[Asterisk-Users] PSTN->voip/sip echo

[JD Austin]

JD wrote:

> I'm still relatively a novice with asterisk and am having issues with 
> echo.
> The calling party that calls a PSTN number doesnt hear the echo, but 
> the answered
> side via sip or forwarded to another PSTN number over voip hears 
> excessive echo that
> makes it difficult to communicate.
>
> I've been playing with the zapata.conf settings for echocancel, 
> echotraining, rxgain, txgain, etc
> and am basically stabbing in the dark (grin)  I've read the wiki about 
> it, but it doesn't go into very
> much detail.
>
> Anyone know which parameters fix this issue?
> Is there an easier way than tweaking settings in zapata.conf, 
> monitoring with ztmonitor, and restarting asterisk over and over?
>
> JD
>
I should add that it's only on calls that are bridged.  Calls inbound to 
PSTN don't hear echo, sip/iax in/calls out don't hear echo but when 
they're bridged (inbound PSTN outbound VOIP) the called party hears echo 
badly.
What can I do about it?

JD

-- 
JD Austin
Twin Geckos Technology Services LLC
email: jd at twingeckos.com
http://www.twingeckos.com
phone/fax: 480.288.8195