[asterisk-users] Bad Echo between SIP calls

[Mojo with Horan & Company, LLC]

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 <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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