# Echo Cancellation

**URL:** <https://community.asterisk.org/t/echo-cancellation/1451>\
**Category:** Asterisk Support\
**Created:** [September 19, 2005, 11:39am UTC](https://community.asterisk.org/t/echo-cancellation/1451 "2005-09-19T11:39:42Z")\
**Posts on this page:** 4\
**Page:** 1

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**Author:** ![chitseong](https://avatars.discourse-cdn.com/v4/letter/c/f1d935/32.png) [@chitseong](https://community.asterisk.org/u/chitseong)\
**Post date:** [September 19, 2005, 11:39am UTC](https://community.asterisk.org/t/echo-cancellation/1451/1 "2005-09-19T11:39:42Z")

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Hi,

I’ve just installed a new \* for one of the department (replacing their old analogue PBX). The users keep complaining that they can hear themself even they are talking within same PBX (e.g. SIP ext to SIP ext) or PSTN call. Initially I thought that was caused by the poor quality phone without echo cancellation feature. But after changing to some other (more expensive) phones, I can still hear my echo.

I’m wondering is there parameter in \* that can eliminate the echo?

below is my zapata.conf

[channels]  
language=en  
context=default  
switchtype=national  
signalling=fxs\_ks  
faxdetext=incoming  
context=from-pstn  
echocancel=yes  
callprogress=yes  
busydetect=yes  
busycount=8  
rxgain=0.0  
txgain=0.0  
echocancelwhebridged=no  
echotraining=800  
group=0  
channel=\>2-4  
channel=\>6-8

rgds  
CS

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**Author:** ![gventer](https://avatars.discourse-cdn.com/v4/letter/g/49beb7/32.png) [@gventer](https://community.asterisk.org/u/gventer)\
**Post date:** [September 19, 2005, 10:18pm UTC](https://community.asterisk.org/t/echo-cancellation/1451/2 "2005-09-19T22:18:18Z")

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Try set your RxGain=1.5 and TxGain=1.0 - it seemed to be better for my environment.

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**Author:** ![mustardman](https://avatars.discourse-cdn.com/v4/letter/m/fbc32d/32.png) [@mustardman](https://community.asterisk.org/u/mustardman)\
**Post date:** [September 19, 2005, 11:55pm UTC](https://community.asterisk.org/t/echo-cancellation/1451/3 "2005-09-19T23:55:12Z")

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Don’t just randomly set the gains. Use ztmonitor using this procedure.  
[asteriskdocs.org/modules/tin … x1695.html](http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html)

That will not cure your SIP to SIP echo though. That can only be caused by the phone or the network or the server. I would check in that order. You said that changing to a different type of phone did not solve the problem so I would look at your network next. Are you using a decent quality switch? Are you going through more than 1 switch? What about specifications of your server. Do you have decent amounts of CPU power and memory?

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**Author:** ![chitseong](https://avatars.discourse-cdn.com/v4/letter/c/f1d935/32.png) [@chitseong](https://community.asterisk.org/u/chitseong)\
**Post date:** [September 20, 2005, 2:07am UTC](https://community.asterisk.org/t/echo-cancellation/1451/4 "2005-09-20T02:07:28Z")

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i’m using one switch for all my phones and \*. The \* is running in a P4 3GHz with 1G RAM. I guess that is good enough for the \* to run decently…

thanks for the ztmonitor tool though.