similar to: usb - audio asterisk crashes

>> 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 >

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

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 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 ... 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

Hi list! Currently I use Asterisk 1.8.30.0 on an OpenWRT-Switch. Now I want to change to Asterisk 13.14.1 on a Banana PI (with Armbian/Debian 9). Well, I copied the configuration and changed what needed, so basically, it works, at least with my tests. But when Asterisk will be started, in the message log I get this error: [Feb 15 08:40:15] ERROR[3971] chan_oss.c: Unable to register channel

> > > 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:

> On Thu, May 28, 2009 at 02:00:15PM -0500, research at businesstz.com wrote: >> Hello >> >> May i please know if asterisk is now supporting sip call encryption. It >> has been a requirement from one of my client to ensure that all >> conversation is well secured from any potential sniffers or inside >> hackers >> >> I have reviewed and shall

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

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

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 have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

Hi all, on my debian box i configured chan_oss to work with /dev/audio device. CLI console command and Dial(CONSOLE/dsp) work perfectly but i notice 2 problems: 1. audio is very low in volume, even if i set 100 the mixer volume (via cmd line setmixer utility) 2. the sound is very crappy: the voice is "vibrant", words sounds like 'ttthhhiiisss iiisss aaa ttteeessstt". Seems

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 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 have a Digium TDM400 in my system and I'm using my main system as my asterisk box at home (very light load). When I start up *, though, it grabs my sound card and I cannot play other music through it (e.g. x/ XMMS). I have moved the alsa.conf and oss.conf files so that there is no configuration for them (though those files seemed to do little), but still the sound card is grabbed. How can

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

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>

I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. I am currently using MixMonitor to append the

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that