[asterisk-users] Bad Echo between SIP calls

[Mojo with Horan & Company, LLC]

First of all, Alex, sorry for not seeing your reply.  Nearly two weeks 
ago now :(

Honestly, with canreinvite=yes, I'm not sure what is meant by "the 
signalling still travels through asterisk"... I would ASSUME that 
includes out-of-band dtmf as well.  Sorry!

Moj

Alex Crow wrote:
> 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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