similar to: chan_alsa - no sound / strange sound - 1.4.9

>> 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, I want to use asterisk with jack audio. I tried the next configurations : 1 ) app_jack.so -> does not work. 2 ) chan_alsa.so without jackd started my .asoundrc : pcm.!default { type hw card 1 } pcm.jack1 { type jack playback_ports { 0 alsa_pcm:playback_1 1 alsa_pcm:playback_2 } capture_ports { 0 alsa_pcm:capture_1 1

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:

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

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb Steve Matzura: > > On 5/24/2023 7:49 AM, Stefan Tichy wrote: > > Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura: > > > > > 1. Still can't register my phone > > > The username and password are correct. I don't know what else to try. > > You can start a sip trace from the asterisk

OK, somebody may have a much better way of doing what I'm attempting. If so, I'm open to suggestions. I am trying to configure confbridge to create a "conference" room with an audio stream coming from my sound card. The idea is for a group of people to be able to call in and listen to someone giving a speech but not necessarily interact. I've got confbridge configured and

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?!

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

How is it possible to get a dialtone using chan_misdn for a ISDN phone connected to a hfc nt-mode card? misdn.conf: [intern] ports=2 context=isdnIntern immediate=yes extensions.conf [isdnIntern] exten => s,1,DigitTimeout(5) I don't want to use answer here because the phone does not show the dialed digits in the display if the call has already been answered. -- Stefan Tichy

Hi, using HFC cards, mISDN/chan_misdn and spandsp lib fax retrieval works, but some log file entries are missing. There should be one of the lines: Fax successfully received. Fax receive not successful. Dail Plan config used: [fax] exten => _.,1,SetVar(FAXFILE=.............) exten => _.,2,SetVar(LOCALSTATIONID=......) exten => _.,3,rxfax(${FAXFILE}) exten => _.,4,NoOp,XYZ exten

WARNING[<pid>]: CAPI[contr3/123456]/178 already has PBX structure?? WARNING[<pid>]: CAPI[contr3/123456]/178 already has a call record?? WARNING[<pid>]: CDR on channel 'CAPI[contr3/12345]/177' already started WARNING[<pid>]: Thread 1109916592 Blocking 'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in procedure ast_waitfor_nandfds

Hi Dominique On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: > So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only > count 2. What can be the reason? It is not silence. Are you sure? The value for silencethreshold (140) is unusually large. -- Stefan Tichy ( asterisk3 at pi4tel dot de )

Hi Stefan, Stefan Tichy <asterisk3 at pi4tel.de> schrieb am Mon, 26. Jan 23:56: > Hi Dominique > > On Mon, Jan 26, 2015 at 04:37:23PM +0100, Dominique Haeber wrote: > > > So, from 15:24:04 to 15:24:10 there are 6 seconds. But asterisk only > > count 2. What can be the reason? It is not silence. > > Are you sure? Yes, im sure. I have looked at the time and

Hi Karsten, On Thu, Apr 30, 2020 at 05:50:39PM +0200, Karsten Wemheuer wrote: > .... The server sends Server Hello, Certificate, Server Key > Exchange and Server Hello Done. Something in that packet seems to be unacceptable for openssl 1.1.1d as it is compiled and configured for Buster. Certificate length, Digest algorithm, ... You my change the system default settings at the

Hi, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

I am having trouble getting wine to work with my sound. When I use winecfg it can not autodetect the right settings, and it only opens a control device, no output device: Code: hammertime boxfire # lsof /dev/snd/* COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME winecfg.e 9086 boxfire 10u CHR 116,32 5684 /dev/snd/controlC1 explorer. 9108 boxfire 10u CHR 116,32 5684

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