All pastes #2664 Raw Edit

Anonymous

public text v1 · immutable
#2664 ·published 2004-12-04 22:05 UTC
rendered paste body
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password@bigserver/local

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20)                                   ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)                            ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1})               ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)                 ; If they press #, return to start

exten => s-BUSY,1,Voicemail(b${ARG1})                   ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1)                             ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1)                              ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})                             ; If they press *, send the user into VoicemailMain

[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,2,Answer                     ; Answer the line
exten => s,3,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10         ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)  ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)  ; Play some instructions

exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
exten => 2,2,Goto(s,6)

exten => 3,1,SetLanguage(fr)            ; Set language to french
exten => 3,2,Goto(s,5)                  ; Start with the congratulations

exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip)         ; "Please hold while..."
                                        ; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234)                ; Right to voicemail

exten => 1236,1,Dial(Console/dsp)               ; Ring forever
exten => 1236,2,Voicemail(u1234)                ; Unless busy

;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks)              ; "Thanks for trying the demo"
exten => #,2,Hangup                     ; Hang them up.

;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1)                  ; If they take too long, give up
exten => i,1,Playback(invalid)          ; "That's not valid, try again"

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default)     ; Call the Asterisk demo
exten => 500,3,Playback(demo-nogo)      ; Couldn't connect to the demo site
exten => 500,4,Goto(s,6)                ; Return to the start over message.

;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,2,Echo                     ; Do the echo test
exten => 600,3,Playback(demo-echodone)  ; Let them know it's over
exten => 600,4,Goto(s,6)                ; Start over

;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,2,Goto(s,5)

;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,2,Background(thanks)                ; "Thanks for calling press 1 for sales, 2 for support, ..."
;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,2,Wait,2
;exten => s,3,Background(submenuopts)   ; "Thanks for calling the sales department.  Press 1 for steve, 2 for..."
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[default]
;
; By default we include the demo.  In a production system, you
; probably don't want to have the demo there.
;
include => demo

;
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)

; Real extensions would go here. Generally you want real extensions to be 4 or 5
; digits long (although there is no such requirement) and start with a single
; digit that is fairly large (like 6 or 7) so that you have plenty of room to
; overlap extensions and menu options without conflict.  You can alias them with
; names, too and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt)   ; permit transfer
;exten => 6245,1,Dial(${HINT},20,rtT)           ; Use hint as listed
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)         ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6394,1,Dial(Local/6275/n)             ; this will dial ${MARK}

;exten => 6275,1,Macro(stdexten,6275,${MARK})   ; assuming ${MARK} is something like Zap/2
;exten => mark,1,Goto(6275|1)                   ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL})    ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,2,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at your
; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you
; use that particular application in this file, the dial plan.
;
exten => 101,1,Dial(SIP/1001)
exten => 102,1,Dial(SIP/1002)