[asterisk-bugs] [Asterisk 0018105]: Playback of audio file (gsm, ulaw) audio and dialplan stops when caller is inbound over iax2

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18105 
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Reported By:                n5yzv
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18105
Category:                   Channels/chan_iax2
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.13 
JIRA:                       SWP-2349 
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:             2010-10-07 11:34 CDT
Last Modified:              2011-06-04 10:25 CDT
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Summary:                    Playback of audio file (gsm,ulaw) audio and dialplan
stops when caller is inbound over iax2
Description: 
Inbound call that arrives from IAX2 channel, when call is answered by a sip
phone (tested with polycom only) works fine.  No issues.  However if call
is not answered, call is dumped to voicemail, about 1 - 3 seconds of the
the greeting (The person at extension...) is played such as "The pers....".
 This is also true for instead of sending call directly so a simple
extension such as:
exten => 888,1,Playback(en/vm-theperson)
The same thing happens.  However, if a local sip phone calls the same
extension or voicemail, everything works as expected.  I have been trying
numerous 1.6.2.x versions and have had to downgrade every time to 1.6.1.12
in order to have a functional box.
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Relationships       ID      Summary
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parent of           0018028 [patch] Exceptionally long queue length...
related to          0018110 Playback stalls when playing demo-congr...
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 (0135710) lmadsen (administrator) - 2011-06-04 10:25
 https://issues.asterisk.org/view.php?id=18105#c135710 
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The usage of res_timing_timerfd should be fixed upon the closure and commit
of code in https://issues.asterisk.org/view.php?id=18028 

Issue History 
Date Modified    Username       Field                    Change               
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2011-06-04 10:25 lmadsen        Note Added: 0135710                          
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