> 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-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
