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<DIV dir=ltr align=left><SPAN class=309064023-11062007><FONT face=Arial
size=2>The echo cancellation card is for SIP->Zap calls only, no echo
cancellation is done in Asterisk for SIP only calls. SIP to SIP, media is
just passed through the server untouched (using media flow through, which
is the option in sip.conf of canreinvite=no) if you are not handling any
translation, even when handling translation between SIP calls there shouldn't be
any echo cancellation done in Asterisk for SIP only calls.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=309064023-11062007><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=309064023-11062007><FONT face=Arial
size=2>The place to look at would be the remote SIP devices which is
typically what is adding the echo, this is usually a gain issue of some sort
depending on which handsets you are using.</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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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> Monday, June 11, 2007 16:37<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>Sounds crazy right? even was I, more over support guy logged in
unloaded the zap modules to test them, still an echo.<BR><BR>Ya, I was clear
saying that we have SIP--- SIP issue ie internal extension echo problem.
It seems the echo with SIP--SIP has many factors. I am just curios to
eliminate any possibility of Asterisk failing to cancel the echo.<BR><BR>OK, one
question here howz the call flow when a SIP---SIP call is established ie.
is the connection between 2 phones when an Internal call is made or does the SIP
call goes via Asterisk once the SIP--SIP call is
establised.<BR><BR>--<BR>Deepak<BR><BR><B><I>Matthew Fredrickson
<creslin@digium.com></I></B> wrote:
<BLOCKQUOTE class=replbq
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid"><BR>On
Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote:<BR><BR>> Hi,<BR>>
We have a PRI
connection & when its was on test networks we <BR>> had echo problems
withoutside line. <BR>><BR>> So I bought a TE212P card resolve the
echo problem. Which did to an <BR>> extent. Its using asterisk 1.2.18
& RHEL4-Update 4.<BR>><BR>><BR>> But now when we are live, there
is a terrible echo between 2 SIP <BR>> calls. If I call the same extension
from outside the voice is clear.<BR>><BR>> I am not sure whats the
problem. Also there's slight echo when <BR>> calling Digium
support.<BR>><BR>> Totally lost Digium says we need to remove the echo
module to resolve <BR>> SIP echo problems. Then ? the heck we pay
for..<BR><BR>Are you sure that they understood that you were having this
problem <BR>between 2 SIP endpoints? That advice only makes sense to test if
one <BR>side is Zap and the other side is SIP.<BR><BR><BR>---<BR>Matthew
Fredrickson<BR>Software Engineer<BR>Digium,
Inc.<BR><BR>_______________________________________________<BR>--Bandwidth and
Colocation provided by Easynews.com --<BR><BR>asterisk-users mailing
list<BR>To UNSUBSCRIBE or update options
visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><BR>
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