[Asterisk-Users] SIP and ECHO

[Brian J. Schrock]

Hello,

I have read the information on echo and SIP in the FAQ and I have
scoured the mailing list for possible solutions, but as yet I have not
been able to get rid of this echo.

I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed
into an asterisk server. If I call between the Sip Phone
(Budgettone-100) and the 4 FXS ports everything sounds great. If I call
out to the PSTN through the FXO cards I get horrible echo, I have even
been able when talking loud enough to get a horrible feedback loop
going. I have tried 4 different echo cancellers in the Makefile for the
Zap drivers and nonoe of them changed the situation.

I have echocancel = (Any where from 1 - 256, I have tried alot of
different values), and I have echocanelwhenbridged = yes.I only hear the
echo start when the call gets bridged onto the outgoing PSTN lines.

Is there anything I can do?

Brian J. Schrock