Hi All, I'm using kamailio with carrierroute to load balance calls to other servers.
I have 2 testing scenarios set up: sipp><kamailio><asterisk server In the above scenario all SIP transactions, dialogs go through kamailio, INVITE, Trying, ACK, OK, BYE, OK Here is another one: phone><asterisk><kamailio><asterisk server In this scenario only the initial INVITE, ACK and OK go through kamailio, then the 2 asterisk servers finish the session directly to each other with ACK, BYE and OK In both asterisk servers, canreinvite=no is set in peers and general section in sip.conf. The kamailio cfg is using t_relay. No errors are coming up anywhere. Here is a pastebin of the 2 asterisk traces and the kamailio config. http://pastebin.com/Uk9qVhX2 Any ideas on why the call is not maintaining SIP session through the proxy? Thanks. JR -- JR Richardson Engineering for the Masses _______________________________________________ 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
