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Documentation Index

Fetch the complete documentation index at: https://docs.famulor.io/llms.txt

Use this file to discover all available pages before exploring further.

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Asterisk / FreePBX Integration

Connect your self-hosted Asterisk or FreePBX PBX to Famulor as a SIP trunk
This guide shows how to add Famulor as a generic outbound SIP trunk on Asterisk (chan_pjsip) or FreePBX. Famulor exposes an FQDN like xxx.sip.livekit.cloud; you create a trunk that registers — or, preferred, statically routes — to that host with credentials you control.
Recommended: use chan_pjsip (the modern Asterisk SIP stack) and static IP-routed trunks without registration. Famulor does not need REGISTER, and avoiding it means one less moving part.

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Prerequisites

  • Asterisk 16+ (or FreePBX 15+).
  • Outbound UDP/TCP 5060 (or TLS 5061) reachable from your PBX to xxx.sip.livekit.cloud.
  • An RTP port range opened on your firewall (Asterisk default 10000–20000).
Find your Famulor SIP address: Famulor dashboard → Phone Numbers → Integrate SIP trunk → Our SIP address.

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Option A: FreePBX (GUI)

  1. Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip) Trunk.
  2. General tab:
    • Trunk Name: famulor
    • Outbound Caller ID: the DID Famulor should present, e.g. +49305550123
  3. pjsip Settings → General:
    • SIP Server: xxx.sip.livekit.cloud
    • SIP Server Port: 5060 (or 5061 with TLS)
    • Transport: udp (or tls if your Famulor project supports it)
    • Authentication: Outbound
    • Registration: None
    • Username and Secret: the credentials you’ll re-use in Famulor
  4. pjsip Settings → Advanced:
    • From Domain: xxx.sip.livekit.cloud
    • From User: the DID’s E.164 number, no +
    • Codecs: ulaw, alaw (force G.711)
  5. Submit → Apply Config.

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Option B: Asterisk pjsip.conf (manual)

Add to /etc/asterisk/pjsip.conf (or split across the conf.d files): 
[famulor]
type=endpoint
context=from-famulor
disallow=all
allow=ulaw
allow=alaw
outbound_auth=famulor-auth
aors=famulor-aor
from_domain=xxx.sip.livekit.cloud
from_user=49305550123
direct_media=no

[famulor-auth]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_STRONG_PASSWORD

[famulor-aor]
type=aor
contact=sip:xxx.sip.livekit.cloud:5060
qualify_frequency=60

[famulor-identify]
type=identify
endpoint=famulor
match=xxx.sip.livekit.cloud
Reload: asterisk -rx 'pjsip reload'.

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Step 2: Outbound route

Send selected destinations through the trunk:
  • FreePBX: Connectivity → Outbound Routes → Add → Trunk Sequence: famulor, dial pattern +. or whatever you want routed to AI.
  • Asterisk dialplan: 
    exten => _+X.,1,Dial(PJSIP/${EXTEN:1}@famulor)

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Step 3: Inbound route

  • FreePBX: Connectivity → Inbound Routes → Add → DID Number: your E.164 DID → destination = the extension or IVR you want the AI to answer as.
  • Asterisk: in the from-famulor context, exten => _+X.,1,Goto(your-ivr,s,1).

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Step 4: Import the number into Famulor

  1. Open your Famulor dashboard and go to Phone Numbers → Integrate SIP trunk.
  2. Fill in: 
    Phone number:   The DID your PBX delivers to Famulor (E.164)
Username:       The same username you set in the PBX
Password:       The same password you set in the PBX
  3. Outbound settings: 
    SIP address:            your-pbx.example.com    (your PBX's public hostname)
Outbound number format: International (with +)
  4. Authorization type: Username and password.
  5. Country: the country your DID belongs to.

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Step 5: Verify

  1. Assign the number to a Receive phone calls assistant. Call the DID — the AI should answer.
  2. Create a Make phone calls assistant with the same number and trigger an outbound test call.

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Common issues

Most common Asterisk problem. In pjsip.conf set external_media_address and external_signaling_address to your PBX’s public IP, and add your LAN to local_net. Make sure RTP UDP 10000–20000 is forwarded to the PBX.
Codec mismatch — leave only ulaw, alaw allowed on the endpoint. Disable G.722, Opus, and any video codecs.
The username/password Asterisk sends doesn’t match what Famulor has. Re-paste the credentials in both places carefully — Asterisk silently strips trailing whitespace from pjsip.conf but FreePBX does not, so confirm by hand.
Re-INVITE / direct-media issue. Add direct_media=no and rewrite_contact=yes to the endpoint. This forces Asterisk to stay in the media path, which avoids the LiveKit ↔ PSTN endpoint trying to talk RTP to a NAT’d address.
Need help with setup? Our support team can review your pjsip.conf snippets and help debug NAT/codec issues.
Related pages: SIP Integration and SIP Troubleshooting.