[asterisk-users] Bad Echo between SIP calls

[Deepak Naidu]

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 <creslin at digium.com> 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.

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