similar to: Asterisk throws error using the alsa, module

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 running pulseaudio on centOS 6.5 audio works fine as the user. running the command aplay /usr/share/sounds/alsa/Front_Center.wav works fine. when I login as root and run the command su user -c "aplay /usr/share/sounds/alsa/Front_Center.wav" it does not work. I thought su actually runs as that user... is there some "magic" to su I don't know about? How might I get

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

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 am trying to get audio to play on Pulse - so just the monitor basically. I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others. The error is always the same "console_request: Console device 'dmix' not found. What is the correct "Console/XXXX" to play on pulse for UBuntu 20.04 LTS ? I can "aplay /usr/share/sounds/alsa/Front_Center.wav" no

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

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

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

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 get a console/dsp application going with 1.4.18 and not hearing any audio. In the CLI I see the call coming in, I see the Dial(Console/dsp) I see <auto answered> I see ALSA default but I hear no audio. What can I do to tell what is happening here. I have in modules.conf: noload chan_oss.so load chan_alsa.so For kicks I tried it the other way to noload chan_alsa.so and load

Hi, I'm a new asterisk user developing an AGI application. As part of my application I'd like to play sounds on the server's speakers, but it seems that I can't do this while asterisk is running. When I try to play sounds using the play or aplay command, it blocks until I stop asterisk. My guess is that asterisk is using the sound device and this means that other programs

> > > Just to verify that you did rerun configure after installing the libraries? > > Doug > Oh that is a good one - I thought I did - but apparently not. menuconfig now shows "*" So is chan_alsa going away ? What is it being replaced with? thank you! Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

> What is the device that you're connecting to? I am trying to just play on PulseAudio actually. This used to work - I have just recently updated to 18.18.0, so I'm puzzled. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230906/eb5e9088/attachment.html>

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

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 ? ? ? ?

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:

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:

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

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are