Home/FreePBX Version 15 PJSIP Trunk Configuration

FreePBX Version 15 PJSIP Trunk Configuration

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Overview

This is a step-by-step guide to configuring a Simtex SIP trunk on FreePBX 15 using the PJSIP (chan_pjsip) channel driver. FreePBX is a widely used, feature-rich graphical interface for Asterisk — freepbx.org

This guide assumes you already have FreePBX installed with the web GUI accessible. If not, download the latest 64-bit ISO from the FreePBX Downloads page.

What you'll need

  • Your Simtex account number (e.g. 214XXXXXXX)
  • Your Simtex SIP password
  • FreePBX 15 installed and accessible
SIP Servers:
  • West Coast (AU): siptcp.simtex.com.au
  • East Coast (AU): siptcpeast.simtex.com.au

Choose the server closest to your location. FreePBX supports SRV record resolution — set the port to 0 and it will automatically discover our geo-redundant infrastructure.

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. If you are unsure, speak to us first.

Step 1 — Enable TCP Transport

Before creating the trunk, you need to enable TCP transport in FreePBX.

  1. Go to Settings → Asterisk SIP Settings → SIP Settings [chan_pjsip] tab
  2. Scroll down to Transports
  3. Click Yes on TCP
  4. Submit and Apply Config

FreePBX enable TCP transport

Step 2 — Create the SIP Trunk

  1. Open the Connectivity menu and select Trunks

FreePBX Connectivity menu - select Trunks

  1. Select Add SIP (chan_pjsip) Trunk

FreePBX Add SIP trunk dropdown

  1. Label your SIP trunk and specify the number of channels

FreePBX SIP trunk general settings

For security, limit the channel count to the number you'll actually need day-to-day. This prevents potential abuse if your system is compromised.

Step 3 — PJSIP Settings

Click the PJSIP Settings tab and configure:

FreePBX PJSIP Settings tab

Username:        214XXXXXXX
Secret:          XXXXXXXX
SIP Server:      siptcp.simtex.com.au
SIP Server Port: 0
Transport:       0.0.0.0-tcp
Port 0 = automatic discovery. FreePBX's PJSIP driver resolves SRV records when the port is set to 0, giving you automatic failover across our geo-redundant server farms. Do not set port 5062 here — let SRV handle it.
East Coast? Replace siptcp.simtex.com.au with siptcpeast.simtex.com.au in the SIP Server field if your server is located on the East Coast of Australia.

Step 4 — Advanced Settings

Click the Advanced tab and configure:

FreePBX PJSIP trunk advanced settings

Contact User:    214XXXXXXX (same as username)
From User:       214XXXXXXX (same as username)
Trust RPID/PAI:  Yes
Send RPID/PAI:   Both

FreePBX PJSIP advanced settings - RPID/PAI

Codec Settings

Click the Codec tab and reorder codecs to have alaw first and ulaw second:

FreePBX codec priority settings

Step 5 — Outbound Routes

Outbound routes tell FreePBX what numbers it's allowed to dial externally and which trunk to use.

Prefix tip: Configure a prefix (e.g. 0) for all external calls. This ensures internal extension numbers never overlap with external destinations — a standard practice in Australia.
If you do not wish to make international calls, leave out the 0011 route pattern to prevent unauthorised international dialling.
  1. Navigate to Connectivity → Outbound Routes → Add Outbound Route

FreePBX Outbound Routes menu

  1. Label your route and select the SIP trunk you created

FreePBX Outbound Route - trunk selection

  1. Click Dial Patterns and configure your routes. The example below accepts a leading 0, strips it off, then sends the remaining digits to the Simtex trunk.

Step 6 — Inbound Routes

Inbound routes direct incoming calls to your allocated DIDs. You can route individual numbers to different extensions, ring groups, IVR menus, etc.

  1. Navigate to Connectivity → Inbound Routes → Add Incoming Route

FreePBX Inbound Routes menu

  1. Create a catch-all route first — this handles any DID sent to your system that doesn't have a specific route. Most users point this at a receptionist or main hunt group.

FreePBX catch-all inbound route

  1. Add entries for individual numbers or blocks of numbers. You can use wildcard patterns such as _618921133XX

FreePBX DID-specific inbound route

When using wildcard patterns, you must prefix with an underscore: _618921133XX

Step 7 — Disable SIP Guest

As a security hardening step, disable SIP guest access to prevent unauthorised calls:

  1. Go to Settings → Asterisk SIP Settings
  2. Under General SIP Settings, find Allow SIP Guests
  3. Set to No

FreePBX disable SIP guest access

Always disable SIP Guests on production systems. Leaving this enabled is one of the most common causes of toll fraud on FreePBX installations.
Last updated 26 March 2026