[asterisk-users] Improve voice quality on Asterisk +
cosmin.prund at adicomsoft.ro
Fri Apr 20 02:42:51 MST 2007
I've implemented my IVR using an FastAGI thing, using the READ
application. "core show application read" shows no information on how
the read function gets it's digits, I assume it does it the right way.
With DTMF clamping off it works, with DTMF clamping on it no longer
works. I've also toggled the "softftfm" setting in capi.conf, no luck
Is there anything else I can try? Did I miss the obvious (it would not
be my first)
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Armin Schindler
> Sent: 20 aprilie 2007 12:32
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Improve voice quality on Asterisk +
> chan_capi+DIVA BRI
> On Fri, 20 Apr 2007, Cosmin Prund wrote:
> > Ok, I've made all those changes, called my operator from an outside
> > and tried alternatively whispering / shouting into the mic, banging
> > 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
> > any way to enable/disable DTMF clamping on a per-call basis? Or
> > disable DTMF only when the call makes it to an operator?
> This is possible, but such a command/feature must be implemented into
> chan-capi first.
> Anyway, even with DTMF clamping the DTMF detection is activated. So
> Asterisk should get the DTMF infos. Or is your IVR doing own DTMF
> detection on voice data? If yes, you should change that.
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-
> > > 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
> > > fine,
> > > > but I wander if there's anything I can do to improve voice
> > > for
> > > > my operators. I'm thinking something along the lines of "auto
> > > 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
> > > such
> > > > functionality?
> > >
> > > Sure, with the Dialogic (Eicon) DIVA Server card DSPs, you have
> > > following possibilities:
> > >
> > > 1. Automatic Gain Control and Active Talker Evaluation in
> > > (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
> > > 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
> > > passing
> > > through gateway in band.
> > > The DTMF tones are suppressed in the way which will not affect
> > > 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
> > > This
> > > feature can be activated using the configuration. This is good
> > > to
> > > activate Part 68 voice signal limiter to protect the people.
> > is
> > > the
> > > dynamic voice signal limiter in accordance with Part 68 of US
> > > requirements.
> > >
> > > The Part 68 Limiter, Audio Recording Automatic Gain Control (AGC
> > > received signal) and the DTMF Clamping (Suppression of DTMF tones)
> > > can be controlled using adapter configuration and do not require
> > > change in the application (but can be controlled on the per call
> > > too, which is not implemented in chan-capi yet).
> > >
> > >
> > > Armin
> > > _______________________________________________
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