Support / Voice / PBX / FreePBX

Add a VoiceTel SIP trunk to FreePBX.

FreePBX 16 / 17 use PJSIP by default. Avoid the legacy chan_sip driver — VoiceTel's STIR/SHAKEN handling is cleaner on PJSIP. Steps below cover both Digest and IP authentication.

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1. Create the trunk

  1. FreePBX UI → Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip) Trunk.
  2. General tab:
    • Trunk Name: voicetel
    • Outbound CallerID: your VoiceTel-assigned DID in E.164, e.g. "VoiceTel" <+12125551234>
    • Maximum Channels: leave blank for unlimited (or set to your concurrent-call cap)

2. PJSIP Settings tab — Digest authentication

  • Username: SIP username from VoiceTel onboarding
  • Secret: SIP password from VoiceTel onboarding
  • Authentication: Outbound
  • Registration: Send
  • SIP Server: the SIP server host shown in your customer portal
  • SIP Server Port: 5060
  • Transport: UDP

Advanced sub-tab:

  • Qualify Frequency: 30
  • From User: SIP username
  • From Domain: the SIP server host shown in your customer portal
  • Send Connected Line: Yes
  • Trust RPID/PAI: Yes
  • Send PAI: Yes
  • Force rport: Yes
  • Symmetric RTP: Yes
  • Rewrite Contact: Yes
  • DTMF Mode: RFC 4733

Codecs sub-tab: enable ulaw, alaw, and g729; move ulaw to the top.

2 (alt). PJSIP Settings — IP authentication

For IP auth trunks, change two things:

  • Authentication: None
  • Registration: None

VoiceTel authorizes inbound INVITEs by source IP based on the ACL you've registered with onboarding. Add the FreePBX server's public IP to that ACL.

3. Submit and apply

Click Submit, then Apply Config in the red banner at the top. The trunk shows up in Reports → Asterisk Info → PJSIP Endpoints; status should be Reachable within a minute.

4. Outbound route

  1. Connectivity → Outbound Routes → Add Outbound Route.
  2. Route Name: voicetel-out
  3. Trunk Sequence: select the voicetel trunk created above
  4. Dial Patterns tab: add patterns for your dialing rules (e.g. 1NXXNXXXXXX for 11-digit US, or NXXNXXXXXX for 10-digit + prepend rule).
  5. Submit + Apply Config.

Verify

# From the FreePBX server CLI
asterisk -rvvvv
*CLI> pjsip show registrations
*CLI> pjsip show endpoints
*CLI> pjsip set logger on    # SIP trace if needed