FreePBX 17 SIP Trunk Configuration

Overview

This guide walks you through configuring a Simtex SIP trunk on FreePBX 17 using the PJSIP (chan_pjsip) channel driver. FreePBX 17 is built on Asterisk 21 and ships with PJSIP as the default SIP stack — the legacy chan_sip driver is no longer available.

FreePBX is the most widely deployed open-source PBX platform, offering a full-featured web GUI for Asterisk — freepbx.org

What you'll need

• Your Simtex account number (e.g. 214XXXXXXX)
• Your Simtex SIP password
• Your allocated DIDs (phone numbers)
• FreePBX 17 installed with web GUI access

Step 1 — Enable TCP Transport

Before creating the trunk, ensure TCP transport is enabled in FreePBX. By default, only UDP is active.

1. Navigate to Settings → Asterisk SIP Settings
2. Click the SIP Settings [chan_pjsip] tab
3. Scroll down to the Transports section
4. Click Yes next to TCP to enable it
5. Click Submit, then Apply Config

Step 2 — Create the SIP Trunk

1. Navigate to Connectivity → Trunks
2. Click Add Trunk
3. Select Add SIP (chan_pjsip) Trunk from the dropdown

General Settings

On the General tab, configure:

Trunk Name:              Simtex SIP
Outbound CallerID:       (your main DID, e.g. 61894883344)
Maximum Channels:        (your purchased channel count)

Step 3 — PJSIP Settings

Click the PJSIP Settings tab and configure the connection to Simtex:

General Tab

Username:          214XXXXXXX
Secret:            XXXXXXXX
Authentication:    Outbound
Registration:      Send
Language Code:     en
SIP Server:        siptcp.simtex.com.au
SIP Server Port:   0
Transport:         0.0.0.0-tcp

Advanced Tab

Click the Advanced tab and configure caller ID passthrough:

Contact User:      214XXXXXXX  (same as username)
From User:         214XXXXXXX  (same as username)
From Domain:       siptcp.simtex.com.au
Trust RPID/PAI:    Yes
Send RPID/PAI:     Both

These settings ensure your outbound caller ID is correctly transmitted via the RPID and P-Asserted-Identity SIP headers, and that Simtex can pass caller ID information back to you on inbound calls.

Codecs Tab

Click the Codecs tab and reorder the codec priority list. Drag to reorder:

Priority 1:  G.711 A-Law  (alaw)
Priority 2:  G.711 U-Law  (ulaw)

Remove any codecs you don't intend to use — a cleaner codec list speeds up call negotiation.

Click Submit, then Apply Config. The trunk should register within a few seconds — you'll see a green indicator next to the trunk name in the Trunks list.

Step 4 — Outbound Routes

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

1. Navigate to Connectivity → Outbound Routes
2. Click Add Outbound Route
3. Enter a Route Name (e.g. "Simtex Outbound")
4. Under Trunk Sequence, select your Simtex SIP trunk as Route 1
5. Click the Dial Patterns tab to configure what numbers this route handles

Step 5 — Inbound Routes

Inbound routes direct incoming calls on your DIDs to the correct destination (extension, ring group, IVR, queue, etc.).

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

Create a catch-all route first

Leave the DID Number field blank to create a catch-all route. This handles any inbound DID that doesn't have a specific route — point it at your receptionist or main ring group.

Then add DID-specific routes

Create additional inbound routes for individual DIDs or DID ranges that need specific routing:

• DID Number: enter the number in E.164 format (e.g. 61894883344)
• Set Destination: choose the target extension, ring group, IVR, or queue

You can use wildcard patterns for blocks of numbers, e.g. _89488334X to match a range.

Step 6 — Security Hardening

After configuring your trunk, take these essential steps to secure your FreePBX installation:

Disable SIP Guest

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

Additional security measures

• Strong extension passwords — FreePBX auto-generates these; do not simplify them
• Responsive Firewall — enable FreePBX's built-in responsive firewall module under Connectivity → Firewall
• Intrusion Detection — enable fail2ban integration to automatically block brute-force attacks
• Keep FreePBX updated — run Admin → Module Admin → Check Online regularly for security patches
• Channel limits — keep Maximum Channels on the trunk set to your actual concurrent call requirement

Testing Your Trunk

Once configured, verify everything works:

1. Check trunk status — navigate to Connectivity → Trunks and confirm the trunk shows a green indicator
2. Make an outbound call — dial an external number from an extension and confirm two-way audio
3. Receive an inbound call — call one of your DIDs from a mobile and confirm it routes correctly
4. Verify caller ID — check your outbound caller ID displays correctly on the receiving end
5. Check voicemail — leave a voicemail to ensure DTMF tones are working correctly through the trunk

Extension-Level Caller ID

By default, outbound calls present the Outbound CallerID set on the trunk. To override per extension:

1. Navigate to Applications → Extensions
2. Edit the extension
3. Set Outbound CID to the DID you want that extension to present in E.164 format (e.g. 61894883344)

The DID must be allocated to your Simtex account — you cannot present arbitrary numbers.

Need Help?

If you run into any issues configuring your FreePBX trunk, our support team can verify your trunk registration status from our side and assist with troubleshooting.

Contact Simtex Support