similar to: alsa issue with 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 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

I found this information elsewhere.... making it /root/.asoundrc helped. # This is .asoundrc # # this makes legacy OSS apps use alsa software mixing dmix pcm.dsp0 { type plug slave.pcm "dmix" } # mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;) ctl.mixer0 { type hw card 0 } # this makes native ALSA apps default to using dmix pcm.!default {

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?!

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

>> 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 >

All, I'm trying to setup asterisk to run off of a raspberry pi and use the alsa module to access the sound card (a usb logitech 270 camera/mic). Everything appears to be working except the audio quality is very (very) poor and there is an error from the asterisk conosle: Loading chan_alsa.so. == Parsing '/etc/asterisk/alsa.conf': Found [Sep 6 17:57:29] WARNING[4562]:

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, I can’t seem to find a solution for my last VM issue. I have distorted sound, when I play the Windows 10 device test sound. Only the Windows sound is affected. Mpd of the host is playing just fine meanwhile. I am using ich9 as emulated card, which is detected and installed correctly. The VM is using vfio-igd passthrough, so vnc or spice are not used. I tried to match the Windows driver

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:

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

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

I use Wine for play World of Warcraft, that support on Windows, 6 channel sounds mixing software. On Ubuntu 8.04 and Wine i get only stereo output. I've tried to do this .asoundrc for "force" use on Wine, of 6channels and I don't get any sound. Code: pcm.!default { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions

Hi, I found the main reason for sound distortions on my system is guest timer configuration. The working one is: <clock offset="localtime"> <timer name="hypervclock" present="yes"/> <timer name="hpet" present="no"/> <timer name="rtc" present="yes" track="guest"

Hey' Can't seem to compile libao. I'm using Slackware-7.1-current-ish, linux-2.4.2, gcc-2.95.2, and alsa-9.0-beta2. This is the first time I've tried to compile libao so I don't know if the problem is related to alsa-9.0 changes. Alsa headers seem to be there: /usr/include/linux> ls -l as*.h -rw-r--r-- 1 root root 35994 Mar 3 10:57 asequencer.h -rw-r--r-- 1

autogen.sh | 2 +- configure.ac | 39 +++++++++++++++++++++++++++++++++------ player/.gitignore | 1 + player/Makefile.am | 28 +++++++++++++++++++--------- player/swfdec_playback_none.c | 38 ++++++++++++++++++++++++++++++++++++++ 5 files changed, 92 insertions(+), 16 deletions(-) New commits: diff-tree

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

I am experiencing what appears to be this same problem. However, I am already running a self-compiled Wine, so that alone cannot be the solution in this case. I am running 64-bit Debian, tracking testing. I have compiled Wine myself, from git revision b05be807583eca27dc0a4de896ba901bac54088c ("Release 1.4-rc5."). As with the previous report, winecfg reports that it's using

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 ? ? ? ?

On 16-01-13 21:52, Clemens Ladisch wrote: > Rene Herman wrote: >> I'm out of touch of with alsa -- but should dmix/dsnoop still be >> enabled by default on hardware that doesn't do hardware mixing? > > Yes. > >> I believe that it should be from /usr/share/alsa/cards/ICH4.conf, >> but I don't get mixing on this setup when using the