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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon May 9 15:40:44 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17873 
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Reported By:                aureliors123
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17873
Category:                   Applications/app_mixmonitor
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.10 
JIRA:                       SWP-2320 
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:             2010-08-16 22:16 CDT
Last Modified:              2011-05-09 15:40 CDT
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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
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---------------------------------------------------------------------- 
 (0134701) iasgoscouk (reporter) - 2011-05-09 15:40
 https://issues.asterisk.org/view.php?id=17873#c134701 
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i have a 8mb connection most of the time and run router network analysis
which doesn't show any bandwith issues - periodic testing shows i have full
bandwith and 700Kbps.  My ISP does not 'shape' 5060 traffic.   But as you
say it's dahdi traffic that's the issue via PSTN = i use a TDM400 card.  
Strangely that's in incoming calls only.   I did see an issue on the log
about audiohook issues with dahdi and audiohook for incoming calls =  for
an outgoing call am assuming there's no bridge =   but for an incoming
call, the dialplan takes the call and then transfers/bridges the call via a
'dial' command to the sip channel/extension/phone for that context. 

There's been alot a issues/changes recently over audiohooks.   Until I
wrote my previous update to dvossel's request for info, i've never looked
at the dates around when 'my' issue starts/stop mixmonitor working, and was
very surprised and cannot more than a co-incidence that it's been every 24
days.

I have the 19th in my diary !

Ian 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-05-09 15:40 iasgoscouk     Note Added: 0134701                          
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