;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI

;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=no
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
;
clearglobalvars=no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
priorityjumping=no
;
; You can include other config files, use the #include command
; (without the ';'). Note that this is different from the "include" command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
; variables,
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;

[globals]

VOICEPULSE_API_KEY=sdajdghajsd
;VOICEPULSE_API_KEY=sdasdasda
VOICEPULSE_API_PREFIX=VOICEPULSE_

DID1=2222222222
DID2=1111111111
CIDNAME1=blarg

    ; .........................................................
    ; Peers
    ; .........................................................

VOICEPULSE_GATEWAY_OUT_A=voicepulse01
VOICEPULSE_GATEWAY_OUT_B=voicepulse02

; ---------------------------------------------------------
; [arbitrary-name] is the context referred to by the
; [voicepulse-in-01] user in iax.conf.  This is where your
; custom incoming call processing should go.
;
; For sample purposes, this section will read back the
; dialed number and then test DTMF by reading back each
; digit pressed by the caller.
; ---------------------------------------------------------

[voicepulse-in]  ; <-- Should match the context you have
                 ;     under [voicepulse-in-01] in iax.conf

exten => _1${DID1},1,Ringing
exten => _1${DID1},2,Wait(6)
exten => _1${DID1},3,Goto(teeheee,s,1)
exten => _1${DID1},4,Hangup

exten => _1${DID2},1,Ringing
exten => _1${DID2},2,Wait(6)
exten => _1${DID2},3,Voicemail(130)
exten => _1${DID2},4,Hangup

exten => a, 1, VoicemailMain(@default)
exten => a, 2, Hangup

exten => t,1,Voicemail(110,u)
exten => i,1,Hangup

;
; teehee Menu
;
[teehee]
include => common
exten => s,1,Answer
exten => s,2,Dial(SIP/100&SIP/110&SIP/111,15,trx)
exten => s,3,Voicemail(110,u)
;exten => s,3,Background(custom/blargh)
;exten => _0,1,Voicemail(100,u)
;exten => _1,1,Voicemail(110,u)
;exten => _2,1,Voicemail(111,u)
exten => 999,1,VoicemailMain
exten => *0*,1,Authenticate(21986129878610249)
exten => *0*,2,Playback(agent-loginok)
exten => *0*,3,Goto(outgoing,s,1)
exten => t,1,Hangup
exten => i,1,Hangup

;
; Main Menu
;
[main]
include => common
exten => s,1,Answer
exten => s,2,Background(custom/blargh)
exten => _0,1,Voicemail(101,u)
exten => _1,1,Dial(SIP/100,15)
exten => _1,2,Voicemail(100,u)
exten => _1,102,Voicemail(100,u)
exten => _2,1,Dial(SIP/100,15)
exten => _2,2,Voicemail(100,u)
exten => _2,102,Voicemail(100,u)
exten => 999,1,VoicemailMain
exten => *0*,1,Authenticate(555)
exten => *0*,2,Playback(agent-loginok)
exten => *0*,3,Goto(outgoing,s,1)
exten => t,1,Hangup
exten => i,1,Hangup


;
;  Select an outgoing context
;
[outgoing]
include => common
include => parkedcalls
exten => s,1,Wait(1)
exten => _1NXXNXXXXXX,1,Goto(outgoing-voicepulse,${EXTEN},1)
exten => _21NXXNXXXXXX,1,Goto(outgoing-voicepulse,${EXTEN:1},1)
exten => 911,1,Dial(SIP/+12129279711@${VOICEPULSE_GATEWAY_OUT_A})
exten => 999,1,Playback(beep)
exten => 999,2,VoicemailMain(${CALLERID(number)},s)
exten => _*91XX,1,Playback(beep)
exten => _*91XX,2,VoiceMailMain(s${EXTEN:2})
exten => _*51XX,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _*51XX,2,Dial(SIP/${EXTEN:2})
exten => _*51XX,3,Hangup
;exten => *0*,1,Authenticate(555)
;exten => *0*,2,Playback(agent-loginok)
;exten => *0*,3,Goto(internal,s,1)
exten => t,1,Hangup
exten => i,1,Hangup


;
; Shared Extensions
;
[common]
exten => _1XX,1,Dial(SIP/${EXTEN},15,X)
exten => _1XX,2,Voicemail(${EXTEN},u)
exten => _1XX,102,Voicemail(${EXTEN},u)
exten => _11XX,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => _11XX,2,Dial(sip/${EXTEN:1})
exten => _11XX,3,Voicemail(${EXTEN:1},u)
exten => _11XX,103,Voicemail(${EXTEN:1},u)
exten => *3,1,Playback(beep)
exten => *3,2,Echo
exten => *7,1,Set(TIMEOUT(digit)=3)
exten => *7,n,Set(TIMEOUT(response)=5)
exten => *7,n,Answer
exten => *7,n,Read(callfrom,beep,11)
exten => *7,n,Read(callto,beep,11)
exten => *7,n,NoOp(${callfrom})
exten => *7,n,NoOp(${callto})
;exten => *7,n,SetCIDNum(${callfrom})
exten => *7,n,Set(CALLERID(name)=Blargggh)
exten => *7,n,Set(CALLERID(number)=${callfrom})
exten => *7,n,Dial(SIP/+${callto}@${VOICEPULSE_GATEWAY_OUT_A})
exten => *7,n,Hangup

;
; Internal Functions
;
[internal]
exten => s,1,Wait(1)
exten => _*1XX,1,Answer()
exten => _*1XX,n,System(/usr/sbin/asterisk -rx "sip notify polycom-check-cfg ${EXTEN:1}")
exten => _*1XX,n,Hangup()
exten => t,1,Hangup
exten => i,1,Hangup

;
; Just Voicemail
;
[voicemail-only]
exten => s,1,Wait(1)
exten => 999,1,Playback(beep)
exten => 999,2,VoicemailMain(${CALLERID(number)},s)
exten => t,1,Hangup
exten => i,1,Hangup

[outgoing-voicepulse]
exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=${DID1},a)
exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=${CIDNAME1},a)
exten => _1NXXNXXXXXX,3,Set(TOUCH_MIXMONITOR_FORMAT=wav)
exten => _1NXXNXXXXXX,4,NoOp(${TOUCH_MIXMONITOR_FORMAT})
exten => _1NXXNXXXXXX,5,Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_A},,X)
exten => _1NXXNXXXXXX,6,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN},10)
exten => _1NXXNXXXXXX,10,Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_B},,X)