[SR-Users] Rewrite SIP INVITE

[Mickael MONSIEUR]

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" ?

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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