# Legacy Asterisk Configuration

This page preserves the legacy DIDWW configuration for Asterisk installations
that still use `chan_sip` and `sip.conf`. `chan_sip` was deprecated in
Asterisk 17, was no longer built by default beginning with Asterisk 19, and was
removed in Asterisk 21.

Warning

Do not use this configuration for new installations. For current Asterisk
releases, see [Asterisk PJSIP Configuration (v20-23)](current-versions.html).

DIDWW SIP trunks can be used with Asterisk for inbound and outbound calls.

## Dialplan example

Add the inbound context to `extensions.conf`:

```
[from-didww]
exten => _X.,1,Ringing
exten => _X.,n,Answer
exten => _X.,n,Echo
exten => _X.,n,Wait(600)
exten => _X.,n,Hangup
```

## Inbound trunk example

Add the following peers to `sip.conf`:

```
[didww-ny]
host=46.19.209.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=invite,port
nat=never
allow=all

[didww-fra]
host=46.19.210.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=invite,port
nat=never
allow=all

[didww-la]
host=46.19.212.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=invite,port
nat=never
allow=all

[didww-mia]
host=46.19.213.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=invite,port
nat=never
allow=all

[didww-sg]
host=46.19.214.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=invite,port
nat=never
allow=all

[didww-hk]
host=46.19.215.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=invite,port
nat=never
allow=all
```

For the current signaling address list, including Amsterdam and IPv6
addresses, see [SIP Signaling Addresses](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip).

## Outbound trunk example

Add the outbound peer to `sip.conf`:

```
[didww-outbound]
type=peer
dtmfmode=rfc2833
dtmf=rfc2833
fromuser=DID_NUMBER
auth=USERNAME:PASSWORD@out.didww.com
secret=PASSWORD
host=any.out.didww.com
fromdomain=ASTERISK_IP_ADDRESS
```

Replace the placeholder values with your DIDWW outbound trunk details. For the
current gateway list and authentication realm, see
[SIP Protocol Details](../../voice/outbound-trunks/outbound-sip-information.html#services-voice-out).

Note

If Asterisk cannot find an extension within the given context, it returns an
unhelpful *No such context/extension* error. Use the catch-all `_X.`
pattern while troubleshooting, then replace it with the routing rules
required by your deployment.

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