[SR-Users] Rewrite SIP INVITE

[Olle E. Johansson]

On 26 Mar 2014, at 16:32, Mickael MONSIEUR <mickael.monsieur at 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 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 kamailioIP:5060>
> Contact: <sip:1053212 at asteriskIP: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
> 
> 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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