Feb 1 22:13:37 VERBOSE[18623] logger.c: -- Zap/2-1 answered SIP/102-9fda Feb 1 22:13:37 DEBUG[18623] channel.c: Avoiding initial deadlock for 'SIP/102-9fda' Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from channel: SIP/102-9fda Feb 1 22:13:43 DEBUG[18623] channel.c: Bridge stops bridging channels SIP/102-9fda and Zap/2-1 Feb 1 22:13:43 VERBOSE[18623] logger.c: -- Hungup 'Zap/2-1' Feb 1 22:13:43 DEBUG[18623] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Can anyone explain why this call dropped? The person dialed a number, the call WAS completed and connected to the PSTN through a PRI, but they never heard audio and the call was disconnected by Asterisk. _______________________________________________ --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
