vicidial.org

hi to all really this vicidial will kill me a day
i have a voip account to call all french number with no prefix
for exemple when i want to call i need use the number directe in my softphone for exemple 0825676675 and no 0033825676675.
when i use softphone directly it work good
so now with asterisk and vicidial it don't want to call
every one i heard "soory you have not a avaible exention please try again"
my config is :vicidial VERSION: 2.0.5-173
CONSTRUCTION: 90320-0424
asterisk-1.2.30.4
vicibox the last version

and i want juste make outboud with no prefix to all french number exemple :
0825676675
(you can see my config maked in vicidial admin-->carriers)

i have read in the forum that i need to change same thing in file
etc/astersik/sip.conf and in etc/asterisk/extensions.conf because for i have just use the vicidial admin to write my trunk not other thing with:
admin-->carriers the i have put my voip config
ok
then i will show you my CLI>
i use manuel dial :

Code: Select all

 == Parsing '/etc/asterisk/meetme.conf': Found
Jun  9 14:21:26 NOTICE[1340]: app_meetme.c:2210 admin_exec: Conference Number not found
    -- Executing Hangup("Local/55558600051@default-b1bb,2", "") in new stack
  == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-b1bb,2'
    -- Executing DeadAGI("Local/55558600051@default-b1bb,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
Jun  9 14:25:24 NOTICE[2030]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc100
  == Manager 'sendcron' logged off from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing MeetMe("Local/8600051@default-85a6,2", "8600051|F") in new stack
  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '8600051'
       > Channel Local/8600051@default-85a6,1 was answered.
  == Starting Local/8600051@default-85a6,1 at default,91492726756,1 failed so falling back to exten 's'
  == Starting Local/8600051@default-85a6,1 at default,s,1 still failed so falling back to context 'default'
    -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-85a6,1
    -- Executing Playback("Local/8600051@default-85a6,1", "invalid") in new stack
    -- Playing 'conf-onlyperson' (language 'en')
    -- Playing 'invalid' (language 'en')
Jun  9 14:25:43 WARNING[2076]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Timeout on Local/8600051@default-85a6,1
  == CDR updated on Local/8600051@default-85a6,1
    -- Executing Goto("Local/8600051@default-85a6,1", "#|1") in new stack
    -- Goto (default,#,1)
    -- Executing Playback("Local/8600051@default-85a6,1", "invalid") in new stack
    -- Playing 'invalid' (language 'en')
    -- Executing Hangup("Local/8600051@default-85a6,1", "") in new stack
  == Spawn extension (default, #, 2) exited non-zero on 'Local/8600051@default-85a6,1'
    -- Executing DeadAGI("Local/8600051@default-85a6,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
    -- Hungup 'Zap/pseudo-1432210701'
  == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-85a6,2'
    -- Executing DeadAGI("Local/8600051@default-85a6,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Refreshing DNS lookups.
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1
standard*CLI>


extensions.conf

Code: Select all

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
TRUNK=Zap/g1                                    ; Trunk interface
TRUNKX=Zap/g2               ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569   ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569   ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com   ; IAX trunk interface
SIPtrunk=SIP/1234:PASSWORD@sip.provider.net   ; SIP trunk
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569   ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569   ; used for testing

#include extensions-vicidial.conf



[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer      ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})



[default]
include => vicidial-auto

; VICI-GROUP DIRECT SUPPORT LINE (VICIHELP[84244357])
exten => _84244XXX,1,Dial(IAX2/vicihelp/${EXTEN:5})

; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


;;;;;;;;;; BEGIN Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;
; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)

; this is used to allow the GUI to send you directly into voicemail
;     don't forget to set GUI variable $voicemail_exten to this extension
exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup

; this is used to allow the GUI to send live calls directly into voicemail
;     don't forget to set GUI variable $voicemail_dump_exten to this extension
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,2,Voicemail(${EXTEN:14}|u)
exten => _85026666666666.,3,Hangup

; prompts for recording AGI script, ID is 4321
; first variable is format (gsm/wav)
; second variable is timeout in milliseconds (default is 720000 [12 minutes])
exten => 8167,1,Answer
exten => 8167,2,AGI(agi-record_prompts.agi,wav-----720000)
exten => 8167,3,Hangup
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----720000)
exten => 8168,3,Hangup

; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => _7851XXXXX,1,WaitForSilence(2000,2) ; AMD got machine.  leave message after recording
exten => _7851XXXXX,2,Playback(${EXTEN:1})
exten => _7851XXXXX,3,AGI(VD_amd_post.agi,${EXTEN:1})
exten => _7851XXXXX,4,Hangup


;;;;;;;;;; END Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;;;


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


; Example phone extensions

; Extension 2000 Sipura/Linksys ATA line 1
exten => 2000,1,Dial(sip/spa2000,30,to)   ; Ring, 30 secs max
exten => 2000,2,Voicemail,u2000           ; Send to voicemail...
; Extension 2001 Sipura/Linksys ATA line 2
exten => 2001,1,Dial(sip/spa2001,30,to)   ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2001           ; Send to voicemail...
; Extension 2102 rings Grandstream phone
exten => 2102,1,Dial(sip/gs102,30,to)    ; Ring, 30 secs max
exten => 2102,2,Voicemail,u2102          ; Send to voicemail...
; Extension 401 rings the firefly softphone
exten => 401,1,Dial((IAX2/firefly01@firefly01/s||t)
exten => 401,2,Hangup

; 100-350 phone extensions now auto-generated
; extensions for other SIP and IAX call center phones
;   cc100-cc150 SIP Phones
;exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
;   cc300-cc350 IAX Phones
;exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)

; extensions if using a T1 channelbank
exten => _19XX,1,Dial(Zap/${EXTEN:2},30,o)
exten => _19XX,2,Hangup

; Extension 4001 rings Zap phone (this example for FXS on Zap port 1)
exten => 4001,1,Dial(Zap/1,30,o)   ; ring Zap device 1
exten => 4001,2,Voicemail,u4001         ; Send to voicemail...


; # timeout invalid rules
exten => #,1,Playback(invalid)              ; "Thanks for trying the demo"
exten => #,2,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"


; Inbound call from BINFONE
; exten => 1112223333,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => 1112223333,2,Dial(sip/gs102,55,o)
; exten => 1112223333,3,Hangup

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery
exten => 7275551212,1,Ringing
exten => 7275551212,2,Wait(1)
exten => 7275551212,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 7275551212,4,Answer
exten => 7275551212,5,Dial(sip/spa2000&sip/spa2001,30,To)
exten => 7275551212,6,Voicemail,u2000

; dial a long distance outbound number to the UK
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten => _901144XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _901144XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,To)
exten => _901144XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten => _901161XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _901161XXXXXXXXX,3,Hangup

; Extensions for performance testing
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TRUNKloop}/${EXTEN:2},,tTo)
exten => _91999NXXXXXX,3,Hangup
exten => 999999999999,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 999999999999,2,Dial(${TRUNKloop}/${EXTEN:1},,tTo)
exten => 999999999999,3,Hangup

; dial an 800 outbound number
exten => _91800NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91800NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91800NXXXXXX,3,Hangup
exten => _91888NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91888NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91888NXXXXXX,3,Hangup
exten => _91877NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91877NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91877NXXXXXX,3,Hangup
exten => _91866NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91866NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91866NXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)
exten => _9NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,To)
exten => _9NXXXXXX,3,Hangup

; dial a local 727 outbound number with area code
exten => _9727NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9727NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)
exten => _9727NXXXXXX,3,Hangup

; dial a long distance outbound number
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

; This is a loopback dialaround to allow for hearing of ringing for 3way calls
exten => _881NXXNXXXXXX,1,Answer
exten => _881NXXNXXXXXX,2,Dial(${TRUNKloop}/9${EXTEN:2},,To)
exten => _881NXXNXXXXXX,3,Hangup

; dial a long distance outbound number through BINFONE
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,To)
; exten => _91NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through a SIP provider
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)
; exten => _91NXXNXXXXXX,3,Hangup
; special extensions for North America to catch invalid phone numbers
; exten => _91XXX[0-1]XXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXX[0-1]XXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXX[0-1]XXXXXX,3,Hangup
; exten => _91[0-1]XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91[0-1]XXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91[0-1]XXXXXXXXX,3,Hangup
; exten => _91XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXX,3,Hangup
; exten => _91XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXX,3,Hangup
; exten => _91XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXXXX,3,Hangup
; exten => _91XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXXXXX,3,Hangup
; exten => _91XXXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXXXXXX,3,Hangup

; parameters for call_inbound.agi (7 fields separated by five dashes "-----"):
; 1. the extension of the phone to ring as defined in the asterisk.phones table
; 2. the phone number that was called, for the live_inbound/_log entry
; 3. a text description of the number that was called in
; 4-7. optional fields, they are also passed as fields in the GUI to web browser
; This is not part of VICIDIAL, it is for astGUIclient agent use only

; Extension 3429 - Inbound 800 number (1-800-555-3429) example of RBS T1
;    with 10 digit ANI and 4 digit DNIS star separated
exten => _**3429,1,Ringing
exten => _**3429,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _**3429,4,Answer
exten => _**3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
exten => _**3429,6,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _*NXXNXXXXXX*3429,4,Answer
exten => _*NXXNXXXXXX*3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
exten => _*NXXNXXXXXX*3429,6,Voicemail,u2000


; parameters for agi-VDAD_ALL_inbound.agi (9 fields separated by five dashes "-----"):
;  1. the method of call handling for the script:
;    - CID -    CID received, add record with phone number
;    - CIDLOOKUP -    Lookup CID to find record in whole system
;    - CIDLOOKUPRL -   Restrict lookup to one list
;    - CIDLOOKUPRC -   Restrict lookup to one campaign's lists
;   - CLOSER -      Closer calls from VICIDIAL fronters
;    - ANI -    ANI received, add record with phone number
;    - ANILOOKUP -    Lookup ANI to find record in whole system
;    - ANILOOKUPRL -   Restrict lookup to one list
;    - 3DIGITID -    Enter 3 digit code to go to agent
;    - 4DIGITID -    Enter 4 digit code to go to agent
;    - 5DIGITID -    Enter 5 digit code to go to agent
;    - 10DIGITID -    Enter 10 digit code to go to agent
; 2. the method of searching for an available agent:
;    - LO - Load Balance Overflow only (priority to home server)
;    - LB - <default> Load Balance total system
;    - SO - Home server only
; 3. the full name of the IN GROUP to be used in vicidial for the inbound call
; 4. the phone number that was called, for the log entry
; 5. the callerID or lead_id of the person that called(usually overridden)
; 6. the park extension audio file name if used
; 7. the status of the call initially(usually not used)
; 8. the list_id to insert the new lead under if it is new (and CID/ANI available)
; 9. the phone dialing code to insert with the new lead if new (and CID/ANI available)
; 10. the campaign_id to search within lists if CIDLOOKUPRC
; 11. the user to queue the call to for AGENTDIRECT in-group calls
; inbound VICIDIAL call with CID delivery through T1 PRI
exten => 1234,1,Answer                  ; Answer the line
exten => 1234,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1)
exten => 1234,3,Hangup

; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
exten => _90009.,1,Answer                  ; Answer the line
exten => _90009.,2,Dial(${TRUNKloop}/9${EXTEN},,to)
exten => _90009.,3,Hangup
exten => _990009.,1,Answer                  ; Answer the line
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
exten => _990009.,3,Hangup
; DID forwarded calls
exten => _99909*.,1,Answer
exten => _99909*.,2,AGI(agi-VDAD_ALL_inbound.agi)
exten => _99909*.,3,Hangup


; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup

; ZapBarge direct channel extensions
exten => _86120XX,1,ZapBarge(${EXTEN:5})


exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten => _X48600XXX,2,Hangup

exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten => _X38600XXX,2,Hangup

exten => _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})
exten => _X28600XXX,2,Hangup

exten => _X18600XXX,1,MeetMeAdmin(${EXTEN:2},M,${EXTEN:0:1})
exten => _X18600XXX,2,Hangup

exten => _55558600XXX,1,MeetMeAdmin(${EXTEN:4},K)
exten => _55558600XXX,2,Hangup
exten => 8300,1,Hangup

; astGUIclient conferences
exten => _86000[0-4]X,1,Meetme,${EXTEN}|q
; VICIDIAL conferences
exten => _86000[5-9]X,1,Meetme,${EXTEN}|F
exten => _8600[1-2]XX,1,Meetme,${EXTEN}|F
; quiet entry and leaving conferences for VICIDIAL (inbound announce and SendDTMF)
exten => _78600XXX,1,Meetme,${EXTEN:1}|Fq
; quiet monitor-only extensions for meetme rooms (for room managers)
exten => _68600XXX,1,Meetme,${EXTEN:1}|Fmq
; quiet monitor-only entry and leaving conferences for VICIDIAL (recording)
exten => _58600XXX,1,Meetme,${EXTEN:1}|Fmq

; voicelab exten
exten => _86009XX,1,Meetme,${EXTEN}|Fmq
; voicelab exten moderator
exten => _986009XX,1,Meetme,${EXTEN:1}



; park channel for client GUI parking, hangup after 30 minutes
;    create a GSM formatted audio file named "park.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup

; park channel for client GUI conferencing, hangup after 30 minutes
;    create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

exten => 8304,1,Answer
exten => 8304,2,Playback(ding)
exten => 8304,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
;    create a GSM formatted audio file complies with safe harbor rules
;    and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
;    Recording is limited to 1 hour, to make longer, just change the Wait,3600
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERIDNAME})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup

; this is used for recording conference calls, the client app sends the filename
;    value as a callerID recordings go to /var/spool/asterisk/monitor (GSM)
;    Recording is limited to 1 hour, to make longer, just change the Wait,3600
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERIDNAME})
exten => 8310,3,Wait,3600
exten => 8310,4,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
;    replace conf with the message file you want to leave
exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine.  leave message after recording
exten => 8320,2,Playback(conf)
exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})
exten => 8320,4,Hangup

; use for selective CallerID hangup by area code(hard-coded)
exten => 8352,1,AGI(agi-VDADselective_CID_hangup.agi,${EXTEN})
exten => 8352,2,Playback(safe_harbor)
exten => 8352,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
;    sends the digits to be played in the callerID field
;    sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; multi-remote-monitor entry extensions
exten => 8162,1,Dial(${TRUNKblind}/34567890123456789,55,to)

exten => 34567890123456789,1,Answer
exten => 34567890123456789,2,Goto(monitor,s,1)

;#### VDAD STANDARD TRANSFER ENTRIES ####
; Below are the parameters needed for the agi-VDAD_ALL_outbound.agi script to be run properly
; 1. the method of call handling for the script:
;    - NORMAL -       <default> Standard outbound routing to agent
;    - TEST -       For performance testing only
;    - BROADCAST -   For no-agent broadcast dialing
;    - SURVEY -      For survery question then on to agent
;    - REMINDER -   Reminder campaign
;    - REMINDX -      Reminder with transfer to agent
; 2. the method of searching for an available agent:
;    - LB - <default> Load Balance total system
;    - LO - Load Balance Overflow only (priority to home server)
;    - SO - Home server only
; 3. the sound file to play when doing a SURVEY, REMINDER, REMINDX campaign
; 4. the acceptible dtmf digits for a SURVEY
; 5. the out-opt digit for a SURVEY (must be in the digit map)
; 6. the sound file to play for a SURVEY when transfering to an agent
; 7. the sound file to play for a SURVEY when DNCing the call
; 8. OPTIN or OPTOUT: if OPTIN call is only sent to agent with button press
;     if OPTOUT call is sent to agent if no button press at all
; 9. the status that is use for a SURVEY when someone opts out
;     if the status is DNC it will also add them to the internal dnc table

; VICIDIAL_auto_dialer transfer script for no-agent campaigns:
exten => 8364,1,Playback(sip-silence)
exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8364,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,Playback(sip-silence)
exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Load Balanced:
exten => 8373,1,Playback(sip-silence)
exten => 8373,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8373,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8373,4,AGI(VD_amd.agi,${EXTEN})
exten => 8373,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,7,Hangup




; PERFORMANCE TESTING
exten => _999XXXXXX1,1,Answer
exten => _999XXXXXX1,2,Wait(2)
exten => _999XXXXXX1,3,Playback(vicidial-welcome)
exten => _999XXXXXX1,4,Hangup

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,2,Answer
exten => _999XX11112,3,Playback(ss-noservice)
exten => _999XX11112,4,Playback(vm-goodbye)
exten => _999XX11112,5,Hangup

exten => _999XX18112,1,Wait(2)
exten => _999XX18112,2,Answer
exten => _999XX18112,3,Playback(vtiger-fax)
exten => _999XX18112,4,Playback(vtiger-fax)
exten => _999XX18112,5,Hangup

exten => _999XX19112,1,Wait(2)
exten => _999XX19112,2,Answer
exten => _999XX19112,3,Playback(vtiger-email)
exten => _999XX19112,4,Playback(vtiger-email)
exten => _999XX19112,5,Hangup

exten => _999XXXX112,1,Wait(5)
exten => _999XXXX112,2,Answer
exten => _999XXXX112,3,Playback(demo-instruct)
exten => _999XXXX112,4,Playback(demo-instruct)
exten => _999XXXX112,5,Hangup

exten => _999XXXXXX2,1,Wait(8)
exten => _999XXXXXX2,2,Answer
exten => _999XXXXXX2,3,Playback(demo-instruct)
exten => _999XXXXXX2,4,Hangup

exten => _999XXXXXX3,1,SetVar(PRI_CAUSE=1)
exten => _999XXXXXX3,2,Hangup

exten => _999XXXXXX4,1,SetVar(PRI_CAUSE=27)
exten => _999XXXXXX4,2,Hangup

exten => _999XXXXXX5,1,Wait(60)
exten => _999XXXXXX5,2,Hangup

exten => _999XXXXXX6,1,Wait(10)
exten => _999XXXXXX6,2,Answer
exten => _999XXXXXX6,3,Playback(demo-instruct)
exten => _999XXXXXX6,4,Hangup

exten => _999XXXXXX7,1,Wait(12)
exten => _999XXXXXX7,2,Answer
exten => _999XXXXXX7,3,Playback(demo-enterkeywords)
exten => _999XXXXXX7,4,Hangup

exten => _999XXXXXX8,1,SetVar(PRI_CAUSE=17)
exten => _999XXXXXX8,2,Hangup

exten => _999XXXXXX9,1,Wait(6)
exten => _999XXXXXX9,2,Answer
exten => _999XXXXXX9,3,Playback(demo-abouttotry)
exten => _999XXXXXX9,4,Hangup

exten => _999XXXXXX0,1,Wait(5)
exten => _999XXXXXX0,2,Answer
exten => _999XXXXXX0,3,Playback(vm-goodbye)
exten => _999XXXXXX0,4,Hangup

exten => 99999999999,1,Answer
exten => 99999999999,2,Playback(conf)
exten => 99999999999,3,Playback(conf)
exten => 99999999999,4,Playback(conf)
exten => 99999999999,5,Playback(conf)
exten => 99999999999,6,Playback(conf)
exten => 99999999999,7,Playback(conf)
exten => 99999999999,8,Playback(conf)
exten => 99999999999,9,Playback(conf)
exten => 99999999999,10,Playback(conf)
exten => 99999999999,11,Playback(conf)
exten => 99999999999,12,Playback(conf)
exten => 99999999999,13,Playback(conf)
exten => 99999999999,14,Hangup


[monitor]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => s,1,SetVar(TIMEOUT(digit)=10)
exten => s,n,SetVar(TIMEOUT(response)=10)
exten => s,n,SetVar(MEETME_EXIT_CONTEXT=monitor_exit)
exten => s,n,Background(vm-extension) ; need audio prompt.
exten => s,n,WaitExten(10)

exten => i,1,Goto(monitor_exit,s,1)
exten => #,1,Goto(monitor_exit,s,1)
exten => t,1,Goto(monitor_exit,s,1)

exten => _8[0-2]XX,1,Meetme(8600${EXTEN:1},FmqX) ; Listen
exten => _99[0-2]XX,1,Meetme(8600${EXTEN:2},FX)  ; Barge

[monitor_exit]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _X,1,Goto(monitor,s,1)

exten => i,1,Goto(monitor,s,1)
exten => #,1,Goto(monitor,s,1)
exten => t,1,Goto(monitor,s,1)


sip.conf

Code: Select all

[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
;realm=mydomain.tld             ; Realm for digest authentication
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld            ; Set default domain for this host
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4                 ; Add IP address as local domain
;allowexternalinvites=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;pedantic=yes                   ; Enable slow, pedantic checking for Pingtel
;tos=184                        ; Set IP QoS to either a keyword or numeric val
tos=lowdelay                    ; lowdelay,throughput,reliability,mincost,none
maxexpiry=3600                  ; Max length of incoming registration we allow
defaultexpiry=120               ; Default length of incoming/outgoing registration
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
;vmexten=voicemail      ; dialplan extension to reach mailbox sets the
;videosupport=yes               ; Turn on support for SIP video
;recordhistory=yes              ; Record SIP history by default
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=gsm                       ;
musicclass=default              ; Sets the default music on hold class for all SIP calls
language=fr                     ; Default language setting for all users/peers
relaxdtmf=yes                   ; Relax dtmf handling
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP activity
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP activity
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
promiscredir = no       ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
;notifyringing = yes            ; Notify subscriptions on RINGING state
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
;regcontext=sipregistrations
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
callevents=no                   ; generate manager events when sip ua performs events (e.g. hold)
;externip = 192.168.1.10     ; Address that we're going to put in outbound SIP messages
;externhost=alphacallcenter.dtdns.net      ; Alternatively you can specify an
;externrefresh=10               ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0     ; Also RFC1918
localnet=172.16.0.0/12          ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes                         ; Global NAT settings  (Affects all peers and users)
canreinvite=no
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
; domain=myasterisk.dom
; domain=customer.com,customer-context
; autodomain=yes
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks


help me please and what can i change or add to do the calling with vicidial please
if you need other information from my server i'am here
thanks[/code]

please please i need all your help here
i want to work it
this is the only place for me to find help no any place
thanks

remove the 0 from the beginning of all your leads and set the prefix/country code to 0.

But from the looks of your extensions.conf, you just copied and pasted the information from the scratch install without actually looking at it. To be honest, I'm not sure how you're able to call out using just your softphone as your extension.conf doesn't seem to support your dialing methods.

Furthermore, what you are asking for help for is not so much a vicidial question, but an asterisk dial plan question. If you want to dial out with vicidial then maybe you could try something like this:

_9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log),Macro(trunkdial,${trunk}/${EXTEN:1},${trunk_cid})

and remove all other entries for that use the _9011, _91X.,_90033. Then for French numbers, you would just load them into the system without the 0 and make the country code/prefix 0. This way Vicidial will dial out using 901234567891 which would be a valid format.

ok if i understand good you say that my extension.conf can not supporte my softphone with this dial plan here i will say you why?
maybe i need change it to auto dial ?
then this line
_9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log),Macro(trunkdial,${trunk}/${EXTEN:1},${trunk_cid})

where can i put it?
directly in the vicidial admin-->carrier
ot in file extension.conf or in sip.conf
because i have not touch the file ;juste i use the admin interface to put my trunk ip
thanks

who can correct me this config ?