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="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <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
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