similar to: adding additional volume to console/dsp

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

I have a case where I need the option g to continue execute after the hangup (I'm using 1.2.1) and I have the following in my extensions: exten => 309,1,System(echo > /tmp/file) exten => 309,2,Dial(Console/dsp,,g) exten => 309,3,System(rm -f /tmp/file) exten => 309,4,Hangup However, after the hangup priority 3 is not executed. Does 'g' not work with console/dsp or do

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

I have an entry for console/dsp in the dialplan. When I call into that extension I get connected to the soundcard and I hear myself etc... everything is fine. However, if I call in and get connected then a second call comes in they also get connected. I was expecting them to get a busy signal or something... Do I have something configured wrong? How do I only get 1 person at a time on

Is there a way to have asterisk setup in ALSA mode (which is working) but also be able to "play file.wav" and hear that also. CONSOLE/dsp is working. If I stop asterisk and "play file.wav" that works. But if asterisk is running I cannot also play a file.wav I am not actually doing them at the same time. Just a couple files here and there to play. But as is I have to stop

In 1.4 I used alsa.conf and Dial(Console/Dsp) In 1.8 this is not working (as I had it) . I know there is a new chan_console I'd like to try both. What is the correct Dial() for ALSA direct? What is the correct Dial() for chan_console? I "thought" if chan_alsa was loaded it would default to old behaviour if chan_console was not loaded. Thanks, jerry -- This is the error I see

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

I have a few commands I wish to run after a hangup. It looks like only the first 2 commands are run after hangup. I am using 1.4.3 How can I get the entire loop to run 10 times. ( I know my example just has noop's but its an example). exten => h,1,Set(i=1) exten => h,n,While($[${i} < 10]) exten => h,n,Noop(jerry) exten => h,n,Set(i=$[${i} + 1]) exten => h,n,EndWhile exten

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port I get a motorboating sound or warble - or - just not clear audio. When I switch that to ALSA direct it sounds just fine. What might be happening with pulse audio that it does not sound clear??? asound.conf below. Thanks, Jerry more /etc/asound.conf # # Place your global alsa-lib configuration here... # @hooks [ {

I am using a extension to dial the console which has autoanswer enabled. I am getting a strange warning, has anyone seen this before? Nothing on Google, or Voip-Info [Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request: oss_request ty <console> data 0x0xb7851e00 <dsp> << Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from

>+1. also, it'd probably be good to state what you're trying to accomplish rather >than if you can throw a technology at vague something or another. I think you'd >find gpg's password input cumbersome for symmetric keys, the output too long in ascii >armor and possibly annoying to deal with public keys... but you don't really say what >you're trying to

I am calling from a polycom phone into asterisk ( 1105 ) on a PC with a speaker attached. When asterisk first starts this works. In fact it works for some time. Then it just stops with this error on the CLI. [Apr 4 15:10:21] NOTICE[4357]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_mediaport105' to extension '1105' rejected because extension not found in

I need to (occasionally) automatically drop out of X11 mode, recompile something, and then reboot. I can drop out of X11 with "init 3", works fine. I use "chvt 1" to ensure I am on virtual terminal 1 My recompile and reboot also work. The piece that doesnt work is: make | tee $LOGFILE I was expecting the output (stdout) to be virtual terminal 1. My LOGFILE has all commands

Hi List; In the extensions.conf file, at the [global] context, there is a variable configured as: CONSOLE=Console/dsp What does it mean that? What dsp mean and it is shortcut for what? How can I use the core to get some data about such thing ambiguous for me? Regards, ---------- Bilal Ghayad ____________________________________________________________________________________ Be a

I have a laptop with windows 10. I went into the Windows disk manager and shrunk the volume to make room for C7. That worked. I also changed the BIOS from secure boot to "both" (secure/legacy) I installed C7, went fine. About the time it was done I realized I never saw anything about "other" boot options (seems I saw that in the past). Anyway sure enough, got done and C7

I am sending an email from my machine devcentos5x64. the transcript below (hangs for 60 seconds) at the line: MAIL From:<root at devcentos5x64.msgnet.com> SIZE=56 AUTH=root at devcentos5x64.msgnet.com The email succeeds - but I am trying to figure out the 60 second delay. Neither email server is busy. Nothing is waiting. the DNS on both machines point to the same nameserver. The DNS

To continue here with problems on this compaq v2000 laptop, I put kernel source on a USB disk and plugged it into the v2000. NOTHING IS recognized. I tried to manually mount the disk and nothing either... I thought USB was well established.... I thought trying to recompile the kernel for realtek support might get my networking going... I am stuck??? Jerry -------------- next part

I am trying to hwclock to set the time. (hwclock -w) this is what I get on standard 5.2 x86_64. hwclock --debug hwclock from util-linux-2.13-pre7 hwclock: Open of /dev/rtc failed, errno=19: No such device. No usable clock interface found. Cannot access the Hardware Clock via any known method. [root at devcentos5x64 src]# ls -l /dev/rtc crw------- 1 root root 10, 135 Feb 6 13:32 /dev/rtc Any

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.