Setting Up a SIP Trunk in FreeSWITCH

This guide provides step-by-step instructions on how to add a SIP trunk in FreeSWITCH.

After signing up with CommPeak and getting your SIP account credentials, your next step is registering your SIP account in CommPeak Softphone, a third-party application, or with your IP phone to make your first call.

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TIP
You can also use the Built-in CommPeak Softphone with your JWT-enabled SIP accounts to place calls right from your browser. You can read more in the Built-in Softphone for JWT-Enabled SIP Accounts article.

Video tutorial

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TIP
With CommPeak Softphone, you get an enterprise-level tool without installing any third-party software.

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IMPORTANT
If you're using third-party SIP clients (not CommPeak's Softphone), you must manually enable SRTP (Secure Real-Time Transport Protocol) when using TLS for SIP transport. This ensures media encryption and compatibility.

This document will guide you 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 CommPeak Portal in VoIP Services > SIP Accounts (Trunks):
User - click the eye icon next tor Username/IP in a SIP account line.
Password - click Reset Password if you forgot it.
Domain - sip.commpeak.com.

SIP account data in CommPeak Portal

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

Create the gateway file

Create a new XML file in this directory for your SIP gateway. You can give it any name, e.g., commpeak_gateway.xml.

Open the file

Open the new file using a text editor. Here we're using nano, but you can use your preferred editor:

sudo nano commpeak_gateway.xml

Step 4: Edit SIP gateway configuration file

Add the configuration

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

1. <include><gateway name="commpeak_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>

Enter your data

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.

Save and exit

Save and exit the file.

Step 5: Reload SIP profile

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

Open the FreeSWITCH CLI

Access the FreeSWITCH CLI:

fs_cli

Reload the SIP profile

Then reload the SIP profile:

reloadxmlsofia 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 commpeak_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.

Open the dialplan directory

Navigate to the dialplan directory:

cd /etc/freeswitch/dialplan

Open the dialplan file

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

sudo nano default.xml

Add an extension

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

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

This example contains the following data:

outbound_to_commpeak_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
commpeak_gateway: the name of your gateway
$1: represents the captured group from the expression (i.e., the dialed number).

Save and exit

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.

Updated 6 days ago

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