Asterisk only supports echocancel on PSTN (zap) interfaces. Echo needs
to be canceled out at the PSTN<->VoIP interface. Calls that are all
VoIP can't have echo. If they have echo then you can, with confidence,
blame the SIP phone.
Here's another example:
Analog Phone -> SIP ATA -> Asterisk -> SIP Phone
The SIP ATA should be the device canceling out the echo.
steve wrote:
I thought the same thing before I made my reply but zapata.conf seems to
be the
only config file that deals with echo at all. From what I understand of
'echotrain'
is that at the beginning of the call it sends a short signal out that
measure echo
in an attempt to try and cancel it. I was wondering if you tried using
it and if so was it of any help?
Sincerely,
Steve
But is Zapata.conf even in the picture when all calls go through a
SIP-provider, since the calls are not being routed through a ZAP channel?
Regards,
Bjorn
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Fra: [EMAIL PROTECTED]
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Sendt: 22. mars 2006 23:11
Til: [email protected]
Emne: [Asterisk-Users] re: Sound issues on SIP-SIP calls
When you hear echo its actually being caused by the other phone. check
your zapata.conf file and see if echotraining=yes.
Echo can be caused by lots of things so this is just a starting point.
For several months now we’ve been experiencing a really strange problem
with
sound which best can be explained as choppy/stuttery, and with a touch of
echo on top. Basically, parts of a conversation might be choppy, but often
combined with some echo as well. The sound problem can only be heard by us,
not by the other party. But this is the strange part:
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