Wilson Pickett a �crit :
What I face is that a SIP call to our GW has from time to time the
behaviour to "loose" audio. Hanging up and retrying can work, but mostly
we wait or use an IAX GW and try again and then it work. Can also take
few hours before it work again.
What RTP ports are used in asterisk and do the match those of the phones?
Asterisk: rtpstart 6970 rtpend 7170
ATA186: RTP 5004
Remember that I face this problem from time to time only -- Daniel _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
