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 > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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