similar to: Asterisk throws error using the alsa module

>> 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 >

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>

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 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 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

Joshua Asterisk 18.14.0 with chan_alsa and Console/dsp works. This does not work in 18.18.0 with chan_console enabled. I am on Ubuntu 20.04 LTS. Is there a howto for the new chan_console ? how can I get this working again ? I am trying to just play audio on pulse audio. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference "FM" is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad

I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry

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"

Hello everybody. I got some strange problem when running wine with ALSA as sound driver. For example: when I run mdk2 (I compiled wine myself version 0.9.22 - newest for now) sound is very noisy and crack'y and I get tons of this errors: err:dsound:DSOUND_MixInBuffer length not a multiple of block size, len = 3122, block size = 4 err:dsound:DSOUND_MixInBuffer length not a multiple of block

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 ? ? ? ?

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

My problem is this. I am connected via fwd's iax protocol. I try to call the echo test, sometimes I get an answer, sometimes not. When I do, sound is about one second lagged - I still hear the ring about a second after I get the answering on console message, and the echo test takes about a second or second and a half to get back the answer. When I use asterisk with dmix and dsnoop, calling

I found "console list available" === === --------------------------------------------------------- === Device Name: default === ---> Default Input Device === ---> Default Output Device === --------------------------------------------------------- === === --------------------------------------------------------- === Device Name: dmix === ---> Output Device ===

Hi all, I'm fairly new to icecast / ices - been using icecast (2.2.0) to stream audio files for a few weeks, it works very well, now I'm trying to add live feeds via ices (v2.0.1). I've built and installed (on SuSE 9.3 platform) with no apparent errors, copied ices-alsa.xml to user area and modified paths, server password etc. When I run ices (using "ices

Hi, I added some nice IP telephone programs and other sound related applications to the next version of WineTools. The problem is that only oss sound is reliable. With it it works very nice. But if during running a program another linux program accesses sound, the sound output of the Wine program is dying often does not come back. This is typical as only one program can access the oss

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

Is it possible ot have multiple contexts in alsa.conf [general] autoanswer=yes input_device=hw:0,0 output_device=hw:0,0 [usbaudio] autoanswer=yes input_device=hw:1,0 output_device=hw:1,0 is Dial(Console/usbaudio) valid? I presume Dial(Console/dsp) uses the default or general context above. how do you direct a call to the second ALSA device usbaudio? Thanks, jerry

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