theoretically, with canreinvite=yes, it's phone <-> phone. with
canreinvite=no, it's phone <-> asterisk <-> phone. BUT there are a few
reasons which canreinvite=yes will not be this way. If for example you
have a T or a t in the Dial string, asterisk will _remain_ in the media
path so it can still detect the DTMF requests for transfer.
Moj
Deepak Naidu wrote:
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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