[Asterisk-Users] Call completes but then drops?

[Matt]

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.