A Trunk is the connection between your FreePBX/Asterisk server and the outside world (your SIP provider). If trunks are misconfigured, you’ll face issues like:

• Inbound calls not reaching FreePBX (DID not matching, provider rejected)
• Outbound calls failing (403/401, “All circuits are busy”, or no route)
• One-way audio (NAT/RTP problems)
• Calls rejected due to missing caller ID or wrong From/Contact headers
• “No matching endpoint found” or repeated authentication challenges

This guide explains trunks in a practical and production-safe way, with the exact places you must configure: Registration (optional), Trunk settings, and Identify/Match.

Where to Create Trunks in FreePBX

Go to:

• Connectivity → Trunks
• Click Add Trunk → choose Add SIP (chan_pjsip) Trunk

Always prefer PJSIP trunks for modern Asterisk/FreePBX deployments. chan_sip is legacy and creates more NAT / compatibility issues.

Understand Two Provider Styles: Registration vs IP Authentication

Providers generally use one of these approaches:

1) Registration-based trunk

• Your PBX authenticates using username/password
• PBX sends REGISTER to provider
• Inbound calls typically arrive because provider knows your PBX is “registered”

2) IP-auth trunk (no registration)

• Provider authenticates by your static public IP
• No REGISTER needed
• Inbound calls arrive from provider IPs; you must configure Identify/Match

Your trunk configuration depends heavily on which one your provider uses. If you set “Registration” when the provider expects IP-auth, or vice versa, calls will fail.

Step 1: General Trunk Details (Naming + Type)

Start with clean trunk naming. Use names that match your provider and purpose:

• {{org}}_{{id}}_Primary

Keep trunk names stable. If you rename frequently, outbound routes and troubleshooting become messy.

Step 2: DIAL Number Manipulation Rules

• Before call goes out system check number and transform it as per rules here.
• Setting up rules here helps in configuring business scenarios such as click to call

Step 3: PJSIP Settings (SIP Server / Authentication / From Domain)

In the trunk’s PJSIP settings, you define:

• SIP Server (provider host/IP)
• SIP Server Port (usually 5060 or 5061 for TLS)
• Username / Secret (only if registration/auth required)
• From Domain (some providers require exact From domain)
• Context / Inbound Destination mapping happens later (Inbound Routes), not here

Production notes

• If provider gave an IP, use IP. If provider gave a hostname, ensure DNS works reliably.
• If using TLS (5061), ensure certificates are valid and firewall allows it.
• Do not guess auth fields—match provider docs exactly.

Step 4: Advance SIP Settings

Advance settings decide how SIP headers are thrown to trunk, control RTP behavior and much more

• Call connects but no audio
• DTMF not working in IVR (“press 1 doesn’t register”)
• Call drops quickly due to negotiation failure

In production, keep codecs minimal and aligned with your provider:

• Most stable: PCMU (G.711 u-law) / PCMA (G.711 A-law)
• If provider supports: Opus can work but depends on path (especially with SBCs)

Recommended DTMF

• RFC4733 (aka RTP Events) is the safest for IVR/DTMF

Common Trunk Troubleshooting Patterns

1) Outbound fails immediately

• Wrong username/password
• Wrong SIP server/port
• Outbound route not pointing to this trunk

2) Inbound never reaches FreePBX

• Missing Identify/Match for provider IPs
• DID mismatch in inbound routes
• Firewall blocking SIP from provider IP

3) Call connects but audio is one-way

• NAT/RTP ports not configured correctly
• Firewall not allowing RTP range
• SIP ALG interfering on router

If audio is the issue, check your FreePBX SIP Settings and network/NAT guide: ensure correct External Address, Local Networks, and open RTP ports.

Production Checklist

• Trunk type is PJSIP and named clearly
• Provider host/port correct
• Registration configured only if required
• Codecs kept minimal (PCMU/PCMA) + RFC4733 DTMF
• Identify/Match includes every provider SBC IP
• Firewall allows SIP from trusted provider IPs + RTP range
• Apply Config after every change

Once trunk is stable, the next steps are: Inbound Routes (DID mapping) and Outbound Routes (dial patterns and trunk selection).