[SR-Users] Rewrite SIP INVITE

Mickael MONSIEUR mickael.monsieur at gmail.com
Wed Mar 26 19:06:54 CET 2014


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 at edvina.net>:

>
> 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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