dialplan: sample or production?

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dialplan: sample or production?

Postby THUFIR » Sun Feb 22, 2015 3:42 pm

I 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..

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.

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Posts: 109
Joined: Fri May 02, 2014 10:46 pm

Re: dialplan: sample or production?

Postby mav2287 » Thu Feb 26, 2015 8:54 am

I have seen that happen when leads don't have a working country code. In your case I would keep in mind the campaigns have a default dial prefix set so in your case they are most likely dialing 9102. If you can post your CLI output I can tell you exactly what is happening. It should work if you get rid of the dial prefix, HOWEVER this is not a good way to test. What would be better is to use a number that is an actual phone number and you control over. For example you are better off loading you cell number in place of 102. Then setup everything like you would for production and test against that.
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