🔧 FreePBX SIP Trunk Setup: Complete Configuration Guide
Summary: This comprehensive guide walks you through configuring a SIP trunk in FreePBX – from choosing the right SIP driver (PJSIP vs chan_sip) to setting up registration, inbound/outbound routing, and troubleshooting common issues. Whether you're connecting to a major provider or a custom SIP carrier, this step-by-step tutorial ensures a smooth setup for your business phone system.
📖 Table of Contents
- 1. Understanding FreePBX SIP Trunks
- 2. PJSIP vs chan_sip: Which Driver to Use?
- 3. Prerequisites & Preparation
- 4. Step-by-Step SIP Trunk Configuration
- 5. Configuring Inbound Routes
- 6. Configuring Outbound Routes
- 7. Testing Your SIP Trunk
- 8. Common Issues & Troubleshooting
- 9. Best Practices
- 10. Frequently Asked Questions
- 11. Related Resources
🔍 Understanding FreePBX SIP Trunks
A SIP trunk in FreePBX is a virtual connection that links your PBX to external networks, such as a telephone service provider or another PBX system[reference:0]. It allows you to make and receive calls over the internet using the Session Initiation Protocol (SIP). FreePBX supports two primary SIP channel drivers: chan_sip (the legacy driver) and chan_pjsip (the modern, recommended driver)[reference:1].
🔹 What a SIP Trunk Does
- Outbound Calling: Routes internal extension calls to external phone numbers via your provider.
- Inbound Calling: Receives external calls and routes them to extensions, ring groups, or IVRs.
- DID Support: Handles multiple Direct Inward Dialing (DID) numbers on a single trunk.
- Cost Savings: Replaces traditional PRI or analog lines with affordable VoIP connectivity.
🔹 When to Use a SIP Trunk
- You need to connect your FreePBX to an external VoIP provider.
- You want to eliminate traditional phone lines and reduce costs.
- You require scalability – add or remove channels as your business grows.
- You need advanced features like failover routing and multiple DIDs.
⚖️ PJSIP vs chan_sip: Which Driver to Use?
FreePBX offers two SIP channel drivers. Here's how they compare:
| Feature | PJSIP (chan_pjsip) | chan_sip (Legacy) |
|---|---|---|
| Status | Modern, actively developed | Legacy, being phased out |
| Performance | Better performance, lower latency[reference:2] | Good but less efficient |
| NAT Handling | Enhanced NAT traversal[reference:3] | Requires more configuration |
| Multiple Registrations | Improved support[reference:4] | Limited |
| Configuration | More intuitive GUI | Requires manual peer/user details |
| Recommendation | ✅ Recommended for all new setups[reference:5] | Only for legacy compatibility |
💡 Recommendation: For all new FreePBX installations, use PJSIP (chan_pjsip). It offers better performance, security, and future-proofing. The legacy chan_sip driver is still available but is gradually being deprecated[reference:6].
📋 Prerequisites & Preparation
Before you begin, ensure you have the following:
✅ System Requirements
- FreePBX installed and running (version 15 or higher recommended)
- Administrative access to the FreePBX web console[reference:7]
- A SIP trunk account from a provider (e.g., NetviaVoice, Telnyx, Twilio)
- Provider credentials: SIP server, username, password (or IP whitelist)
- At least one DID (Direct Inward Dialing) number assigned
🔍 Network & Firewall
- SIP Port: UDP/TCP 5060 (default) – ensure open[reference:8]
- RTP Ports: 10000-20000 or 20000-27999 (configurable)[reference:9]
- SIP ALG: Disable SIP ALG on your router if present[reference:10]
- NAT: Enable Consistent NAT if behind NAT[reference:11]
- Firewall: FreePBX firewall auto-discovers trunks[reference:12]
⚠️ Important: If your router has a SIP ALG (Application Layer Gateway) option, it is almost always better to TURN IT OFF. SIP ALG often causes more problems than it solves[reference:13].
📝 Step-by-Step SIP Trunk Configuration
Follow these steps to add a SIP trunk to your FreePBX system using the recommended PJSIP driver[reference:14][reference:15]:
- 1 Log in to FreePBX Admin – Open your FreePBX web interface and log in with administrator credentials[reference:16].
- 2 Navigate to Trunks – Go to Connectivity → Trunks from the top menu[reference:17].
- 3 Add a New Trunk – Click Add Trunk and select Add SIP (chan_pjsip) Trunk[reference:18][reference:19].
- 4 General Settings – On the General tab:
- Trunk Name: Enter a descriptive name (e.g., "NetviaVoice-Primary")[reference:20].
- Outbound Caller ID: Set your business caller ID number[reference:21].
- Maximum Channels: Leave blank for unlimited (or set a limit)[reference:22].
- 5 PJSIP Settings – On the pjsip Settings tab:
- Username: Your SIP provider username[reference:23].
- Secret: Your SIP provider password[reference:24].
- SIP Server: Your provider's SIP domain or IP address[reference:25].
- Context: Set to
from-internalfor internal-origin calls[reference:26]. - Authentication: Set to "Outbound"[reference:27].
- Registration: Set to "Send" to enable registration[reference:28].
- Port: Default is 5060[reference:29].
- 6 Codecs – Scroll down and allow ulaw and alaw codecs (at minimum)[reference:30].
- 7 Submit & Apply – Click Submit, then Apply Config to save changes[reference:31].
💡 IP-Based Trunks: If your provider uses IP-based authentication (whitelisting), set Registration to "None" and ensure your public IP is whitelisted[reference:32].
📞 Configuring Inbound Routes
Inbound routes tell FreePBX what to do when a call comes in on your DID number[reference:33]:
- 1 Go to Connectivity → Inbound Routes[reference:34].
- 2 Click Add Inbound Route[reference:35].
- 3 Under General:
- Description: A descriptive name (e.g., "Main Line")[reference:36].
- DID Number: Your 10-digit incoming number[reference:37].
- Set Destination: Choose where to route calls – extension, ring group, IVR, or queue[reference:38].
- 4 Click Submit, then Apply Config[reference:39].
⚠️ Note: If your DID is set to "Server" in your provider's portal, create the inbound route using the DID in E.164 format (e.g., 12125551234)[reference:40]. If set to "Device", use the SIP endpoint username[reference:41].
📤 Configuring Outbound Routes
Outbound routes define how calls from extensions are routed to the outside world via your SIP trunk[reference:42]:
- 1 Go to Connectivity → Outbound Routes[reference:43].
- 2 Click Add Outbound Route[reference:44].
- 3 Under Route Settings:
- Route Name: A descriptive name (e.g., "Default Route")[reference:45].
- CID: Your outbound caller ID (must match a number on the trunk)[reference:46].
- 4 Trunk Sequence: Select your newly created trunk[reference:47].
- 5 Dial Patterns: Add patterns for numbers to route:
1NXXNXXXXXX– US 10-digit numbers[reference:48]NXXNXXXXXX– 7-digit local numbers[reference:49][3-9]11– Emergency/services (911, 411)[reference:50]011.– International calls (optional)[reference:51]
- 6 Click Submit, then Apply Config[reference:52].
🧪 Testing Your SIP Trunk
After configuration, test your trunk to ensure everything works[reference:53]:
📞 Inbound Test
- Dial your DID number from an external phone (mobile or landline).
- Check if the call reaches the destination you set in the inbound route.
- Verify caller ID display and audio quality.
📤 Outbound Test
- Dial an external number from any extension.
- Confirm the call connects and audio works both ways.
- Check that the correct outbound route is applied.
💡 Pro Tip: Check Reports → Asterisk Log or /var/log/asterisk/full for detailed call logs and error messages if something isn't working.
🚀 Need Expert Help with FreePBX? NetviaVoice Can Assist
Our VoIP specialists can help you configure SIP trunks, optimize your FreePBX system, and troubleshoot any issues – ensuring a reliable and cost-effective phone system.
Visit NetviaVoice🔧 Common Issues & Troubleshooting
| Issue | Possible Cause | Solution |
|---|---|---|
| Trunk not registering | Incorrect username/password, wrong SIP server, firewall blocking port 5060 | Double-check credentials. Verify SIP server domain. Ensure port 5060 is open[reference:54]. |
| Inbound calls not working | DID not configured correctly, inbound route missing or misconfigured | Verify DID number in trunk and inbound route. Check provider's DID routing[reference:55]. |
| Outbound calls fail | Dial pattern mismatch, outbound route not assigned to trunk | Review dial patterns. Ensure trunk is selected in the outbound route[reference:56]. |
| Audio issues (one-way or no audio) | Firewall blocking RTP ports, NAT issues | Open RTP ports (10000-20000). Configure NAT settings in Asterisk SIP Settings[reference:57]. |
| Registration failed – authentication error | Incorrect secret or username format | Verify the username and secret exactly as provided by your carrier[reference:58]. |
💡 Quick Troubleshooting: Always check the Asterisk Log (Reports → Asterisk Log or /var/log/asterisk/full) for detailed error messages – this is the best place to start diagnosing issues.
🏆 Best Practices for FreePBX SIP Trunks
🔹 Security
- Use TLS/SIPS for encrypted SIP signaling.
- Enable SRTP for encrypted media.
- Set strong passwords for SIP credentials.
- Disable SIP ALG on your router[reference:59].
- Use failover trunks for redundancy.
🔹 Performance
- Use PJSIP driver for better performance[reference:60].
- Configure codec preferences (ulaw/alaw first).
- Set appropriate RTP port ranges in Asterisk SIP Settings.
- Monitor call quality and adjust as needed.
🔹 Maintenance
- Keep FreePBX modules updated.
- Regularly test failover trunks (if configured).
- Review CDR (Call Detail Records) for billing and analysis.
- Document trunk configuration for future reference.
🔹 Network
- Use a static public IP or reliable DDNS for IP-based trunks.
- Configure QoS to prioritize SIP/RTP traffic.
- Ensure sufficient bandwidth for concurrent calls.
- Consider VPN for secure inter-site trunking.
❓ Frequently Asked Questions
PJSIP is the modern SIP channel driver in FreePBX, offering better performance, enhanced NAT handling, and improved support for multiple registrations. chan_sip is the older driver, still supported but gradually being phased out. PJSIP is recommended for all new FreePBX installations[reference:61].
Log in to FreePBX Admin, go to Connectivity > Trunks, click Add Trunk, and select Add SIP (chan_pjsip) Trunk. Enter the trunk name, username, password, SIP server, and configure authentication settings. Then create inbound and outbound routes to complete the setup[reference:62][reference:63].
SIP signaling uses port 5060 (UDP/TCP) by default. RTP media ports typically range from 10000-20000 or 20000-27999 (configurable in Asterisk SIP Settings). Ensure these ports are open in your firewall for proper call flow[reference:64].
Common causes include incorrect username/password, wrong SIP server domain, firewall blocking port 5060, or NAT issues. Check the Asterisk log (/var/log/asterisk/full) for detailed error messages. Also verify that your provider's IP is whitelisted if using IP-based authentication[reference:65].
Go to Connectivity > Inbound Routes, click Add Inbound Route. Enter a description, set the DID Number (your incoming number), and choose a destination (extension, ring group, IVR, etc.). Save and apply configuration. Repeat for each DID number you have[reference:66].
📢 Need help setting up your FreePBX SIP trunk? Let NetviaVoice guide you.