similar to: Transfer call placed from console (with chan_alsa)

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?!

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 Managing Director TAAN Consultants Ltd http://www.global-gateway.net/

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

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

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

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

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

Hello there I have a serious issue with the AVM Fritz PCI V2 I have the following setup and the problem is, that the kernel freezes hard after about 16 hours. 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

Hello there I have a serious issue with the AVM Fritz PCI V2 I have the following setup and the problem is, that the kernel freezes hard after about 16 hours. 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

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 hear the same choppy music

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

The Asterisk Development Team has announced the release of Asterisk 11.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.21.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

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 guys I just did a ps -Af | grep asterisk on the machine and got several screens full of this: root 6970 6946 0 13:10 ? 00:00:00 rasterisk rxcore show channels verbose root 6987 6948 0 13:10 ? 00:00:00 rasterisk rxcore show channels verbose root 7005 6985 0 13:10 ? 00:00:00 rasterisk rxcore show channels verbose root 7021 7003 0 13:10 ?

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

I've been trying to get a console channel working without success. The sound card, which is built into the motherboard, is a VIA Technologies, Inc. VT82C686 AC97 Audio Controller. Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I just get beeps and screeches. Using alsa drivers (snd-via82cxx) and chan_oss (using the alsa oss emulation), playing sound works, but

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

I need to set up the sound card of a server to use in an overhead paging system, as normal I am testing this on my home machine first (which has slightly different Hardware). I'm using chan_alsa with the Intel HD Audio driver on an Intel 82801G (ICH7 Family) sound card. I am running Asterisk 1.4.17 and have a fully loaded TDM400P as a timing source. When calling console/dsp (using

The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF integration feature that can be enabled for dedicated voice mail ports. What I want to do is connect an X100P FXO port to a jack on the Panasonic and make use of the Panasonic's DTMF call progress tones that it provides in DTMF integration mode. The integration works well when a Panasonic extension is forwarding into