similar to: fix for typo in latest cvs in channels/chan_alsa.c

Is it possible to give a false ringbakc on asterisk ? -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or

is there any particular reason why there is no option to configure the codec framesizes in iax2 ? It would come rathrer handy to decide if you want less bandwidth or more robustness on the payload side ... -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is

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?!

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

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

Hi everyone. Having experimented a but with a prototype of a system I described in an earlier thread (Reading DTMF sent by callee during a SIP call), I decided 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,

-----BEGIN PGP SIGNED MESSAGE----- Greetings, I've just about got Asterisk up and running and am wondering the following. Currently, I subscribe to both Vonage and Broadvoice and as such, I've got a Sipura and Cisco ATA186. Although I'm sure this is expressly prohibited somewhere in my service agreements, can I reprogram these devices to access my own asterisk server rather than

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

Hello, I just checked out the latest CVS and compiled and now get the following error: [res_config_mysql.so] => (MySQL RealTime Configuration Driver) Jan 26 13:03:51 WARNING[27081]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/res_mysql.conf': Found Jan 26 13:03:51 WARNING[27081]: res_config_mysql.c:561 parse_config: MySQL

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>

>> 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 >