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Done for you — any region, any country (our EU partner carrier)If you prefer not to configure a SIP trunk or import numbers yourself, our EU partner carrier can supply local numbers in any region and any country, set up the SIP trunk, and import the connection into Famulor for you. Provisioning runs end-to-end through the partner (carrier-side), so you skip DIY trunk and number import in Famulor.

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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.