# Asterisk

Send and receive calls through Vonyx with PJSIP.

Connect Asterisk to Vonyx with a PJSIP trunk. You get outbound calls on your Vonyx numbers, and inbound calls into your dialplan.

## Before you start

- A Vonyx [SIP trunk](/docs/console/sip-trunks) with a [number](/docs/console/numbers) attached
- Asterisk 16 or later with `res_pjsip` (PJSIP)
- A public IP or hostname for Asterisk, with UDP `5060` and your RTP range open to `43.204.9.41`

## Outbound calls

<Steps>
<Step>

### Add the trunk to `pjsip.conf`

Replace the username and password with your trunk's values. There is no `registration` section: Vonyx checks credentials on each call and doesn't need Asterisk to register.

```ini title="pjsip.conf"
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
; Behind NAT only: your public IP and your LAN
;external_media_address = 203.0.113.10
;external_signaling_address = 203.0.113.10
;local_net = 192.168.1.0/24

[vonyx-auth]
type = auth
auth_type = userpass
username = your_trunk_username
password = your_trunk_password

[vonyx]
type = aor
contact = sip:43.204.9.41:5060

[vonyx]
type = endpoint
transport = transport-udp
context = from-vonyx
disallow = all
allow = ulaw,alaw
outbound_auth = vonyx-auth
aors = vonyx
direct_media = no
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes

[vonyx]
type = identify
endpoint = vonyx
match = 43.204.9.41
```

`ulaw` and `alaw` are PCMU and PCMA, the only codecs Vonyx supports. On recent Asterisk releases `auth_type = userpass` is deprecated and treated as `digest`; either works.

</Step>
<Step>

### Set NAT if Asterisk is behind a router

If Asterisk has a private IP, uncomment the three NAT lines on the transport:

- `external_media_address` and `external_signaling_address`: your public IP. Asterisk writes it into SIP headers and the SDP for traffic leaving `local_net`.
- `local_net`: your LAN range. Add one line per range.

Forward UDP `5060` and your RTP range to Asterisk, 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, set in `rtp.conf`, must be open to `43.204.9.41`:

```ini title="rtp.conf"
[general]
rtpstart = 10000
rtpend = 20000
```

</Step>
<Step>

### Add the outbound dialplan

Outbound calls must use one of your Vonyx numbers as caller ID.

```ini title="extensions.conf"
[outbound]
exten => _91XXXXXXXXXX,1,Set(CALLERID(num)=91XXXXXXXXXX)
 same => n,Dial(PJSIP/${EXTEN}@vonyx,60)
 same => n,Hangup()
```

Include `outbound` in the context your phones use.

</Step>
</Steps>

## Inbound calls

<Steps>
<Step>

### Add the inbound context

The `identify` section matches calls from `43.204.9.41` to the `vonyx` endpoint, which sends them to `from-vonyx`.

```ini title="extensions.conf"
[from-vonyx]
exten => _X.,1,Answer()
 same => n,Playback(hello-world)
 same => n,Hangup()
```

Replace these steps with your IVR, queue or agent.

</Step>
<Step>

### Point Vonyx at Asterisk

In the Vonyx console, open the trunk, turn on **Inbound calls**, and set **Inbound address** to Asterisk's public host, for example `pbx.example.com:5060`. See [Inbound calls](/docs/console/inbound-calls).

</Step>
</Steps>

## Test the connection

```bash
asterisk -rx "core reload"
asterisk -rx "pjsip show endpoint vonyx"
asterisk -rx "channel originate PJSIP/91XXXXXXXXXX@vonyx application Playback hello-world"
```

The `originate` call rings `91XXXXXXXXXX` and plays a prompt when answered. Then call your Vonyx number from a mobile and check that it reaches `from-vonyx`. Use `pjsip set logger on` to see the SIP messages.

## Limitations

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

## Troubleshooting

| Problem | Fix |
| --- | --- |
| `401` or `407` that never succeeds | Check `username` and `password` in `vonyx-auth`, and that the endpoint has `outbound_auth = vonyx-auth`. |
| `429` after auth succeeds | The caller ID isn't one of your Vonyx numbers. |
| Inbound calls rejected or sent to the wrong context | Check the `identify` section matches `43.204.9.41` and that `pjsip show identifies` lists it. |
| `488 Not Acceptable Here` | Allow only `ulaw` and `alaw` on the endpoint. |
| One-way or no audio | Set `external_media_address`, `external_signaling_address` and `local_net` behind NAT, and open your `rtp.conf` range to `43.204.9.41`. |

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