Re: [asterisk-users] Default extension

2014-03-27 Thread Olle E. Johansson

On 26 Mar 2014, at 19:14, Mickael MONSIEUR mickael.monsi...@gmail.com wrote:

 Hello,
 
 When I get a SIP INVITE as follows: 
 INVITE sip:s@10.1.0.191:5060 SIP/2.0
 Max-Forwards: 69
 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
 To: sip:02XX@IP:5060
 Contact: sip:1053212@IP:5060
 Call-ID: 344d42bd16975a54141d11f635bdf...@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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[asterisk-users] Default extension

2014-03-26 Thread Mickael MONSIEUR
Hello,

When I get a SIP INVITE as follows:

INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
To: sip:02XX@IP:5060
Contact: sip:1053212@IP:5060
Call-ID: 344d42bd16975a54141d11f635bdf...@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
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Re: [asterisk-users] Default extension

2014-03-26 Thread Rusty Newton
On Wed, Mar 26, 2014 at 1:14 PM, Mickael MONSIEUR
mickael.monsi...@gmail.com wrote:
 Hello,

 When I get a SIP INVITE as follows:

 INVITE sip:s@10.1.0.191:5060 SIP/2.0
 Max-Forwards: 69
 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
 To: sip:02XX@IP:5060
 Contact: sip:1053212@IP:5060
 Call-ID: 344d42bd16975a54141d11f635bdf...@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.

-- 
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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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