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.
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