similar to: strange sound when bridging Zap

Jean-Marc Valin (jean-marc.valin@hermes.usherb.ca) wrote: > > > I've been using Speex in my voice-over-IP program on Win32, in > > wideband (16kHz) mode. I just starting using VBR recently and > > have run into something that might be a problem within Speex: > > Are you turning on DTX in addition to VBR? Also, what version are you > using. As of 1.0, DTX is

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

I'm running FreeBSD 4.2. I downloaded and installed Icecast 1.3.10, Liveice, mpg123, and Lame. I know the Icecast streamer is working because I can feed it from WinAMP on my PC and listen to the stream somewhere else and it sounds fine. I know liveice is working too because Icecast shows that it connects and is sending audio to it. However when I listen to the stream on my PC with WinAMP

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

Hi, I've been using Speex in my voice-over-IP program on Win32, in wideband (16kHz) mode. I just starting using VBR recently and have run into something that might be a problem within Speex: If someone hasn't spoken for a little while, and the bitrate drops to very low, sometimes the high half of the spectrum becomes frozen with a looping sound. The bottom half of the spectrum is

I've used a Sound Blaster AWE32 / Liveice combo before. Sounds like your record volume is set WAY to high or more likely you are recording from the wrong input. Make sure that it's the _line_in_ that's selected for recording in you mixer app. Also If i remember you have to use HALF_DUPLEX mode (in liveice.cfg) at least if you're using MIXER mode... I'm not sure about in

The code below illustrates some points about results from tapply that I find strange. I wonder if they are intended and if so why it is so. 1) When you make a table the dimnames is a *named* list, tapply returns an unnamed list. 2) data.frame behaves differently on an array and a table. Is this an intended feature? 3) For tables class(TAB) and attr(TAB,"class") both return

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

I've got a real problem with the debian (unstable) wine 0.0.200202228-1 Whenever I try to run a windows executable - of any sort, although it's Jeskola Buzz in the example I've attached - wine cheerfully spits the dummy. I realise that the DRIVE_Init warnings are ignorable, but #warn:dosfs:DOSFS_FindUnixName 'config\system' not found in ... #warn:dosfs:DOSFS_FindUnixName

Your distro probably came with an sound mixer application. I use aumix on the console which i had to go get, but you may already have, see if you have a man page. Else you can search freshmeat.net for it. You will need to make sure _line_in_ is set for recording and set the _rec_ volume at 1/2 to start and adjust by trial and error. NOTE: the _line_in_ Volume adjusts how loud you HEAR the source,

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) + } >

How do I change the record volume. I am doing all this from the console, no GUI. I have not seen anything in any of the config files to set the record volume. Matt -----Original Message----- From: owner-icecast@xiph.org [mailto:owner-icecast@xiph.org]On Behalf Of harvey smith Sent: Wednesday, March 21, 2001 10:43 PM To: icecast@xiph.org Subject: Re: [icecast] Liveice & Icecast...help

Hi, I am a newbie in LLVM. I am very impressed with this forum and appreciate your help and time. I am trying to include llvm IR parser in my codebase, the way I wanna do is generate llvm's shared object (.so) file and use it in my project. So far I haven't been able to generate correct .so's. When I build a debug build with gmake (have llvm and clang in my sandbox), I get the

I was thinking about running Wine on Interix (the POSIX layer of Microsoft's Services for Unix). There are a few apparent unknows: Would Wine load the .dll.so's? Since Interix uses PE for it's native format (running on Windows, duh), Would Wine load the .dll.so's? Would it reject them being in PE format and having the 'POSIX Layer' attribute set, or could we hack on a

hello, please 'scuse the slightly offtopic question, but i see a lot of posts about the adit600, used as a channel bank, but from what i understand it can be used as a PRI interface as well. If anyone who is using the adit600 to interface to 4 T1/E1's has feedback, i would appreciate it, specifically involving the asterisk interface, echo, and DTMF. thanks, yair

Rich, Thanks a bunch, totally understand now and that actually makes total sense. (no need for schematics). This also explains why I used an TA750 to go into a Nortel MICS system, using FXO and no buzz. Totally balanced load from the analog ports on the Nortel across the 5 feet of CAT5 to the FXO on the adtran. Now I need to get rid of some Adtrans --- Anyone lookin to buy? :) Thanks

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of

All: Any have sample config's for an ADIT600 w/ CMG card connected to an Asterisk box via MGCP? Thanks, Garrett Smith <mailto:gsmith@voipsupply.com> gsmith@voipsupply.com VoIPSupply.com -- a division of B2 Technologies, LLC (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales VoIPSupply.com is North America's largest VAR

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