If the clients are not doing the echo cancellation properly, you can always use
a centralized echo cancellation software for VoIP networks.
On Wednesday, August 27, 2014 5:25 PM, Dennis Guse
dennis.g...@alumni.tu-berlin.de wrote:
On VoIP echo cancellation is basically: hope that the
It sounds to me like you should first discuss it with adtran. The standard echo
cancellation for Asterisk have a hard time cancellingecho generated at the far
end, especially if the echo tail/delay is notminimal.
If adtran can not solve the problem at their end, you can use a server-side
echo
From my experience, clicking noise can be originated from loss of audio frames.
First, verify your CPU is not loaded, then measure the frame loss to see if
this is the source of the problem.
From: Léopold Baillard leobaill...@leoserveur.org
To:
20:02, Valer Nur a écrit :
From my experience, clicking noise can be originated from loss of audio
frames.
First, verify your CPU is not loaded,
It is not.
then measure the frame loss to see if this is the source of the problem.
How can I do
Carlos,
Echo might be a possible cause of the noise but it is strange you hear it also
on internal calls since they have very low latency.
Can you record a short sample with this noise or at least ask your customer to
provide a more detailed description of the noise ?
Hi Carlos,
To solve the echo problem from your 96 analog ports, you can use the PBXMate.
Valer.
From: Carlos Alvarez car...@televolve.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 25,
Gary,
It sounds to me that this issue might relate to the quality of the hardware.
Anyway, to analyze/solve the problem you can do the following:
Install PBXMate. Modify your dial-plan to route conference-bridge calls, to go
to the PBXMate. So the flow will be:Phone-PBXMate (Sip Trunk) -
You can try PBXMate. It is more of speech improvement software (i.e. noise
removal etc.) but it also gives you speech quality statistics.
It is not a free tool but I think there is a free evaluation version.
http://www.solicall.com/products.html#PBXMate
From:
Interestingly, that isn't completely true. If it goes out a SIP trunk
to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300
(where the T1 goes), it has the same problem. This was leading me to
believe that the problem was on the 8300.
Well, that doesn't disprove my
Hi Bilal,
High volume is always a big for echo cancellation. The problem is that the
signal reaches saturation and therefore reduce the effectiveness of the
detection/convergence. If your existing echo cancellation can not handle it,
you might want to try a different algorithm for echo
Mike, if the delay in your calls (i.e. end-to-end) is over 100ms echo might be
a
problem.
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