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.
Matt wrote:> 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.Very difficult to guess without any information about your system. If you are using Asterisk 1.2.2, this is a known problem. If not, we'll need a lot more information to be able to even try to help you.