similar to: ALSA Module Sample Rate

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

>> 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 just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

Hi! I am interisted to dial out from the console with chan_alsa. Can somebody of you help me according this problem?! I added user the asterisk to "pulse" and "pulse-access", and it didn't change anything. alsa applications are routed by default to pulse. cat /etc/asound.conf pcm.!default { type pulse } ctl.!default { type pulse } What might be the problem?!

Hello Everybody I got some little issues with the sound. Logged in as my normal user - I've the testsound (winecfg) on the internal device. I can't change it. If I log in as root I've the testsound on my external USB speakers (the way I would prefer it works). I can't Imagine me - why there is a different between root and my normal user. Please find below some details. Code:

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

asterisk-1.4.7, Fedora 7, intel emt64 - nocona: == Parsing '/etc/asterisk/alsa.conf': Found ALSA lib pcm_dsnoop.c:558:(snd_pcm_dsnoop_open) unable to open slave [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:365 alsa_card_init: snd_pcm_open failed: No such file or directory [Jul 15 10:12:23] ERROR[24420]: chan_alsa.c:481 soundcard_init: Problem opening alsa I/O devices == No sound

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

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

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

I recently upgraded a server to Asterisk 1.4.22 with OpenR2. Previously I was using 1.4.18. It seems that 1.4.22 has a big bug using chan_alsa.so for overhead paging. After rebooting the server it would work once or twice and then I just got an error on the CLI: [Jan 7 10:35:14] ERROR[26164]: chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable I had to switch to chan_oss

Hello, 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 have and audio card ac97 chipset intel i810 in motherboard. The list of the modules loaded is: namor:/etc/asterisk# lsmod Module ? ? ? ? ? ? ? ? ?Size ?Used by ? ?Not tainted snd-pcm-oss ? ? ? ? ? ?35652 ? 0 snd-mixer-oss ? ? ? ?

Hi All, I updated * from CVS today (ran make clean, install successfuly ) and when I start it in console mode , it hangs forever trying to the load chan_alsa. A snippet of the console log is provided below,after the alsa module description, it justs sits . I had asterisk working before to todays version. Does anyone else have a similar problem, I am running stock RH9, with ALSA drivers for

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

Hi, i want send sound to my yamaha amplifier trough digital output from my audio-card. When I make: speaker-test -Dplughw:0,1 -c2 i can hear the sound. If I set the DevicePCM1 to default i can hear sounds from analog output trying to set hw:0 hw:0,1 plughw:0,1 didn't produced any sound (analog nor digital). How should I setup Alsa Driver in the wine registry? asound.conf and .asoundrc is

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

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

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

hi wine community, i recently wrote to this list in my effort to get traktor dj studio to work with wine. many thanks to robert reif, who got it fully working with the wineoss driver and native oss. the problem was, that wineoss needs mmap access to the soundcard to correctly initilize dsound. dsound is needed for 4channel playback (master&headphone) my configuration is fc2 with kernel

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