On 26 Mar 2014, at 19:06, Mickael MONSIEUR <mickael.monsi...@gmail.com> wrote:

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

> How?
Read the docs... ;-)

/O
> 
> 
> 2014-03-26 16:40 GMT+01:00 Olle E. Johansson <o...@edvina.net>:
> 
> On 26 Mar 2014, at 16:32, Mickael MONSIEUR <mickael.monsi...@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:1053...@sip.domain.com>;tag=as7df9ab18
>> To: <sip:02XXXXXX@kamailioIP:5060>
>> Contact: <sip:1053212@asteriskIP: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
>> 
>> 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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>> sr-users@lists.sip-router.org
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> 
> 
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