Hi,
When Asterisk create a register it uses by default extension "s".
register => user[:secret[:authuser]]@host[:port][/extension]
But that we do not care. I want to replace the 's' from my SIP INVITE. 
How?


2014-03-26 16:40 GMT+01:00 Olle E. Johansson <oej@edvina.net>:

On 26 Mar 2014, at 16:32, Mickael MONSIEUR <mickael.monsieur@gmail.com> wrote:

Hello,

I installed kamailio and asterisk with the tutorial of asipto. 
For alias numbers I configured the module alias_db. 

Everything works because Asterisk outgoing call is directed at Kamailio, then Kamailio is sending to the alias registered (table location) 

Only the number alias appears in the "To" (02XXXXXX) and not in the INVITE URI. It only shows "s" ... is it possible to force writing 02XXXXXX instead of "s" ?

Who sets the "s" in the request URI? Asterisk use "s" when there's no extension given in the dialplan. 

/O

Example:

INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: "0475XXXXXX" <sip:1053212@sip.domain.com>;tag=as7df9ab18
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

Because without it, Asterisk servers behind Kamailio not will route the call to the correct extension but to the "s". Asterisk ignores the "To" apparently this is strange ...

Thank you,
Mickael
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