I am using asterisk-1.4.21 and it is configured to pass media through it for SIP calls. I have observed that if the callee answers the call and starts speaking immediately for e.g. 'Hello one two three', the caller would get to hear only 'one two three'. From packet captures I can see that asterisk receives all the RTPs from the callee but it truncates the 'Hello' word from the voice path when passing the stream on the other side.
The signaling gets complete between caller and callee, so asterisk should bridge the channels immediately. I am using canreinvite=no and nat=yes option in sip.conf. Has anyone observed this issue why asterisk is cutting of the initial voice? ---Mayur
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
