group can still be picked up by doing a directed call
picked to the ringing extension, which works whether or not this is checked.
Skip Busy Agent When checked, agents who are on an occupied phone will skipped as if the line
were returning busy. This means that call waiting or multi-line phones will not be
presented with the call and in the various hunt style ring strategies, the next agent
will be attempted.
Confirm Calls Enable this if you’re calling external numbers that need confirmation-eg, a mobile
phone may go to voicemail which will pick up the call. Enabling this requires the
remote side push 1 on their phone before the call is put through. This feature only
works with the ringall ring strategy.

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Remote Message to be played to the person RECEIVING the call, if ‘Confirm Calls’ is
Announce enabled.
To add additional recordings use the “System Recordings” MENU to the left

Too-Late Message to be played to the person RECEIVING the call, if the call has already
Announce been accepted before they push 1.
To add additional recordings use the “System Recordings” MENU to the left

Mode Default: Transmits the Callers CID if allowed by the trunk.

Fixed CID Value: Always transmit the Fixed CID Value below.

Outside Calls Fixed CID Value: Transmit the Fixed CID Value below on calls will

continue to operate in default mode.

Use Dialed Number: Transmit the number that was dialed as the CID for calls

coming from outside. Internal extension to extension calls will continue to operate
in default mode. There must be a DID on the inbound route for this. This will be
BLOCKED on trunks that block foreign Caller ID
Force Dialed Number: Transmit the number that was dialed as the CID for calls

coming from outside. Internal extension to extension calls will be continue to

operate in default mode. There must be a DID on the inbound route for this. This
WILL be transmitted on trunks that block foreign CallerID
Fixed CID Value Fixed value to replace the CID with used with some of the modes above. Should be
in a format of digits only with an option of E164 format using a leading “+”.
Record Calls You can always record calls that come into ring group, never record them, or allow
the extension that answers to do on-demand recording. If recording is denied then
one-touch on demand recording will be blocked.

3.1.3 Follow Me

Follow Me (also known as Find Me / Follow Me or FMFM) allows you to redirect a call that is
placed to one of your extensions to another location. You can program the system to ring the
extension alone for a certain period of time, then ring some other destination(s), such as a mobile
phone or a related extension, and then go to the original extension's voicemail if the call is not
answered. Follow Me can also be used to divert calls to another extension without ringing the
primary extension.

Select the PBX -> PBX Configuration -> Follow Me.

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Figure 3-1-4 Follow Me

interface Select the extensions that you want to define.

Figure 3-1-5 Follow Me User interface

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Table 3-1-3 Definition of Follow Me

Item Definition
Basic
Extension Edited extension
Disable By default (not checked) any call to this extension will go to this Follow-Me
instead, including directory calls by name from IVRs. If checked, calls will go
only to the extension.
However, destinations that specify FollowMe will come here.
Checking this box is often used in conjunction with VmX Locater, where you
want a call to ring the extension, and then only if the caller chooses to find you
do you want it to come here.
Initial Ring Time This is the number of seconds to ring the primary extension prior to proceeding
to the follow-me list. The extension can also be included in the follow-me list.
A 0 setting will bypass this
Ring Strategy Ringallv2: ring Extension for duration set in Initial Ring Time, and then, while
continuing call to extension, ring Follow-Me List for duration set in Ring
Time.
Ringall: ring Extension for duration set in Initial Ring Time, and then,

terminate call to extension, ring Follow-Me List for duration set in Ring Time.
Hunt: take turns ringing each available extension

Memoryhunt: ring first extension in the list, then ring the 1st and 2nd

extension, then ring 1st 2nd and 3rd extension in the list…. etc.
*-prim: these mode act as described above. However, if the primary extension

(first in the list) is occupied, the other extensions will not be rung. If the
primary is DND, it won’t be rung. If the primary is CF unconditional, then all
will be rung
Firstavailable: ring only the first available channel

Firstavailable: ring only the first channel which is not off hook-ignore CW

Ring Time (max Time in second that the phones will ring. For all hunt style ring strategies, this
60 sec) is the time for each iteration of phone(s) that are rung
Destination if no Choose a destination when there is no answer.
answer
Follow-Me List List extensions to ring, one per line, or use the Extension Quick Pick below.
You can include an extension on a remote system, or an external number by
suffixing a number with a pound (#). Ex:2448089# would dial 2448089 on the
appropriate trunk （see Outbound Routing）.

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Announcement Message to be played to the caller before dialing this group.

To add additional recordings please use the “System Recordings” MENU to
the left.
Play Music On If you select a Music on Hold class to play, instead of ‘Ring’, they will hear
Hold that instead of Ringing while they are waiting for someone to pick up.
CID Name You can optionally prefix the Caller ID name when ringing extensions in this
Prefix group. Ie: if you prefix with “Sales:”, a call from John Doe would display as
“Sales: John Doe” on the extensions that ring
Alert Info You can optionally include an Alert Info which can create distinctive ring on
SIP phones.
Advanced
Confirm Calls Enable this if you’re calling external numbers that need confirmation, eg, a
mobile phone may go to voicemail which pick up the call. Enabling this
require the remote side push 1 on their phone before the calls is put through.
This feature only works with the ringall/ringall-prim ring strategy.
Remote Message to be played to the person RECEIVING the call, if ‘Confirm Calls” is
Announce enabled.
To add additional recordings use the ‘System Recordings” MENU to the left

Too-Late Message to be played to the person RECEIVING the call, if the call has
Announce already been accepted before they push 1.
To add additional recordings use the ‘System Recordings” MENU to the left

Mode Default: Transmits the Caller CID if allowed by the trunk.

Fixed CID Value: Always transmit the Fixed CID Value below.

Outside Calls Fixed CID Value: Transmit the Fixed CID Value below on calls

will continue to operate in default mode.

Use Dialed Number: Transmit the number that was dialed as the CID for calls

coming from outside. Internal extension to extension calls will continue to

operate in default mode. There must be a DID on the inbound route for this.
This will be BLOCKED on trunks that block foreign Caller ID
Force Dialed Number: Transmit the number that was dialed as the CID for

calls coming from outside. Internal extension to extension calls will be

continue to operate in default mode. There must be a DID on the inbound route
for this. This WILL be transmitted on trunks that block foreign CallerID
Fixed CID Value Fixed value to replace the CID with used with some of the modes above.
Should be in a format of digits only with an option of E164 format using a
leading “+”.

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3.2 Trunks
The "Trunks Module" is used to connect your FreePBX/Asterisk system to another VOIP system
or VOIP device so that you can send calls out to and receive calls in from that system/device. You
can create connections with Internet Telephone Service Providers ("ITSPs"), with other FreePBX/
Asterisk systems, with commercial VOIP phone systems, with FXO Gateways (a device that
connects an ordinary telephone line with a VOIP phone system using a network connection), and
with FXO cards (cards that are installed in your computer and allow you to connect a standard
telephone line).

If you don't have a Trunk set-up, you can still make calls, but only to other extensions on your
same phone system.

Figure 3-2-1 Add trunk interface

Figure 3-2-2 Add SIP Trunk

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Table 3-2-1 Definition of add a SIP trunk

Item Definition
Basic
Enable Trunk Check this to disable this trunk in all routes where it is used.
Trunk Mode Authentication mode of this trunk.
Usually, this will be set to "Outbound", which authenticates calls going out, and
Authentication allows unauthenticated calls in from the other server. If you select "None", all
calls from or to the specified SIP Server are unauthenticated.
Trunk Name Descriptive Name for this trunk.
Host Host settings for this device, almost always dynamic for endpoint.
Transport Transports which the device supports.
From user Rewrite the caller id
From Domain Example: proxy.provider.domain
Enable NAT Check this to enable or disable NAT
Codec Allow specified codecs, the available codecs are on the left options bar and the
selected on the right.
Advanced
DTMF Mode Types of DTMF.
Outbound CallerID for calls placed out on this trunk
Format: <#######>. You can also use the format: “hidden” <#######> to hide
CallerID
the CallerID sent out over Digital lines if supported (SIP/IAX).
Controls the maximum number of outbound channels (simultaneous calls) that
Maximum can be used on this trunk. To count inbound calls against this maximum, use the
Channels auto-generated context: as the inbound trunk's context. (see
extensions_additional.conf) Leave blank to specify no maximum.
Permanent Auth Determines whether failed authentication challenges are treated as permanent.
Rejection
Forbidden Retry How long to wait before retry when receiving a 403 Forbidden response.
Interval
Fatal Retry How long to wait before retry when receiving a fatal response.
Interval
General Retry The interval between two registered request packets.
Interval
Expiration Expiration time for registrations in seconds.
Max Retries The times asterisk will attempt to register before give up.
Qualify Interval between two qualifies.
Frequency
Qualify Timeout Timeout of qualify
Contact User Contact user to use in request.
AOR Contact Permanent contacts assigned to AoR.
Support Path When the button is enabled, registering request of outbound will advertise

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support for path header.

Support T.38 Allow the device to support T.38 UDPTL
UDPTL
T.38 UDPTL T.38 UDPTL error correction method
Error Correction
T.38 UDPTL Whether NAT support is enabled on UDPTL sessions
NAT
Fax Detect When a CNG is detected, the session will be sent to the fax extension.
Inband Progress Determine whether chan_sip indicates ringing using inbound progress.
Direct Media Method for building direct media between endpoints.
Method
Trust Connected Accept Connected Line updates from this endpoint.
Line
Send Connected Send Connected Line updates to this endpoint
Line
Connected Line Method used when updating connected line information.
Method
Direct Media Determines whether media may flow directly between endpoints.
RTP Symmetric Enforce that RTP must be symmetric.
Rewrite Contact Allow contact header to be rewritten
Asterisk Trunk Asterisk Dial command options to be used when calling out this trunk. To
override the Advanced Settings default, check the box and then provide the
Dial Options
required options for this trunk
Context (Experts Only) Set the context that calls will originate from. Leaving this as
from-internal unless you know what you’re doing.
Normally the next trunk is only tried upon a trunk being ‘Congested’ in some
Continue if Busy form, or unavailable. Checking this box will force a failed call to always
continue to the next configured trunk or destination even when the channel
reports BUSY or INVALID NUMBER.

Figure 3-2-3 Add FXO Trunk

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Table 3-2-2 Definition Add FXO Trunk

Item Definition
Basic
Enable Trunk Check this to disable this trunk in all routes where it is used.
Trunk Name Descriptive Name for this trunk.
Group ID FXO channels are referenced either by a group number or channel number
(which is defined in chan_dahdi.conf). The default setting is g0 (group zero).
Policy Used to make FXO trunks decisions, help determine the ringing order among
multiple members of group
Member of Adding FXO ports into trunk groups allow automatic selection of the selected
Groups idle port for outgoing calls.
Advanced
Outbound CallerID for calls placed out on this trunk
CallerID Format: <#######>. You can also use the format: “hidden” <#######> to hide
the CallerID sent out over Digital lines if supported (SIP/IAX).
CID Options Determines what CIDs will be allowed out this trunk. IMPORTANT:
EMERGENCY
CIDs defined on an extension/device will ALWAYS be used if this trunk is part

of an EMERGENCY Route regardless of these settings.

Allow Any CID: all CIDs including foreign CIDS from forwarded external

calls will be transmitted.

Block Foreign CIDs: blocks any CID that is the result of a forwarded call from

off the system. CIDs defined for extensions/users are transmitted.

Remove CNAM: this will remove CNAM from any CID sent out this trunk

Force Trunk CID: Always use the CID defined for this trunk except if part of

any EMERGENCY Route with an EMERGENCY CID defined for the

extension/device. Intra-Company Routes will always transmit an extension’s
internal number and name.
Maximum Controls the maximum number of outbound channels (simultaneous calls) that
Channels can be used on this trunk. Inbound calls are not counted against the maximum.
Leave blank to specify no maximum.
Asterisk Trunk Asterisk Dial command options to be used when calling out this trunk. To
Dial Options override the Advanced Settings default, check the box and then provide the
required options for this trunk
Context (Experts Only) Set the context that calls will originate from. Leaving this as
from-internal unless you know what you’re doing.
Continue if Normally the next trunk is only tried upon a trunk being ‘Congested’ in some

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Busy form, or unavailable. Checking this box will force a failed call to always
continue to the next configured trunk or destination even when the channel
reports BUSY or INVALID NUMBER.

Figure 3-2-4 Add IAX2 Trunk

Table 3-2-3 Definition of Add IAX2 Trunk

Item Definition
Basic
Enable Trunk Check this to disable this trunk in all routes where it is used.
Trunk Mode Authentication mode of this trunk.
Trunk Name Descriptive Name for this trunk
Host Host settings for this device, almost always dynamic for endpoint.
Type Asterisk connection type. There are three type you can choose, friend, peer
and user.usually friend for endpoint.
Trunk Use IAX2 trunk with this host.
Advanced
Outbound CallerID CallerID for calls placed out on this trunk
Format: <#######>. You can also use the format: “hidden” <#######> to

hide the CallerID sent out over Digital lines if supported (SIP/IAX).
CID Options Determines what CIDs will be allowed out this trunk. IMPORTANT:
EMERGENCY
CIDs defined on an extension/device will ALWAYS be used if this trunk is

part of an EMERGENCY Route regardless of these settings.

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Allow Any CID: all CIDs including foreign CIDS from forwarded external
calls will be transmitted.
Block Foreign CIDs: blocks any CID that is the result of a forwarded call

from off the system. CIDs defined for extensions/users are transmitted.
Remove CNAM: this will remove CNAM from any CID sent out this trunk

Force Trunk CID: Always use the CID defined for this trunk except if part of

any EMERGENCY Route with an EMERGENCY CID defined for the

extension/device. Intra-Company Routes will always transmit an extension’s
internal number and name.
Maximum Channels Controls the maximum number of outbound channels (simultaneous calls)
that can be used on this trunk. To count inbound calls against this maximum,
use auto-generated context: from-trunk-[trunkname] as the inbound trunk’s
context. (see extesions_additional.conf) Leave blank to specify no maximum.
Outbound Dial The outbound dialing prefix is used to prefix a dialing string to all outbound
Prefix calls placed on this trunk. For example, if this trunk is behind another PBX or
is a Centrex line, then you would put 9 here to access an outbound line.
Another common use is to prefix calls with 'w' on a POTS line that need time
to obtain dial tone to avoid eating digits.
Most users should leave this option blank.

Qualifyfreq OK Frequency in milliseconds to send qualify messages to the endpoint.

Qualifyfreq Not OK Frequency in milliseconds to send qualify messages to the endpoint.
Qualify Setting to yes (equivalent to 2000 msec) will send an OPTIONS packet to the
endpoint periodically (default every minute). Used to monitor the health of
the endpoint. If delays are longer then the quality time, the endpoint will be
taken offline and considered unreachable. Can be set to a value which is the
msec threshold. Setting to no will turn this off. Can also be helpful to keep
NAT pinholes open.
Asterisk Trunk Dial Asterisk Dial command options to be used when calling out this trunk. To
Options override the Advanced Settings default, check the box and then provide the
required options for this trunk.
Context (Experts Only) Set the context that calls will originate from. Leaving this as
from-internal unless you know what you’re doing.
Continue if Busy Normally the next trunk is only tried upon a trunk being ‘Congested’ in some
form, or unavailable. Checking this box will force a failed call to always
continue to the next configured trunk or destination even when the channel
reports BUSY or INVALID NUMBER.
Codec
Audio Codecs You can choose specific audio codecs here

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Video Codecs You can choose specific video codecs here

Figure 3-2-5 Add CUSTOM Trunk interface

Table 3-2-4 Definition of Add CUSTOM Trunk

Item Definition
Basic
Enable Trunk Check this to disable this trunk in all routes where it is used.
Trunk Name Descriptive Name for this trunk
Custom Dial String Define the custom Dial String. Include the token $OUTNUM$ wherever
the number to dial should go.
examples:

CAPI/XXXXXXXX/$OUTNUM$
H323/[email protected]
OH323/[email protected]:XXXX
vpb/1-1/$OUTNUM$
Advanced
Outbound CallerID CallerID for calls placed out on this trunk
Format: <#######>. You can also use the format: “hidden” <#######> to

hide the CallerID sent out over Digital lines if supported (SIP/IAX).
CID Options Determines what CIDs will be allowed out this trunk. IMPORTANT:
EMERGENCY
CIDs defined on an extension/device will ALWAYS be used if this trunk is

part of an EMERGENCY Route regardless of these settings.

Allow Any CID: all CIDs including foreign CIDS from forwarded external

calls will be transmitted.

Block Foreign CIDs: blocks any CID that is the result of a forwarded call

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from off the system. CIDs defined for extensions/users are transmitted.
Remove CNAM: this will remove CNAM from any CID sent out this trunk

Force Trunk CID: Always use the CID defined for this trunk except if part

of any EMERGENCY Route with an EMERGENCY CID defined for the

extension/device. Intra-Company Routes will always transmit an extension’s
internal number and name.
Maximum Channels Controls the maximum number of outbound channels (simultaneous calls)
that can be used on this trunk. Inbound calls are not counted against the
maximum. Leave blank to specify no maximum.
Asterisk Trunk Dial Asterisk Dial command options to be used when calling out this trunk. To
Options override the Advanced Settings default, check the box and then provide the
required options for this trunk
Context (Experts Only) Set the context that calls will originate from. Leaving this as
from-internal unless you know what you’re doing.
Continue if Busy Normally the next trunk is only tried upon a trunk being ‘Congested’ in some
form, or unavailable. Checking this box will force a failed call to always
continue to the next configured trunk or destination even when the channel
reports BUSY or INVALID NUMBER.

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3.3 Call Control

3.3.1 Inbound Routes

When a call comes into your system from the outside, it will usually arrive along with
information about the telephone number that was dialed (also known as the "DID") and the Caller
ID of the person who called.

The Inbound Routes module is used to tell your system what to do with calls that come into
your system on any trunk that has the "context=from-trunk" parameter in the PEER details.

Figure 3-3-1 Add incoming Route interface

Table 3-3-1 Definition of Add incoming Route

Item Definition
Basic
Description Provide a meaningful description of what this incoming route is
DID Number Define the expected DID Number if your trunk passes DID on incoming calls.
Leaving this blank to match calls with any or no DID info.

You can also use a pattern match (eg_2[345]X) to match a range of numbers.

CallerID Define the CallerID Number to be matched on incoming calls.

Number
Leave this field blank to match any or no CID info. In addition to standard dial
sequences, you can also put Private, Blocked, Unknown, Restricted, Anonymous
and Unavailable in order to catch these special cases if the Telco transmits them.
CID Priority This effects CID ONLY routes where no DID is specified. If checked, calls with
Route this CID will routed to this route, even if there is a route to the DID that was
called. Normal behavior is for the DID route to take the calls. If there is a specific

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DID/CID route for this CID, that route will still take the call when that DID is
called.
Inbound Indicates extension, Ring Group, Voicemail or other destination to which the call
Destination is suppoesd to be directed when the outside callers have called specified DID
Number
Advanced
Alert Info ALERT_INFO can be used for distinctive ring with SIP devices.
CID name You can optionally prefix the CallerID name. ie: If you prefix with “Sales:”, a call
prefix from john Doe would display as “Sales: John Doe” on the extension that ring
Music On Set the MoH class that will be used for calls that come in on this route. For
Hold example, choose a type appropriate for routes coming in from a country which
may have announcements in their language.
Signal Some devices or providers require RINGING to be sent before ANSWER. You’ll
RINGING notice this happening if you can send calls directly to a phone, but if you send it to
an IVR, it won’t connect the call.
Pause Before An optional delay to wait before processing this route. Setting this value will delay
Answer the channel from answering the call. This may be handy if external fax equipment
or security systems are installed in parallel and you would like them to be able to
seize the line.
Privacy If no CallerID has been received, Privacy Manager will ask the caller to enter their
Manager phone number. If an user/extension has Call Screening enabled, the incoming
caller will be prompted to say their name when the call reaches the user/extension.
Source Source can be added in Caller Name Lookup Sources section.
Language Allows you to set the language for this DID.
Fax Detect Attempt to detect faxes on this DID.

No: No attempts are made to auto-determine the call type; all calls sent to
destination below. Use this option if this DID is used exclusively for voice OR
fax.

Yes: try to auto determine the type of call; route to the fax destination if call is
a fax, otherwise send to regular destination. Use this option if you receive
both voice and fax calls on this line.

3.3.2 Outbound Routes

The Outbound Routes Module is used to tell your FreePBX/Asterisk system which numbers your
phones are permitted to call and which Trunk to send the calls to.

Generally, a FreePBX/Asterisk system will have a Restricted route which designates certain numbers
that can never be dialed (such as 900 and 976 numbers), an Emergency route to use for routing110 calls,
and a route for ordinary calls. A phone system might also have special routes for interoffice

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calls, international calls, and other special circumstances

Figure 3-3-2 Outbound Routes interface

Table 3-3-2 Definition of Outbound Routes

Item Definition
Basic
Route Name Name of this route. Should be used to describe what type of calls this route
matches (for example, 'local' or 'longdistance').
Route CID Optional Route CID to be used for this route. If set, this will override all CIDS
specified except:
extension/device EMERGENCY CIDs if this route is checked as an

EMERGENCY Route
trunk CID if trunk is set to force it's CID
Forwarded call CIDs (CF, Follow Me, Ring Groups, etc)
Extension/User CIDs if checked

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Route Optional: A route can prompt users for a password before allowing calls to
Password progress. This is useful for restricting calls to international destinations or 1-
900 numbers.
A numerical password, or the path to an Authenticate password file can be used.

Leave this field blank to not prompt for password.

Dial Patterns A Dial Pattern is a unique set of digits that will select this route and send the call
that will use to the designated trunks. If a dialed pattern matches this route, no subsequent
this Route routes will be tried. If Time Groups are enabled, subsequent routes will be
checked for matches outside of the designated time(s).
Rules:

X matches any digit from 0-9

Z matches any digit from 1-9

N matches any digit from 2-9
[1237-9] matches any digit in the brackets (example: 1,2,3,7,8,9). wildcard,

matches one or more dialed digits

Prepend: Digits to prepend to a successful match. If the dialed number

matches the patterns specified by the subsequent columns, then this will be
prepended before sending to the trunks.
Prefix: Prefix to remove on a successful match. The dialed number is

compared to this and the subsequent columns for a match. Upon a match, this
prefix is removed from the dialed number before sending it to the trunks.
Match pattern: The dialed number will be compared against the prefix + this

match pattern. Upon a match, the match pattern portion of the dialed number
will be sent to the trunks.
CallerID: If CallerID is supplied, the dialed number will only match the prefix

+ match pattern if the CallerID being transmitted matches this. When extensions
make outbound calls, the CallerID will be their extension number and NOT their
Outbound CID. The above special matching sequences can be used for CallerID
matching similar to other number matches.
Dial patterns These options provide a quick way to add outbound dialing rules. Follow the
wizards prompts for each.
Lookup local prefixes This looks up your local number on

ww.localcallingguide.com (NA-only), and sets up so you can dial either 7, 10 or

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11 digits (5551234, 6135551234, 16135551234) to access this route.

Upload from CSV Upload patterns from a CSV file replacing existing entries.

If there are no headers then the file must have 4 columns of patterns in the same
order as in the GUI. You can also supply headers: prepend, prefix, match
pattern and callerid in the first row. If there are less than 4 recognized headers
then the remaining columns will be blank.
Add Trunks Trunks used by this outbound route, the available trunks are on the left options
bar and the selected on the right.
Advanced
Route Type Optional: Selecting Emergency will enforce the use of a device
Music On Hold You can choose which music category to use. For example, choose a type
appropriate for a destination country which may have announcements in the
appropriate language.
Time Group If this route should only be available during certain times then Select a Time
Group created under Time Groups. The route will be ignored outside of times
specified in that Time Group. If left as default of Permanent Route then it will
always be available.
Route Position Where to insert this route or relocate it relative to the other routes.
PIN Set Optional: Select a PIN set to use. If using this option, leave the Route Password
field blank.
Optional If all the trunks fail because of Asterisk ‘CONGESTION’ dial status you can
Destination on optionally go to a destination such as a unique recorded message or anywhere
Congestion else. This destination will NOT be engaged if the trunk is reporting busy, invalid
numbers or anything else that would imply the trunk was able to make an
‘intelligent’ choice about the number that was dialed. The ‘Normal Congestion’
behavior is to play the ‘ALL Circuits Busy’ recording or other options
configured in the route Congestion Messages module when installed.

3.3.3 Call Restrictions

Black List

The blacklist module is used to add a phone number to a blacklist or remove a phone number from
a blacklist. You can also choose to blacklist any blocked or unknown calls.

When a number is blacklisted, any calls with that number in the Caller ID field received by
the system will be routed to the disconnected record.

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Figure 3-3-3 Blacklist interface

Table 3-3-3 Definition of Blacklist

Item Definition
Name Name of this blacklist rule.
Type Which type the rule applies to, including Inbound/Outbound/Both
Number Enter the number you want to block, you can input sets of digits
that match the Dial Pattern Rules.

White List

Figure 3-3-4 Whitelist interface

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Table 3-3-4 Definition of Whitelist

Item Definition
Name Name of this whitelist rule.
Type Which type the rule applies to, including Inbound/Outbound/Both
Number Enter the number you want to add into whitelist, you can input sets
of digits that match the Dial Pattern Rules.

3.3.4 Set CallerID

You can change name and number of incoming call display.

Figure 3-3-5 Set CallerID interface

Table 3-3-5 Definition of CallerID

Item Definition
Description Provide a title for it
CallerID Name The callserID name will be changed to it.
CallerID Number The callserID number will be changed to it.
Destination Destination the call will be sent to after CID has been processed.

3.3.5 Call Flow Control

The Call Flow Control module is used to create a single destination that can act as a switch that can
be toggled by anyone who has access to a local phone. It is commonly used to allow phone system
users to manually switch between "Daytime Mode" and "Nighttime Mode."

Call Flow Control should not be confused with Time Conditions. While both of these modules
relate to call flow, Call Flow Control is designed to be a manual switch, while a Time Condition is

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designed to be a scheduled, automatic switch.

Figure 3-3-6 Call flow control interface

Table 3-3-6 Definition of Call flow control

Item Definition
Feature Code Index There are a total of 10 Feature code objects,0-9, each can control a call flow
and be toggled using the call flow toggled feature code plus the index
Name Description for this Call Flow Toggle Control
Current Mode This will change the current state for this Call Flow Toggle Control, or set
the initial state when creating a new one.
Recording for Message to be played in normal mode (Green/BLF off)
Normal Mode To add additional recordings use the “System Recordings” MENU to the
left
Recording for Message to be played in override mode (Green/BLF off)
Override Mode To add additional recordings use the “System Recordings” MENU to the
left
Optional Password You can optionally include a password to authenticate before toggling the
call flow. If left blank anyone can use the feature code and it will be un-
protected
Normal Flow Destination to use when set to Normal Flow (Green/BLF off) mode
(Green/BLF off)

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Override Flow Destination to use when set to Override Flow (Red/BLF off) mode
(Red/BLF on)

3.3.6 Time Conditions

You can create various time conditions and use these time conditions in conjunction with
your Inbound Route to individualize each of the incoming trunk’s behavior.

Figure 3-3-7 Time Conditions interface

Table 3-3-7 Definition of add Time Conditions

Item Definition
Time Condition name Give this Time Condition a brief name to help you identify it.
Time Group Select a time group created under Time Groups. Matching times
will be sent to matching destination. If no group is selected, call
will always go to no-match destination.
Destination if time matches The destination the call will be sent to when the time matches.
Destination if time does not The destination the call will be sent to when the time doesn’t
match match.

3.3.7 Time Groups

The Time Groups Module is used to define periods of time that can then be selected in the
Time Conditions module or Outbound Routes module.

For example, you might create a Time Group called "Lunch" that might start at 12:00 p.m and end
at 1:00 p.m. You could then create a Time Condition that would use the Lunch Time Group to send
calls to voicemail during lunch, and to a ring group at other times.

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Figure 3-3-8 Time Groups interface

Table 3-3-8 Definition of Time Conditions

Item Definition
Name It will display as the name of time group
Time Choose a time interval.
Week Day Start To Finish Start and end times of one week.
Month Day Start To Finish Start and end times of one month.
Month Start To Finish Start and end times of one year.

3.3.8 PIN Sets

UC Series allows you to require callers to dial a password before an outbound call will go through.
You can require a password on all calls, or only on calls to certain numbers.

The PIN Sets Module allows you to create define groups and then assign a list of passwords to each
group. You can then restrict certain calls to certain groups by going to the Outbound Routes Module
and limiting the route to a certain PIN Set group. Each Outbound Route can be limited to just one
PIN Set group. So, if you want to allow more than one PIN Set group to make a certain type of call,
just create a duplicate Outbound Route and assign the second Outbound Route to a different PIN Set
Group.

Figure 3-3-9 PIN Sets Interface

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Table 3-3-9 Definition of add PIN Set

Item Definition
Name Name of the pin sets.
Record In Select this box if you would like to record the PIN in the
CDR call detail records when used.
PIN List Enter a list of one more PINs. One PIN per line.

3.3.9 FXO Channels DIDs

The FXO Channel DIDs module allows you to assign a DID or phone number to specific
analog channels.

Unlike SIP or PRI trunks, analog lines do not send a DID or dialed number to the PBX. Since the
PBX routes all inbound calls based on the DID or number dialed, we need to map each analog port
or channel to a fake number so we can match that number to an Inbound Route number and route
your calls.

Each channel can be mapped to the same phone number if you want all calls on the analog lines to
go to the same destination. This would be a common scenario if you have multiple POTS lines that
are on a hunt group from your provider.

Figure 3-3-10 Add FXO Channel interface

Table 3-3-10 Definition of Add FXO Channel

Item Definition
Channel The FXO Channel number to map to a DID
Description A useful description this channel
DID The DID that this channel represents. The incoming call on this
channel will be treated as if it came in with this DID and can be
managed with Inbound Routing on DIDs

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3.3.10 AutoCLIP Route

Generally, in the enterprise's telephony system, incoming calls are routed to IVR, ring groups,
queues, and so on instead of specific extensions. AutoCLIP can redirect calls to the extension of
the original caller instead of the automated attendant or the default ring group.
You may encounter situations that when you use an internal extension to call a client or colleague and
they don't answer the call in time. By the time he/she dials back, the IP telephony system directs him/her
to the default inbound routing destination such as IVR, making it difficult for callers who are not in touch
to find you. AutoCLIP deals with this by ignoring the routing destination and redirecting this call to the
original extension (your IP Phone line) according to stored records of outgoing calls in the AutoCLIP
route table. This feature will retain many opportunities and possibilities for customers.

Figure 3-3-11 Add FXO Channel interface

Table 3-3-11 Definition of AutoCLIP Route

Item Definition

Delete Used If enabled, when an AutoCLIP record is matched, it will be automatically

Records deleted afterwards.
Record Keep Time This sets how long each record will be kept in the AutoCLIP List.

Only Keep Missed If enabled, the system will only keep records of calls that are not answered by
Call Records the called party in the AutoCLIP list.
Note: PSTN line will keep records of all calls whether this option is enabled

or disabled.
Digits Match Define how many digits from the last digit of the incoming phone number will
be used to match the AutoCLIP record. If the number has fewer digits than the

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value defined here, all digits will be matched.

Match Outgoing If enabled, only the incoming call that came to the PBX through the same
Trunk trunk which made the call will be match against the AutoCLIP List.
Member Trunks This defines AutoCLIP Route will apply to which trunk and which trunk's
record will be kept in the AutoCLIP list. If no trunk's selected, AutoCLIP will
stop working.

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3.4 Call Features

3.4.1 IVR

The IVR module allows you to create one or more IVRs ("Interactive Voice Response" systems or
Auto Attendants). You can then route calls to the IVR and play a recording prompting callers what
options to enter, such as “press 1 for sales and press 2 for the company directory.” An IVR can also
route calls to another IVR, or in other words, a sub-menu. As a general rule, you never want more
than five or six options in a single IVR, or it will become too confusing to navigate. It is better to
only include a few options at a single menu level, and route callers to a sub-menu for more choices.

Figure 3-4-1 IVR interface

Table 3-4-1 Definition of add IVR

Item Definition
Basic
IVR Name Name of this IVR
Prompt The prompt will be played when a call reaches the IVR.
Prompt Repeat The number of times that the prompt will be played.
Count
Response The number of seconds to wait for a digit input after prompt.
Timeout (s)
Dial Extensions Allow the caller to dial extension directly.
Dial to Check If enabled, the caller will be allowed to dial '*97' to check voicemail.
Voicemail
Advanced
Invalid Retries Number of time to retry when receiving an invalid/unmatched response from
the caller

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Invalid Retry Prompt to be played when an invalid/unmatched response is received, before

Recording prompt the caller to try again
Append After playing Invalid Retry Recording the system will replay mail IVR
Announcement Announcement
on Invalid
Return on Check this box to have this option return to a parent IVR if it was called
Invalid from a parent IVR. If not, it will go to the chosen destination.
The return path will be to any IVR that was in the call path prior to this IVR

which could lead to strange result if there was an IVR called in the call path
but not immediately before this.
Invalid Prompt to be played before sending the caller to an alternate destination due
Recording to the caller pressing 0 or receiving the maximum amount of
invalid/unmatched responses (as determined by Invalid Retries)
Timeout Retry Prompt to be played when a timeout occurs, before prompting the caller to
Recording try again
Append After playing the Timeout Retry Recording the system will replay the main
Announcement IVR Announcement.
on Timeout
Return on Check this box to have this option return to a parent IVR if it was called
Timeout from a parent IVR. If not, it will go to the chosen destination.
The return path will be to any IVR that was in the call path prior to this IVR

which could lead to strange result if there was an IVR called in the call path
but not immediately before this
Timeout Prompt to be played before sending the caller to an alternate destination due
Recording to the caller pressing 0 or receiving the maximum amount of
invalid/unmatched responses (as determined by Invalid Retries)
Return to IVR If checked, upon exiting voicemail a caller will be returned to this IVR if
after VM they got a user voicemail

3.4.2 Queues

The Queues module is a more advanced version of the Ring Groups module. Like the Ring Groups
module, the Queues module is used to create an extension number that your users can dial in order to
ring multiple extensions at the same time. It also creates a destination to which you can send calls
that will ring those multiple extensions.

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Figure 3-4-2 Queues interface

Table 3-4-2 Definition of Queues

Item Definition
Basic
Queue Number Use this number to dial into the queue, or transfer callers to this number to put
them into the queue.
Agents will dial this queue number plus* to log the queue, and this queue

number plus** to log out the queue.

For example, if the queue number is 123:

123*=log in

123**=log out

Queue Name Give the queue a brief name to help you identify it.
Queue Password You can require agents to enter a password before they can log in to this

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queue.
This setting is optional.

The password is only used when logging in with the legacy queue no* code.

When using the toggle codes, you must use the Restrict Dynamic Agents
option in conjunction with the Dynamic Members list to control access.
Generate Device If checked, individual hints and dialplan will be generated for each SIP and
Hints IAX2 device that could be part of this queue. These are used in conjunction
with programmable BLF status as to the current state, the format of this hints
is
*45ddd*qqq

Where *45 is the currently define toggle feature code, ddd is the device

number (typically the same as the extension number) and qqq is this queue’s
number
Call Confirm If checked, any queue member that is actually an outside telephone number, or
any extension Follow-Me or call forwarding that are pursued and leave the
PBX will be forced into Call Confirmation mode where the member must
acknowledge the call before it is answered and delivered.
Call Confirm Announcement played to the Queue Member announcing the Queue call and
Announce requesting confirmation prior to answering. If set to default, the standard call
confirmation default message will be played unless the number is reached
through a Follow-Me and this is an alternate message provided in the Follow-
Me. This message will override any other message specified.
To add additional recordings please use the “System Recordings” MENU.

CID Name Prefix You can optionally prefix the CallerID name of callers to the queue. ie: If you
prefix with “Sales:”, a call from John Doe would display as “Sales: John Doe”
on the extensions that ring.
Wait Time Prefix When set to Yes, the CID Name will be prefix with the total wait time in the
queue so the answering agent is aware how long they have waited. It will be
rounded to the nearest minute, in the form of Mnn: where nn is the number of
minutes.
If the call is subsequently transferred, the wait time will reflect the time since

it first entered the queue or reset if the call is transferred to another queue with
this feature set.
Alert Info ALERT_INFO can be used for distinctive ring with SIP device.
Static Agents Static agents are extensions that are assumed to always be on the queue.
Static agents do not need to 'log in' to the queue, and cannot 'log out' of the

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queue.
List extensions to ring, one per line.

You can include an extension on a remote system, or an external number

(Outbound
Routing must contain a valid route for external numbers). You can put a ","

after the agent followed by a penalty value, see Asterisk documentation

concerning penalties.
An advanced mode has been added which allows you to prefix an agent

number with S, X, Z, D or A. This will force the agent number to be dialed as

an Asterisk device of type SIP, IAX2, ZAP, DAHDi or Agent respectively.
This mode is for advanced users and can cause known issues in PBX as you
are by-passing the normal dialplan. If your 'Agent Restrictions' are not set to
'Extension Only' you will have problems with subsequent transfers to
voicemail and other issues may also exist.
(Channel Agent is deprecated starting with Asterisk 1.4 and gone in 1.6+.)

Dynamic Dynamic Members are extensions or callback numbers that can log in and out
Members of the queue. When a member logs in to a queue, their penalty in the queue
will be as specified here. Extensions included here will NOT automatically be
logged in to the queue.
Restrict Dynamic Restrict dynamic queue member logins to only those listed in the Dynamic
Agents Members list above. When set to Yes, members not listed will be DENIED
ACCESS to the queue.
Agent When set to ‘Call as Dialed’ the queue will call an extension just as if the
Restrictions queue were another user. Any Follow-Me or Call Forward states active on the
extension will result in the queue call following these call paths. This behavior
has been the standard queue behavior on past PBX versions.
When set to ‘No Follow-Me or Call Forward’, all agents that are extensions on

the system will be limited to ring their extensions only. Follow-Me and Call
Forward settings will be ignored. Any other agent will be called as dialed. This
behavior is similar to how extensions are dialed in ringgroups
When set to ‘Extensions Only’ the queue will dial Extensions as described for

‘No Follow –Me or Call Forward’. Any other number entered for an agent that
is NOT a valid extension will be ignored. No error checking is provided when
entering a static agent or when logging on as a dynamic agent, the call will
simply be blocked when the queue tries to call it. For dynamic agents, see the
‘Agent Regex filter’ to provide some validation.

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General Queue Options

Ring Strategy Ringall: ring all available agents until one answers (default)
Leastrecent: ring agent which was least recently called by this queue

Fewestcalls: ring the agent with fewest completed calls from this queue

Random: ring random agent

Rrmemory: round robin with memory, remember where we left off last ring

pass
Rrordered: same as rrmemory, except the queue member where order from

config file is preserved

Linear: rings agents in the order specified, for dynamic agents in the order

they logged in
Wrandom: random using the member’s penalty as a weighting factor, see

asterisk documentation for specifics.

Autofill Starting with Asterisk 1.4, if this is checked, and multiple agents are available,
Asterisk will send one call to each waiting agent(depending on the ring
strategy). Otherwise, it will hold all calls while it tries to find an agent for the
top call in the queue making other calls wait. This was the behavior in Asterisk
1.2 and has no effect in 1.2. See Asterisk documentation for more details of
this feature.
Skip Busy When set to 'Yes' agents who are on an occupied phone will be skipped as if
Agents the line were returning busy. This means that Call Waiting or multi-line phones
will not be presented with the call and in the various hunt style ring strategies,
the next agent will be attempted.
When set to 'Yes + (ringinuse=no)' the queue configuration flag 'ringinuse=no'

is set for this queue in addition to the phone's device status being monitored.
This results in the queue tracking remote agents (agents who are a remote
PSTN phone, called through Follow-Me, and other means) as well as PBX
connected agents, so the queue will not attempt to send another call if they are
already on a call from any queue.
When set to 'Queue calls only (ringinuse=no)' the queue configuration flag

'ringinuse=no' is set for this queue also but the device status of locally
connected agents is not monitored. The behavior is to limit an agent belonging
to one or more queues to a single queue call. If they are occupied from other
calls, such as outbound calls they initiated, the queue will consider them

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available and ring them since the device state is not monitored with this
option.
WARNING: When using the settings that set the 'ringinuse=no' flag, there is a

NEGATIVE side effect. An agent who transfers a queue call will remain
unavailable by any queue until that call is terminated as the call still appears as
'inuse' to the queue UNLESS 'Agent Restrictions' is set to 'Extensions Only'.
Queue Weight Gives queue a ‘weight’ option, to ensure calls waiting in a higher priority
queue will deliver its calls first if there are agents common to both queues.
Music on Hold Music (MoH) played to the caller while they wait in line for an available
Class agent. Choose “inherit” if you want the MoH class to be what is currently
selected, such as by the inbound route. MoH Only will play music until the
agent answers. Agent Ringing will play MoH until an agent’s phone is
presented with the call and is ringing. If they don’t answer MoH will return.
Ring only makes callers hear a ringing tone instead of MoH ignoring any MoH
class selected as well as any configured periodic announcements. This music
is defined in the “Music on Hold” Menu.
Join Announcement played to callers prior to joining the queue. This can be
Announcement skipped if there are agents ready to answer a call (meaning they still may be
wrapping up from a previous call) or when they are free to answer the call
right now. To add additional recordings please use the “System Recordings”
MENU.
Caller Volume Adjust the recording volume of the caller.
Adjustment
Agent Volume Adjust the recording volume of the queue member (Agent).
Adjustment
Mark calls Enabling this option, all calls are marked as ‘answered elsewhere’ when
answered cancelled. The effect is that missed queue calls are *not* shown on the
elsewhere phone(if the phone support it)
Timing & Agent Options
Max Wait Time The maximum number of seconds a caller can wait in a queue before being
pulled out.(0 for unlimited).
Max Wait Time Asterisk timeoutpriority. In ‘Strict’ mode, when the ‘Max Wait Time’ of a
Mode caller is hit, they will be pulled out of the queue immediately. In ‘Loose’
mode, if a queue stops ringing with this call, then we will wait until the queue
stops ringing this queue number or otherwise the call is rejected by the queue
member before taking the caller out of the queue. This means that the ‘Max
Wait Time’ could be as long as ‘Max Wait Time’+’Agent Timeout’ combined.
Agent Timeout The number of seconds an agent’s phone can ring before we consider it a
timeout. Unlimited or other timeout values may still be limited by system
ringtime or individual extension defaults.
Agent Timeout If timeout restart is set to yes, then the time out for an agent to answer is reset

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Restart if a BUSY or CONGESTION is received. This can be useful if agents are able
to cancel a call with reject or similar
Retry The number of seconds we wait before trying all the phones again. Choosing
“No Retry” will exit the queue and go to the fail-over destination as soon as
the first attempted agent time-out, additional agents will not be attempted.
Wrap-Up-Time After a successful call, how many seconds to wait before sending a potentially
free agent another call (default is 0, or no delay) If using Asterisk 1.6+, you
can also set the ‘Honor Wrapup Time Across Queues setting (Asterisk:
shared_lastcall) on the Advanced Settings page so that this is honored across
queues for members logged on to multiple queues.
Member Delay If you wish to have a delay before the member is connected to the caller (or
before the member hears any announcement messages), set this to the number
of seconds to delay.
Agent Announcement played to the Agent prior to bridging in the caller.
Announcement
Example : ”the Following call is from the Sales Queue” or “This call is from
the Technical Support Queue”.
To add additional recordings please use the “System Recordings” MENU.

Compound recordings composed of 2 or more sound files are not displayed as

options since this feature can not accept such recordings.
Report Hold If you wish to report the caller’s hold time to the member before they are
Time connected to the caller, set this to yes.
Auto Pause Auto Pause an agent in this queue (or all queues they are a member of ) if they
don’t answer a call. Specific behavior can be modified by the Auto Pause
Delay as well Auto Pause Busy/Unavailable settings if supported on this
version of Asterisk.
Auto Pause on When set to Yes agents devices that report busy upon a call attempt will be
Busy considered as a missed call and auto paused immediately or after the auto
pause delay if configured
Auto Pause on When set to Yes agents devices that report congestion upon a call attempt will
Unavailable be considered as a missed call and paused immediately or after that auto pause
delay if configured
Auto Pause This setting will delay the auto pause of an agent by auto pause delay seconds
Delay from when it last took a call. For example, if this were set to 120 seconds, and
a new call is presented to the agent 90 seconds after they last took a call, will
not be auto paused if they don’t answer the call. If presented with a call 120
seconds or later after answering the last calls, this will have no effect.
Capacity Options
Max Callers Maximum number of people waiting in the queue (0 for unlimited)
Join Empty Determines if new callers will be admitted to the Queue, if not, the failover

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destination will be immediately pursued. The options include:

Yes Always allows the caller to join the Queue.

Strict Same as Yes but stricter. Simply speaking, if no agent could answer
the phone then don’t admit them. If agents are infused or ringing someone
else, caller will still be admitted.
Ultra Strict Same as Strict plus a queue member must be able to answer
the phone ‘now’ to let them in. simply speaking, any ‘available’ agents
that could answer but are currently on the phone or ringing on behalf of
another caller will be considered unavailable.
No Callers will not be admitted if all agents are paused, show an invalid
status for their device, or have penalty values less than
QUEUE_MAX_PENALTY (not currently set in dialplan).
Loose Same as No except Callers will be admitted if there are paused
agents who could become available.

Leave Empty Determines if callers should be exited prematurely from the queue in situations
where it appears no one is currently available to take the call. The options
include:
Yes Callers will exit if all agents are paused, show an invalid state for

their device or have penalty values less than QUEUE_MAX_PENALTY

(not currently set in dialplan)
Strict Same as Yes but stricter. Simply speaking, if no agent could answer
the phone then have them leave the queue. If agents are inuse or ringing
someone else, caller will still be held.
Ultra Strict Same as Strict plus a queue member must be able to answer
the phone ‘now’ to let them remain. simply speaking, any ‘available’
agents that could answer but are currently on the phone or ringing on
behalf of another caller will be considered unavailable.
Loose Same as No except Callers will remain in the queue, if there are
paused agents who could become available.
No never have a caller leave the Queue until the Max Wait Time has
expired.

Penalty Members Asterisk: penalty members limit. A limit can be set to disregard penalty
Limit settings, allowing all members to be tried, when the queue has too fewer
members. No penalty will be weight in if there are only X or fewer queue
members.
Frequency How often to announce queue position and estimated holdtime (0 to Dis able
Announcements).
Announce Announce position of caller in the queue
Position

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Announce Hold Should we include estimated hold time in position announcements? Either yes,
Time no, or only once; hold time will not be announced if <1 minute.
IVR Break Out You can optionally present an existing IVR as a ‘break out’ menu.
Menu
This IVR must only contain single-digit ‘dialed options’ . The recording set for
the IVR will be played at intervals specified in ‘Repeat Frequency’, below.
Repeat How often to announce a voice menu to the caller (0 disable Announcements)
Frequency
Event When When this option is set to YES, the following manager events will be
Called generated: AgentCalled, AgentDump, AgentConnect and AgentComplete.
Member Status When set to YES, the following manager event will be generated:
Event QueueMemberStatus.
Service Level Used for service level statistics (calls answered within service level time
frame)
Agent Regex Provides an optional regex expression that will be applied against the agent
Filter callback number. If the callback number does not pass the regex filter then it
will be treated as invalid. This can be used to restrict agents to extensions
within a range, not allow callbacks to include keys like *, or any other use that
may be appropriate. An example input might be:
^([2-4][0-9]{3})$

This would restrict agents to extensions 2000-4999. Or

^([0-9]+)$ would allow any number of any length, but restrict the * key.

WARNING: make sure you understand what you are doing or otherwise leave

this blank!
Run Select how often to reset queue stats. The following schedule will be followed
for all but custom:
Hourly Run once an hour, beginning of hour

Daily Run once a day, at midnight

Weekly Run once a week, midnight on Sun
Monthly Run once a month, midnight, first of month
Annually Run once a year, midnight, Jan.1
Reboot Run at startup of the server OP of the cron deamon (i.e. after every
service cron restart)
If Randomize is selected, a similar frequency will be followed, only the exact

times will be randomized (avoiding peak business hours, when possible).

Please note: randomized schedules will be rescheduled (randomly) every time
ANY backup is saved.

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3.4.3 Phonebook

With the Phonebook module, we can have a centralized list of numbers that can be accessed by
the users. Each number of this list has a special code in order to dial it quicker than by dialing the
number itself.

Figure 3-4-3 Phonebook interface

Navigate to PBX > Call Control > Phonebook, add a speed dial number by using the following
information.

Table 3-4-3 Definition of Phonebook

Item Definition
Speed Dial This option must be checked
Name Name of the speed dial
Number Destination external number
Speed dial code A number to associate this code to the external number
to dial
To dial this speed dial number, we dial *088, where *0 is to access the speed dial system's
feature and 88 is the speed dial code we entered.

Some actions that we can perform on the speed dial administration web page are as follows:

Export in CSV: If we click on this link, we can download the current speed dial list.

Import from CSV: We can upload a CSV file with the format: “Name”; Number; Speeddial

Navigate to PBX > Settings > Functions Code, switch the Speeddial prefix to Enabled.

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3.4.4 DISA

DISA (Direct Inward System Access) allows you to dial in from outside to the Asterisk switch
(PBX) to obtain an "internal" system dial tone. You can place calls from it as if they were placed
from within.

Figure 3-4-4 DISA Interface

When you choose the DISA option to call a number, you will be greeted with “Please enter your
password followed by the pound key” and after entering your password, you will then get a dial
tone. You may start dialing the telephone number.

Table 3-4-4 Definition of add DISA

Item Definition
DISA name Give this DISA a brief name to help you identify it.
PIN The user will be prompted for this number. If you wish to have multiple
PIN’s, separate them with commas.
Response The maximum amount of time it will before hanging up if the user has dialed
Timeout an incomplete or invalid number. Default of 10 seconds.
Digit Timeout The maximum amount of time permitted between digits when the user is
typing in an extension. Default of 5.
Require Require Confirmation before prompting for password. Used when your PSTN
Confirmation connection appears to answer the call immediately.
Caller ID (Optional) When using this DISA, the users CallerID will be set to this.
Format is “User Name” <5551234>
Context (Experts Only) Set the context that calls will originate from. Leaving this as
from-internal unless you know what you’re doing.
Allow Hangup Allow the current call to be disconnected and dial tone presented for a new
call by pressing the Handup feature code: ** while in a call.
Caller ID Determine if we keep the Caller ID being presented or if we override it.
Override Default is Enable.

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3.4.5 Conference

The Conference option is used to create a single extension number that your users can dial so that
they can talk to each other in a conference call. It also creates a destination to which you can send
calls so that they can participate in the conference call.

For example, you could create a Conference that will allow your local phones to dial 800, and
then enter into a conference call.

Figure 3-4-5 Conference interface

Below a description of each parameter:

Table 3-4-5 Definition of add Conference

Item Definition
Basic
Conference Use this number to dial into the conference.
Number
Conference Give this conference a brief name to help you identify it.
Name
User PIN You can require callers to enter a password before they can enter this conference.
This setting is optional.

If either PIN is entered, the user will be prompted to enter a PIN.

Admin PIN Enter a PIN number for the admin user.

This setting is optional unless the ‘leader wait’ option is in use, then this PIN will

identify the leader.

Advanced

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Join Message Message to be played to the caller before joining the conference.
To add additional recordings use the “System Recordings” MENU to the left

Leader Wait Wait until the conference leader (admin user) arrives before starting the conference
Talker Turn on talker optimization. With talker optimization, Asterisk treats talkers who
Optimization are not speaking as being muted, meaning that no encoding is done on transmission
and that received audio that is not registered as talking is omitted, causing no
buildup in background noise.
Talker Sets talker detection. Asterisk will sends events on the Manager Interface
Detection identifying the channel that is talking. The talker will also be identified on the
output of the meetme list CLT command.
Quiet Mode Quiet mode (do not play enter/leave sounds)
User Count Announce user(s) count on joining conference
User join/leave Announce user join/leave
Music on Hold Enable Music on Hold when the conference has single caller
Music on Hold Music (or Commercial) played to the caller while they wait line for the conference
Class to start. Choose “inherit” if you want the MoH class to be what is currently
selected, such as by the inbound route.
This music is defined in the “Music on Hold” to the left.

Allow Menu Present Menu (user or admin) when ‘*’ is received (‘send’ to menu).
Record Record the conference call
Conference
Maximum Maximum Number of users allowed to join this conference.
Participants
Mute on Join Mute everyone when they initially join the conference. Please note that if you do
not have ‘Leader Wait’ set to yes you must have ‘Allow Menu’ set to Yes to
unmute yourself.

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3.4.6 Callback

Callback is where you make a call to your IP-PBX and when reached you will be disconnected, but
it does not end there. Your PBX will in turn call your mobile and reconnect you relieving you of the
cost of the lengthy Mobile phone call that you will otherwise be up for.

Let’s take this step by step.

1. Setup DISA

Figure 3-4-6 Set on DISA

a. DISA name: MyMobile

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b. Response Timeout:10

c. Digit Timeout:5

d. Caller ID:0400123456 (My Mobile Number)

e. Context: from-internal

2. Setup Callback

Figure 3-4-7 Callback interface

a. Callback Description: My Mobile

b. Callback Number: 0400123456 (My mobile Number)

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c. Delay Before Callback:10

d. Destination after Callback: IVR – Residence (or Office IVR)

3. Inbound Routes

Figure 3-4-8 Inbound Routes interface

a. Description: Callback-MyMobile

b. DID Number:61247324100 (My DID number)

c. Caller ID Number: 0400123456 (My mobile Number)

d. Set Destination to: Callback – MyMobile

Click Save button then Click on the red circle at the top & follow on screen prompts

Now enable send caller ID on your mobile and call your DID number. When connected you will
get one beep and then followed by silence. Hang up your mobile and wait for approximately10
seconds and your mobile will ring.

When you answer your mobile, you will hear your IVR playing with the various options. One of
the silent options in my IVR is DISA. If I need to make an external call using my PBX. If I know
the option and select it, I will be then get DISA where I can make an external call at no cost to my
Mobile.

Table 3-4-6 Definition of add Callback

Item Definition
Callback Enter a description for this callback
Description

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Callback Number Optional: Enter the number to dial for the callback. Leave this blank to just
dial the incoming CallerID Number.
Delay Before Optional: Enter the number of seconds the system should wait before calling
Callback back.
Destination Destination of a callback.
Item Definition
Callback Enter a description for this callback
Description
Callback Number Optional: Enter the number to dial for the callback. Leave this blank to just
dial the incoming CallerID Number.
Delay Before Optional: Enter the number of seconds the system should wait before calling
Callback back.

3.4.7 Parking Lot

Figure 3-4-9 Parking Lot interface

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Table 3-4-7 Definition of Parking Lot

Item Definition
Basic
Parking Lot This is the extension where you will transfer a call to park it
Extension
Parking Lot Name Name of the parking Lot.
Parking Lot Starting The starting postion of the parking lot.
Position
Number of Slots The total number of parking lot spaces to configure.
Parking Timeout The timeout period in seconds that a parked call will attempt to ring back
(seconds) the original parker if not answered.
Parked Music Class This is the music class that will be played to a parked call while in the
parking lot UNLESS the call flow prior to parking the call explicitly set a
different music class
BLF Capabilities Enable this to have Asterisk “hints” generated to use with BLF buttons.
Find Slot If you want the parking lot to seek the next sequential parking slot relative
to the the last parked call instead of seeking the first available slot.
Destination Destination of Parking Lot.
Advanced
Pickup Courtesy Whom to play the courtesy tone to when a parked call is retrieved.
Tone
CallerID Prepend String to prepend to the current Caller ID associated with the parked call
prior to sending back to the Originator or the Alternate Destination.
Transfer Capability parkedcalltransfers. Enables or disables DTMF based transfers when
picking up a parked call.
Parking Alert-Info Alert-Info to add to the call prior to sending back to the Originator or to the
Alternate Destination.
Re-Parking parkedcallreparking. Enables or disables DTMF based parking when
Capability picking up a parked call
Auto CallerID These options will be appended after CallerID Prepend if set.
Prepend
Announcement Optional message to be played to the call prior to send back to the
originator.
Come Back to Where to send a parked call that has timed out. If set to yes then the parked
Origin call will be sent back to the originating device that sent the call to this
parking lot.

This module is used to configure Parking Lot(s) in Asterisk.

Simply transfer the call to said parking lot extension. Asterisk will then read back the parking
lot number the call has been placed in. To retrieve the call simply dial that number back.

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Table 3-4-8 Example usage of Parking Lot

*2nn: Attended Transfer call into Park lot nnn

(It will announce the slot back to you)
nn: Park Yourself into Parking lot nnn
(Announcing your parked slot to you)

3.4.8 Voicemail Blasting

Voicemail blasting lets you send a voicemail message to multiple users at the same time. The
Voicemail Blasting module is used to create a group of users and assign a number to the group. A
user can dial this number to leave a voicemail message for the group. All members of the group
will receive the message in their voicemail boxes.

Figure 3-4-10 Voicemail Blasting interface

Table 3-4-9 Definition of add VMBlast Group

Item Definition
VMBlast Number The number users will dial to voicemail boxes in this VMBlast group
Group Description Provide a descriptive title for this VMBlast Group.
Audio Label Paly this message to the caller so they can confirm they have dialed the proper

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voice mail group number, or have the system simply read the group number.
Optional Password You can optionally include a password to authenticate before providing access
to this group voicemail list.
Voicemail Box List Select voice mail boxes to add to this group. Use Ctrl key to select multiple.
Default VMBlast Each PBX system cam have a single Default VOICEMAIL Blast Group. If
Group specified, extensions can be automatically added (or removed) from this
default group in the Extensions (or Users) tab.
Making this group the default will uncheck the option from the current default

group if specified.

3.4.9 Paging and Intercom

The Paging and Intercom module is used to set up an extension number that your users can dial in
order to place an intercom call to multiple phones on your system at the same time.

For example, in a small office, you might set up a page group with extension number "100." When
100 is dialed by a local user, all of the phones in the office would go off-hook, and you could speak
to everyone at every extension at the same time. Alternatively, you could set up page groups with
different extension numbers for each department in the office, i.e. 100 for sales, 110 for service,
and so on.

This module is for specific phones that are capable of Paging or Intercom. This section is for
configuring group paging, intercom is configured through Feature Codes. Intercom must be enabled
on a handset before it will allow incoming calls. It is possible to restrict incoming intercom calls to
specific extensions only, or to allow intercom calls from all extensions but explicitly deny from
specific extensions.

This module should work with Aastra, Grandstream, Linksys/Sipura, Mitel, Polycom, SNOM, and
possibly other SIP phones (not ATAs). Any phone that is always set to auto-answer should also
work (such as the console extension if configured). Intercom mode is currently disabled, it can be
enabled in the Feature Codes Panel.

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Figure 3-4-11 Paging and Intercom interface

Table 3-4-10 Definition of Paging and Intercom

Item Definition
Paging The number users will dial to page this group.
Extension
Group Provide a descriptive title for this VMBlast Group.
Description
Device List Choose extensions.
Busy Skip will not page any busy extension. All other extensions will be paged as
Extensions normal Force will not check if the device is in use before paging it. This means
conversations can be interrupted by a page (depending on how the device handles
it).
Duplex Paging is typically one way for announcements only Checking this will make the
paging duplex, allowing all phones in the paging group to be able to talk and be
heard by all.

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3.4.10 Scheduled Broadcast

You can broadcast some audio by setting up the scheduled broadcast feature to inform the group.

Figure 3-4-12 Scheduled Broadcast interface

Table 3-4-11 Definition of Scheduled Broadcast

Item Definition
Paging/Intercom Select the desired paging group or intercom group.
Prompt Random If enabled, the prompt should be played randomly.
Prompt Select the desired prompt.
Custom Date You can select a time to play the scheduled broadcast on a special day.

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3.4.11 Wakeup Service

User can enable the Wakeup service and set the time and date, members, and receive the call
reminder after the time. The wake-up service will ring for 30 seconds every 30 seconds for
the duration of the wake-up service.

In the following example, the wakeup service is set up for extension101-110 from Monday to
Friday on 7:40AM.

Figure 3-4-13 Wakeup Service Interface

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3.5 Voice Prompts

3.5.1 Languages

The Languages module is used to allow calls to be routed to localized or alternate

language recordings.

Figure 3-5-1 Languages interface

Languages allow you to change the language of the call flow and then continue on to the desired
destination. For example, you may have an IVR option that says "For French Press 5 now". You
would then create a French language instance and point it's destination at a French IVR. The
language of the call's channel will now be in French. This will result in French sounds being
chosen if installed.

Table 3-5-1 Definition of add Language

Item Definition
Description The descriptive name of this language instance. For example, “French Main IVR”
Language The Asterisk language code you want to change to. For example, “fr” for French.
Code
Destination Indicates extension, Ring Group, Voicemail or other destination to which the call is
supposed to be directed when the outside callers have called specified

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3.5.2 System Recordings

The System Recordings module is used to record or upload messages that can then be played back
to callers in other modules. It can also be used to make pre-installed Asterisk recordings available
for use in other modules.

For example, you might create a recording called "Main Menu" and then play that message in an
IVR before a caller is asked to make a selection. Or, you might record a recording called "Holiday
Message" and then use that message in an Announcement. You would then route incoming calls
to the Announcement or IVR using the Inbound Routes Module.

Figure 3-5-2 System Recordings interface

Figure 3-5-3 Custom Prompts Interface

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3.5.3 Announcement

The Announcements Module is used to create a destination that will play an informational
message to a caller. After the message is played, the call will proceed to another destination.

For example, you might create an Announcement that plays the address, fax number, and the web-
site of your business. A caller could reach that message by pressing the number 2 from the
company's main menu. After hearing the message, the call might be routed back to the company's
main menu and allowed to make another selection.

Figure 3-5-4 Announcements interface

Table 3-5-2 Definition of Announcements

Item Definition
Name The name of this announcement
Recording Message to be played.
To add additional recordings, use the “System Recordings” MENU to the left

Repeat Key to press that will allow for the message to be replayed. If you choose this
option there will be a short delay inserted after the message. If a longer delay is
needed it should be incorporated into the recording.
Allow Skip If the caller is allowed to press a key to skip the message
Return to If the announcement came from an IVR and this box is checked, the destination
IVR below will be ignored and instead it will be return to the calling IVR. Otherwise,
the destination below will be taken. Don’t check if not using in this mode.
The IVR return location will be to the last IVR in the call chain that was called so

be careful to only check when needed. For example, if an IVR directs a call to

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another destination which eventually calls this announcement and this box is
checked, it will return to that IVR which may not be the expected behavior.
Don't Answer Check this to keep the channel from explicitly being answered. When checked, the
Channel message will be played and if the channel supports that. When not checked, the
channel is answered followed by a 1 second delay. When using an announcement
from an IVR or other sources that have already answered the channel, that 1
second delay may not be desired.
Destination Indicates extension, Ring Group, Voicemail or other destination to which the call
is supposed to be directed when the outside callers have called specified

3.5.4 Route Congestion

Figure 3-5-5 Route Congestion Messages interface

Table 3-5-3 Definition of Route Congestion Messages

Item Definition
No Routes Available
Standard Message or tone to be played if no trunks are available.
Routes
Intra-Company Message or tone to be played if no trunks are available. Used on routes marked
Routes as intra-company only.
Emergency Message or tone to be played if no trunks are available. Used on all emergency
Routes routes. Consider a message instructing caller to find an alternative means of
calling emergency services such as a cell phone or alarm system panel.
Trunk Failures

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No Answer Message or tone to be played if there was no answer. Default message is: “The
number is not answering." Hangupcause is 18 or 19
Number or Message or tone to be played if trunk reports Number or Address Incomplete.
Address Usually this means that the number you have dialed is to short. Default message
Incomplete is: “The number you have dialed is not in service. Please check the number and
try again."Hangupcause is 28

3.5.5 Music On Hold

The volume adjustment is a linear value. Since loudness is logarithmic, the linear lever will be less of
an adjustment. You should test out the installed music to assure it is at the correct volume. This
feature will convert MP3 files to WAV files. If you do not have mpg123 installed, you can set the
parameter: Convert Music Files to WAV to false in Advanced Settings.

Figure 3-5-6 Music on Hold Interface

You can add a custom music on hold playlist and upload your audio files to the PBX.

1. Add a Custom music on hold Playlist.

Go to PBX > Voice Prompts > Music on Hold page, click Create New Playlist. On
the configuration page, set the playlist name and the playlist order, click Save.

2. Upload the audio file.

Click Browse to choose an audio file from your local PC, and then click Upload

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3.6 Settings

3.6.1 Global Settings

Figure 3-6-1 Global Settings Interface

Table 3-6-1 Instruction of Global Settings/Basic

Options Definition
Asterisk Manager

Asterisk Manager Password for accessing the Asterisk Manager Interface (AMI), this will be
Password automatically updated in manager.conf.
Asterisk Manager Username for accessing the Asterisk Manager Interface (AMI), this will be
User automatically updated in manager.conf.

System Setup

Aggressively
Check for Aggressively Check for Duplicate Extensions
Duplicate

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Sets the extension behavior in UC device. If set to extensions, Devices and

User & Devices Users are administered together as a unified Extension, and appear on a
Mode single page. If set to deviceanduser, Devices and Users will be administered
separately.

Call Recording Format to save recoreded calls for most call recording unless specified
Format differently in specific applications.

When set to false, the MP3 files can be loaded and WAV files converted to

Convert Music MP3 in the MoH module. The default behavior of true assumes you have
mpg123 loaded as well as sox and will convert MP3 files to WAV. This is
Files to WAV
highly recommended as MP3 files heavily tax the system and can cause
instability on a busy phone system

Voice Prompts

Default language The default language for Voice Prompts.

3.6.2 Analog Settings

Basic setting

Figure 3-6-2 Analog Settings/Basic/General Configuration

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Table 3-6-2 Instruction of General

Options Definition

Tone duration How long generated tones (DTMF and MF) will be played on the
channel. (in milliseconds)
Codec Set the global encoding: ulaw, alaw.

Impedance Configuration for impedance.

Echo Cancellation Enable/Disable

EC Taps 128/256/512/1024

Flash/Wink Turn on/off Flash/wink.

Min flash time Min flash time. (in milliseconds).

Max flash time Max flash time. (in milliseconds).

Ending Dial Key Turn on/off Ending Dial Key (#).

Hang up on polarity switch Turn on/off Hangup on polarity switch

Figure 3-6-3 Analog Settings/Basic/Hardware gain

Table 3-6-3 Instruction of Analog Settings/Basic/Hardware gain

Options Definition
FXO Rx gain Set the FXO port Rx gain. Range: from -150 to 120.
FXO Tx gain Set the FXO port Tx gain. Range: from -150 to 120.
FXS Rx gain Set the FXS port Rx gain. Range: -35, 0 or 35.
FXS Tx gain Set the FXS port Tx gain. Range: -35, 0 or 35.

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Figure 3-6-4 Analog Settings/Basic/Gain Settings

Table 3-6-4 Instruction of Analog Settings/Basic/Gain

Options Definition
Rx gain Gain for the rx (receive) channel. Default: 0.0
Tx gain Gain for the tx (transmit) channel. Default: 0.0

Figure 3-6-5 Analog Settings/Basic/Fax

Table 3-6-5 Definition of Analog Settings/Basic/Fax

Options Definition
Maximum Transmission Rate Set the maximum transmission rate
Minimum Transmission Rate Set the minimum transmission rate
Ecm Enable/disable T.30 ECM (error correction mode) by default.
Caller ID setting

Figure 3-6-6 Analog Settings/Caller ID/Send Caller ID

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Table 3-6-6 Instruction of Analog Settings/Caller ID/Send Caller ID

Option Description
The pattern of sending Some countries (UK) have ring tones with different ring
CID tones(ring-ring), which means the caller ID needs to be set later
on, and not just after the first ring, as per the default (1).
Waiting time before How long we will be waiting before sending the CID on the
sending CID channel. (in milliseconds).
Sending polarity Send polarity reversal before sending the CID on the channel.
reversal (DTMF Only)
Start code (DTMF Only) Start code.
Stop code (DTMF Only) Stop code.

Figure 3-6-7 Analog Settings/Caller ID/CallerID Detection

Table 3-6-7 Instruction of Analog Settings/Caller ID/CallerID detect

Options Definition
Use CallerID Turn on/off CallerID detect function
Hide CallerID Turn on/off CallerID detect function
Callerid Caller ID can be set to "asreceived" or a specific number if you want to
override it. Note that "asreceived" only applies to trunk interfaces.
CID Signalling Type of caller ID signalling in use

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bell = bell202 as used in US (default)

v23 = v23 as used in the UK

dtmf = DTMF as used in Denmark, Sweden and Netherlands

CID Start What signals the start of caller ID

ring=a ring signals the start (default)

polarity=polarity reversal signals the start

Figure 3-6-8 Analog Settings/Tone/Country

Table 3-6-8 Definition of Analog Settings/Tone/Country

Options Definition
Country Configuration for location specific tone indications.
Dial tone Set of tones to be played when one picks up the hook.
Busy tone Set of tones played when the receiving end is busy.
Congestion tone Set of tones played when there is some congestion.
Record tone Set of tones played when call recording is in progress.
Ring cadence List of durations the physical bell rings.

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Ring tone Set of tones to be played when the receiving end is ringing.
Call waiting tone Set of tones played when there is a call waiting in the background.
Dial recall tone Many phone systems play a recall dial tone after hook flash.
Info tone Set of tones played with special information messages (e.g., number is
out of service.)

Figure 3-6-9 Analog Settings/Tone/Silence detect

Table 3-6-9 Definition of Analog Settings/Tone/Silence detect

Options Definition
Silence detect Turn on/off silence detect function
Silence threshold What we consider silence: the lower, the more sensitive, eg:250 is
250ms. Range: 100 to 500(100 to 500ms), default: 250
Silence length How many silence threshold of silence before hanging up(eg: 16 is
250ms*16=4s). Range: 2 to 1020 (200ms to 512s), default: 80(20s)
Silence Rx threshold Range: -20 dBm0 to -40 dBm0, default: 20(-20 dBm0), all values are
framesize understood to be negative.
Tx threshold Range: -20 dBm0 to -40 dBm0, default: 20(-20 dBm0), all values are
understood to be negative.

Figure 3-6-10 Analog Settings/Tone/Special tone

Table 3-6-10 Instruction of Analog Settings/Tone/Special tone

Options Definition
Custom Busy Tone detect Turn on/off busy detect function
Busy Tone count How many busy tones to wait for before hanging up. The
default is 3, but it might be safer to set to 6 or even 8.

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Busy Tone pattern Set the busy detect country

3.6.3 SIP Settings

Figure 3-6-11 SIP Settings/Basic/Misc PJSip Settings

Table 3-6-11 Instruction of SIP Settings/Basic/Misc PJSip Settings

Options Definition
User Agent Value used in User-Agent header for SIP requests and Server header for
SIP responses.
Realm When generates a challenge, the digest realm will be set to this value if
there is no better option (such as auth/realm) to be used.
Allow Guests When set Asterisk will allow Guest SIP calls and send them to the Default
SIP context. Turning this off will keep anonymous SIP calls from entering
the system. Doing such will also stop 'Allow Anonymous Inbound SIP
Calls' from functioning. Allowing guest calls but rejecting the Anonymous
SIP calls below will enable you to see the call attempts and debug
incoming calls that may be mis-configured and appearing as guests.
Domain The Typically used with SIP calling. Example user@domain, where domain is
Transport Comes the value that would be entered here
From
Local networks Local network settings in the form of ip/cidr or ip/netmask. For networks
with more than 1 LAN subnets, use the Add Local Network Field button
for more fields. Blank fields will be ignored.

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External Media External IP address to use in RTP handling.

Address
External External address for SIP signaling.
Signaling Address
External External port for SIP signaling.
Signaling Port

Figure 3-6-12 SIP Settings/Basic/NAT STUN Settings

Table 3-6-12 Instruction of SIP Settings/Basic/NAT STUN Settings

Options Definition
Enable If enabled, the Nat Stun will be enabled.
STUN Server Address of the STUN server to query.
Custom valid form:[(hostname | IP-address)]
STUN Port The port defaults to the standard STUN port (3478).
Refresh Time Number of seconds between STUN refreshes, Default is 30.

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Figure 3-6-13 SIP Settings/Transports

Table 3-6-13 Instruction of SIP Settings/Transports

Options Definition
UDP
Enable USE 0.0.0.0 - All
Bind Host You can customize the UDP bind host, the default is 0.0.0.0
Port To Listen On The port that this transport should listen on
TCP
Bind Host You can customize the TCP bind host, the default is 0.0.0.0
Port To Listen On The port that this transport should listen on
TLS
Bind Host You can customize the TLS bind host, the default is 0.0.0.0
Port To Listen On The port that this transport should listen on
Certificate Manager Select a certificate to use for the TLS transport. These are
configured in the module Certificate Manager1
SSL Method Method of SSL transport (TLS ONLY). The default is currently
TLSv1, but may change with future releases.1
Verify Client Require verification of server certificate (TLS ONLY).

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Verify Server Require verification of server certificate (TLS ONLY).

WS
Bind Host You can customize the WS bind host, the default is 0.0.0.0
WSS
Bind Host You can customize the WSS bind host, the default is 0.0.0.0

3.6.4 IAX2 Settings

Figure 3-6-14 IAX2 Settings/Audio Codecs

Table 3-6-14 Instruction of IAX2 Settings/Audio Codecs

Options Definition
Asterisk: codecpriority. Controls the codec negotiation of an inbound IAX call. This
option is inherited to all user entities. It can also be defined in each user entity
Codec separately which will override the setting here. The valid values are:host - Consider
the host's preferred order ahead of the caller's.caller - consider callers host's. disabled
Priority
disable consideration codec preference altogether. (this is original behavior before
preferences were added)reqonly same as disabled, only do not capabilities if
requested format available call will be accepted available.
Bandwidth Asterisk: bandwidth. Specify bandwidth of low, medium, or high to control which
codecs are used in general.
Video Check to enable and then choose allowed codecs. If you clear each codec and then

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Support add them one at a time, submitting with each addition, they will be added in order
which will effect the codec priority.
Audio Check the desired codecs, all others will be disabled unless explicitly enabled in a
Codecs device or trunks configuration. Drag to re-order.

Figure 3-6-15 IAX2 Settings/Jitter Buffer Settings

Table 3-6-15 Instruction of IAX2 Settings/Jitter Buffer Settings

Options Definition
Registration Settings
Registration Asterisk: minregexpire, maxregexpire. Minimum and maximum length of time
Times that IAX peers can request as a registration expiration interval (in seconds).
Jitter Buffer Settings
Asterisk: jitterbuffer. You can adjust several parameters relating to the jitter buffer.
Jitter Buffer The jitter buffer’ s function is to compensate for varying network delay. the jitter
buffer works incoming audio - outbound will be dejittered by at other end.

Figure 3-6-16 IAX2 Settings/Advanced General Settings

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Table 3-6-16 Instruction of IAX2 Settings/Advanced General Settings

Options Definition
Language Default Language for a channel, Asterisk: language
Asterisk: bindaddr. The IP address to bind to and listen for calls on the Bind Port. If
Bind set to 0.0.0.0 Asterisk will listen on all addresses. To bind to multiple IP addresses or
Address ports, use the Other iax settings' fields where you can put settings such
as:bindaddr='192.168.10.100:4555.' it is recommended to leave this blank.
Asterisk: bindport. Local incoming UDP Port that Asterisk will bind to and listen for
Bind Port IAX messages. The IAX standard is 4569 and in most cases this is what you want. It
is recommended to leave this blank.
Delay Auth Asterisk: delayreject. For increased security against brute force password attacks
enable this which will delay the sending of authentication reject for REGREQ or
Rejects
AUTHREP if there is a password.
You may set any other IAX settings not present here that are allowed to be
Other IAX configured in the General section of iax.conf. There will be no error checking against
these settings so check them carefully. They should be entered as: [setting] =
Settings
[value]in the boxes below. Click the Add Field box to add additional fields. Blank
boxes will be deleted when submitted.

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3.6.5 RTP Settings

Figure 3-6-17 RTP Settings

Table 3-6-17 Definition of RTP Settings Options

Options Definition
Strict RTP Enable strict RTP protection. This will drop RTP packets that do not come from
the source of the RTP stream. This option is disabled by default.
RTP Whether to enable or disable UDP checksums on RTP traffic
Checksums
Whether to enable ICE support. Defaults to no. ICE (Interactive Connectivity
ICE Support Establishment) is a protocol for network address Translator (NAT) traversal for
UDP-based multimedia sessions established with the offer/answer model. This
option is commonly enabled in WebRTC setups
RTP Start Start of range of port numbers to be used for RTP. Defaults is 10000.
RTP End End of range of port numbers to be used for RTP. Defaults is 20000.

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yes: standard reinvites; (yes = update + nonat)

no: never;

Reinvite nonat: An additional option is to allow media path redirection (reinvite) but only

Behavior when the peer where the media is being sent is known to not be behind a NAT (as
the RTP core can determine it based on the apparent IP address the media arrives
from;
update: use UPDATE for media path redirection, instead of INVITE.

The call is terminated when there is no RTP or RTCP activity on the audio channel
RTP Time Out for a period of time (that is, the set timeout period). This is to be able to hang up
the call in case of network interruption (not on hold)
RTP Hold If there is no RTP or RTCP activity on the audio channel for a period of time (that
is, the set hold timeout period), the call will be terminated (in hold state). This value
Time Out
must be greater than the timeout period.
RTP Keep Send Keepalive in RTP stream to keep NAT open (default is off)
Alive
Configure the STUN server address. STUN is a Client/Server protocol and also a
STUN Server Request/Response protocol. It is used here to check the connectivity between two
terminals, like a way to maintain NAT binding entries Keep-alive agreement.
Configure the TURN server address, STUN can handle most of the NAT problems.
TURN Server TURN is an enhanced version of the STUN protocol, dedicated to dealing with
symmetric NAT problems.
TURN Server Configure the TURN Server name
Name
TURN Server Configure the TURN Server password
Password

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3.6.6 Recording Settings

Figure 3-6-18 Recording Settings

Table 3-6-18 Definition of Recording Settings Options

Options Definition
Enable Recording of Internal Check this option, and all internal calls made by the selected
Calls extensions will be recorded automatically.
Internal Call Being Recorded If the internal call has enabled call recording, this prompt will
Prompt notify the called party that the call is being recorded.
Outbound/Inbound Calls If the external call (outbound/inbound) has enabled call
recording, this prompt will notify the external party that the call
Being Recorded Prompt
is being recorded.
Record Trunks When a call reaches the selected trunk, it will be recorded.

Record Extensions The selected extensions will be recorded.

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Record Conferences The selected conferences will be recorded.

3.6.6 Misc Destinations

The Misc Destinations Module is used to create a miscellaneous destination to which you can route
calls from another module.

For example, you might create a misc destination called "My Mobile Phone" that dials your
mobile telephone number. Then, you could set up an IVR so that if a caller presses 9, they would
be routed to "Misc Destinations:My Mobile Phone."

Misc Destinations are for adding destinations that can be used by other FreePBX modules,
generally used to route incoming calls. If you want to create feature codes that can be dialed by
internal users and go to various destinations, please see the Misc Applications module. If you need
access to a Feature Code, such as *98 to dial voicemail or a Time Condition toggle, these
destinations are now provided as Feature Code Admin destinations. For upgrade compatibility, if
you previously had configured such a destination, it will still work but the Feature Code short cuts
select list is not longer provided.

Figure 3-6-19 Misc Destinations interface

Table 3-6-19 Definition of add Misc Destination

Item Definition
Description Give this Misc Destination a brief name to help you identify it.
Dial Enter the number this destination will simulate dialing, exactly as you would
dial it from an internal phone. When you route a call to this destination, it
will be as if the caller dialed this number from an internal phone.

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3.6.7 Functions Code

The Feature Codes Module is used to enable and disable certain features available in your PBX and
Asterisk, and to set the codes that local users will dial on their phones to use that particular feature.

For example, the Feature Codes Module can be used to set the code that a user will dial to activate or
deactivate Call Forwarding. It can also be used to set a Code that can be used to enter into an Echo
Test, to hear your extension number, or to hear the time of day.

Figure 3-6-20 Feature code admin interface

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3.6.8 AMI

Figure 3-6-21 Manager User interface

Table 3-6-20 Definition of Manager User

Item Definition
Manager name Name of the manager without space.
Manager secret Password for the manager.
Deny If you want to deny many hosts or networks, use & char as separator.
Example: 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0

Permit If you want to permit many hosts or networks, use & char as
separator. Look at deny example.

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Figure 3-6-22 AMI Setting interface

Table 3-6-21 Definition of AMI Setting Options

Item Definition
AMI Port Sets the port number to listen on for AMI connections. Port numbers
can not be less than 1024. The default is 7777.
TLS Enable Enables listening for AMI connections using TLS. The default is no.
TLS Port Sets the port to listen on for TLS connections to the AMI. The default
is 5039.
TLS Bind Address Sets the address to listen on for TLS-based AMI connections. The
default is to listen on all addresses (0.0.0.0).
Timestamp Events Add a Unix epoch timestamp to events (but not action responses.
Write Timeout Write Timeout

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3.7 Recording

3.7.1 Call Recordings

The option "Calls Recordings" of the Menu "Recordings" in UC series lets us view a list with details
of all recorded calls for the extension associated to the connected user. The administrator account
can see all the recordings.

Figure 3-7-1 Calls Recordings interface

3.7.2 VoiceMails

The option "Voicemail" of the Menu “Recordings” in UC series lets us view a list with details of
the voicemails for the extension of the logged user.

Figure 3-7-2 Voicemails interface

The report will change depending on the values of the filter:

Table 3-7-1 Definition of Show Filter

Parameter Description
Start Date Start date for the selection of voicemails.
End Date End date for the selection of voicemails.

To delete a voicemail, just select the voicemail from the list and click on "Delete" button.

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3.7.3 VoiceMails Admin

The option "Voicemail Admin" of the Menu “Recordings” lets us view or modify some
voicemail configuration.

Figure 3-7-3 Voicemails Admin interface

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3.8 Tools

3.8.1 Asterisk-Cli

The option “Asterisk-Cli” of the Menu “Tools” in UC series lets us execute Asterisk commands.

Figure 3-8-1 Asterisk-Cli interface

To execute a command, input the same in the Command field and click on the button.

The module "Asterisk File Editor" of the Menu "Tools" in UC series lets us edit easily the
configuration files of UC series, while you have to enter the developer mode before use it. The
path of the files you can modify is /etc/asterisk/.

Figure 3-8-2 Asterisk File Editor interface

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Editing a file
You can find a file by entering the name in the filter field. To edit the file, click on the name to go
to the edit mode. Click on "Save" button to save changes and "Reload Asterisk" if necessary.

Figure 3-8-3 Editing a file interface

Creating a file
Also you can create a new file by clicking on "New File" button. This file will be created with
the extension ".conf" in /etc/asterisk/.

Figure 3-8-4 Create a file interface

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3.8.2 AI TTS

Text can be converted to audio in the "AI TTS" function module. The output format of this file can
be ".wav". Write the information you want to convert, select the output format, and click the
"Generate Audio File" button. The system will automatically save the file to the location of your
hard drive as you requested.

Figure 3-8-5 Text to Wav interface

3.8.3 API

This VoIP PBX provides the API interfaces for you to integrate a third-party software or device. You
need enable API to access on the pbx and Set the Username and Password, click Save and Apply. The
3rd-party software should use the user name and the password to connect to the PBX API.

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Figure 3-8-6 API Interface

In Extension tab, you can set whether to monitor extension status.

If the extension status is changed, the PBX will send report to the 3rd-party application server.

Figure 3-8-7 Extension Status Interface

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3.9 Auto Provision

The "Endpoint Configurator" module enables automatic remote configuration of supported
endpoints. With this module, the UC series administrator can point supported endpoints to the
UC series as their telephony server.

Figure 3-9-1 Endpoint Configurator interface

Interface description
Main listing

This is the listing of all endpoints that have been detected or entered. Unlike the old
implementation, any endpoints detected or uploaded in past sessions will be kept and displayed until
they are explicitly erased. The main listing contains the following columns:

Table 3-9-1 Description of Endpoint Configurator Interface

Item Description
Status This displays the status of the endpoint as one or more icons. The available
flags are as follows:
Scroll icon: the endpoint has not been scanned, but rather defined in an

upload.
Disk icon: the endpoint configuration has been updated in the database but

not yet applied to its configuration files.

Person icon: the endpoint has at least one endpoint assigned.

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MAC Address This is the main identifier for the endpoint. Configurations in the database
and uploaded files are considered to refer to the same endpoint if they
reference the same MAC address.
Current IP If the endpoint was detected through a scan, this field will show the IP at
which the endpoint was found. This field is a link to the HTTP
configuration interface (if supported) of the phone.
Manufacturer This displays the detected manufacturer of the endpoint.
Model This displays the detected model of the endpoint. Since automatic model
detection is not (yet) implemented for some manufacturers, this field allows
the user to correct the model via a drop-down list. Accurate model detection
is required for many other features (such as account assignment) to work.
Options This link displays a modal dialog on which common options for the
endpoint can be manually configured.

Endpoint scan toolbar button

This widget contains a textbox with a network/netmask definition, and a magnifying glass icon. By
default, the network definition will be filled with the network definition of the first ethernet interface
of the server. The user may correct this definition to restrict the scan, and then click on the icon to
start the scan. When scanning, the toolbar will change to a spinning icon and a Cancel button. As
endpoints are detected, they will be added to the main listing, along with their detected manufacturer
and model. The toolbar will revert to its default state when the scan is done, or if the scan is aborted
with the Cancel button.

Endpoint configuration toolbar button

Select a phone that needs to be configured, click and the following window will pop up, you can
clearly see some of the phone's attributes:

Figure 3-9-2 Endpoint Information

Click to set the extension, and check the line 1/2/3/4 to set the line independently
without affecting other lines.

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You can also click to adjust the network parameters of the phone. When selecting
static IP, please make sure to manually enter IP and other parameters.

Configuration apply toolbar button

Clicking on this button will start applying the configuration for all selected endpoints (all endpoints
for which the checkbox is set). When applying the configuration, the toolbar will change to a
progress bar. As endpoints are configured, the progress bar will update, and the toolbar will revert to
the default state when the configuration is done. During configuration, a log is generated, and can be
viewed by clicking on the Configuration Log toolbar button.

Configuration Log toolbar button

Clicking on this button will open a modal dialog in which a log of the last configuration run will be
shown. This is useful for diagnosing issues with the module failing to configure an endpoint.

Remove configuration toolbar button

Clicking on this button will (after a confirmation dialog) remove the database records for the selected
endpoints, as well as any generated configuration files for these endpoints. It will NOT, however,
contact the endpoints themselves in any way.

Download toolbar button

Clicking on this button will display a list of links to download the list of endpoints stored on the
database, in three different formats. The supported formats are:

• CSV (Legacy). This is the format used by the old Endpoint Configurator.
• XML. This format allows the definition of endpoints with multiple accounts and properties, as
an XML document.
• CSV (Nested). This format can be generated by careful editing in a spreadsheet, and uses
indentation to group multiple accounts and properties per endpoint.

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3.10 PBX Monitor

The " PBX Monitor " module of the menu "PBX" in UC IPPBX allows managing the telephony
operations. You can control inbound calls, outbound calls, the order in which the calls are taken,
the area that is designated to attend a call, etc.

This module is useful for receptionists who have a general view of the queues, conferences,
parking lots, internal extensions, trunks. Here the receptionist can start a call or transfer a call by
dragging one extension to another, or include several extensions to a conference room, or a queue.
The receptionist can also see the busy extensions, the elapsed time and the caller ID.

Figure 3-10-1 Operator Panel interface

Click to forcibly hang up the calling extension.

Click to jump to PBX>Recording>Voicemail

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3.11 Conference Panel

The conference panel can help you easily manage and monitor the conference and realize multi-party
calls.

3.11.1 Conference List

In the Conference List, you can check how many conferences are created on the PBX, and monitor
the status of the conferences.

Figure 3-11-1 Conference List Interface

Click the icon to enter the operation panel, as shown below:

Figure 3-11-2 Conference Operation Interface

Click the icon to invite the extension into the conference room. Of course, you can also select
multiple extensions and click one key to invite the selected extensions. In addition, extensions can
also directly dial the conference room number to actively join the conference room.

Click the icon to kick the extension out of the conference room.

Click the icon, the extension will be muted, and other extensions will not hear the sound of this
extension.

Click the icon, the extension will be unmuted, and other extensions can hear the voice of this
extension.

Click the icon to delete the extension from the conference operation panel. Of course, you can

also select multiple extensions and the selected extension with one click.

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Click to change the password of the participant (non-administrator). Leave it

blank to indicate that no password is required to enter the meeting. After the change, you need to
click Apply to take effect.

Click to add a new participant member, as shown below:

Figure 3-11-3 Add Contacts/Extension

Of course, you can also check Custom to enter other numbers. This number can be a mobile
phone number, and the number can be called from an outside line.

Figure 3-11-4 Add Contacts/Custom

Click , you can select a contact group member to import into the current conference
in batches.

Click to save the participants of the current conference to a contact group, as

shown in the figure below:

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Figure 3-11-5 Save Contact Group

3.11.2 Conference Contacts

In this module, you can manage and add contact groups.

Figure 3-11-6 Conference Contact

Click to add a contact group, as shown below:

Figure 3-11-7 Add Contact Group

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4 Fax

4.1 Virtual Fax

4.1.1 Virtual Fax List

The option Virtual Fax List of the Menu FAX in UC series lets us verify the list of all the
virtual faxes, including the status of each one.

Figure 4-1-1 Virtual Fax List interface

Clicking on the name of the Virtual Fax will jump to a page displaying its information, in which
you can Edit and Delete the Virtual Fax.

Figure 4-1-2 New Virtual Fax interface

Click you can create a new virtual fax. You should have previously created an IAX

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extension in PBX > Extensions > Add IAX2 Extension.

Figure 4-1-3 New Virtual Fax interface

To create a new virtual fax, enter the name, e-mail, extension, secret code, country code and area code
for the virtual fax (these are the mandatory fields). After this information is added, click on the

button to save the virtual fax.

4.1.2 Send Fax

The option Send Fax of the menu Fax in UC series allows sending faxes to one or more numbers.

Here you can enter the text you want to send and click on button.

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Figure 4-1-4 Send fax with text

information Also, you can send files in the format .pdf, .tiff and .txt

Figure 4-1-5 Send fax with File Upload

4.1.3 Fax Queue

The option Fax Queue from the Menu FAX in UC series shows the list of faxes that are awaiting its
turn to be sent. All the jobs have an ID and a status so you can monitor the sending of the faxes. If

you want to cancel a job, just select the job and click on button.

Figure 4-1-6 Fax Queue interface

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4.2 Fax Master

The option "Fax Master" of the Menu "FAX" in UC series lets us input the email address of the
administrator of the Fax, and this email will receive notifications of the messages received, errors
and other activities of the Fax Server.

Figure 4-2-1 Fax Master Interface

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4.3 Fax Clients

The option "Fax Clients" of the Menu "FAX" in UC series lets us input the IPs that have
permission to send faxes through UC series.

Figure 4-3-1 Fax Client interface

Proceed to input the IPs, one IP per line and click on the button.

It is recommended that you input the IP 127.0.0.1 and localhost in the configuration because
some processes might need to use them.

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4.4 Fax Viewer

The option "Fax Viewer" of the Menu "Fax" shows a list with all the faxes that have been sent
and received in the virtual Faxes. You can download the faxes if you click on the name of the file.

Figure 4-4-1 Fax Viewer interface

By the default all the files are shown, but you can filter according to company name, company fax,
fax date or type fax.

Figure 4-4-2 Fax Viewer show filter

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5 Reports

5.1 CDR Report

The option CDR Reports of the Menu Reports in UC series lets us view a list with the details of
the calls. You can download this list in different format files such as CSV, XLS and PDF.

Figure 5-1-1 CDR Report interface

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5.2 Channels Usage

The option Channels Usage of the menu Reports in UC series allows us view graphically
the number of simultaneous calls for each channel.

Figure 5-2-1 Channel usage interface

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5.3 Graphic Report

The option Graphic Report of the Reports module allows visualizing graphically information about
the number of calls, queues and trunks of the system both in quantity and percentage.

Figure 5-3-1 Graphic report interface

To see the information of a specific extension, select Extension (Number) and then click on the link

fill in the extension number and click .

It is possible to generate a graphic of Number of Calls in Queues. To do this just select Queue from

the dropdown menu .

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5.4 Summary
The option Summary of the menu Reports in UC series shows a report of each Extension registered
in the server. You can see the number of incoming and outgoing calls, the duration of the calls, the
caller id and the dialing number. Use the filter to find an extension or user.

Figure 5-4-1 Summary interface

Click on Call Details to see more information of an extension.

Figure 5-4-2 View Extension Call Details

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5.5 Missed Calls

The option Missed Calls of the menu Reports in UC series shows a report of the missed calls of
all extensions so you can know when an extension has been receiving calls. You can download this
report by clicking on "Download" button. The available formats for this file are csv, xml and pdf

You can filter the results by:

• Start Date: Find missed calls from this date.

• End Date: Find missed calls until this date.
• Search: You can filter the results by these
parameters: o Source: Number that made the call.
o Destination: Number that received the call.

Figure 5-5-1 Missed calls interface

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5.6 Downloads
In Downloads, users can find and download reports generated in previous modules by themselves,
including CDRs, call recordings, event logs, missed calls, weak keys, audits, and more.

Figure 5-6-1 Downloads interface

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6 Extras

6.1 Calling Cards

Calling Cards is a full-featured, flexible VoIP billing system and terminal platform that provides
the fastest and lowest cost service for VoIP billing customers on the market today. This feature has
been added to the new UC, and users who need it can enter the account password to log in.

Figure 6-1-1 Calling Cards interface

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6.2 Video Conference

Users can create video conferences in the IPPBX system, allowing multiple people to participate
at the same time.

Figure 6-2-1 Video Conference interface

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6.3 Hotel

6.3.1 Information

On the Extras > Hotel > Information page, you can see some information directly on this page.

The total rooms, how many available rooms (rooms free), or not (rooms busy), if you have some
booking today, if your hotel is full or potentially full (caused by the booking).

If there's a booking today, just click on the button to going to Booking list
menu directly.

Figure 6-3-1 Hotel Information

6.3.2 Service

Room List

You can see the status of the rooms currently in your hotel. The guest name, the room name, if
it's free or busy, cleaned or not. If the guest used the mini-bar or not. If the room is on DND (Do
Not Disturb) status or not. And you can see if the room is included in a group or not.

If the phone device is a SIP phone, you can know if the phone is connected or not. In this case,
you have a small yellow triangle beside the phone number.

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Figure 6-3-2 Room List

Check In

You can for a new customer. You can see some fields to enter different values.

Figure 6-3-3 Check-In option

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Date, it's the Check-in date (the current date by default). Date Checkout, is needed to have a
reference in the case where another room will be booked. This information is not used for billing.
Room, displays all available rooms into this list. Of course, you must enter a Last Name and
First Name to making a check-in.

The other fields below are optional. However, one field is needed in the case where you want to
sending the billing by mail. In this case, you must enter the Mail field. No billing will be sent by
fax yet.

Once the guest is checked-in, you can click and view the Guest’s info. Here you can see the
customer's name, check-in time, check-out time, room price and other information.

Figure 6-3-4 Guest Info

When the customer needs to change the room, select the icon.

Figure 6-3-5 Transfer

If the customer has purchased a Mini-Bar item or made an outside call, the system will automatically
transfer the customer's purchase and call costs to the new room.

And you also can check-in for a customer who have booked in Booking List:

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Figure 6-3-6 Check-in/Booking

List Or check-in for regular customers in Customers List:

Figure 6-3-7 Check-in/Customers List

Check Out

You can do 2 types of checkout. A classic checkout by room, and a checkout by group.

When the customer needs to check out, select Check Out icon.

Figure 6-3-8 Check out by room

Figure 6-3-9 Check out by room options

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If paid is checked, the billing is paid by the guest, else, this billing is tagged like not paid.

If you want to have all calls details for the room, check .

If you want that guest receive its billing by mail, check .

After checking out, you can check the billing report in Report > Billing Report.

Group List

Group List is used for unified management of customers who check-in in groups.

Figure 6-3-10 Group List

Here, you can see all group already existing, and you can add lots of checked rooms into a group in
the same time. Just selecting several rooms maintaining, press the shift key and click on the rooms
that you want.

Figure 6-3-11 Group List options

Check Out

Checkout by group will take all room in group, and will make the checkout, room by room.

Check the group you want check out and click . Checking out of 10.1-10.3

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group means that both Room 302 and Room 303 will check out.

Figure 6-3-12 Check out by group

option When checking out, check Details to see the call bill.

Booking List

Here, you have all booking which currently entered into Hotel. You can do a view between 2 dates.

Figure 6-3-13 Booking List

Booking room for a new customer:

Figure 6-3-14 Add Booking

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To make a checking on a booked room, click the Checkin box, and if you want to cancel a
booking, check the Canceled box.

Figure 6-3-15 Check-in/Booking

List Booking room for a regular customer in the Customers List:

Figure 6-3-16 Check-in/Customers List

Customers List

All check-in and booking information will be entered into the customer list. This module can
also customize customer information. You can book room or check-in for them.

Figure 6-3-17 Customers List

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Also, you can add a new customer.

Figure 6-3-18 Add a New Customer

Wake Up

This module allows you to set up a wake-up call service for specific customers. The wake-up
service will ring for 30 seconds every 30 seconds for the duration of the wake-up service.

In the following example, the wakeup service is set up for Room110's customers from 2020-08-
04 11:46 to 2020-08-04 11:52.

Figure 6-3-19 Wake Up options

Figure 6-3-20 Wake Up Interface

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The extension bound to Room110 will ring from 2020-08-04 11:46 to 2020-08-04 11:52

Figure 6-3-21 Wakeup Service

It can also be viewed or edited in the Room List module.

Figure 6-3-22 Room List/Wake Up

6.3.3 Configuration

Billing Settings

When the customer needs to use the extension to make an outside call, the service charge for the
call is calculated based on the set call rate.

Billing Rates

If there is no matching rate list, the default rate will be used.

After initialized the rates, you can create a new billing rate or edit the existing rate.

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Figure 6-3-23 Billing Rates

Note: Only trunks allowed by Billing Trunk can be displayed in the tariff list.

Then create a new billing Rates:

Figure 6-3-24 Add/Edit Billing Rates

Billing Setup

You should initialize the billing rates before setting billing rules:

Figure 6-3-25 Billing Setup

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Default Rate and Default Rate Offset set the rate of the Default billing rate. Billing Trunk sets
the billing trunks allowed when creating new billing rates.

Room Setting

Room Setup

Once we have created the room type, we can generate the hotel room. Don't forget, try to prepare a
good list of names for each room. (e.g.: room 100, room 101.etc). This name will be use by Hotel
if no name is entered.

Click and create room.

Figure 6-3-26 Add Room

Besides, you can click to batch create rooms.

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Figure 6-3-27 Batch initialization of rooms

Room Type

Room Type display all types already recorded into Hotel configuration.

Before to add any room, you must create some room type to putting them on each room. You can
create hotel room types, such as common Standard Rooms, Double Rooms, Business Room,
King Rooms, etc.

You could delete a room or more just selecting the checkbox at left of row.

Figure 6-3-28 Room Types

Just putting a type with its price, enter a price to additional guest if you want, and select the
V.A.T used by this room. (2 V.A.T. are enabled).

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Figure 6-3-29 Room Types options

General

Before we begin, we should initialize the configuration of the hotel system, which includes
customizing the company logo, company information, configuring emails, etc.

Here, you could select 2 operating mode (Hotel and Hospital). Now, only one operating mode
is enabled.

Figure 6-3-30 General/Operating

Mode You could select 3 basic functions in Hotel.

• Locked when checkout. When the room will be billed, this room will be locked. So impossible to
calling a number.

• Calling between rooms: When checked, the room is able to call another room, but only if
this room is included into the same group as the called room.

• Room must be clean: The room appear into the list of available room only if the room is cleaned.
Else, the room will not appear into this list. However, this room could appear if you need to make a
booking about this room.

Figure 6-3-31 General/Functions

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You can customize your company header, like the logo (png, or jpg file extension), the
company address, and the professional mail of company.

Figure 6-3-32 General/Company

Note: Mail is used to send Booking and Check-out reminders, and you need to configure the SMTP
service in System->Email before using it.

Figure 6-3-33 SMTP settings

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You can customize or change the prefix of each hotel function. 3 Prefix exist right now.

• Mini-bar, is able to add some drinks on the room, and will used during the billing. When the
chambermaid will clean the room, she could check the mini-bar and enter all drink used by the guest.
• Room Clean Prefix will used when the room will cleaned by the chambermaid.
• Reception is here to giving a phone number to the reception..

Figure 6-3-34 General/Hotel Dial Plan

Two tax values can be entered. The first value is used by the outbound calls during the billing.

Figure 6-3-35 General/Tax

Figure 6-3-36 General/Discount

Mini Bar

Set up the items and VAT in the Mini-Bar. The waiter can dial to record the items purchased by the
customer, dial the prefix (*37 by default) and press the number of the product used, ending with the
* key. If you do not press the * key, the purchase will not be recorded. For example: if the
customer has purchased three copies of Sprite, use the room’ extension to dial *37222*

This menu affecting a product on each key with its price. You can enter 10 different products on
this module. 2 V.A.T can be selected.

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Figure 6-3-37 Mini Bar

If a customer has purchased an item, we can check it in the room list.

Figure 6-3-38 Room List/Mini Bar

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Remote Action Control

Figure 6-3-39 Remote Action Control

Booking Email Template

Figure 6-3-40 Booking Email Template

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6.3.4 Report

Billing Report

You can check customer consumption in the Billing Report after checking out.

Figure 6-3-41 Billing Report list

Click you can view the billing report, and click can export this report.

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Figure 6-3-42 Billing Report

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Call Billing Report

After the external call is over, the call record will be displayed in Hotel->Report->Call
Billing Report.

Figure 6-3-43 Call Billing Report list

Company Report

You can realize some company report, like how many checks in and checkout by day between two dates.
Type of report include Check-in and Check-out info, Sum Rooms, mini-bar, calls, and billings.

Figure 6-3-44 Company Report option

You can realize some company report, like how many check-in and checkout by day between
two dates.

Figure 6-3-45 Company Report

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7 Logs

7.1 Logs Settings

Figure 7-1-1 Logs Settings interface

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7.2 System Logs

Figure 7-2-1 System Logs interface

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7.3 Asterisk Logs

Figure 7-3-1 Asterisk logs interface

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7.4 DAHDI Logs

Figure 7-4-1 DAHDI logs interface

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7.5 FXO Monitor Logs

Figure 7-5-1 FXO Monitor logs interface

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7.6 VPN Logs

Figure 7-6-1 VPN logs interface

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8 Me Bar
As mentioned in 2.4 User Permission, all SIP extensions will be given Me module permissions by
default. Enter Me Bar interface requires login with extension account. On the login page of IPPBX,
enter the Extension/User Password and click login. Note that the extension login uses the User
Password, not the Registration Password. After the extension is logged in, as shown below

Figure 8-1 Me Bar interface

If the extension has set permissions in System>User Permission, the corresponding module will
appear in the menu after the extension logs in. You can set the extension permissions flexibly
and reduce the burden of the UC administrator. For details, see 2.4 User Permission.

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8.1 Extension Number

Most of the parameters on this page can also be configured under the Administrator's
management page (PBX>Extension>Edit).

You can set the basic information of the extension.

Figure 8-1-1 User Information

You can also set the Voicemail feature of the current extension:

Figure 8-1-2 Voicemails

You can also set the Call Forwarding feature for the current phone, click the drop-down list to select
Voicemails/Extension:

Figure 8-1-3 Call Forwarding

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You can also set whether to enable the Mobility Extension feature (i.e. mobile phone number) for
the current extension.

Figure 8-1-4 Mobility Extension

You can also set other functions of this extension.

Figure 8-1-5 Others

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8.2 Call and Recordings

You can view the call records and recordings related to the current extension

Figure 8-2-1 CDR & Records

When the extension is given permission to download CDRs, button will appear,
which can generate call detail records and you can download them on the Downloads page.

When the extension is given permission to delete the CDR, the button will appear.
You can click the call log in the check box and click the button to delete it.

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8.3 Voicemail
You can also check the voice messages of the changed extension

Figure 8-3-1 VoiceMail interface

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8.4 Password Settings

You can reset the login password of the extension

Figure 8-4-1 Password setting

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8.5 Black List

The Black List module is the same as the blacklist function under the admin account. You can add
a phone number to a blacklist or remove a phone number from a blacklist. You can also choose to
blacklist any blocked or unknown calls.

When a number is blacklisted, any calls with that number in the Caller ID field received by
the system will be routed to the disconnected record.

Figure 8-5-1 Black List

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8.6 White List

The White List module is the same as the whitelist function under the admin account.

Figure 8-6-1 White List

If enabled White Only, the incoming call will be limited. For example, if you add a 1000 in
whitelist, and the type is Inbound, then only 1000 can dial in and reach this extension.

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8.7 Downloads
The call records generated on the CDR page or the CDR & Records page of the Me Bar can
be downloaded on the Downloads page.

If the extension is given permission to view and download, the download content of the
specified extension can also be viewed on the page. For details, see 2.4 User Permission.

Figure 8-7-1 Downloads

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9 Web Phone
If you have enabled Web Phone in PBX > Extensions > Extensions module under the admin
account, you can enter the WebPhone module which supports all the functions of Me Bar, and use
it to make calls directly. It should be noted that, since the underlying transmission of the VoIP uses
the WSS protocol, this means that you cannot use WebPhone and other phones at the same time.

Recommend using Chrome browser.

Figure 9-1 WebPhone page

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9.1 Web Dialing

When entering the module for the first time, the extension is not registered and the dial cannot be

used. You need to slide the switch to turn it on. After it is turned on, the extension's
transmission protocol will automatically become wss and be registered.

Figure 9-1-1 Dial

You can tap the dial pad on the page to dial, or input the number you want to dial, and then
tap to initiate the call. If the browser asks whether to enable the microphone, please allow it.

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9.2 Contacts
Contacts can be understood as a phone book, and you can add frequently used contacts to this phone
book to achieve speed dialing. The added contact is only visible to the current extension.

Figure 9-2-1 Contacts

Click to add a new contact, "Contact Name" and "Phone Number" options are required.

Figure 9-2-2 Add Contact

After checking the box, click to delete contacts in batches. Of course, you can also click

to delete this contact.

Click to directly dial the contact without entering the number.

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Click to edit this contact.

In this interface, you can also search for a contact, enter his/her name in the input box, or enter the

phone number, and click the button. If the contact is in the "Phonebook", the contact will be
displayed, otherwise it will prompt "No records match the filter criteria".
The Extension tab will display all extensions in the IPPBX system, and will display the status of the
extensions (Idle, Offline, Busy).

Click to directly dial the currently online extension.

Figure 9-2-3 Extension

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9.3 Settings
This setting page is basically the same as the Extension page in Me Bar.

Figure 9-3-1 Settings

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9.4 CDR
You can view the call details records and related recordings of the current extension

Figure 9-4-1 CDR

When the extension is given the permission to download call records, button will
appear, you can generate call records and download them in the download content.

When the extension is given the permission to delete call records, button will appear,
you can select call records and delete.

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9.5 VoiceMails
You can view the VoiceMail of the current extension.

Figure 9-5-1 VoiceMails

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9.6 Password Settings

You can modify the login password of the extension.

Figure 9-6-1 Password Settings

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9.7 Functions Code

The function code page lists all the function codes that can be used on IPPBX. You can quickly
configure/use the basic functions of system according to different function codes. For details, please
refer to PBX>Settings>Function Codes.

Figure 9-7-1 Functions Code

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9.8 Downloads
The call records generated on the CDR page or the CDR & Records page of the Me Bar can be
downloaded on the Downloads page.

If the extension is given permission to view and download, the download content of the
specified extension can also be viewed on the page. For details, see 2.4 User Permission.

Figure 9-8-1 Downloads

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