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...files !!------------------------ !!System wide config file (/etc/asound.conf) # # Place your global alsa-lib configuration here... # @hooks [ { func load files [ "/etc/alsa/pulse-default.conf" ] errors false } ] !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: CMI8738 [C-Media CMI8738], device 0: CMI8738-MC6 [C-Media PCI DAC/ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CMI8738 [C-Media CMI8738], device 1: CMI8738-MC6 [C-Media PCI 2nd DAC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CMI8738 [C-Media...

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm) exten=s,4,Playback(/var/lib/asterisk/sounds/record/deneme.gsm) but everytime i got the error msg file.c:635 ast_openstream_full: File /record/deneme.gsm does not exist in any format etc... It is a bug or am i...

Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a number is being played? Here is a test I am running. I am playing multiple portions (sounds and numbers). curl -v -u asterisk:asterisk -X POST http://localhost:8088/ari/channels/1596750578.46/play?media=sound:hello-world,sound:tt-monkeys,number:553,...

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk. <SIP/6000-0000000a>AGI Rx << EXEC playback "num:333456,say" -- AG...

I'm trying to devise a way to playback more than one file per call when I copy my file 'Test.call' into .. var/spool/asterisk/outgoing Channel: Zap/1/put_your_phone_number_here Application: Playback Data: demo-thanks + a-second-file + a-third-file Is there some way to do this? Charlie Hatchette chatchette@generalcare...

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten => 600,3,Echo...

...vaya extentions but when i press 1627 then it is wait for 5 second and then rining start alternative press '#' what is the method to break this space of waiting after dialing my extention.conf ;North Delhi NOC Extention exten => _16XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _16XX,2,Playback(vm-nobodyavail) exten => _16XX,102,Playback(all-allbusy) ;Mumbai NOC extention exten => _22XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _22XX,2,Playback(vm-nobodyavail) exten => _22XX,102,Playback(all-allbusy) exten => _17XX,1,Dial(SIP/mediant/${EXTEN},60) exten => _17XX,2,Playba...

Hi All, Can I overlap Playback() with Dial() in a dialplan? For example, I have this scenario: A call comes in, Asterisk picks it up, does Background(enter_number), then does Playback(bulletin_message), and while the Playback() is still going, I want to execute Dial() to the target extension so it overlaps with the Playback()...

When I first install wine using the default repository in Ubuntu 9.04 and run winecfg I get the following error: wine: created the configuration directory '/home/bill/.wine' err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element err:alsa:ALSA_CheckSetVolume Could not find 'PCM Playback Volume' element fixme:system:SetProcessDPIAware stub! fixme:iphlpapi:NotifyAddrChange (Handle 0x7d0049e8, overlapped 0x7d0049cc): stub fixme:shell:DllCanUnloadNow stub wine: configuration in '/home/bill/.wine&...

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten...

Hello, I have dozens of video files that were encoded using the speex format of audio and can not seem to find any video playback devices that support speex lol. Literally every playback device I've encountered (mac, 10.10.2) plays the videos w no sound. Do you know of any playback devices for the mac format that support speex? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://li...

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm) exten => *0,3,Playback(fcopba1) exten => *0,4,Hangup() include => extentions include => pasvalide [extentions] exten => 1001,1,Dial(SIP/sipchan1001,10) exten => 1001,2,Voicemail(u1001...

...st 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 the console picks up. What i'm doing right now is calling Playback after the Dial. However, No playback sound or background sound is being heard over the console speakers or are any error messages appearing in the command line. The extensions.conf entry looks like this: [access-internal] include => parkedcalls exten => 31,1,Dial(SIP/line1,30,t) exten =>...

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a single D...

I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.....) [custom-inbound] exten => _556,1,answer exten => _556,n,playback(beep) exten => _557,1,answer exten => _557,n,playback(beep) exten => _558,1,answer exten => _558,n,playback(beep) exten => _989,1,answer exten => _989,n,playback(beep) If there are like 100s of different prefixes, this list gets really big. Not desired. How can I have a more e...

...${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1. *Set(DB(blacklist/${blacknr})=1)* [pbx_config] 2. Playback(num-was-successfully) [pbx_config] 3. Playback(added) [pbx_config] 4. Wait(1) [pbx_config] 5. Hangup() [pbx_config] 's...

Hi, we are using Asterisk to record and playback. Both services are working well independently but it seems we can't start playback of a file while we are still recording it, even if the file is already in the hard disk. Is it possible to playback while recording the same audio file? Or is there a way to enable it? Regards, Jahnavi. -------...

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated cheers

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten => confmenu,1,Background(to-listen-to-it&press-1&to-rerecord-it&press-star|m||macro-systemrecording) exten => confmenu,n,Read(RECRESULT||1|||4) exten => confmenu,n,...