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mickael.monsieur at gm...
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 Posted: Wed Mar 26, 2014 1:14 pm Post subject: [asterisk-users] Default extension |
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Hello,
When I get a SIP INVITE as follows:
| Quote: | INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: "0475XXXXXX" <sip:1053212@sip.domain.com ([email]sip%3A1053212@sip.domain.com[/email])>;tag=as7df9ab18
To: <sip:02XXXXXX@IP:5060>
Contact: <sip:1053212@IP:5060>
Call-ID: 344d42bd16975a54141d11f635bdfc71@sip.domain.com (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
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Asterisk considers that the extension is 's'. (The Register)
How to make the extension number that is shown in the 'To' ??
Thank you,
Mickael |
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rnewton at digium.com
Guest
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| Posted: Wed Mar 26, 2014 5:14 pm Post subject: [asterisk-users] Default extension |
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On Wed, Mar 26, 2014 at 1:14 PM, Mickael MONSIEUR
<mickael.monsieur@gmail.com> wrote:
| Quote: | 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" <sip:1053212@sip.domain.com>;tag=as7df9ab18
To: <sip:02XXXXXX@IP:5060>
Contact: <sip:1053212@IP:5060>
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' ??
|
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.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
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oej at edvina.net
Guest
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| Posted: Thu Mar 27, 2014 4:29 am Post subject: [asterisk-users] Default extension |
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On 26 Mar 2014, at 19:14, Mickael MONSIEUR <mickael.monsieur@gmail.com (mickael.monsieur@gmail.com)> wrote:
| Quote: | Hello,
When I get a SIP INVITE as follows:
| Quote: | INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: "0475XXXXXX" <sip:1053212@sip.domain.com ([email]sip%3A1053212@sip.domain.com[/email])>;tag=as7df9ab18
To: <sip:02XXXXXX@IP:5060>
Contact: <sip:1053212@IP:5060>
Call-ID: 344d42bd16975a54141d11f635bdfc71@sip.domain.com (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' ??
|
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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