[Asterisk-Users] feature - VM gain adjust?

Chris Shaw chriss at watertech.com
Mon Jul 12 08:43:19 MST 2004


Hmmm... I don't know if playing with the * code would really be the best
here... Although if it was a plug-in app like app_volume or something I
guess it couldn't hurt... It really sounds like you have a line issue here.
You said that adjusting the gain on your card introduced echo issues. It
sounds like you have an impedance mismatch/imbalance. Like your telco is
trying to cut corners going from a 4-pair to 2-pair or doing some creative
splitting... Do you possibly know where the source of the echo might be
coming from? Maybe somewhere under your control? If not it can be a pain
getting the telco to acknowledge/fix the problem.

Most proprietary PBXs even would have this problem, although they usually
don't introduce so much attenuation as your FXO card seems to be doing... I
know I know * is way better than a PBX and it should be more flexible. I'm
just saying that normally there's no way short of getting the damn telco to
fix the problem or getting your own ISDN (T1 if you're in the
Telco-Logically backward USA like me) with channel bank... Even then they
don't always work...

Just my $0.2 ...



----- Original Message -----
From: "Rich Adamson" <radamson at routers.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, July 12, 2004 5:46 AM
Subject: Re: [Asterisk-Users] feature - VM gain adjust?


> > > Are you suggesting such a thing exists, or that that would be a
> > > proposed future application?
> >
> > I propose to think if an AGC / dynamic compressor could be used instead
of
> > a config variable.
> >
> > Most sound editors have modules for this.
>
> So how would you detect the remote caller is 14.7 db away from *
> and adjust the 'outbound' voice message to be at some higher
> audio level?
>
> I like the AGC approach, but I'm not sure its realistic in terms of
> consistently being able to identify the transmission loss from
> each and every vm call. Since we know what the loss is for each
> pstn line (to the central office), it would appear that static
> value would be a good starting point and the user could adjust from
> there. Much easier (and more likely) to implement.
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list