Setting up a SIP Trunk in FreeSWITCH

This guide provides step-by-step instructions on how to add a SIP trunk in FreeSWITCH (also known as a gateway in FreeSWITCH terminology).

Prerequisites

Make sure you have the following:

• an installed and running FreeSWITCH server
• a SIP account from CommPeak

The SIP credentials and domain details that you need are available in your account at my.commpeak.com in Setup -> SIP Accounts:

• User - click SHOW under Username/IP in a SIP account line
• Password - click Reset Password if you forgot it 
• Domain - sip.commpeak.com

Adding a SIP Trunk

Step 1: Access FreeSWITCH

First, you need to access your FreeSWITCH server. This can usually be done via SSH. Run the command similar to the following:

ssh user@your_freeswitch_server_ip

Step 2: Locate SIP Profile Directory

FreeSWITCH configurations are stored in XML files located under /etc/freeswitch. Navigate to the sip_profiles directory, which contains files for external SIP profiles.

cd /etc/freeswitch/sip_profiles/external

Step 3: Create SIP Gateway Configuration File

1. Create a new XML file in this directory for your SIP gateway. You can give it any name, e.g., my_sip_gateway.xml.
2. Open the new file using a text editor. Here we're using nano, but you can use your preferred editor:

sudo nano my_sip_gateway.xml

Step 4: Edit SIP Gateway Configuration File

1. In the newly created my_sip_gateway.xml file, include the following configuration:

   <include>
   <gateway name="my_sip_gateway">
     <param name="username" value="yourusername"/>
     <param name="password" value="yoursecret"/>
     <param name="proxy" value="sip.commpeak.com"/>
     <param name="register" value="true"/>
   </gateway>
   </include>

2. Enter the following data in this configuration:

   • my_sip_gateway is the name of your SIP gateway 
   • yourusername is your SIP username 
   • yoursecret is your SIP password 
   • sip.commpeak.com is your SIP domain
3. Save and exit the file.

Step 5: Reload SIP Profile

To apply the changes, you must reload the SIP profile in FreeSWITCH.

1. Access the FreeSWITCH CLI:

   fs_cli

2. Then reload the SIP profile:

   reloadxml
   sofia profile external rescan

Step 6: Verify SIP Gateway Registration

After reloading, you can verify if the gateway registration was successful by checking the SIP gateways:

sofia status gateway my_sip_gateway

The status should be REGED if the gateway is registered correctly.

Step 7: Configure Dialplan

You'll need to set up your dialplan to use this SIP Gateway. The dialplan in FreeSWITCH controls how incoming and outgoing calls are handled and routed.

1. Navigate to the dialplan directory:

   cd /etc/freeswitch/dialplan

2. Open the default.xml file or any specific dialplan file you're using:

   sudo nano default.xml

3. Add a new extension to route calls to your new SIP gateway. An XML example might look as follows:

   <extension name="outbound_to_my_sip_gateway">
    <condition field="destination_number" expression="^(\d+)$">
      <action application="bridge" data="sofia/gateway/my_sip_gateway/$1"/>
    </condition>
   </extension>

   This example contains the following data:

   • outbound_to_my_sip_gateway: the extension name 
   • condition: the field that checks the destination number
   • ^(\d+)$: the expression matches any number
   • bridge: the action used to connect the call to the SIP gateway
   • my_sip_gateway: the name of your gateway
   • $1: represents the captured group from the expression (i.e., the dialed number).
4. Save and exit the file.

Step 8: Reload Dialplan

Now you need to reload the dialplan for the changes to take effect. In the FreeSWITCH CLI, execute:

reloadxml

Step 9: Test the SIP Gateway

Finally, you can make a test call to verify everything works correctly. You can do this from a registered SIP device (like a softphone or IP phone) configured to use your FreeSWITCH server.

Troubleshooting

If you encounter any issues, you can use the FreeSWITCH CLI for troubleshooting by executing the following command:

sofia loglevel all 9

It will provide detailed information about SIP communication and allow you to identify any issues.