[asterisk-bugs] [Asterisk 0017151]: Playback deadlock?

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17151 
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Reported By:                RobinFood
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17151
Category:                   Applications/app_playback
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.6 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-04-08 12:24 CDT
Last Modified:              2010-04-13 18:57 CDT
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Summary:                    Playback deadlock?
Description: 
My issue is very close to the issue id 0014412. 

I'm using the Local channel to make a dialout calls.
I'm using the Playback application to play a gsm-file.
Sometimes (approximately 1 from 10 Playback calls) Playback application
stops to play after 1-5 seconds of a voice. Next-priority extensions does
not start. MixMonitor records a silence from * and a voice from a called
party. 
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 (0120369) RobinFood (reporter) - 2010-04-13 18:57
 https://issues.asterisk.org/view.php?id=17151#c120369 
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Thank you.

I'm attached the full debug log.
There was an one little difference between this and previous problems: now
there was no voice after starting a playback - only a silence (maybe there
was a voice, but is was very short in this case, less than one second).

Sorry, but there are some other calls in the log, which may not affect
this issue. I leaved them in the full log because I may be wrong, and these
other calls really may affect this issue. The problematic playback was
started at 01:22:25.

Sorry again, but I replaced all IP-addresses and all phone numbers by
different placeholders (i.e. 'aXX.bXX.cXX.dXX' instead of real IPs, and
'phoneX' instead of real phone numbers - only the phone number 790 left
unchanged). IP of my asterisk was replaced by a01.b01.c01.d01, and IP of
the 'alc' peer was replaced by 'a02.b02.c02.d02' 

Issue History 
Date Modified    Username       Field                    Change               
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2010-04-13 18:57 RobinFood      Note Added: 0120369                          
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