<div>I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls.</div> <div> </div> <div>To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial & no echo. So ya dont know whats wrong.</div> <div> </div> <div>Thanks all for your inputs & sharing ur experience.</div> <div> </div> <div>--</div> <div>Deepak<BR><BR><B><I>Darryl Dunkin <ddunkin@netos.net></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px;
MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"> <META content="MSHTML 6.00.2900.3059" name=GENERATOR> <DIV dir=ltr align=left><SPAN class=704192400-13062007><FONT face=Arial size=2>This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver.</FONT></SPAN><BR></DIV> <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left> <HR tabIndex=-1> <FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Matt<BR><B>Sent:</B> Tuesday, June 12, 2007 16:44<BR><B>To:</B> Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re: [asterisk-users] Bad Echo between SIP calls<BR></FONT><BR></DIV> <DIV></DIV>I don't see this listed anywhere here in the replies so.<BR><BR>In your zapata.conf file try changing:<BR>echocancelwhenbridged=no<BR><BR>to:<BR>echocancelwhenbridged=yes<BR>_______________________________________________<BR>--Bandwidth
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