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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13801 
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Reported By:                justdave
Assigned To:                Corydon76
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Project:                    Asterisk
Issue ID:                   13801
Category:                   Applications/app_meetme
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     confirmed
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:              
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Date Submitted:             2008-10-29 13:47 CDT
Last Modified:              2008-12-04 15:00 CST
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Summary:                    [patch] 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.
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 (0095795) justdave (reporter) - 2008-12-04 15:00
 http://bugs.digium.com/view.php?id=13801#c95795 
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meh, I wrote up stuff about my environment in here yesterday in response to
DEA, but obviously didn't submit the form before closing the window or
something. :(

Anyhow, in answer to mdu113, yes, it's definitely related to the talker
optimization (or at least something triggered by it) because when I leave
out the 'o' option to MeetMe, the problem doesn't happen at all.

As for our environment, the meetings in question where people notice this
have calls coming in from all of ZAP, IAX2, and SIP, using an assortment of
Polycom phones (both desk phones from offices and speakerphones in two
physical conference rooms), softphone clients, cell phones, and landlines,
typically about 20 to 40 phone connections on the call (about 80 or 100
individuals counting people physically present in the conference rooms). 
Callers are typically auto-muted and have to dial *1 before they can talk. 
The two physical conference rooms typically dial the *1 as soon as they
join since they're conference rooms.  The problem has been observed both
with people talking slightly too far away from the phone in a conference
room and people on remote phones who seem to be talking normal, but it cuts
them off frequently anyway. 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-04 15:00 justdave       Note Added: 0095795                          
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