[asterisk-bugs] [Asterisk 0017873]: Record audio channels gets out of sync

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17873 
====================================================================== 
Reported By:                aureliors123
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17873
Category:                   Applications/app_mixmonitor
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.10 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-08-16 22:16 CDT
Last Modified:              2010-08-17 14:02 CDT
====================================================================== 
Summary:                    Record audio channels gets out of sync
Description: 
When recording with mixmonitor, some record files channels gets out of
sync.
The scenario was:
- Medium call traffic: More than 60 and less than 120 concurrent calls.
- All calls are recorded
- Codec: Alaw
- Record File format: WAV GSM
- Processor load: Never more than 50%
- Asterisk Version: 1.6.2.10 (also happened in older versions, since
1.4.19)
- SO Debian 5.0.3 - 2.6.26-2-686
====================================================================== 

---------------------------------------------------------------------- 
 (0126028) lmadsen (administrator) - 2010-08-17 14:02
 https://issues.asterisk.org/view.php?id=17873#c126028 
---------------------------------------------------------------------- 
As with the last issue please provide the dialplan, sip.conf, and scenario
so we can reproduce the issue. I'd also be curious to know what kind of
system you're using. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-17 14:02 lmadsen        Note Added: 0126028                          
2010-08-17 14:02 lmadsen        Status                   new => feedback     
======================================================================