> ## 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.

# Asterisk / FreePBX Integration

> Connect a self-hosted Asterisk or FreePBX PBX to Famulor via SIP

<Warning>
  **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.

  * **Custom number region / full-service onboarding:** [Famulor Custom Number Region form](https://tally.so/r/w4bN2Y)
  * **More detail or bespoke options:** [support@famulor.de](mailto:support@famulor.de)
</Warning>

# 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.

<Tip>
  **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.
</Tip>

## 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`).

<Note>
  **Find your Famulor SIP address:** Famulor dashboard → **Phone Numbers → Integrate SIP trunk → Our SIP address**.
</Note>

## 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**.

## Option B: Asterisk pjsip.conf (manual)

Add to `/etc/asterisk/pjsip.conf` (or split across the conf.d files):

```ini theme={null}
[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'`.

## 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)
  ```

## 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)`.

## 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.

## 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.

## Common issues

<AccordionGroup>
  <Accordion title="One-way audio (NAT)">
    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.
  </Accordion>

  <Accordion title="488 Not Acceptable Here">
    Codec mismatch — leave only `ulaw, alaw` allowed on the endpoint. Disable G.722, Opus, and any video codecs.
  </Accordion>

  <Accordion title="401 / 407 on outbound">
    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.
  </Accordion>

  <Accordion title="Calls drop after 30 seconds">
    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.
  </Accordion>
</AccordionGroup>

***

<Note>
  Need help with setup? Our support team can review your `pjsip.conf` snippets and help debug NAT/codec issues.
</Note>

<Tip>
  Related pages: [SIP Integration](/en/provisioning/sip-ai/sip-integration) and [SIP Troubleshooting](/en/provisioning/sip-ai/sip-troubleshooting).
</Tip>