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Darryl Dunkin wrote:
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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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Thats true. Echo usually appear if your end devices uses additional
voice
amplifier. Try try to set -2 db on audio input if your gateways have
internal gain control.<br>
<br>
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cite="mid:EC5DC6B7B2274B47A28A30C5FCF87440019463DB@exchange.nosi.netos.com"
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<hr tabindex="-1"><font face="Tahoma" size="2"><b>From:</b>
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <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>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 <a class="moz-txt-link-rfc2396E" href="mailto:ddunkin@netos.net"><ddunkin@netos.net></a></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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lang="en-us">
<hr tabindex="-1"> <font face="Tahoma" size="2"><b>From:</b>
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <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>
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>
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