[Asterisk-Dev] Volume control ?

Kris Boutilier Kris.Boutilier at scrd.bc.ca
Wed Oct 12 15:50:41 MST 2005


> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of José Pablo
> Ezequiel Fernández
> Sent: Wednesday, October 12, 2005 2:46 PM
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] Volume control ?
> 
> 
> Since we get calls from our Zap card at very different volumes (from time to 
> time the other party) can't be heard, we decided to implement RTP volume 
> control, similar to that of meetme, but after some days reading and 
> commenting code, I still don't see a way to do it. I am tring 
> to do something that will be too hard ?
> Any ideas or tips as to how to implement it will be very welcome.

If it's specific to zaptel interfaces you can patch Asterisk to be able to fiddle with the channels gains on the fly. I contributed something along these lines, but for console based realtime gain control, in bug http://bugs.digium.com/bug_view_page.php?bug_id=0002784  

There has also been ongoing discussion of implementing a channel independant, per-call gain control for use with Voicemail (see http://bugs.digium.com/bug_view_page.php?bug_id=0002023). 

Neither of these items were ever accepted, rather the recommendation was to resolve the root cause of amplitude deviations - specifically bug #2023 provided ample evidence of issues with volume level when switching between the different recording formats. 

Hope that helps.

Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District


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