Greeting, I have a Kamailio server connected to two network, Network one their an Asterisk server, at network two we have the client one a client call , the call will have no audio voice how I can solve this using Kamailio and rtp server ?
Thanks for your help Regards _______________________________________________ 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
