On Fri, 20 Apr 2007, Cosmin Prund wrote: > 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 > ether way. > > Is there anything else I can try? Did I miss the obvious (it would not > be my first)
Can you please create a capi log: set verbose 5 capi debug to see what really happens via the interface? Armin > > -----Original Message----- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] 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 > > 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? > > > > 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. > > > > Armin > > > > > > -----Original Message----- > > > > From: [EMAIL PROTECTED] [mailto:asterisk- > > users- > > > > [EMAIL PROTECTED] 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 > > > _______________________________________________ > > > --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 > > > > > _______________________________________________ > > --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 > _______________________________________________ > --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 > _______________________________________________ --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
