> Actully ethereal OK... Try canreinvite=yes in the [general] section; this makes it the default setting for all peers unless specified otherwise. Do the same for nat=no in [general] to rule out all NAT'ing related issues. You don't have tT in your Dial() statement, that's good. You say you verified that no transcoding is needed (i.e. both ends use the same codec). Well, then it should work!
Once you get it to work, you can individualize the accounts and no longer use a global setting. But that's down the line. > asterisk always creates a 'native bridge' and seems to hold on for dear > life so far as I have seen :-) It says "Attempting Native Bridge" but it doesn't tell if you if it succeeded or not; there was once a notice saying the the bridge could not be established (failed?) but it caused even more confusion. You could add some statements to the ast_rtp_bridge() code in rtp.c and give yourself some feedback -- succeeded, failed because X / Y / Z. Hope that helps... --Luki _______________________________________________ --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
