similar to: RTP mixer in Asterisk

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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

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

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

-----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.

Hi, Matt and Asterisk gurus I encountered the same problem in my asterisk meetme. Whenever the 3rd person joins the meeting, it creates echo in the meeting, while 2 person meeting is fine. I am wondering if you can give me more hint on how to configure the mixer to have echo cancelled. We are using analog phones connected to asterisk TDM cards. Thanks a lot. Michael

I know this is not the most appropriated list to this, but I will try: Does anyone know what is the criteria to the generation of the transaction identifiers in MGCP? I mean, are they generated by a randomic method? I'm using Asterisk and MGCP eyeP Phone and observed that the RSIP and NTFY (methods created by the gateway) use high values in the transaction identifiers, while the RQNT, AUEP,

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

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, I've created a test at "extensions.conf" like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the

Hello All, Has anyone ever seen this before. This only happens when i'm on phone call -- Zap/2-1 is ringing -- SIP/2203-c48d is ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan

Is anyone working on Vorbis over RTP? It's been silent on this frontier for a while. I have been hearing from representatives of the IETF that they were not impressed by the previous draft, and that they were not coherent with RTP thinking and other RTP transport specs for various content. There exist a textbook on RTP which was released recently, if someone wants to learn it from the ground

Hello Vorbis Developers. I'm in the proccess of writing an OggVorbis audio player using libao and libvorbisfile. I've greatly appreciated the simplicity of the API and good documentation with the souce of the libraries. I've descided that I really want a mixer on my player, and I'm unsure how to best go about it. LibAO (unless I'm absolutely blind) doesn't