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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Mar 7 08:20:41 CST 2011


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:              2011-03-07 08:20 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
----------------------------------------------------------------------
related to          0018110 Playback stalls when playing demo-congr...
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---------------------------------------------------------------------- 
 (0132687) n5yzv (reporter) - 2011-03-07 08:20
 https://issues.asterisk.org/view.php?id=18105#c132687 
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Note:  Jitter buffer settings/on/off had no affect on my issue.  I wish it
would have been that simple.  I am pretty sure it was the integration of
the new timing "setup" introduced in 1.6.1.13 (as or last functional
version was 1.6.1.12).

Ok, after being patient, we suspected that there was a patch in the works.
 I have loaded and tested 1.8.2.4 on a x86_64 on Gentoo 2.6.34-hardened-r1.
 All is fine.  The initial issue is gone.  I will be doing quite a bit of
testing in the next week, so if the power that be can leave this open for a
few more weeks that would be nice.  Looking good, thanks!  Oh, to those
that are "not happy" to go to 1.8.x, it has been very stable for me in a
production environment.  We have only delayed upgrading to it due to this
issue.  I guess we have no more excuses. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-07 08:20 n5yzv          Note Added: 0132687                          
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