similar to: console/dsp 1.4.11

In theory ParkAndAnnounce has a lot of usefulness, however, that we've had very little success with application... Our application is similiar to the local Walgreens pharmacy.. Dr. Calls in, selects the "Im a doctor with a prescription option"... call is parked, and announcement overhead is given.. "Doctor holding on ${EXTEN}" exten =>

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

I have a USB to serial converter attached to my box. pl2303: Prolific PL2303 USB to serial adaptor driver if I login to the box and send/receive serial commands over this unit it works without error EVERY time. however, if I run the same command set from with-in the extensions.conf with System() I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - failed submitting read urb, error

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'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

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 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

I'm trying to setup a primitive announcement-paging system in my house using the line-out from my * box to a cheap amplifier that runs to speakers on our first and second floors from the basement. I have a extension that connects to Console, and console is set to auto-pickup. I'm using alsa drivers. This all works great, except for one thing. I want to play a tone over the console after

Hello everyone, I'm streaming audio on CentOS 5.8 with no problem, even on a cheap sound card using DarkIce as the input tool. For the input under CentOS5, I use: device = /dev/dsp # OSS DSP soundcard device for the audio input But under CentOS 6.2, there is no such device. I see /dev/snd, and it has: controlC0 hwC0D2 midiC0D1 pcmC0D0p pcmC0D2p pcmC1D0p seq controlC1 hwC1D0

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

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

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

Hey all, I tried to do some searching but I found snippets and I am having trouble putting it all together. I want to have an option off the IVR that plays back the announcement for the day. At the end of the message, I want the caller to get kicked back to the previous menu. The conditions are that I want the recorder to dial a feature code that prompts him to record the message. He

Hi all, I am trying to set up DND in my asterisk, I am using the following context: [app-naoperturbe] exten => *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten => *11,2,GotoIf($["${DND}" = "YES"]?*11,3:*11,101)exten => *11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten => *11,4,Playback(beep)exten => *11,5,Hangup()exten =>

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.

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

Is there a setting that can add additional volume to the Console/Dsp output? Jerry

Dear folks, I have a problem with console/dsp using ALSA. I dont know why the output sound is choppy sometimes and also the input one has an awful delay. Is there anyone here with experince about ALSA channels or not? I would be highly appreciated if anyone could help me. Regards. M. Shokuie Nia _________________________________________________________________ Don't just search. Find.

Hi, I am trying to connect the outbound dialing with mysql with the following code, exten => 88,1,MYSQL(Connect connid hostname username password dbname) exten => 88,2,GotoIf($["${connid}" = ""]?error,1) exten => 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\ <tablename>\ WHERE\ phone =${a}) exten => 88,4,MYSQL(Fetch fetchid ${resultid} ph\ sa)