> I have canreinvite and directmedia to 'no' - and there is no NAT > between the phones and asterisk...
Hmm. In that case, I'm not sure. You could take a look at the output of "rtp set debug on" when the call is going on to see what is going on with the audio. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
