[asterisk-bugs] [Asterisk 0017232]: Extensions with playback stalls in different points in bridged calls

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17232 
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Reported By:                EduFrazao
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17232
Category:                   Applications/app_playback
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
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:             2010-04-22 16:15 CDT
Last Modified:              2010-04-23 10:44 CDT
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Summary:                    Extensions with playback stalls in different points
in bridged calls
Description: 
Hi all.
Im tryng to connect Two Asterisk Boxes, via IAX, both with version
1.6.2.6, or any 1.6.2.x ( same result ).

This is the scenario: SIP Phone -> Local Asterisk (IAX) Remote Asterisk ->
Sip Phone.

All works fine with the calls, when it terminates on a remote SIP Phone.
But, When I try to use any application on the remote server Like a
Voicemail, or Meetme, o even a simple extension that uses some Playback
statements, the dialplan stalls before few playbacks.

Absolutelly nothing is reported on the CLI, even with verbose and debug in
999.

Anyway, ive made this tests:
1) Downgrade Asterisk to 1.6.1.18, and run it, with exactly same
configuration files. For my surprise, all works, perfecly!
2) Build a VPN, and connects a SIP Phone, directly to the remote server,
and dial the test extension, made by some Playback statements. It works
nice!

To make more tests, I upgraded the local server to 1.6.2.6 again, and call
the playback tests again, on the remote server, that is using 1.6.1.18.
Works too. If I start a call from 1.6.1.18 to 1.6.2.6, the problem occours.
It means, that the problem is with the playback on the bridged call, from
1.6.2.6 box.

Ive tried to connect the servers with SIP, and test if this problem is
only with the IAX trunk, but the problem persists even with SIP, so, ive
back to my IAX Configuration.

About my System:
Two boxes, on a Dual XEON E5410, with kernel 2.6.31, 2GB RAM, and GCC
4.3.4.
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 (0120828) EduFrazao (reporter) - 2010-04-23 10:44
 https://issues.asterisk.org/view.php?id=17232#c120828 
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Ive read the HowTo, and attached the Logs..

The core show locks does not show nothing: This is the output of the two
servers, when the dialplan stalls:

pbx01*CLI> core show locks                                                
   
pbx01*CLI>                                                                
   
=======================================================================   
   
=== Currently Held Locks ==============================================   
   
=======================================================================   
   
===                                                                       
   
=== <pending> <lock#> (<file>): <lock type> <line num> <function> <lock
name> <lock addr> (times locked)
===                                                                       
                             
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Issue History 
Date Modified    Username       Field                    Change               
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2010-04-23 10:44 EduFrazao      Note Added: 0120828                          
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