Hi Marwan, i still try to do the same (with public ip / private ip) with kamailio. Thats is what i figured out (its not 100% but better then nothing):
turn mhomed=1 in your kamailio.cfg and do "rtpproxy -l network.1-ip network.2-ip" for bridging. Kamailio should listen:network-ip1 and listen:network-ip2 Then you must call rtpproxy_manage with the right params like OCFEI, IP-Address-1 for example. This should be god starting points with examples that bridges ipv4 to ipv6 or act as ALG. <http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6> <http://openser.svn.sourceforge.net/viewvc/openser/trunk/modules/nathelper/examples/alg.cfg?revision=2&view=markup> 2012/4/5 Marwan Idriss <marwan.idr...@gmail.com>: > 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 -- Mit freundlichen Grüßen *Karsten Horsmann* _______________________________________________ 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
