Default Extension

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Asterisk Users 2 Comments

Hello,

When I get a SIP INVITE as follows:

INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: “0475XXXXXX” ;tag=as7df9ab18
To:
Contact:
Call-ID: 344d42bd16975a54141d11f635bdfc71@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’ ??

Thank you, Mickael

2 thoughts on - Default Extension

  • What version of Asterisk are you using?

    It would help to show how you are performing the dial in dialplan or otherwise. If you are dialing a user/peer present in sip.conf or a database then show that configuration as well. Based on that someone could make a suggestion.

  • 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