[asterisk-users] Improve voice quality on Asterisk + chan_capi+ DIVA BRI

[Cosmin Prund]

Ok, I've made all those changes, called my operator from an outside line
and tried alternatively whispering / shouting into the mic, banging the
microphone with a metal object and pressing DTMF digits.

So far - so good, it seems to work.

I've now got an other problem. Clamping DTMF disabled my IVR! Is there
any way to enable/disable DTMF clamping on a per-call basis? Or better,
disable DTMF only when the call makes it to an operator?

--
Thanks,
Cosmin Prund

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Armin Schindler
> Sent: 19 aprilie 2007 14:35
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Improve voice quality on Asterisk +
> chan_capi+ DIVA BRI
> 
> On Thu, 19 Apr 2007, Cosmin Prund wrote:
> > Hello everyone!
> >
> > I've got a Eicon Diva Server BRI card into my "*" box working just
> fine,
> > but I wander if there's anything I can do to improve voice quality
> for
> > my operators. I'm thinking something along the lines of "auto gain"
> and
> > sudden noise suppression (like when you hit a fax machine or the
> other
> > party accidently touches the dial pad on the phone).
> >
> > Does one of Asterisk, chan_capi or the Diva driver have support for
> such
> > functionality?
> 
> Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have the
> following possibilities:
> 
> 1. Automatic Gain Control and Active Talker Evaluation in conference
> (by
>    default automatically activated with three or more parties)
> 2. Recording Stream Automatic Gain Control
> 3. Manual Control of Signal Level
> 4. Manual control of the signal pitch and/or bitrate (rate conversion)
> 5. Suppression of DTMF tones. This feature can be activated using
> adapter
>    configuration (for all calls) or on per call basis
>    This is always good to activate this feature for operators to
> protect
>    people from signals or in one gateway to prevent DTMF tones from
> passing
>    through gateway in band.
>    The DTMF tones are suppressed in the way which will not affect the
>    quality of the voice signal in case voice signal and DTMF tones
> overlap.
> 6. Part 68 Voice Signal Limiter (Required in US, by default
deactivated
> in
>    Europe). This protects the ears from "clicks" and too loud signals.
> This
>    feature can be activated using the configuration. This is good idea
> to
>    activate Part 68 voice signal limiter to protect the people. This
is
> the
>    dynamic voice signal limiter in accordance with Part 68 of US
>    requirements.
> 
> The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC of
> received signal) and the DTMF Clamping (Suppression of DTMF tones) are
> can be controlled using adapter configuration and do not require any
> change in the application (but can be controlled on the per call basis
> too, which is not implemented in chan-capi yet).
> 
> 
> Armin
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users