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Discussion: asterisk: configuration pour pré décroché automatique.

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  1. #1
    Membre Junior
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    septembre 2016
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    Suite du fichier extensions.conf :
    exten => _X.,50000(stdexten),NoOp(Start stdexten)
    exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
    exten => _X.,n,Set(LOCAL(dev)=${ARG1})
    exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
    exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
    exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
    exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
    exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
    exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
    exten => stdexten-BUSY,n,Return() ; If they press #, return to start
    exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
    exten => a,n,Return()

    [stdPrivacyexten]
    ; Standard extension subroutine:
    ; ${ARG1} - Extension
    ; ${ARG2} - Device(s) to ring
    ; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
    ; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
    ; ${ARG5} - Context in voicemail (if empty, then "default")
    exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
    exten => _X.,n,Set(LOCAL(ext)=${ARG1})
    exten => _X.,n,Set(LOCAL(dev)=${ARG2})
    exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
    exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
    exten => _X.,n,Set(LOCAL(cntx)=${ARG5})

    exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""])
    exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
    ; option (or use P for databased call _X.creening)
    exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
    exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
    exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
    exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
    exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
    exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
    exten => stdexten-BUSY,n,Return() ; If they press #, return to start
    exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite "Don't call again" script.
    exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.
    exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer
    exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
    exten => a,n,Return

    [macro-page];
    ; ${ARG1} - Device to page
    exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
    exten => s,n,GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail)
    exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
    exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
    exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
    exten => s,n,Dial(${ARG1})
    exten => s,n(fail),Hangup

    [demo]
    include => stdexten
    exten => s,1,Wait(1) ; Wait a second, just for fun
    exten => s,n,Answer ; Answer the line
    exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
    exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
    exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
    exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
    exten => s,n,WaitExten ; Wait for an extension to be dialed.
    exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
    exten => 2,n,Goto(s,instruct)
    exten => 3,1,Set(CHANNEL(language)=fr) ; Set language to french
    exten => 3,n,Goto(s,restart) ; Start with the congratulations
    exten => 1000,1,Goto(default,s,1)
    exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
    ; (but skip if channel is not up)
    exten => 1234,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}) )
    exten => 1234,n,Goto(default,s,1) ; exited Voicemail
    exten => 1235,1,Voicemail(1234,u) ; Right to voicemail
    exten => 1236,1,Dial(Console/dsp) ; Ring forever
    exten => 1236,n,Voicemail(1234,b) ; Unless busy
    exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
    exten => #,n,Hangup ; Hang them up.
    exten => t,1,Goto(#,1) ; If they take too long, give up
    exten => i,1,Playback(invalid) ; "That's not valid, try again"
    exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
    exten => 500,n,Dial(IAX2/guest@pbx.digium.com/s@default) ; Call the Asterisk demo
    exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
    exten => 500,n,Goto(s,6) ; Return to the start over message.
    exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
    exten => 600,n,Echo ; Do the echo test
    exten => 600,n,Playback(demo-echodone) ; Let them know it's over
    exten => 600,n,Goto(s,6) ; Start over
    exten => 76245,1,Macro(page,SIP/Grandstream1)
    exten => _7XXX,1,Macro(page,SIP/${EXTEN})
    exten => 7999,1,Set(TIMEOUT(absolute)=60)
    exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
    exten => 8500,1,VoicemailMain
    exten => 8500,n,Goto(s,6)
    ;exten => 1265,1,Dial(Phone/phone0,15)
    ;exten => 1265,n,Goto(s,5)

    [page]
    exten => _X.,1,Macro(page,SIP/${EXTEN})
    ;[mainmenu]
    ;exten => s,1,Answer
    ;exten => s,n,Background(thanks) ; "Thanks for calling press 1 for sales, 2 for support, ..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(submenu,s,1)
    ;exten => 2,1,Hangup
    ;include => default
    ;[submenu]
    ;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
    ;exten => s,n,Wait,2
    ;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales department. Press 1 for steve, 2 for..."
    ;exten => s,n,WaitExten
    ;exten => 1,1,Goto(default,steve,1)
    ;exten => 2,1,Goto(default,mark,2)

    [public]
    ; ATTENTION: If your Asterisk is connected to the internet and you do
    ; not have allowguest=no in sip.conf, everybody out there may use your
    ; public context without authentication. In that case you want to
    ; double check which services you offer to the world.
    ;
    include => demo
    [default]
    [local]
    exten => _1XX, 1, Dial(SIP/${EXTEN}, 15) ; Compose 101 appelle franky etc
    exten => _1XX, n, VoiceMail(${EXTEN}) ; Voicemail apres 15 secondes
    exten => 90,1,VoiceMailMain(${CALLERID(num)}) ; Messagerie
    exten => 300, 1, Meetme(300)

    include => demo
    ;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)
    ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1(Joe Schmoe) ; Channel hints for presence
    ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
    ;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
    ;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
    ;exten => 6245,s+1,Hangup ; s+1, same as n
    ;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
    ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
    ;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
    ;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
    ;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org) ;Dial via jingle using asterisk as the transport and calling mogorman.
    ;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

    ;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
    ; assuming ${MARK} is something like DAHDI/2
    ;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
    ;exten => mark,1,Goto(6275,1) ; alias mark to 6275
    ;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
    ; Ditto for wil
    ;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
    ;exten => wil,1,Goto(6236,1)
    ;exten => 6600,hint,park:701@parkedcalls
    ;exten => 6600,1,noop
    ; You can also monitor the status of a queue by providing a hint for a
    ; particular queue name.
    ;exten => 8502,hint,Queue:markq
    ;exten => 8502,1,Queue(markq)

    ;To subscribe to the availability of a free member in the 'markq' queue.
    ;Note: '_avail' is added to the QueueName
    ;exten => 8501,hint,Queue:markq_avail
    ;exten => 8501,1,Queue(markq)
    ; Some other handy things are an extension for checking voicemail via
    ; voicemailmain
    ;exten => 8500,1,VoicemailMain
    ;exten => 8500,n,Hangup
    ;exten => 8600,1,Meetme(1234)
    ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
    ;[acme-incoming]
    ;exten => s,1,Wait(1)
    ;exten => s,n,Answer()
    ;exten => s,n(menu),Playback(acme/vm-brief-menu)
    ;exten => s,n(exten),Background(vm-enter-num-to-call)
    ;exten => s,n,WaitExten(5)
    ;exten => s,n(goodbye),Playback(vm-goodbye)
    ;exten => s,n(end),Hangup()
    ;include => acme-extens
    ;exten => i,1,Playback(vm-invalid)
    ;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
    ;exten => t,1,Goto(s,goodbye)
    ; this is the context our internal SIP hardphones use (see sip.conf)
    ;[acme-internal]
    ;exten => s,1,Answer()
    ;exten => s,n(exten),Background(vm-enter-num-to-call)
    ;exten => s,n,WaitExten(5)
    ;exten => s,n(goodbye),Playback(vm-goodbye)
    ;exten => s,n(end),Hangup()
    ;include => trunkint
    ;include => trunkld
    ;include => trunklocal
    ;include => acme-extens
    ;exten => 777,1,DISA(no-password,acme-incoming)
    ;[acme-extens]
    ;include => stdexten
    ;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
    ;exten => 111,n,Goto(s,exten)
    ;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
    ;exten => 112,n,Goto(s,end)
    Pour le prédécroché apparemment :
    pour le prédécroché c'est l'option "m" de la commande "dial": exten => 1,1,Dial(SIP/toto,,m(musique))
    Dernière modification par Hub49 ; 01/11/2016 à 20h32.

  2. #2
    Membre Senior
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    septembre 2010
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    Where the sun shines
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    - le fichier extensions.ael a été installé avec les paquets, dans l'absolu, tu peux soit le vider, soit l'effacer - ca contient du code qui si utilisé, apporte des fonctionnalités d'un pabx

    - attention, softphone + asterisk sur le meme serveur = conflit pour utiliser le port 5060.... il faut t'assurer que le sofptohne n'essaie pas d'utiliser le port 5060 comme port source

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