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<DIV><FONT face=Arial size=2>hi ..</FONT></DIV>
<DIV><FONT face=Arial size=2>i have the exact same problem you have
..</FONT></DIV>
<DIV><FONT face=Arial size=2>seems to be related to Budgettone phones in my
prob.</FONT></DIV>
<DIV><FONT face=Arial size=2>I *tried* selling an asterisk exchange to a
client</FONT></DIV>
<DIV><FONT face=Arial size=2>and today he phoned telling me he is very
unsatisfied</FONT></DIV>
<DIV><FONT face=Arial size=2>& I risk being thrown out .. suggestions would
be </FONT></DIV>
<DIV><FONT face=Arial size=2>welcome! i've tried *everything* that has gone
</FONT></DIV>
<DIV><FONT face=Arial size=2>in your correspondence on the list, and a
few</FONT></DIV>
<DIV><FONT face=Arial size=2>of my own .. no luck! seems a hardware
problem.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>arghhh!!!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>the other budgettone problem is it won't do a
</FONT></DIV>
<DIV><FONT face=Arial size=2>consultative transfer ie. you answer an incoming
call,</FONT></DIV>
<DIV><FONT face=Arial size=2>speak to someone else on a different extension
</FONT></DIV>
<DIV><FONT face=Arial size=2>and then pass the call .. only way I have found
to</FONT></DIV>
<DIV><FONT face=Arial size=2>"emulate" that is using call parking which is
VERY</FONT></DIV>
<DIV><FONT face=Arial size=2>messy!!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>well .. </FONT></DIV>
<DIV><FONT face=Arial size=2>maybe it's consoling to know you are not
alone!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>cheers,</FONT></DIV>
<DIV><FONT face=Arial size=2>Dave</FONT></DIV>
<DIV> </DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=dandre@iris-tech.fr href="mailto:dandre@iris-tech.fr">Daniel
ANDRE</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, August 29, 2003 5:56
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] SIP and
ECHO</DIV>
<DIV><BR></DIV>Hello,<BR><BR>Brian West a écrit:<BR>
<BLOCKQUOTE cite=midPine.LNX.4.53.0308282050210.1144@www.bkw.org type="cite"><PRE wrap="">I get no echo on mine.. but you can check to make sure your line isn't
reversed. A reverse wired jack can do that.</PRE></BLOCKQUOTE>I don't think
so but I have tested reversed and it doesn't solve my echo
problem<BR><BR>Daniel<BR>
<BLOCKQUOTE cite=midPine.LNX.4.53.0308282050210.1144@www.bkw.org type="cite"><PRE wrap="">
bkw
On Thu, 28 Aug 2003, Brian J. Schrock wrote:
</PRE>
<BLOCKQUOTE type="cite"><PRE wrap="">I can minimize doing those tricks, but I cannot seem to get it to go
away.
On Thu, 2003-08-28 at 11:33, Dan wrote:
</PRE>
<BLOCKQUOTE type="cite"><PRE wrap="">----- Original Message -----
From: "Brian J. Schrock" <A class=moz-txt-link-rfc2396E href="mailto:brians@anistonetech.com"><brians@anistonetech.com></A>
To: <A class=moz-txt-link-rfc2396E href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></A>
Sent: Thursday, August 28, 2003 6:16 PM
Subject: [Asterisk-Users] SIP and ECHO
</PRE>
<BLOCKQUOTE type="cite"><PRE wrap="">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
</PRE></BLOCKQUOTE><PRE wrap="">Hi,
For me:
rxgain=0.8
txgain=0.8
in zapata conf do the trick.
Now the echo is allmost inexistant. Maybe the sound is not very strong but
the quality is very good.
I have the default echo canceller (no modification in the source files).
Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711),
Cisco 79x0) and one X100P card.
BR,
Dan
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</PRE></BLOCKQUOTE><BR><PRE class=moz-signature cols="72">--
Daniel ANDRE (<A class=moz-txt-link-freetext href="mailto:dandre@iris-tech.fr">mailto:dandre@iris-tech.fr</A>)
IRIS Technologies - <A class=moz-txt-link-freetext href="http://www.iris-tech.com">http://www.iris-tech.com</A>
Serveur kwartz - <A class=moz-txt-link-freetext href="http://www.kwartz.com">http://www.kwartz.com</A>
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