[asterisk-users] Bad Echo between SIP calls

[Deepak Naidu]

Ya, I have done that, below is zapata.conf.  Also we had an TMP card with analog lines. & SIP cals were great on them. & now when we switched over. SIP calls have echo.. which shouldnt be at all.

[channels]
language=en
#include zapata_additional.conf
context=from-pstn
switchtype=national
pridialplan=national
signalling=pri_cpe
faxdetect=incoming
usecallerid=yes
echocancel=yes
callerid=asreceived
echocancelwhenbridged=no
echotraining=128
;rxgain=-3.0
;txgain=-7.0
group=0
channel=1-23

--
Deepak

Alex Balashov <abalashov at evaristesys.com> wrote: On Sat, 9 Jun 2007, Deepak Naidu wrote:

> 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.

   My impression is that the transcoding that takes place between two
purely software SIP calls never goes through the TE212P card.

   There are probably echo cancellation options you can enable that are
relevant to software channels.  I distantly recall there even being some
stuff youc an uncomment in the source.

> Totally lost Digium says we need to remove the echo module to resolve 
> SIP echo problems. Then ? the heck we pay for...

   Not sure why Digium would say that.

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671
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