dial in ok; dial out results in invalid.gsm playing

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dial in ok; dial out results in invalid.gsm playing

Postby THUFIR » Mon Apr 06, 2015 7:13 pm

Incoming works fine, it gets routed to the SIP phone logged into 200. For outgoing, Asterisk is playing invalid.gsm when dialing my cellphone from the vicidial dialer, which is in the NANP as 1nnnnnnnnnn, which I attribute to the dialplan. The provider gave the connection parameters broken into incoming and outgoing, and they're using port 5065.

The specific audio is "I'm sorry that's not a valid extension..." which makes me think it would actually work for dialling internal extensions from the dialler, which would be a nice success, although not exactly knocking the ball out of the park.

I'm trying to put the in/out contexts to a single context. Will that simplify troubleshooting?

Why does extension 200 have its own context? I'd like to make that default. At present, my only concern is dialing out. It's more than ironic that receiving calls works fine, when that's the least desired feature.



Code: Select all
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
register => 12345678901@sip.babytel.ca:jfkldjf543543jdslfds:12345678901@nat5.babytel.ca:5065/12345678901

; VICIDIAL Carrier: BABYTEL - babytel
; Babytel
[babytel_in]
type=peer
qualify=no
host=nat5.babytel.ca
port=5065
context=inbound-calls

[babytel_out]
type=peer
username=12345678901
host=nat5.babytel.net
outboundproxy=nat5.babytel.ca:5065
secret=jfkldjf543543jdslfds
canreinvite=no
insecure=invite



[200]
username=200
secret=password
accountcode=200
callerid="" <200>
mailbox=200
context=local_200
type=friend
host=dynamic

[201]
username=201
secret=password
accountcode=201
callerid="" <201>
mailbox=201
context=default
type=friend
host=dynamic

[202]
username=202
secret=password
accountcode=202
callerid="" <202>
mailbox=202
context=default
type=friend
host=dynamic

[gs102]
username=gs102
secret=password
accountcode=gs102
callerid="Test Admin Phone" <>
mailbox=102
context=default
type=friend
host=dynamic


; END OF FILE    Last Forced System Reload: 2015-04-03 17:14:22



Code: Select all
; 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
TESTSIPTRUNK = SIP/babytel_out



; 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: 192.168.0.19
exten => _192*168*000*019*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*000*019*.,2,Hangup()
exten => _**192*168*000*019*.,1,Goto(default,${EXTEN:18},1)
exten => _**192*168*000*019*.,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 Carrier: BABYTEL - babytel
; Babytel
[general]

[inbound-calls]
exten => 12345678901,1,Dial(SIP/200)

[local_200]
exten => _9x.,1,Set(CALLERID(all)="Ali Baba" <1234567890>)
exten => _9x.,1,Dial(SIP/${EXTEN:1}@babytel_out)
exten => 201,1,Dial(SIP/201)

[local_201]
exten => 200,1,Dial(SIP/200)

[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 => 200,1,Dial(SIP/200,60,)
exten => 200,2,Goto(default,85026666666666200,1)
exten => 200,3,Hangup()
exten => 201,1,Dial(SIP/201,60,)
exten => 201,2,Goto(default,85026666666666201,1)
exten => 201,3,Hangup()
exten => 202,1,Dial(SIP/202,60,)
exten => 202,2,Goto(default,85026666666666202,1)
exten => 202,3,Hangup()
exten => 102,1,Dial(SIP/gs102,60,)
exten => 102,2,Goto(default,85026666666666102,1)
exten => 102,3,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: 2015-04-03 17:14:22



In the local_200 context, I'm trying to find EXTEN:1, I suspect that EXTEN is what's causing difficulty in dialing out...?


Page 176 of the definitive Asterisk guide has this line:

Code: Select all
exten => _1NXXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider)


I'm not clear on the distinction between "exten" above versus "EXTEN", but that line seems critical. The comment from the book on that line of configuration is that this is essential in order to send calls to the service provider (SIP trunk carrier).
ViciBox Redux v.6.0.3-141118 from .iso | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | AMD Phenom(tm) II X6 1090T Processor | 8GiB RAM
THUFIR
 
Posts: 109
Joined: Fri May 02, 2014 10:46 pm

Re: dial in ok; dial out results in invalid.gsm playing

Postby bobchaos » Fri May 01, 2015 5:14 pm

Sounds to me like you need to setup a carrier under Admin --> Carriers. That should all be in the manual.
bobchaos
 
Posts: 171
Joined: Fri Jan 06, 2012 12:46 pm


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