[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:              2011-02-21 07:46 CST
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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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related to          0018110 Playback stalls when playing demo-congr...
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 (0132238) timking (reporter) - 2011-02-21 07:46
 https://issues.asterisk.org/view.php?id=18105#c132238 
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This is happening to me as well in 1.6.2.1x, but it's more apparent when
saving voicemail. If you have an incoming IAX (trunk) and jitter buffer
enabled go directly to Voicemail then the message saved is garbled, and it
sounds like it's been played back too fast (eg recorded too slowly). I have
tried changing timing sources and it makes no difference. You can get back
perfect recordings when your turn the jitterbuffer off. I didn't think jb
was even in the loop in 1.6.2? 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-21 07:46 timking        Note Added: 0132238                          
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