[asterisk-bugs] [Asterisk 0017066]: Adaptive Jitter Buffer issue

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17066 
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Reported By:                globalnetinc
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17066
Category:                   Core/Jitterbuffer
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.6 
JIRA:                       SWP-1129 
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:             2010-03-20 19:32 CDT
Last Modified:              2010-09-11 11:13 CDT
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Summary:                    Adaptive Jitter Buffer issue
Description: 
After upgrade to 1.6.2.6 the adaptive part of th jitter buffer works for
the first time in 1.6.2.x.  It does however have some issues. The first
word or two are broken up.  In the log these errors are created for every
call.
[Mar 20 18:28:21] WARNING[25522]: abstract_jb.c:469 create_jb: Failed to
put first frame in the jitterbuffer on channel
'SIP/VoIP360Residential-00000001'
[Mar 20 18:28:21] WARNING[25522]: abstract_jb.c:469 create_jb: Failed to
put first frame in the jitterbuffer on channel 'SIP/4065511212-00000000'
[Mar 20 18:28:21] WARNING[25522]: chan_iax2.c:1094 jb_warning_output:
Resyncing the jb. last_delay 0, this delay -40544659, threshold 500, new
offset 40544659
[Mar 20 18:28:21] WARNING[25522]: chan_iax2.c:1094 jb_warning_output:
Resyncing the jb. last_delay 0, this delay -13990, threshold 500, new
offset 13990

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 (0126856) avalentin (reporter) - 2010-09-11 11:13
 https://issues.asterisk.org/view.php?id=17066#c126856 
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Hi!

I can confirm the problem, too.
Asterisk:
1.8.0 Beta4 / SVN 286266

Aasta 55i 57i
SNOM 360

Leads to no audio.
It happens if I do an attended transfer of a dahdi channel from one sip
phone to another. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-09-11 11:13 avalentin      Note Added: 0126856                          
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