similar to: Underwater in 10 - 20 seconds

I know the recommendation it to not run more than 2 T/E100P's in a system but what about X100P's.. Usually there are 5 PCI slots in a system, has anyone tried 5 x X100P's in a system? Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

*Top 10 Buildings of the world<http://www.awpedia.com/newstuff/highest-building/> Tuffest, largest, highest, underwater and underground buildings * <http://www.awpedia.com/newstuff/highest-building/> [image: Top 10 Records] <http://www.awpedia.com/newstuff/highest-building/> World Most Tuff Building Earthquake Proove buildings American Steel Buildings Allied steel building The

2 things, first an idea... next a question. QUALITY MEASUREMENT IDEA: I find it difficult to hear 2 voice samples and tell which is nearer the original, especially if the background hiss is slightly different. So what if you actually subtract the post-compression sound from the original and then listen to the DIFFERENCE. If you can't hear any voice except background noise and some hiss from

Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

Hi all, I started testing the RC1 encoder at ftp://sjeng.sourceforge.net/pub/sjeng/oggdrop.exe (based on branch_monty_20010708) On the songs I have tested so far (not much :) I did not hear any stereo issues, but there are some very noticeable problems with the produced files. Songs without much high-end will suddenly have one when encoded. (you'd expect it the other way around) It

I was doing some testing with RC2 and I noticed that RC2 doesn't encode past 19kHz with this clip (-b256 and -b350). There are no problems with this clip like it was before, but this clip contains signal past 19kHz which is audible as a faint high-frequency hiss - and that hiss is gone in the encoded file since RC2 cuts off at 19kHz. I think that -b256 and -b350 should encode at least up to

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Guys (and girls), I\'ve been testing Ogg Vorbis quality at home, and I thought I\'d share the results with you. For those who want the conclusion, Vorbis 1.0 RC3 sounds better than LAME 3.70 vbr/cbr (which I\'m told now that it\'s broken for VBR), at less bitrate, however, it\'s not perfect (and probably never will, since it\'s a lossy audio codec) and could use some

Hi, I have speex running as a part of a voice conferencing app. Well, one under development anyway. I'm running VBR at quality 3 and get a "hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help

I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm having a hard time getting the kind of audio quality that I'd like. I'm hoping to be able to use SIP phones to make calls through Asterisk and have the same quality as a regular analog phone connected to the PSTN. Are my expectations too high? Is it that once I'm using Asterisk as a gateway to the PSTN, I

I think you misunderstood my quality measurement idea. I mean if you subtract the original and the one after, the LESS voice that is less over or the LESS you can tell when someone is speaking, the better the compression. This is still subjective but I think its easier to tell this way because its easier to tell how much voice is remaining than to tell how much the compressed voice is missing from

Heya all, I'm experiencing some problems with a TDM40P and was wondering if anyone else on this list has similar experiences, or maybe even a possible solution. My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected to an analog line to my telco, and a TDM40P with analog phones connected. The TDM-card is not sharing any interrupts, but the X100P is, with the 2 Adaptec SCSI

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

So, what is the story about 4KB disk sectors? Should such disks be avoided with ZFS? Or, no problem? Or, need to modify some config file before usage? -- This message posted from opensolaris.org

Hi List! Suppose I have the following, please: > setClass("buzz",representation(x="numeric"),S3methods=TRUE) [1] "buzz" > x <- rnorm(10) > class(x) <- "buzz" > plot.buzz <- function(x,y,...) { + plot.default(x,type="l",col="blue") + } > f <- function(x) { + return(x^2) + } >

I'm using CWRSync 2.6.9 protocol 29 on a Windows 2003 Standard R2 server. I'm trying to copy files onto a local usb harddrive. Both filesystems are NTFS. Rsync is failing to copy some files due to the file name length. I kind of assumed that Rsync could handle the same file name length that Windows users are allowed to enter. :-) Is there *anything* I can do with Rsync to allow these

It has long been a nuisance to me not being able to form margins on multiway tables in a simple fashion, so i wrote margins(). In my opinion it should go into the base package. The code and the documentation is in: http://www.biostat.ku.dk/~bxc/R/margins/ Please help yourself, and enhance and rename as you see fit. Bendix ---------------------- Bendix Carstensen Senior Statistician Steno

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750