Home/FreePBX Version 15 PJSIP Trunk Configuration

FreePBX Version 15 PJSIP Trunk Configuration

This is a step-by-step guide to configure your FreePBX 15 installation with a Simtex SIP trunk.
FreePBX 15 is a widely used, stable and feature-rich graphical user interface for Asterisk – https://www.freepbx.org/

Our guide assumes you have already set the server up, and have the web-based GUI ready to go. If not, grab an ISO image of FreePBX14 from here: FreePBX Distro

Search for the latest ISO within “64 BIT DOWNLOADS” on the FreePBX Download page

Do not deploy FreePBX with an external IP address.

In 99.9% of cases you do not require any ports forwarded on your router or firewall to make FreePBX talk to us.

If you are unsure, speak to us.

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For security reasons, it’s best to limit the quantity of channels to the amount you will actually need in day to day use

Note: By default TCP transport is not enabled, to enable you will need to go to Settings -> Asterisk SIP settings -> then the tab SIP settings [chan_pjsip] and scroll down till you see Transports. Click Yes on TCP

In order to make outbound calls, your system needs to know what it’s allowed to route and where the calls are to be routed. We’ll set these up to guard against people making unauthorised calls on your system.

If you do not wish to make international calls, leave out the 0011 route pattern.

To save yourself from grief later down the track, configure a prefix to route all external calls. In Australia, we use 0 as a standard. This will ensure internal numbers will never overlap external destinations.

Note: If using wildcard patterns, you must use an underscore: _618921133XX

Last updated 25 February 2026