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UniFi UVP
Quick Start Manual
24 pgs10.1 Mb0
User Manual
36 pgs6.9 Mb0

Ubiquiti UniFi UVP, UniFi UVP-Pro, UniFi UVP-Executive User Manual

Quick Provisioning Guide for Third-Party PBX
Table of Contents Quick Provisioning Guide
Table of Contents
Chapter 1: Overview ................................................1
Chapter 2: Asterisk Configuration ...................................2
Creating a Phone Extension on Asterisk ...........................................2
Configuring the Phone’s SIP Settings ..............................................2
Configuring Calls Between Phones ................................................4
Chapter 3: FreeSwitch Configuration ................................6
Creating a Phone Extension on FreeSWITCH .......................................6
Configuring the Phone’s SIP Settings ..............................................6
Chapter 4: 3CX Configuration .......................................9
Creating a Phone Extension on 3CX ...............................................9
Chapter 5: Elastix Configuration ...................................13
Creating a Phone Extension on Elastix ............................................13
Configuring the Phone’s SIP Settings .............................................16
Chapter 6: FreePBX Configuration .................................17
Creating a Phone Extension on FreePBX ..........................................17
Configuring the Phone’s SIP Settings .............................................20
Chapter 7: FusionPBX Configuration ...............................21
Creating a Phone Extension on FusionPBX .......................................21
Configuring the Phone’s SIP Settings .............................................25
Chapter 8: General Phone Configuration ...........................26
Initial Configuration .............................................................26
Audio Codecs and DTMF Configuration (Optional) ...............................27
Incoming Call Action ............................................................27
Removing an Account (Optional) ................................................28
Appendix A: Important Warning Regarding Emergency Calls .......29
Appendix B: Contact Information ..................................30
Ubiquiti Networks Support ......................................................30
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Chapter 1: OverviewQuick Provisioning Guide
Chapter 1: Overview
This User Guide provides intructions on how to configure a third-party Private Branch eXchange (PBX) for use with the UniFi® VoIP Phone (models UVP, UVP-Pro, and UVP-Executive). Throughout the rest of this document, UniFi VoIP Phone refers to any and all models of the UniFi VoIP Phone.
This configuration involves two steps:
• Configure the PBX with the extension of each phone.
• Configure each UniFi VoIP Phone’s SIP settings so that it can connect to the PBX.
For detailed instructions, refer to the appropriate section for your PBX:
“Asterisk Configuration” on page 2
“FreeSwitch Configuration” on page 6
“3CX Configuration” on page 9
“Elastix Configuration” on page 13
“FreePBX Configuration” on page 17
“FusionPBX Configuration” on page 21
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Chapter 2: Asterisk Configuration Quick Provisioning Guide
Chapter 2: Asterisk Configuration
This chapter provides detailed instructions on how to configure a UniFi VoIP Phone on the Asterisk® (http://www.asterisk.org/) open‑source PBX. This process, which must be done once for each phone, involves modifying the SIP configuration file for the phone, editing the dial plan file to enable calls to the extension, and configuring the SIP settings on the phone.
Creating a Phone Extension on Asterisk
Each PBX comes with a default configuration that contains a dial plan, extensions, and all initial settings needed. By default, the file /etc/asterisk/sip.conf contains the extensions. To add extension 100, add the following text snippet to this file (bold italic text indicates user‑specified values):
[sip-temp](!)
type=friend
host=dynamic
disallow=all
allow=ulaw
qualify=1000
canreinvite=no
nat=force_rport
dtmfmode=rfc2833
context=from-internal
[100](sip-temp)
username=100
callerid=Your Name <100>
secret=password
dial=SIP/100
To be able to call this extension, you need to hook it up to the corresponding dial plan (found in file /etc/asterisk/extensions.conf by default). To do so, create the context from-internal that is specified as the outbound context for the SIP extension. For extension 100, the lines to be added are:
[from-internal]
exten => 100,1,Dial(SIP/${EXTEN}|40|Ttr)
Configuring the Phone’s SIP Settings
Before you can configure the UniFi VoIP Phone’s SIP settings, perform initial configuration on the phone by following the instructions in “Initial Configuration” on page 26. Then, configure the phone’s SIP account by following these steps:
1. Press the Settings icon at the bottom left of the Welcome screen to display the Phone Settings page.
2. Press SIP service.
3. Press SIP accounts.
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4. Press Add account. The SIP Settings page is displayed.
Chapter 2: Asterisk ConfigurationQuick Provisioning Guide
5. Complete the following SIP account information and press OK:
Field Description
Server SIP server IP address or hostname for registration
IP-based Select this option if you are using IP‑based authentication
User name SIP user ID
Password SIP password
Authentication name (optional) The username used for authentication, if required
Display name (optional) The name that will be displayed to other users as the Caller ID
Display extension (optional) The extension that will be displayed to other users
Voicemail number The extension where the voicemail service can be reached at your PBX
Outbound proxy (optional) The SIP proxy used for outbound calls, if required
The minimum Asterisk SIP configuration requirements are:
• Server Set this to the IP or hostname of your Asterisk server.
• User name Set this to the SIP username.
• Password Set this to the SIP password.
Enter the SIP settings that you configured in Asterisk in “Creating a Phone Extension on Asterisk” on page 2. For the Password field, use the setting of the secret option.
6. The new account will be displayed on the SIP accounts page.
If the new account was properly configured, Receiving calls will be displayed.
If it was not properly configured, Not connected to server (error: 408) will be displayed.
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Chapter 2: Asterisk Configuration Quick Provisioning Guide
If configuration was successful:
• The Asterisk log will show the following to indicate that the UniFi VoIP Phone has
registered:
-- Registered SIP ‘100’ at XX.XX.XX.XX:1024
[Jun 3 13:36:06] NOTICE[4093]: chan_sip.c:23522 handle_response_
peerpoke: Peer ‘100’ is now Reachable. (139ms / 5000ms)
• The UniFi VoIP Phone’s Dialer screen will show the phone as connected, and will allow you
to make calls if your Asterisk server is set up for outbound calls (SIP, IAX, PRI, etc.).
Configuring Calls Between Phones
To enable calls between UniFi VoIP Phones (extensions 100 and 101 in this example), first add the following lines to the sip.conf file (bold italic text indicates user‑specified
values):
[sip-temp](!)
type=friend
host=dynamic
disallow=all
allow=ulaw
qualify=1000
canreinvite=no
nat=force_rport
dtmfmode=rfc2833
context=internal-calls
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Chapter 2: Asterisk ConfigurationQuick Provisioning Guide
[100](sip-temp)
username=100
callerid=Your Name <100>
secret=Password
[101](sip-temp)
username=101
callerid=Your Name <101>
secret=Password
Then, add the following lines to the /etc/asterisk/extensions.conf file:
[100-in]
exten => s,1,Dial(SIP/100,30)
exten => s,2,Hangup()
[101-in]
exten => s,1,Dial(SIP/101,30)
exten => s,2,Hangup()
[internal-calls]
exten => 100,1,GoTo(100-in,s,1) ;Extension 100
exten => 101,1,GoTo(101-in,s,1) ;Extension 101
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Chapter 3: FreeSwitch Configuration Quick Provisioning Guide
Chapter 3: FreeSwitch Configuration
This chapter provides detailed instructions on how to configure a UniFi VoIP Phone on the FreeSWITCH® (http://www.freeswitch.org/) open‑source PBX. This process, which must be done once for each phone, and involves creating a unique configuration file for the phone, editing the dial plan file to enable calls to the extension, and configuring the SIP settings on the phone.
Creating a Phone Extension on FreeSWITCH
Each PBX comes with a default configuration that contains a dial plan, extensions and all initial settings needed. By default, the files containing the extensions in FreeSWITCH are in the /etc/freeswitch/directory/default directory. To add extension 100 to FreeSWITCH, create a file named 100.xml with the following contents (bold italic text indicates user‑specified values):
<include>
<user id=“100”>
<params>
<param name=”password” value=“password”/>
<param name=”vm-enabled” value=”true”/>
<param name=“vm-password” value=“8761”/>
</params>
<variables>
<variable name=”user_context” value=”default”/>
<variable name=”effective_caller_id_name” value=“Your Name”/>
<variable name=”effective_caller_id_number” value=“100”/>
</variables>
</user>
</include>
To be able to call this extension, you need to edit the FreeSWITCH dial plan (found in file
/etc/freeswitch/dial plan/default.xml by default) and add the following
contents:
<extension name=”100”>
<condition field=”destination_number” expression=”^100$”>
<action application=”bridge” data=“user/100”/>
</condition>
</extension>
Once you have completed these steps and configured your UniFi VoIP Phone, you will be ready to make and receive calls with it.
Configuring the Phone’s SIP Settings
Before you can configure the UniFi VoIP Phone’s SIP settings, perform initial configuration on the phone by following the instructions in “Initial Configuration” on page 26. Then, configure the phone’s SIP account by following these steps:
1. Press the Settings icon at the bottom left of the Welcome screen to display the Phone Settings page.
2. Press SIP service.
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3. Press SIP accounts.
4. Press Add account. The SIP Settings page is displayed.
Chapter 3: FreeSwitch ConfigurationQuick Provisioning Guide
5. Complete the following SIP account information and press OK:
Field Description
Server SIP server IP address or hostname for registration
IP-based Select this option if you are using IP‑based authentication
User name SIP user ID (ask your PBX administrator)
Password SIP password (ask your PBX administrator)
Authentication name (optional) The username used for authentication, if required
Display name (optional) The name that will be displayed to other users as the CallerID
Display extension (optional) The extension that will be displayed to other users
Voicemail number The extension where the voicemail service can be reached at your PBX
Outbound proxy (optional) The SIP proxy used for outbound calls, if required
The minimum FreeSWITCH SIP configuration requirements are:
• Server Set this to the IP or hostname of your FreeSWITCH server.
• User name Set this to the SIP Username.
• Password Set this to the SIP Password (secret above).
Enter the SIP settings that you configured in FreeSWITCH in “Creating a Phone Extension on FreeSWITCH” on page 6.
6. The new account will be displayed on the SIP accounts page.
If the new account was properly configured, Receiving calls will be displayed.
If it was not properly configured, Not connected to server (error: 408) will be displayed.
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