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Asterisk SIP Trunk Guide

In this article


At the end of this article, you will be able to configure a SIP trunk from your Asterisk PBX to the network. Should you need any assistance, feel free to contact our support department.


Configuring a SIP trunk to Asterisk PBX


The first process to getting your Asterisk PBX online is to log into your customer portal, then select the order services tab. From here expand the SIP trunk menu, add the number of channels you require and add a new SIP trunk, as outlined in the screenshot below. Ensure you accept the service terms and conditions then submit the order before continuing. 




Next, click on the PBX tab located in the top menu bar, drag and drop the SIP trunk object onto the main screen from the left-hand toolbox. Now, click on the settings icon and change the mode to create SIP registration.

After that, the SIP trunk details will be displayed as shown below. Please note the "allow override existing contact" checkbox. When enabled, any new SIP trunk registration will replace the old connection, preventing multiple registration attempts. This feature is useful when changes are made often to the SIP trunk
. However, be aware that it might compromise security.




Now you are ready to configure the SIP trunk on your Asterisk PBX. For this, you should edit the sip.conf file using your preferred text editor.




In the sip.conf file, you should use the following code with the information collected from the SIP trunk details, as shown in the image below. Please note that it is important to specify the port in the host field.


[sip trunk]
username=[SIP username]
host=[SIP server address]


You will also need to configure the registration string as shown below.

register => sipusername:sippassword@FQDN:7060

FQDN is SIP server address from the settings of your SIP-trunk, eg



Once you have edited your sip.conf file reload the configuration to force the new changes. The status of the SIP trunk on inside your customer portal should change to reachable. The final process is to test inbound and outbound calls to ensure they are working correctly, should you have issues with calls it is important you check you have configured the inbound and outbound routes correctly on your Asterisk PBX.




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