Hai everyone,
I have used the SpeechBackground application in asterisk. I used the version 1.6.0.6.
I have a entry like,
;;SpeechCreate
exten => s,1,SpeechCreate()
exten => s,2,SpeechActivateGrammar(yesno)
exten => s,3,SpeechStart()
exten => s,4,SpeechBackground(demo-instruct)
exten => s,5,SpeechDeactivateGrammar(yesno)
I don't know which file I meed to give in SpeechBackground application. Please give me any idea. I have given the sound file from /sounds directory.
If I call to 's' the call will be immediately released.I didn't get any audio sound.
Please any one help me...
It doesn't appear that you have Answer()'d the channel.
;; SpeechCreate
exten => s,1,Answer()
exten => s,n,SpeechCreate()
exten => s,n,SpeechActivateGrammar(yesno)
exten => s,n,SpeechStart()
exten => s,n,SpeechBackground(demo-instruct)
exten => s,n,SpeechDeactivateGrammar(yesno)
Related
I am trying to originate a call using rawman, sending a phone number (57522666) as a parameter and playing a message to the answering party.
This is what I have so far:
http://192.168.11.11:8088/rawman?action=originate&channel=????&context=outboundmsg1&exten=s&priority=1&timeout=30000
And in extensions.conf:
[outboundmsg1]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Background(custom/message)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup
I have a SIP ATA as trunk with the following name: 23656429
I have no clue what to put in the channel parameter. I have tried SIP/23656429#57522666, but I don't get the outbound call.
In case someone else runs into this question... I got it to work by using the following:
http://192.168.11.11:8088/rawman?action=originate&channel=SIP/23656429/57522666&context=outboundmsg1&exten=s&priority=1&timeout=30000&callerid=23656429
And in extensions.conf
[outboundmsg1]
exten => s,1,Answer
exten => s,2,WaitExten()
exten => s,n,Wait(1)
exten => s,n,Playback(custom/elcodigo) ; "play outbound msg"
exten => s,n,SayDigits(1498)
exten => 2,n, Wait(1)
exten => s,n,Hangup
I am trying to learn ASterisk PBX and i have a problem that i did not saw in any tutorial.
I want to have and extension to automatically anser the call and transfert to extension 101 or 102 depending on the user selection from the softphone keyboard or voice. How can i implement that?
[default]
Exten => 101,1,Answer
Exten => 101,n,NoOp(Test101)
Exten => 101,n,Dial(SIP/101,12,r)
Exten => 101,n,VoiceMail(101#interno-ipca)
Exten => 101,n,Hangup()
Exten => 102,1,Answer
Exten => 102,n,NoOp(Test102)
Exten => 102,n,Dial(SIP/102,12,r)
Exten => 102,n,VoiceMail(102#interno-ipca)
Exten => 102,n,Hangup()
Hi if you want to anwser any call you can use
exten => _X,1,Answer()
exten => _X,2,Read($Number,,[max digits])
exten => _X,3,Dial(SIP/$Number)
You can use [max digits], otherwise asterisk waits until the maximum of digits is reached, or you end the typing with "#"
i have problem with my dial plan. I am using some php codes for looping and i am getting dynamic data in array. Ex: One username have maybe 2 or more accounts. I am getting account names and i am choosing a account.
i am using background option in loop for playing account names and context. When i press button 2,3 or 4 dial plan going to context and context pattern. So its working. But button 1 not working. I am getting extension error. I have no idea about why?
[2015-09-15 17:30:31] WARNING[26175][C-0000077d]: pbx.c:6696 __ast_pbx_run: Invalid extension '1', but no rule 'i' or 'e' in context 'Net-IVR-EN-ID'
and this is my config file;
exten => multi_wdsl,n,AGI(n_get_adres_wdsl.php,ara=${acc_number})
exten => multi_wdsl,n(lbl_Net-IVR-EN-ID_16),Set(counterplus=${MATH(${counter}+1,i)})
exten => multi_wdsl,n(lbl_Net-IVR-EN-ID_17),Set(TIMEOUT(response)=10)
exten => multi_wdsl,n,Set(TIMEOUT(digit)=10)
exten => multi_wdsl,n,Background(${ses${counter}}&/var/lib/asterisk/sounds/e/i/${counterplus},,en,WDSL_MENU)
exten => multi_wdsl,n,Set(counter=${MATH(${counter}+1,i)})
exten => multi_wdsl,n,GotoIf($["${wdsl_total}" = "${counter}"]?:lbl_Net-IVR-EN-ID_16)
exten => multi_wdsl,n,Set(TIMEOUT(response)=10)
exten => multi_wdsl,n,Set(TIMEOUT(digit)=10)
exten => multi_wdsl,n,Background(5sec&custom/custom/se,,en,WDSL_MENU)
exten => multi_wdsl,n,Goto(Net-IVR-EN-ID,multi_wdsl,1)
exten => i,1,Goto(multi_wdsl,lbl_Net-IVR-EN-ID_17)
exten => multi_wdsl__,1,Read(ir,custom/custom/ir,1,,5,10)
exten => multi_wdsl__,n,GotoIf($["${ir}" = "*"]?:lbl_Net-IVR-EN-ID_18)
exten => multi_wdsl__,n,Goto(Net-IVR-EN-ID,multi_wdsl,1)
exten => multi_wdsl__,n(lbl_Net-IVR-EN-ID_18),GotoIf($["${ir}" = "0"]?:lbl_Net-IVR-EN-ID_19)
exten => multi_wdsl__,n,Goto(Net-IVR-EN,multi_wdsl,1)
exten => multi_wdsl__,n(lbl_Net-IVR-EN-ID_19),Playback(custom/se/wrong_button)
exten => multi_wdsl__,n,Goto(Net-IVR-EN,multi_wdsl,1)
Thank you.
Dialplan you shown not have any option for 1,2,3 or 4
Background app usually used like this:
; this one just play file, not collect any input
exten => multi_wdsl,n,Background(5sec&custom/custom/se,,en,WDSL_MENU)
exten => multi_wdsl,n,WaitExten(5); on this stage it wait for input and go extension you entered
exten => 1,1,Noop(do something for 1)
exten => 2,1,Noop(do somethign for 2)
exten => i,1,Noop(invalid input);
Also you have Read app, which play and collect digit.
So background app not collect anything. Issue is not background app.
I am setting up IVR and encountered following trouble there is no dealing tone when client gets to "Dial", it just silently dialing. How to fix this?
exten => s,1,Background(custom/welcome-msg)
exten => s,n,WaitExten(10)
exten => s,n,Dial(SIP/0773,20)
exten => _X,1,Read(NUMBER,,4,n,,10)
exten => _X,n,Dial(SIP/${NUMBER},20)
exten => _X,n,HangUp()
You need add 'r' option in params.
exten => _X,n,Dial(SIP/${NUMBER},20,r)
https://wiki.asterisk.org/wiki/display/AST/Dial+Application
I have some extensions for which i have written dialplans.Now i want to add voicemail feature in it but i dont know how to add it in the existing dialplan.What are the other settings i need to do to achieve this.I am running asterisk on Ubuntu machine.Here is my extention.conf file ..
[others]
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
Please help me as i am very novice in this field.
Thanks in advance.
When I was working with Asterisk the WIKI saved me so many times. Here is an example of how to do the extension configuration
exten => 6000,1,Answer(500)
exten => 6000,n,Playback(hello-world)
exten => 6000,n,Hangup()
exten => 6001,1,Dial(SIP/demo-alice,20)
exten => 6001,n,VoiceMail(6001#vm-demo,u)
exten => 6002,1,Dial(SIP/demo-bob,20)
exten => 6002,n,VoiceMail(6002#vm-demo,u)
exten => 6500,1,Answer(500)
exten => 6500,n,VoiceMailMain(#vm-demo)
And here if you need to know more about how to configure asterisk for voicemail