# FreeSWITCH

Add Vonyx as a gateway on the FreeSWITCH external profile.

Add Vonyx as a Sofia gateway on the `external` profile. You get outbound calls on your Vonyx numbers, and inbound calls into the `public` dialplan.

## Before you start

- A Vonyx [SIP trunk](/docs/console/sip-trunks) with a [number](/docs/console/numbers) attached
- FreeSWITCH with the default (vanilla) config, where the `external` profile listens on port `5080`
- A public IP or hostname for FreeSWITCH, with UDP `5080` and your RTP range open to `43.204.9.41`

## Outbound calls

<Steps>
<Step>

### Create the gateway

```xml title="conf/sip_profiles/external/vonyx.xml"
<include>
  <gateway name="vonyx">
    <param name="proxy" value="43.204.9.41:5060"/>
    <param name="realm" value="43.204.9.41"/>
    <param name="username" value="your_trunk_username"/>
    <param name="password" value="your_trunk_password"/>
    <param name="register" value="false"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>
```

`register` is `false`: Vonyx checks credentials on each call and doesn't need FreeSWITCH to register. `caller-id-in-from` puts the caller ID in the From header.

</Step>
<Step>

### Use G.711

Vonyx supports PCMU and PCMA only. The `external` profile reads its codec lists from these variables. Change the existing lines:

```xml title="conf/vars.xml"
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>
```

These variables are shared with the `internal` profile. To change only the Vonyx side, set `inbound-codec-prefs` and `outbound-codec-prefs` to `PCMU,PCMA` in `sip_profiles/external.xml` instead.

</Step>
<Step>

### Set NAT if FreeSWITCH is behind a router

The `external` profile uses `ext-rtp-ip` and `ext-sip-ip`, which default to `$${external_rtp_ip}` and `$${external_sip_ip}` (found by STUN). Behind NAT, set them to your public IP:

```xml title="conf/vars.xml"
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=203.0.113.10"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=203.0.113.10"/>
```

Replace the existing `stun-set` lines for these two variables. Forward UDP `5080` and your RTP range to FreeSWITCH, and turn off SIP ALG on the router.

</Step>
<Step>

### Set the RTP port range

Vonyx sends RTP from UDP `10000–20000`. Your own range must be open to `43.204.9.41`. It is set in `switch.conf.xml`:

```xml title="conf/autoload_configs/switch.conf.xml"
<param name="rtp-start-port" value="16384"/>
<param name="rtp-end-port" value="32768"/>
```

These are the defaults shown in the sample config. Change them to match your firewall.

</Step>
<Step>

### Add an outbound route

Caller ID must be one of your Vonyx numbers.

```xml title="conf/dialplan/default/01_vonyx.xml"
<include>
  <extension name="vonyx_out">
    <condition field="destination_number" expression="^(91\d{10})$">
      <action application="set" data="effective_caller_id_number=91XXXXXXXXXX"/>
      <action application="bridge" data="sofia/gateway/vonyx/$1"/>
    </condition>
  </extension>
</include>
```

</Step>
</Steps>

## Inbound calls

<Steps>
<Step>

### Route the number in the public dialplan

The `external` profile sends calls to the `public` context and doesn't ask callers to authenticate. Map your Vonyx number to an extension:

```xml title="conf/dialplan/public/01_vonyx.xml"
<include>
  <extension name="vonyx_in">
    <condition field="destination_number" expression="^(91XXXXXXXXXX)$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="transfer" data="1000 XML default"/>
    </condition>
  </extension>
</include>
```

Because port `5080` accepts calls without auth, allow only `43.204.9.41` (and your other carriers) to reach it in your firewall.

</Step>
<Step>

### Point Vonyx at FreeSWITCH

In the Vonyx console, open the trunk, turn on **Inbound calls**, and set **Inbound address** to your FreeSWITCH host with the `external` port, for example `fs.example.com:5080`. See [Inbound calls](/docs/console/inbound-calls).

</Step>
</Steps>

## Test the connection

Changes to `vars.xml` and the profile need the profile restarted. A new gateway file only needs a rescan.

```bash
fs_cli -x "reloadxml"
fs_cli -x "sofia profile external restart"
fs_cli -x "sofia status gateway vonyx"
fs_cli -x "originate {origination_caller_id_number=91XXXXXXXXXX}sofia/gateway/vonyx/91XXXXXXXXXX &echo"
```

`origination_caller_id_number` must be your Vonyx number. The callee hears their own voice echoed back. Then call your Vonyx number from a mobile and check it reaches extension `1000`. Use `sofia global siptrace on` to see the SIP messages.

## Limitations

- DTMF and call transfer are not available on Vonyx yet. Don't rely on IVR key presses or `deflect` / REFER on Vonyx calls.
- New accounts make 1 call at a time ([raise limits](/docs/sip/limits)).
- No TLS or SRTP: use UDP or TCP on port `5060` towards Vonyx.

## Troubleshooting

| Problem | Fix |
| --- | --- |
| `401` or `407` that never succeeds | Check `username` and `password` in the gateway. |
| `429` after auth succeeds | The caller ID isn't one of your Vonyx numbers. |
| `INCOMPATIBLE_DESTINATION` / `488` | Limit the codec lists to `PCMU,PCMA`. |
| Inbound calls hit the wrong extension or fail | Check the `public` dialplan matches the number as Vonyx sends it, and that the inbound address uses port `5080`. |
| One-way or no audio | Set `external_rtp_ip` / `external_sip_ip` behind NAT, and open your RTP range to `43.204.9.41`. |

More in [SIP troubleshooting](/docs/sip/troubleshooting).