Hello, Could somebody tell me is it possible to use asterisk without RTP proxying in SIP<->H323 calls? I mean exactly what canreinvite=yes option do in SIP<->SIP calls. I don't need a transcoding, only a signaling conversion, and this is possible with some softswitches, so i wondering what about asterisk. Same question about H323<->H323 calls I'm using NuFone Network's H323 cahhel
Thanks -- Sincerely, Elman Efendiyev PROTECH INC. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
