FreePBX PJSIP Trunk Setup for a Flat-Rate SIP Provider

1. Why PJSIP Instead of chan_sip

FreePBX 15 and later default to PJSIP (chan_pjsip) for a reason: it handles multiple registrations per trunk, supports IPv6, and gives you per-transport TLS with SRTP without patching Asterisk. For outbound-heavy deployments — call centers running 20–200 simultaneous channels per seat — PJSIP's connection pooling under load outperforms chan_sip's single-socket model. If your FreePBX is still on chan_sip, see the companion article on chan_sip vs PJSIP for outbound workloads.

UnlimCall's network accepts both protocols, but PJSIP is the tested path for anything above 10 concurrent calls per trunk.

2. Prerequisites

Before touching FreePBX:

• FreePBX 15 or 16 with Asterisk 18 or 20
• A confirmed UnlimCall account — seat pricing starts at $99/seat/month in the US and Canada (flat, no overage)
• Outbound caller ID numbers provisioned for your target markets — UnlimCall generates them on demand across 33 live markets; there is no pre-bought pool sitting idle
• Your server's public IP allowlisted on the UnlimCall side (IP authentication is the default for trunks; SIP registration is available as a fallback)

3. Create the PJSIP Trunk in FreePBX

Navigate to Connectivity > Trunks > Add Trunk > Add SIP (chan_pjsip) Trunk.

General tab:

• Trunk Name: unlimcall-us (or region-specific)
• Outbound CallerID: leave blank here; set per outbound route or campaign instead

pjsip Settings — General tab:

• Username: your assigned SIP username
• SIP Server: your assigned SIP proxy hostname (do not hardcode an IP; use the DNS name so failover works)
• SIP Server Port: 5060 (UDP) or 5061 (TLS)
• Context: from-trunk
• Transport: 0.0.0.0-udp for standard, or create a TLS transport if your compliance posture requires encrypted signaling

pjsip Settings — Advanced tab:

• From Domain: your SIP proxy hostname (match SIP Server)
• Contact User: your SIP username
• DTMF Mode: RFC 4733 (rfc2833)
• Qualify Frequency: 60 (sends OPTIONS every 60 seconds to verify trunk health)

Codecs tab:

Enable G.711 ulaw first (required for US/CA PSTN compatibility and STIR/SHAKEN pass-through). Add G.711 alaw for international legs. OPUS is supported on the UnlimCall network for compressed-bandwidth environments but adds transcoding overhead on the Asterisk side — benchmark before enabling in production.

Save and click Submit, then Apply Config.

4. Verify Registration or IP Auth

If you are using IP authentication (recommended for static-IP servers):

`` asterisk -rx "pjsip show endpoint unlimcall-us" ``

You should see state: Available within 60 seconds of the first OPTIONS exchange.

If you are using SIP registration:

`` asterisk -rx "pjsip show registrations" ``

Look for Status: Registered on your trunk endpoint.

5. Create an Outbound Route

Navigate to Connectivity > Outbound Routes > Add Outbound Route.

• Route Name: unlimcall-outbound
• Route CID: leave blank (caller ID set per DID assignment, not route — keeps it flexible for multi-campaign setups)
• Trunk Sequence: select unlimcall-us as position 0; add a PSTN failover trunk at position 1 if you have one

Dial Patterns tab:

For US/Canada 10-digit dialing:

International patterns depend on which of the 33 markets you are dialing into — see multi-country outbound configuration for pattern tables.

6. Test an Outbound Call

From the Asterisk CLI:

`` asterisk -rx "channel originate PJSIP/unlimcall-us/15551234567 application Playback demo-congrats" ``

Confirm the call connects, the codec negotiated is ulaw, and the SIP trace shows your caller ID presented correctly. Check asterisk -rx "pjsip show channels" during the call to confirm the channel is up on your trunk.