Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer. Here's my network topology: +---> [asterisk1] [public_ip] | 10.50.10.131 [router] <---NAT---> [kamailio] <---+ 10.50.10.1 10.50.10.120 | +---> [asterisk2] 10.50.10.132 In my setup i planned to use UAC and DISPATCHER modules. I started from the "kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER. All is working fine when i do a test call from a softphone inside network 10.50.10.0/24. When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP. I understand that i need to rewrite SDP in that case, but i actually don't know how/where. I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible. For security reasons, i would like to force the RTP through RTPProxy. I'm missing something, and need your help me to understand my errors. Best Regards, Bruno _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users