similar to: is multiple contexts in alsa.conf possible

Hi, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

Hi, I want to use asterisk with jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

Hello, I can't get the sound over alsa to work with Asterisk. My current version is 1.4.21.2~dfsg-3 running on debian stable. All settings are the default ones with exception of: /etc/asterisk/modules.conf: load => chan_alsa.so noload => chan_oss.so /etc/asterisk/alsa.conf: input_device=default output_device=default asterisk is started up and doesn't complain about alsa in

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav > Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit > Little Endian, Rate: 48000 Hz, mono > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >

I have setup asterisk alsa.conf to read the null device for ALSA and console/dsp input. asound.conf is pcm.nullpcm { type null } alsa.conf has input_device=plug:nullpcm Then when I call into the Console/dsp I get very choppy audio. I dont need and data from the microphone. I just want data out on console/dsp. Have I not setup the device properly? why would I be getting choppy audio

Hi some problem with chan_alsa. Depending on the configuration I don't get any sound output (output_device not set in alsa.conf - same as output_device=default) or very strange output (output_device=hw:0,0) when dialing into something like exten => 10,1,Answer exten => 10,n,Playback(soundfile) exten => 10,n,Hangup Other alsa applictions do work without problems and for example this

In console.conf.sample it says run the command "console list available" CLI command. It does not seem to be present: console list available No such command 'console list available' These are the only console commands I see: console answer Answer an incoming console call console autoanswer Sets/displays autoanswer console dial

I have searched all around and combined items in an /etc/asound.conf file to have by default a capture and multiple playback for alsa. I am looking for a way to: something like "aplay --nonblock test.wav" having at least 2 active one time and at that same time do "arecord --nonblock -d 1 input.wav" I have something like: pcm.!dmixer { type dmix

I was trying to compile alsa 1.0.14rc3 on centos 4.92 and there are errors. I have informed alsa. They dont seem to happy redhat has once again messed up alsa with their custom kernel. Just FYI. Jerry CC [M] /usr/src/alsa-driver-1.0.14rc3/misc/ac97_bus.o In file included from /usr/src/alsa-driver-1.0.14rc3/misc/ac97_bus.c:1: /usr/src/alsa-driver-1.0.14rc3/misc/../alsa-kernel/ac97_bus.c:30:

I am using alsa with asterisk. The asound.conf is below. When I start asterisk with /etc/asound.conf present I get errors on the console that: chan_alsa.c:304 alsa_card_init: snd_pcm_open failed: Invalid argument If I remove the asound.conf asterisk starts up and works. However I NEED the asound.conf for another application. What might be the issue here? THanks jerry /etc/asound.conf

I am having trouble getting wine to work with my sound. When I use winecfg it can not autodetect the right settings, and it only opens a control device, no output device: Code: hammertime boxfire # lsof /dev/snd/* COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME winecfg.e 9086 boxfire 10u CHR 116,32 5684 /dev/snd/controlC1 explorer. 9108 boxfire 10u CHR 116,32 5684

So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230908/dee530c8/attachment.html>

Thank you Rsay, it was a nice try, but I could not get this to work with my Logitech mic. For the record I have it (the microphone, that is) set up so that it works fine in Audacity. After opining Wine configure, dmesg gives me lots of these: [ yadda. yadda*] ALSA /build/buildd/linux-ubuntu-modules-2.6.24-2.6.24/debian/build/build-generic/sound/alsa-driver/usb/usbaudio.c:1345: 4:1:1:

Package: logcheck-database Version: 1.2.32 Severity: wishlist Ignore rules to supress messages generated from pugging in, and then removing, a USB headset (one speaker). ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: drivers\/usb\/class\/audio\.c: v1.0.0:USB Audio Class driver$ ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: usbaudio: assuming that a stereo channel connected directly to a mixer is

OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions. I am trying to configure confbridge to create a "conference" room with an audio stream coming from my sound card. The idea is for a group of people to be able to call in and listen to someone giving a speech but not necessarily interact. I've got confbridge configured and

Hello, I only know a nice tool for JACK, called silentjack. So if by any chance, you find the strength to use JACK anyway, here is the source code : http://www.aelius.com/njh/silentjack/ The code is very light and simple : if you have some programming skills, you might even adapt it for ALSA. Sorry for this OT answer. Hoggins! Le 09/08/2014 02:39, Dean Sauer a ?crit : > Looking to

Hello friends, I'm trying to make a simple call from asterisk CLI, but is quite confuse i followed the information here: http://www.voip-info.org/wiki/view/Asterisk+CLI+dial and changed my extensions.conf like this: alsa.conf [general] autoanswer=no context=consolecontext extension=100 By the way, how do i know if my console is using the channel driver ALSA or OSS? then, in

Hi all, Just did a fresh checkout, compiled ok and when * starts it bails with the following message: [chan_alsa.so] => (ALSA Console Channel Driver) asterisk: pcm.c:5476: snd_pcm_sw_params_set_silence_threshold: Assertion `val < pcm->buffer_size' failed. Alsa rpms installed on this RH9 box: alsa-lib-devel-0.9.3-2 alsa-utils-debuginfo-0.9.3-2 kernel-module-alsa-0.9.3a-2_2.4.20_9

Any one try alsa 1.0.11rc3 on AMD 64? I download drivers, library and utils. I did ./configure; make; make install for drivers. worked fine. I did ./configure; make; make install for libs. worked fine. I did ./configure on utils worked fine. on doing a make I get the following errors below. I did try the alsa mailing list first... All I got there was it worked on their linux system. Any