similar to: Sound not working?

Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have

Hi, I searched the Internet to find an answers, but I didn't find anything useful, so I'm turning to you, maybe someone has the answer. I want to make a realtime broadcast from a Linux box. The source is the soundcard's line-in, and it sends the stream to an Icecast server. I would like to have realtime compressor/limiter functionalities on this Linux box, so the outgoing signal

On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: > The fact that the AEC takes a few seconds to converge is normal. The Do you think there might be a way to reduce this? > fact that it needs to completely re-converge in the middle of a call > probably indicates that something went "wrong" in the audio > capture/playback. For example, that could be an

Can anyone share some ideas or point me to some documentation on how I can easily begin streaming live audio? My input would be /dev/audio (my microphone or soundcard input). Is there some type of small HTTP server that can use this /dev/audio piped through oggenc (or the ogg vorbis library built into the httpd)? Or do I use a FIFO and any regular httpd? (I am running NetBSD.) Thanks,

Darkice and Icecast2 are running, I have a cd in the box next to my server trasmitting to the soundcard line in... I connect to my IP :8000/mountpoint and get nothing, I see nothing wrong in my logs....shouldn't I be getting something? I have my soundcard settings at Line : Locked and Record selected. Here are my latest logs [2004-03-15 11:45:10] INFO main/main icecast server started

Hi, there is an interesting case when the FLAC encoder (using 1.2.0) is given simple waves. Simple waves means: I have a list of {frequency, duration, pause} tuples that define the monophonic tune. In other words, exactly one frequency is played at a time. This is the original dataset from 1989 (driving a PC speaker back then): $ ls -l ihold.sd -rw-r--r-- 1 jengelh users 20616 Aug 14 00:57

* EvilOverlord (eviloverlord@kucs.net) écrivait : > In whatever you use as mixer control for the soundcard (alsamixer, > aumix, etc) what is set as the "capture" channel? Your soundcard may > not support capturing what is being played. If it helps : my soundcard is a ES1988 Allegro-1, the module is "maestro3". I use kmix as mixer, but there is nothing in it about a

Hi, Earlier this day I started a thread about Skype not working with outgoing sound. Now I wonder if it's not a problem with my soundcard. I had a vague suspicion about this, but since it's way off the original topic, I decided to start a new thread about this. Here's what the soundcard looks like: [root at grossebertha etc]# lspci | grep -i audio 00:1b.0 Audio device: Intel

Perhaps you could add a warm-start to the AEC, such that the parameters start near the correct values on all but the first use? Stuart On 05/25/2011 08:22 AM, Andras Kadinger wrote: > 2011.05.25. 4:52 keltez?ssel, Arun Raghavan ?rta: >> On Tue, 2011-05-24 at 11:09 -0400, Jean-Marc Valin wrote: >>> The fact that the AEC takes a few seconds to converge is normal. The >> Do

Has anyone ever experienced this problem? Every 12-24 hours *one of* my mongrel processes gets a TERM signal and quits. There is all I get in my mongrel log.. ** Mongrel available at 0.0.0.0:8003 ** TERM signal received. And my rails-log is silent too. No request starting. Just the end of a successful request and then nothingness. The other processes are doing just fine and stay up for good.

This afternoon, I bought a pair of ear plugs (the large flat kind that hook onto my ears). Plugged them in to test and they work perfectly. I'm listening to streamaudio.com as I type this. However, when I tried to watch a youtube.com video (my wife exercise dancing) just silence. She had the same video on to test and the sound was fine on her box. Then, I tried system-config-soundcard and

Hi. I have two CDs that the artists inteded for simultaneous listening. They even provide instructions on how to start two CD players with the minimal time shift. (for those interested - the CDs are Neurosis/Times Of Grace and Tribes Of Neurot/Grace) I've heard that Vorbis can encode multiple streams. I've already encoded both albums into regular 2-channel .ogg files. The questions

Hello I am running Xen 3.3.1 on Debian x86 with 2.6.26 Dom0. I have a PV guest running another Debian with 2.6.18 kernel. all kernels are precompiled Debian ones. Everything work quite well. I have a modified hypervisor running to intercept syscalls issued by the PV guest and analyze the behavior of a given PID. But this is not an issue, as it works fine. My concern is about the applications

Hi, I have a RH9 system with an onboard VIA sound chip. According to the archives, VIA won't work for asterisk. So, I disabled the VIA and I purchased a Creative Labs "Soundblaster PCI 128-Voice" soundcard ($13). This card is on the "approved" RedHat list. However, the documentation inside the package says "VIBRA 128". Anyhow, kudzu doesn't see then

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > No "R" on the pcm line, so that is probably the problem : my soundcard can > > only record line1... > doesn't this mean that pcm is captured at the moment? > just try this: use aumix, go to mic, pcm or master and press space, then > this might be captured (a R appears left to the bar). I

Thanks Geoff, it's becoming more clear to me now... > So, assuming it does, you could try: > aumix -w r [yann@raglou yann]$ aumix -w R [yann@raglou yann]$ aumix -q vol 100, 100 pcm 100, 100 speaker 0, 0 line 0, 0, P mic 4, 0, P cd 0, 0, P igain 0, 0, P line1 0, 0, R phin 0, 0, P phout 0, 0 video 0, 0, P No "R" on the pcm line, so that is probably the problem : my soundcard

Hi, First at alll, I beg your pardon because maybe I explained bad my questions (because my low level english) I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2 compiled and installed. I have modules alsa 0.9.8 compiled and installed My PC has an audio card ac97 chipset intel i810 in its motherboard. I want to use asterisk in this way: |PC linux | --ethernet--

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

Hello, We've integrated the speex echo canceller into PulseAudio, and coupled with the AGC/denoiser, it works reasonably well and has been a breeze to plug in, so thank you! One thing we're seeing, though, is that the canceller seems to take some time (a few seconds) to "learn", so initially the echo is clearly audible and it slowly fades out. This sometimes occurs in the middle

Sounds like a sound driver issue. Might want to try the SMP version of OSS (www.opensound.com). Do other sound apps work OK? -bg > On Sat, 20 Oct 2001, Akos Maroy wrote: > > /me cries, bummer deal. > Yes, running as root. > The load was in the 'teens as I remember, I looked @ that too. > One CPU was like 15% and the other 11%, something like that. > 2.2.19ext3 #2 SMP,