search for: 10db

OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line

...handle incoming calls from the PSTN and have managed to eliminate pretty much most of the echo. My big problem is getting the outbound calls to work. When I get ZAP to dial out it won't connect and I get what I think is the Congestion signal - like a busy signal but with what appears to be a 10db alternating gain shift. (From zaptel/zonedata.c) { ZT_TONE_BUSY, "425/375,0/375" }, { ZT_TONE_RINGTONE, "413+438/400,0/200,413+438/400,0/2000" }, /* XXX Congestion: Should reduce by 10 db every other cadence XXX */ { ZT_TONE_CONGESTION, "425/375,0/375,420/375,8/375" }...

Hi Everyone, I cannot seem to get the voicemail gain option g(#) work in Asterisk 1.4.11. I am using it like so... Voicemail(4444 at mycontext,bg(10)) ; for busy announce and 10dB record gain This has absolutely NO affect on the resulting voicemail wav file. I have also tried using "format=wav" instead of wav49 in voicemail.conf to increase the volume as well. This also has no affect on the volume of the resulting wav file. Any help would be greatly appreciated,...

Hi, I've added visualization of the speex noise cancellation to my program. I did this by taking st->noise[] and st->ps[], scale both by 1.0/(st->ps_size * 32768.0) (to get a value between 0.0 and 1.0), and then draw them as a realtime lineplot. This works well, and my users like being able to see roughly what frequency bands they have noise in and compare it to their input

...ringer units. I bought a "Composer Pro" and found that it have artifacts that are associated with lower cost units. I can't be surprised as the street price is about $150.. If you don't push these units you can get some work out of them. I found that their sweet spot is only a 10dB of range. Much beyond that they start to sound crappy. I may have mentioned this on the list in the past, but Aphex makes some rather nice boxes that do very well by having a wide dynamic range and very little artifacts. You can usually find the Compellor (compressor) and Dominator (limiter) on...

I have a simple question: I want to mix N samples of 20ms into one sample of 20ms...is the algorithm above exact? : short inputsamples[160][N]; short outputsample[160]; int i,j; for(i=0;i<160;++i){ outputsample[i] = 0; for(j=0;j<N;++j){ outputsample[i] += inputsample[i][j]; } outputsample[i] /= N; }

...lity taking in account that callers will use cell phones that are most likely in 3G network coverage. So it means, I believe, either AMR NB or GSM EFR. Any hints on pre-filtering and volume normalization techniques that could be beneficiary in this case? Currently, a Sony Sound Forge speech preset -10dB is applied to normalize the volume (AFAIK) and then audio is re-sampled with SOX -t raw -e a-law -r 8000 -c 1 Free and commercial software recommendations are fine. It would be essential to get your comments (in email or by leaving a voice message) about sound quality if you could call the menu at...

Hi, I think that it would be a good thing to know more about those 2 projects (and also the future patent free format). I think that many people as me know about Lame, but not about Vorbis, and vice-versa. It would be fine that someone (perhaps the maintainer) of every project would introduce to both group of people those projects. 2 things would be interesting (to my mind): - to know about the

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues

..." request has been entered as bug #2023. > > It also appears that VM has an issue (by itself) with recording/playing > volume. Transmitting a 1004hz tone at 0db through a ata186 (set for > -1db fxs loss), and then retreiving the same VM results in that tone > measured at ~ -10db. Doing the same from a pstn location (via TDM FXO) > suggests the same -10db loss (in addition to the pstn loss). Zapata.conf > rxgain and txgain set to 0. Using CVS-HEAD-07/12/04, but same result > with CVS-HEAD-07/1/04. Entered as bug #2022. > > Add comments to either if you...

>Fromwhat you describe, your comp/limiter can't possibly be working correctly. It should be the last unit in line before the sound card, and needs to be adjusted properly. You also need to balance the levels on your mixing board (so that the correc t level comes at predictable place on the slider). It might be worthwhile to find someone with some sound-mixing or radio engineering experience