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ruben.zamora at zys.co... Guest
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Posted: Fri Apr 25, 2008 6:40 pm Post subject: [asterisk-users] choopy audio when both side talk at the sam |
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Hi
I have a server with the last version of asterisk branches, zaptel
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10
Grandstream GXP2000.
zapata.conf
echocancel=64
rxgain=0
txgain=0
when i place a call o receive a call, I finish a sentence i hear a
ssssssss, AND when the both side talks at
the same time i have choppy audio.
Any help i appreciate.
Thanks Ruben |
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mwatson at becon.org Guest
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Posted: Fri Apr 25, 2008 7:20 pm Post subject: [asterisk-users] choopy audio when both side talk at the sam |
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You might want to begin with tuning your rxgain and txgain settings... there are a few methods for doing this on the internet, unfortunatly nobody can give you exactly values to use for tx/rxgain as they will be not only specific to your install, but specific to every single analog line you have... you can probably get away with setting it once for all of your lines, but i'd recomend setting it for every one.
http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html
That is the best guide i;ve seen to doing it, but there may be better ones.
After you get rxgain and txgain tuned for every one of your lines, you'll probably notice a dramatic decrease in echo right away, but you can also tune your echocancel= and echotraining= after that.
You can set these values on a per-channel basis by doing it like:
rxgain=6.3
txgain=-1.0
channel => 9
rxgain=7.595
txgain=-2.0
channel => 10
rxgain=6.3
txgain=-1.136
channel => 11
etc.
First step IMHO is getting your rx/txgain set properly... don't underestimate how important those values are... I learned that the hard way.
--
Matt
________________________________________
From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Ruben Zamora [ruben.zamora at zys.com.mx]
Sent: Friday, April 25, 2008 7:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] choopy audio when both side talk at the same time
Hi
I have a server with the last version of asterisk branches, zaptel
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10
Grandstream GXP2000.
zapata.conf
echocancel=64
rxgain=0
txgain=0
when i place a call o receive a call, I finish a sentence i hear a
ssssssss, AND when the both side talks at
the same time i have choppy audio.
Any help i appreciate.
Thanks Ruben
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