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UniFi VoIP - FreeSWITCH: SIP Configuration

Overview


Readers will learn how to configure a SIP account in FreeSWITCH, and configure SIP settings in the UVP.

 

Creating an extension for the UVP on FreeSWITCH


The files containing the extensions in FreeSWITCH are normally in the /etc/freeswitch/directory/default directory. As in the previous example, to add extension 100 to FreeSWITCH you'd have to create a file named 100.xml with the following contents:

 

<include>
  <user id=“100">
	<params>
  	<param name="password" value=“password"/>
  	<param name="vm-enabled" value="false”/>
	</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 dialplan, that can usually be found in the /etc/freeswitch/dialplan/default.xml file, 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 UVP you will be ready to make and receive calls with it.

 

Configuring the UVP with SIP Settings


If you have not already followed the Initial Configuration steps in this article Standalone UniFi VoIP Phone Configuration Guide please complete these before continuing.  If you have any questions about the following settings or what they mean please refer to the article above in the SIP Configuration section. 

 

Minimum FreeSWITCH SIP Configuration Requirements:

Server

User name

Password

 

Enter the SIP settings that you configured in FreeSWITCH above.

Server = Server IP(or hostname) for FreeSWITCH server

User name = SIP Username

Password = SIP Password

 

Navigate back to the UVP App home screen and it should be connected and allow you to call if your FreeSWITCH server is set up for outbound calls(SIP, IAX, PRI, etc)