similar to: sound

hello, if i use pasdp to run winecfg it opens but if i try it with anything else i get bash: pasdp: command not found, does anyone know why that happens?

I have sound problems in Ubuntu Studio, starting yesterday. These may be caused by a Ubuntu bug, but I wanted to run them by wine users and see if anyone else is having a similar problem. My configuration: I use Ubuntu Studio. I installed it clean this morning, with default options. Then I purged pulseaudio (because pulseaudio does not work with wine) asoundconf list shows the following

Change -- The program I test with is Dragon NaturallySpeaking. Yesterday's Ubuntu updates produced a rash of pulseaudio changes, and suddenly alsa works with wine, without stumbling. Today's secret to making it work: In System/Preferences/Startup applications, uncheck pulseaudio sudo apt-get install esound sudo nano /etc/pulse/client.conf autospawn = no Then every time your computer

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

I have a USB soundcard that I have tried to run with Dragon NaturallySpeaking. It doesn't work unless I set winecfg to oss instead of alsa. Is this normal? Am I doing something wrong? How do I get alsa over USB? Thanks, Susan Cragin

* 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

here at wkdu.org our minds are blown trying to figure out what our problem is, we are using a dell optiplex gx1 with a soundblaster live card. we are getting a nasty hum on the line in. the thing is we plug a cd player directly into the soundcard the hum goes away and when we plug the line in into anything else there is no hum, so the question i ask where is the hum coming from? we are

Hi, Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non admins from using toggle_mute to unmute themselves? If there isn't such an option on there any devs here that can ping me off line what it would

After hours of searching, I give up and will ask for directions. runing wine 1.1.27 on Ubuntu 9.04 on a T-23 Thinkpad using the onboard sound card and a Plantronics DSP-500 headset. After days of searching I finally got both to work, but limitation in Ubuntu will not allow both to work at the "same time". One or the other. Plantronics only works in a mixed mode ALSA and OSS

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > I use kmix as mixer, but there is nothing in it about a > > "capture channel". How could I find where it is defined ? > then try alsamixer or amixer and watch out for capture and unmute and apply > this for the according channel (mic, line-in, pcm or master) Hum... the problem is that RH 9 uses

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Asoundconf has been my preferred method of changing my default card, to use wine outside pulseaudio. However, Asoundconf has been removed in karmic, in anticipation of the new GNOME volume control + pulse configuration. Now, if we need asoundconf, we must download alsa-utils from jaunty, and extract asoundconf from that. For further info see this bug. https://bugs.launchpad.net/bugs/376024

http://bugs.freedesktop.org/show_bug.cgi?id=17896 Summary: no sound with 0.8.0-1 hardy w/ usb audio Product: swfdec Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: major Priority: medium Component: plugin AssignedTo: swfdec at lists.freedesktop.org

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

Hi: If you can record your mic but it's not working in XMMS, this suggests an issue on the XMMS side. Are you invoking the XMMS Recorder plugin as detailed in the Readme? Geoff. <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org'

Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

Gentleman, Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me. I had some difficult looking for a Asterisk software that provide me some functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i decided to develop than. In a few weeks i'll deploy a Beta version of this software and i'd like to know if is somebody available to try this