similar to: meetme mixer configuration

I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo cancellation on the POTS lines?

I have no sound in wine. I open my application (Dragon NaturallySpeaking) and it says the device is not responding. (Neither the mike jack nor the USB pod.) When I run winecfg I get the following message. fixme:wave:ALSA_ComputeCaps Device has a minimum of 2 channels fixme:mixer:ALSA_MixerInit No master control found on ThinkPad Console Audio Control, disabling mixer So it appears that my mixer

Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, Tim

Hi there! I've just downloaded a shiny new demo version of the rocking flight simulator X-Plane 7.30 [1]. Wine seems to like it, except for two flaws. It would be nice if any of you felt good enough to track down and solve any of them: - Mixer underrun: The sound mixer buffer seems to have some problems with this game when changing engine speed. I guess there this is a multi-wav mixing

Just bringing up a Centos-6 (x86_64, FWIW) and I can't find the mixer app... on my C5 box(es), double click on the speaker icon in the top panel and I get a mixer app appears on the screen. My new C6 box doesn't do that. I've installed all the PULSEAUDIO RPMS I can find, but still nothing. what am I missing here? thanks! Fred

Do es anyone know what the following means for sound under Wine? fixme:mixer:ALSA_MixerInit No master control found on MPU-401 UART, disabling mixer Severa of us are trying to get WOW internal voice chat and teamspeak to work (different uses - wow for groups, teamspeak for guild). Im wondering if the above output has a workaround until a fix is made or even if it's related to our partial

Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, Is there a way to force Wine to use a particular mixer? I'm not getting any sound via my USB audio in Ubuntu, and I'm wondering if this is because WINE is attempting to use my VIA audio mixer rather than USB Audio (which it does list within winecfg's audio tab for both ALSA and OSS). Any ideas of other things I can try to

hello, i'm looking for a mixer application as source for icecast. the app should be scriptable, e.g. it should take 2 or 3 input channels (files) at various volume levels and downmix it to one source stream for icecast. pretty much the same as a normal mixer does, but i want to control it by a kind of predefined timeline. any idea anybody? thanks, uno

Hi, I've been struggling to get everything going fine for a while now. Finally I can talk clearly and hear people. Problem is it was coming through my speakers connected to an external USB Sound Card; if I set it to come through my USB Headset, it comes through speakers anyway. I unplugged USB sound card and restarted computer. Now I tried again. This time I simply do not hear sound.

I've just installed 4.0_rc1 to evaluate and compare with FC3 for a local project. The installation and initial usage of the two distros is almost identical. up2date and yum work well on either. The one problem I have is with sound. CentOS has loaded snd_intel8x0 and all the usual suspects, but the mixers (GNOME volume controls, etc., do not work because there is no /dev/xxx for the mixer

Well, I've got multiple sound devices on my system (integrated HDA SB ATI sound card, headset connected to USB sound card, and my webcam's built-in mic). Unfortunately, Wine does not always use my integrated sound device for some reason. So, instead it will do one of two things: go to the headset or the webcam. Going to the webcam is an auto-fail, and if it goes to the headset it

Hi list, I have a customer which will organize a conference in a big meeting room which has a sound system. He would like to connect this sound system to a MeetMe room so participant in the MeetMe can act as if they where on site. My idea is to take a barbone or Notebook, connect it to the sound system using the soundcard and run a softphone on it. Does some of you already have success in

I will implement a SIP application and I'm considering using Asterisk for mixing the media streams (audio). Does anybody know if Asterisk supports or contains a RTP mixer? If so, how to use it? Just to be a little more clearer: I will send to Asterisk more than one RTP stream and they must be mixed. The result must be a single stream to be forwarded to a SIP phone or to the PSTN. Thanks,

I'm looking for some sort of mixer library to use with libao. Does anyone have any suggestions on what to use beyond what was said in August 2003[1]? My application is console-only and mixes OGGs, MODs, and AIFFs which could be of any channel count, bitness, and sample rate. [1] http://lists.xiph.org/pipermail/vorbis-dev/2003-August/016439.html -- David Griffith dave at 661.org

Hi there I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not hear their voice half a second later (because meetme mixes the voice and sends to everyone in the conference). Now because of requirements

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

I am hoping maybe some of you have come across these before in your experience with web meetme. Below are the messages im receiveing when I load the web meetme home page. Notice: Undefined variable: s in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 9 Notice: Undefined variable: logoff_section in /usr/local/apache2/htdocs/web-meetme/meetme_control.php on line 12 Notice:

I see in the archives a brief thread between Barton and w last November 2003 about streaming to the Internet. I'd like to use an Asterisk to mediate multiple VOIP calls originated from the Internet to the studio to be mixed then passed out to an encoding PC thence back to Internet {~~~~~~~~~} +---------------+ +-----------+ +---------------+ +---------------+ {

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>