[asterisk-bugs] [Asterisk 0013801]: No way to tune talker optimization in meetme, causes users to get cut off while they're still talking

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Oct 29 15:44:40 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13801 
====================================================================== 
Reported By:                justdave
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13801
Category:                   Applications/app_meetme
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.22 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-10-29 13:47 CDT
Last Modified:              2008-10-29 15:44 CDT
====================================================================== 
Summary:                    No way to tune talker optimization in meetme, causes
users to get cut off while they're still talking
Description: 
I enabled 'o' talker optimization on my conference rooms because the
documentation in 1.4 says the feature will be permanently enabled in
Asterisk 1.6 with no way to disable it, so I figured we should probably get
used to it. However, if it works like this we'll have to never upgrade to
1.6.  We get constant complaints about people getting cut off while still
talking in the conferences, and I can't find any way to tune what it
considers "talking". If the feature is going to be permanently enabled, we
at least need some way to tune how sensitive it is.
====================================================================== 

---------------------------------------------------------------------- 
 (0094378) mdu113 (reporter) - 2008-10-29 15:44
 http://bugs.digium.com/view.php?id=13801#c94378 
---------------------------------------------------------------------- 
I can confirm the problem. It's actually so bad that after a few
experiments we have completely given up on using that optimization feature.
It may be ok for providing free/cheap conferences, but not for the cases
when you need good voice quality.
I vote for making it optional. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-10-29 15:44 mdu113         Note Added: 0094378                          
======================================================================




More information about the asterisk-bugs mailing list