I hate to think of the possible echo if you change the volume and a
sip device didn't know about it. It wouldn't effectively use the echo
cancellation on board, I am not an expert with echo cancellation
however.
It should be possible to just multiply each byte of a waveform by a
percentage if
Hi,
On Fryday, 2007-02-02 François Delawarde wrote :
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes through it?
François.
On a SIP-to-SIP-Call Asterisk is not neccessarily in the voice flow,
so this does not
Hi
Is there a way to control volume in VoIP calls just like the gain
parameters for ZAP lines?
Thanks,
François.
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Yes. This is a function of the VoIP endpoint devices, not Asterisk.
François Delawarde wrote:
Hi
Is there a way to control volume in VoIP calls just like the gain
parameters for ZAP lines?
Thanks,
François.
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Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes through it?
François.
Eric ManxPower Wieling wrote:
Yes. This is a function of the VoIP endpoint devices, not Asterisk.
François Delawarde wrote:
Hi
Is there a
Perhaps you can write the functionality? I'm sure you can do a quick
hack of you modify app_voicechangedial.
On 2/2/07, François Delawarde [EMAIL PROTECTED] wrote:
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes
From: Andrew Joakimsen [EMAIL PROTECTED]
Perhaps you can write the functionality? I'm sure you can do a quick
hack of you modify app_voicechangedial.
Not sure if this is a good idea. How do you handle situations where no
transcoding is required? You don't want unnecessary heavy lifting.
