[asterisk-users] Bad Echo between SIP calls

Alex Crow acrow at integrafin.co.uk
Tue Jun 12 13:28:26 CDT 2007


Moj,

Does this mean that even out-of-band DTMF still gets sent
SIP-phone<-->SIP-phone without Asterisk hearing them? (eg RFCxxxx DTMF,
can't remember the number right now)

Forgive me for butting into this thread but this is interesting...

Cheers

Alex


On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan & Company, LLC wrote:
> 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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