> ## Documentation Index
> Fetch the complete documentation index at: https://docs.getsmartalex.com/llms.txt
> Use this file to discover all available pages before exploring further.

# FreePBX

> Connect FreePBX (Asterisk with GUI) to SmartAlex via chan_pjsip SIP trunk.

<Info>
  **Audience**: FreePBX admin. **Time**: 10 to 15 minutes. **Driver**: chan\_pjsip (recommended, not legacy chan\_sip).
</Info>

## Prerequisites

* FreePBX admin access
* Outbound UDP 5060 allowed on the firewall
* SIP ALG disabled on the router (mandatory)
* PBX SIP Domain (FreePBX's public FQDN or IP)

## Step 1: create the trunk in SmartAlex

Phone Numbers, then Connect Your Carrier, then SIP Trunk, then pick FreePBX. Save, copy the credentials.

## Step 2: add the pjsip trunk in FreePBX

<Steps>
  <Step title="Open Trunks">
    FreePBX admin, then Connectivity, then Trunks, then Add SIP (chan\_pjsip) Trunk.
  </Step>

  <Step title="General tab">
    * **Trunk Name**: `smartalex`
    * **Hide CallerID**: No
    * **Outbound CallerID**: your DID
    * **Maximum Channels**: match the concurrent-call limit in SmartAlex (default 10)
  </Step>

  <Step title="pjsip Settings: General tab">
    * **Username**: (from SmartAlex)
    * **Secret**: (from SmartAlex)
    * **Authentication**: Outbound
    * **Registration**: Send
    * **SIP Server**: `sip.voice.getsmartalex.com`
    * **SIP Server Port**: `5060`
    * **Context**: `from-pstn`
    * **Transport**: `0.0.0.0-udp`
  </Step>

  <Step title="Codecs tab">
    Enable `ulaw` and `alaw`. Disable `g729` unless licensed.
  </Step>

  <Step title="Submit and Apply Config">
    Click **Submit**, then the **Apply Config** button in the red bar at the top.
  </Step>

  <Step title="Verify registration">
    Admin, then Reports, then Asterisk Info, then Registrations. The trunk should show **Registered**.
  </Step>
</Steps>

## Step 3: inbound route

Connectivity, then Inbound Routes, then Add. Set the DID, destination as your SmartAlex trunk (outbound leg).

Alternatively if using a SmartAlex DID, skip this step.

## Step 4: outbound route (optional)

Only if FreePBX extensions should dial out through SmartAlex. Connectivity, then Outbound Routes, then add a dial pattern that strips the prefix and uses the SmartAlex trunk.

## Step 5: extensions and agent linking

Add extensions in SmartAlex Settings, then PBX. Link agent in Agent Studio.

## Step 6: test

Dial the DID. Ask for a transfer.

## Alternative: Native Extension Mode

The flow above has FreePBX register out to SmartAlex via a trunk. The inverse, SmartAlex registers as an extension on YOUR FreePBX, is often simpler when FreePBX is your source-of-truth. Provision a normal pjsip extension in **Applications → Extensions**, give it to SmartAlex, and we register exactly like any softphone. From there, route DIDs through your usual inbound-route + ring-group rules. See [Native Extension Mode](/telephony/native-extension-mode).

## FreePBX-specific notes

* **SIP Debug**: Admin, then Asterisk CLI. Run `pjsip set logger on` to trace SIP messages in real time.
* **NAT handling**: pjsip handles NAT well when `rewrite_contact` is enabled (default).
* **Call recording conflicts**: if FreePBX records calls at the trunk level, it will record the SmartAlex leg separately. Decide whether that's wanted.
* **Distro vs Incredible PBX vs stock FreePBX**: configuration is identical across distributions.

## Next steps

<CardGroup cols={2}>
  <Card title="Asterisk" href="/telephony/asterisk">
    Raw Asterisk setup if you don't run FreePBX.
  </Card>

  <Card title="Troubleshooting" href="/telephony/troubleshooting">
    Error reference.
  </Card>
</CardGroup>