[asterisk-users] Default extension

[Olle E. Johansson]

On 26 Mar 2014, at 19:14, Mickael MONSIEUR <mickael.monsieur at gmail.com> wrote:

> Hello,
> 
> When I get a SIP INVITE as follows: 
>> INVITE sip:s at 10.1.0.191:5060 SIP/2.0
>> Max-Forwards: 69
>> From: "0475XXXXXX" <sip:1053212 at sip.domain.com>;tag=as7df9ab18
>> To: <sip:02XXXXXX at IP:5060>
>> Contact: <sip:1053212 at IP:5060>
>> Call-ID: 344d42bd16975a54141d11f635bdfc71 at sip.domain.com
>> CSeq: 102 INVITE
>> Date: Wed, 26 Mar 2014 15:06:01 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
>> Supported: replaces, timer
>> Content-Type: application/sdp
>> Content-Length: 252
>> 
> 
> Asterisk considers that the extension is 's'. (The Register) 
> How to make the extension number that is shown in the 'To' ??

You never route calls on the To: header in SIP. You route on the request URI. Unless this is something where you used the REGISTER statement in sip.conf and forgot to add an extension or you register once for multiple DIDs.

I would suggest changing your register statement to include an extension. In that extension you read the To: header with the SIP_HEADER() dialplan function and issue a goto so you end up with the extension in the To header.

The IETF has with help of the SIP forum written a standard extension to SIP to handle this use-case, something called GIN. It's now part of the SIPConnect specification. using the gin extension, you would get the called phone number in the r-uri.

/O

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