Re: [Kamailio-Users] Send all traffic including SIP to SIP to Asterisk or PSTN Gateway

From: Iñaki Baz Castillo <ibc(a)aliax.net> Subject: Re: [Kamailio-Users] Send all traffic including SIP to SIP to Asterisk or PSTN Gateway To: Cc: users(a)lists.kamailio.org Date: Sunday, August 9, 2009, 9:31 AM 2009/8/8 Raju Abhyankar <kf6rzt(a)yahoo.com>om>: traffic is forwarded to Asterisk which then terminates the call. What I would like to do is forward all SIP to SIP calls also to Asterisk? This implies I would like to turn off the call look up on Kamailio (loose route?) and blindly forward to Asterisk. Can some one suggest how this could be done. "I would like to turn off the call look up on Kamailio (loose route?)" ¿?¿ If you want to pass all the traffic to Asterisk it to send it back to kamailio: - Call from alice to bob. - Kamailio checks if bob exists => does_uri_exist() function. - If true, route the call to Asterisk (without changing the entire RURI or keeping the RURI username). - Asterisk generates a call to "sip:bob@kamailio_IP" and sends it to Kamailio. - Kamailio does lookup("location"). Note: This wouldn't work in a multidomain scenario as Asterisk doesn't support real multidomain. -- Iñaki Baz Castillo <ibc(a)aliax.net> _______________________________________________ Kamailio (OpenSER) - Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users