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<DIV dir=ltr align=left><SPAN class=095254102-13062007><FONT face=Arial
size=2>What are the end devices? That seems to have been lost here. The real
issue is the handsets as those are the devices introducing the echo (the only
analog players here). Most likely a volume or gain issue on those handsets, what
SIP devices are the echo issues between? If both people hear echo, both devices
are at fault, if one person hears it, it is the other end at
fault.</FONT></SPAN></DIV><BR>
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<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>Deepak
Naidu<BR><B>Sent:</B> Tuesday, June 12, 2007 19:28<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>
<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>
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<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>
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<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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