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. 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.
________________________________ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Monday, June 11, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. 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. 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. -- Deepak Matthew Fredrickson <[EMAIL PROTECTED]> wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: > Hi, > We have a PRI connection & when its was on test networks we > had echo problems withoutside line. > > So I bought a TE212P card resolve the echo problem. Which did to an > extent. Its using asterisk 1.2.18 & RHEL4-Update 4. > > > But now when we are live, there is a terrible echo between 2 SIP > calls. If I call the same extension from outside the voice is clear. > > I am not sure whats the problem. Also there's slight echo when > calling Digium support. > > Totally lost Digium says we need to remove the echo module to resolve > SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ________________________________ Yahoo! Answers - Get better answers from someone who knows. Try it now <http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU> .
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