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Echo on Linksys SP941 IP Phone

Posted on 2006-06-13
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Last Modified: 2012-05-05
Hi, I'm running an Asterisk server and configured outside calling.  When i call in using and outside phone and pick up with the Linksys SP941 SIP phone, I hear major echo.  Does anyone know how to troubleshoot/resolve such an issue?  Is it a setting on the phone or is it on the Asterisk server?

I'll greatly appreciate the help...
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Question by:jetli87
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4 Comments
 
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Expert Comment

by:christsis
ID: 16898840
When you say you're calling in is this on an FXO port in the asterisk box? T1? SIP?

If you're calling in on a zaptel port (FXO, T1, or any other PSTN connect) you'll want to check a couple things.
Make sure you get the latest zaptel driver (svn is the best but stable is fine as well)
Edit the zconfig.h and make sure you are using
#define ECHO_CAN_MG2
as the echo anceller and comment out the rest.

Recompile the zaptel driver, unload and reload the modules, and restart asterisk.

MG2 has made some huge echo cancelling improvements.

Also make sure in your zapata.conf you have
echocancel=yes
echotraining=yes (or 800)
rxgain=0.0
txgain=0.0

This is where you want to start for testing.

From here place a call and run ztmonitor from the zaptel source directory:
./ztmonitor <zaptel channel number> -v

This will show you the RX and TX gains. The general setting would be to have both about in the middle, but it varies what will help echo. Watch the ### movement to see if you are hitting extremely high or extremely low while talking. From there adjust the rxgain and txgain in the zapata.conf to increase or decrease the db as needed. By properly getting the rxgain and txgain set you will see a huge decrease in echo.

Normal situations you will see rxgain require a higher value than the txgain. It's just going to be a large amount of trial and error to find something that works on your lines.

Make sure you restart asterisk after editing the zapata.conf as a simple reload does not make the new changes take effect.

Chris
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by:jetli87
ID: 16898880
thanks in advance chrsitsis...I'll try everything you listed here.

btw, this is my first asterisk server so bare with me since i'm still a noob.

AS for calling, I'm currently testing my settings on an FXO port but will be deploying the Asterisk server using a T1 line in a few weeks.

will the echo settings have different results on either a T1 or FXO port?
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christsis earned 500 total points
ID: 16898910
No problem at all.

As far as the T1/PRI vs FXO, absolutely... however it will all depend on the telco. The T1 may or may not have echo cancellation alerady on the line. So you will pretty much have to go through the same process to find the right settings for the T1 as it's unlikely they will be the same as your POTS FXO port.

Also make sure you do lots of RTFMing as it will help your understand who, what, why echo occurs...

http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation
http://www.voip-info.org/wiki/view/Causes+of+Echo
http://www.voip-info.org/wiki/view/Asterisk+echo+avoidance

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by:jetli87
ID: 16898914
thanks for the prompt help!
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