[asterisk-bugs] [Asterisk 0018475]: call files: Calls get stalled during stream_file, asterisk "hangs up" without hanging up

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=18475 
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Reported By:                haakon
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18475
Category:                   PBX/pbx_spool
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.15 
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:             2010-12-15 02:27 CST
Last Modified:              2010-12-16 10:53 CST
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Summary:                    call files: Calls get stalled during stream_file,
asterisk "hangs up" without hanging up
Description: 
I created a outgoing call with an agi that were to do the following:

1. Play informative message
2. Record message (nothing gets recorded)
3. Play menu for handling message (asterisk bails out here after a few
ms)

If I use this AGI when calling _in_ to asterisk there is no problems.
But if I use a call file and put it in /var/spool/asterisk/outgoing, the
following happends:

The message is not really recorded to the file, and when exiting "record
message" with #, it starts to play the menu message, but after just 100ms
or so, it suddenly stops transmitting audio.
The next peculiar thing that happens is that asterisk tries to hang up the
call when the file is done playing, but the call stays up.

I have tried this with the following scenarios:
1. call file channel: Local// which calls via DAHDI
2. call file channel: DAHDI
3. call file channel: SIP to telco
4. Rewrite entire agi in a much simpler form
5. Change audio files played with other working ones.
6. Call file Application: agi
7. call file Context which runs agi
8. call file channel, SIP to another server with the script as incoming
agi.
9. Tried to move everything to another server

All the mentioned tests have been done, and the result is always the
same.
I have tried this with 1.6.1.x and 1.6.2.15, and also 1.8.x.

I can't figure out how this even happens over SIP to the external machine.
But it seems like asterisk doesn't like to record on an outgoing call leg.
Somehow, it is different to asterisk if its an incoming call, or of it is
an outgoing call via AGI.
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Issue History 
Date Modified    Username       Field                    Change               
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2010-12-16 10:53 lmadsen        Status                   new => feedback     
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