search for: chan_alsa

...ssible that there's some sort of access-permission problem in pulseaudio and it's refusing to allow connections from asterisk for some reason. > > i've acticated the alsa plugin for asterisk: > > puppy:/etc/asterisk# grep -E 'alsa|oss' modules.conf > load => chan_alsa.so > noload => chan_oss.so > > puppy:/etc/asterisk# grep default alsa.conf > input_device=default > output_device=default Might try setting these to "pulse" rather than "default", perhaps? > i'm running pulseaudio on top of alsa. through setting /etc/...

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 pul...

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 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20 18:28:34 ERROR[8192]: chan_alsa...

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...

Index: channels/chan_alsa.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_alsa.c,v retrieving revision 1.2 diff -r1.2 chan_alsa.c 1042c1042 < if ((cfg = ast_load(config)) { --- > if ((cfg = ast_load(config))) { -- Michael Bielicki Managi...

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 at...

On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis <jerry.geis at gmail.com> wrote: > 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 ? > I'm not aware of one. The module itself has existed since at least Asterisk 1.8 > how can I get this working again...

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 card detected -- console channel will be unavailable But: ls /dev/snd -l total 0 crw-rw-rw-+ 1 root root 116, 7...

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>

Hi Does anyone used asterisk as a operator phone,with multiple lines and features like transfer forward and etc.I used chan_alsa driver to make asterisk as SIP Phone,but it has limitation,we cant make or receive multiple calls,and will not able to do any features like transfer forward etc. Is any other application available in asterisk to do this . Thanks Nikhil

...s. The second problem is, that the S-Bus gets jammed as well, so you can't even use a analog phone! on the NT Kernel 2.4.21rc2 with ACPI Patch and of course capi are there any reasons why this configuration should not work? And the second thing is could any of the skilled C programers get the chan_alsa to work with the latest alsa version , I tried really hard to do it myself but my C is simply not sufficient :-(.. I think the problem has something to do with the buffer readout because I am able to get the playback working, but not the recording so the switching/mixing of those 2 is not really...

...s. The second problem is, that the S-Bus gets jammed as well, so you can't even use a analog phone! on the NT Kernel 2.4.21rc2 with ACPI Patch and of course capi are there any reasons why this configuration should not work? And the second thing is could any of the skilled C programers get the chan_alsa to work with the latest alsa version , I tried really hard to do it myself but my C is simply not sufficient :-(.. I think the problem has something to do with the buffer readout because I am able to get the playback working, but not the recording so the switching/mixing of those 2 is not really...

Some unsuccessfull attempts to make console calls working. If a sip phone is called, the other side will hear nothing. If I try to record some sound the application will not finish. There is a sound file, but it is empty (0 bytes). "Record(${FILE}:gsm|10|30|skip)" is used in the dial plan. After hangup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format:

If anyone else is trying to use asterisk with the sound port AND use something else like mplayer my experience was asterisk BLOCKS the port. I added a bug item this morning to suggest a parameter control in alsa.conf and 1 line program change to chan_alsa.c of: snd_pcm_nonblock(handle, 1); Note this will always set NONBLOCK which is what I want at this time. The paramter in alsa.conf is more flexible. Took quite a while to find this (for me) hope it is valuable to others.... Jerry

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...

Hi folks, Its the forth day I'm sticking to a problem with chan_alsa, The sound played or captured from the device is choppy time to time. I mean when talking on a console/dsp microphone the other side hear my sound choppy and I'm hearing hers the same but not all the time during a call, sound sometimes are clear. Even when I'm putting the sip side on hold i...

I am using asterisk 1.4.21and svn-124910 and getting the chan_alsa:693 resource temporarily unavailable message. The audio is working but I dont recall getting any error message in the past. Is this something to be concerned about? Jerry

...cided to implement my requirement by transferring the call to another extension. This way, the callee can open the door by pressing #1, and the dial plan for extension 1 takes care of the rest. This works when I make a typical SIP to SIP call, but it doesn't when I call from the console, using chan_alsa. I can see that the transfer feature is inactive: rasterisk*CLI> core show channeltype console -- Info about channel driver: Console -- Device State: no Indication: yes Transfer : no Capabilities: 0x40 (slin) Digit Begin: no Digit End: yes Send HTML : no Image Support:...

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

On 9/8/2023 8:18 AM, Jerry Geis wrote: > But when I do a second test. Asterisk HANGS on ChanIsAvail() > > Then I thought lets SKIP that - and just let it do the Dial() - I > stopped everything - got it running again. - and then the Dial() hangs > on the second call. > > So both ChanIsAvail() or Dial() both hang on the second call in. > > So only 1 call in will work.