Hi Shaun! Your problem description is too short to give you any good help.
Use tcpdump (or other tools) to capture the scenario with Asterisk and Kamailio. Then compare them to find out why it doesn't work.
Is media sent directly to Asterisk then it ca not be the problem of Kamailio.
I hope the mobile client is smart enough to also send a reINVITE when getting the new IP address (of the mobile connection) with proper Contact header - otherwise it can not receive SIP requests from Asterisk.
regards Klaus On 20.06.2012 18:07, Shaun Clark wrote:
The use case is that I have a SIP client registered to Kamailio talking to an Asterisk box connected to the PSTN. The client is a mobile phone and the user is connected to wifi. The user then steps out of wifi range and the phone drops the connection and picks up the 3g data connection. I want the media stream to reconnect to the client and the call to resume without having to redial. This works now if the client is directly connected to the Asterisk machine, but not when I am routing through my Kamailio server. How do I go about this, examples are always appreciated, thanks! _______________________________________________ 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
_______________________________________________ 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
