dialplan: sample or production?
Posted: Sun Feb 22, 2015 3:42 pmI loaded sample leads, except that I went through and changed the phone numbers to all be "102" so that every time a call is made from the dialer, it just calls 102 in the dialplan. All well and good, select "call next number" starts a call..
but then I hear:
Congratulations. You have successfully installed and executed the Asterisk open source PBX. You have also installed a set of sample sounds and configuration files that should help you to get started. Like a normal PBX, you will navigate this demonstration by dialing digits. If you are using a console channel driver instead of a real phone you can use the dial, answer, and hang up commands to simulate the actions of a standard telephone.
So, I look in the dialplan, and see:
is the problem that, at least for 102, that the dialplan doesn't handle how to answer the phone??
Yet, when I just dial 101 from 102, or vice versa, that works like a charm.
- ViciBox Redux v.6.0.3-141118
- Code: Select all
mysql>
mysql> SELECT last_name, address1, phone_number FROM asterisk.vicidial_list;
+-----------+-----------------+--------------+
| last_name | address1 | phone_number |
+-----------+-----------------+--------------+
| lead01 | 1234 Fake St. | 102 |
| lead02 | 1234 Fake St. | 102 |
| lead03 | 1234 Fake St. | 102 |
| lead04 | 1234 Fake St. | 102 |
| lead05 | 1234 Fake St. | 102 |
| lead06 | 1234 Fake St. | 102 |
| lead07 | 1234 Fake St. | 102 |
| SMITH01 | 249 MUNDON ROAD | 102 |
| SMITH02 | 249 MUNDON ROAD | 102 |
| SMITH03 | 249 MUNDON ROAD | 102 |
| SMITH04 | 249 MUNDON ROAD | 102 |
| SMITH05 | 249 MUNDON ROAD | 102 |
| SMITH06 | 249 MUNDON ROAD | 102 |
| SMITH07 | 249 MUNDON ROAD | 102 |
| SMITH08 | 249 MUNDON ROAD | 102 |
| SMITH09 | 249 MUNDON ROAD | 102 |
| SMITH10 | 249 MUNDON ROAD | 102 |
| SMITH11 | 249 MUNDON ROAD | 102 |
| SMITH12 | 249 MUNDON ROAD | 102 |
| SMITH13 | 249 MUNDON ROAD | 102 |
| SMITH14 | 249 MUNDON ROAD | 102 |
+-----------+-----------------+--------------+
21 rows in set (0.00 sec)
mysql>
but then I hear:
Congratulations. You have successfully installed and executed the Asterisk open source PBX. You have also installed a set of sample sounds and configuration files that should help you to get started. Like a normal PBX, you will navigate this demonstration by dialing digits. If you are using a console channel driver instead of a real phone you can use the dial, answer, and hang up commands to simulate the actions of a standard telephone.
So, I look in the dialplan, and see:
- Code: Select all
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions-vicidial.conf
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
TRUNKloop = IAX2/ASTloop:password@127.0.0.1:40569
TRUNKblind = IAX2/ASTblind:password@127.0.0.1:41569
TRUNKplay = IAX2/ASTplay:password@127.0.0.1:42569
; agent phones restricted to only internal extensions
[default---agent]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----default---agent-------------------------NO)
exten => s,n,Set(INVCOUNT=0)
exten => s,n,Background(sip-silence)
exten => s,n,WaitExten(20)
; hangup
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => i,1,Goto(s,4)
exten => i,n,Hangup()
; hangup
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; custom dialplan entries
include => vicidial-auto-internal
include => vicidial-auto-phones
; logging of all outbound calls from agent phones
[defaultlog]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----defaultlog-------------------------NO)
exten => s,n,Set(INVCOUNT=0)
exten => s,n,Background(sip-silence)
exten => s,n,WaitExten(20)
; hangup
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => i,1,Goto(s,4)
exten => i,n,Hangup()
; hangup
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; custom dialplan entries
exten => _X.,1,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => _X.,n,Goto(default,${EXTEN},1)
[vicidial-auto-external]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Local Server: 192.168.1.2
exten => _192*168*001*002*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*001*002*.,2,Hangup()
exten => _**192*168*001*002*.,1,Goto(default,${EXTEN:18},1)
exten => _**192*168*001*002*.,2,Hangup()
; Agent session audio playback meetme entry
exten => _473782178600XXX,1,Meetme(${EXTEN:8},q)
exten => _473782178600XXX,n,Hangup()
; Agent session audio playback loop
exten => _473782168600XXX,1,Dial(${TRUNKplay}/47378217${EXTEN:8},5,To)
exten => _473782168600XXX,n,Hangup()
; Agent session audio playback extension
exten => 473782158521111,1,Answer
exten => 473782158521111,n,ControlPlayback(${CALLERID(name)},99999,0,1,2,3,4)
exten => 473782158521111,n,Hangup()
; SendDTMF to playback channel to control it
exten => _473782148521111.,1,Answer
exten => _473782148521111.,n,SendDTMF(${CALLERID(num)},250,250,IAX2/ASTplay-${EXTEN:15})
exten => _473782148521111.,n,Hangup()
; Silent wait channel for DTMFsend
exten => 473782138521111,1,Answer
exten => 473782138521111,n,Wait(5)
exten => 473782138521111,n,Hangup()
[vicidial-auto-internal]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Voicemail Extensions:
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,n,Voicemail(${EXTEN:14},u)
exten => _85026666666666.,n,Hangup()
exten => _85026666666667.,1,Wait(1)
exten => _85026666666667.,n,Voicemail(${EXTEN:14},su)
exten => _85026666666667.,n,Hangup()
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
exten => 8500,3,Hangup()
exten => 8501,1,VoicemailMain(s${CALLERID(num)})
exten => 8501,2,Hangup()
; Prompt Extensions:
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()
; 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 server
; setting ViciDial Recording Limit
; this is the WAV verison, default
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERID(name)})
exten => 8309,3,Wait(3600)
exten => 8309,4,Hangup()
; this is the GSM verison
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERID(name)})
exten => 8310,3,Wait(3600)
exten => 8310,4,Hangup()
; agent alert extension
exten => 83047777777777,1,Answer
exten => 83047777777777,2,Playback(${CALLERID(name)})
exten => 83047777777777,3,Hangup()
; This is a loopback dial-around to allow for immediate answer of outbound calls
exten => _8305888888888888.,1,Answer
exten => _8305888888888888.,n,Wait(${EXTEN:16:1})
exten => _8305888888888888.,n,Dial(${TRUNKloop}/${EXTEN:17},,To)
exten => _8305888888888888.,n,Hangup()
; No-call silence extension
exten => _8305888888888888X999,1,Answer
exten => _8305888888888888X999,n,Wait(3600)
exten => _8305888888888888X999,n,Hangup()
[vicidial-auto-phones]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Phones direct dial extensions:
exten => 101,1,Dial(SIP/101,60,)
exten => 101,2,Goto(default,85026666666666101,1)
exten => 101,3,Hangup()
exten => 102,1,Dial(SIP/102,60,)
exten => 102,2,Goto(default,85026666666666102,1)
exten => 102,3,Hangup()
exten => 103,1,Dial(SIP/103,60,)
exten => 103,2,Goto(default,85026666666666103,1)
exten => 103,3,Hangup()
exten => 104,1,Dial(SIP/104,60,)
exten => 104,2,Goto(default,85026666666666104,1)
exten => 104,3,Hangup()
[vicidial-auto]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
include => vicidial-auto-internal
include => vicidial-auto-phones
include => vicidial-auto-external
; END OF FILE Last Forced System Reload: 2015-02-20 16:49:28
tleilax:~ #
is the problem that, at least for 102, that the dialplan doesn't handle how to answer the phone??
Yet, when I just dial 101 from 102, or vice versa, that works like a charm.
- ViciBox Redux v.6.0.3-141118