Dial TESTSIPTRUNK does what, exactly?

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Dial TESTSIPTRUNK does what, exactly?

Postby THUFIR » Sun Apr 12, 2015 6:17 pm

When vicidial tries to dial out using testcarrier, or at least this is the intention, this results in invalid.gsm playing -- but why? It should be dialing using:

Code: Select all
exten => _91999NXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,To)


which, to my reading, says to dial using the value of TESTSIPTRUNK, passing the current extension, minus the first two digits. Isn't that correct?


I think it's because context shows as trunkinbound:

Code: Select all
vici:~ #
vici:~ # asterisk -rx "sip show peers"
Name/username             Host                                    Dyn Forcerport ACL Port     Status     
300/300                   192.168.0.24                             D   N             51934    UNREACHABLE
301/301                   (Unspecified)                            D   N             0        UNKNOWN   
302/302                   192.168.0.24                             D   N             51150    UNREACHABLE
gs102/gs102               (Unspecified)                            D   N             0        UNKNOWN   
testcarrier/19876543210   198.38.7.34                                  N             5065     OK (81 ms)
5 sip peers [Monitored: 1 online, 4 offline Unmonitored: 0 online, 0 offline]
vici:~ #
vici:~ # asterisk -rx "sip show peer testcarrier"


  * Name       : testcarrier
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : trunkinbound
  Subscr.Cont. : <Not set>
  Language     : en
  AMA flags    : Unknown
  Netborder CPD: No
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  MOH Suggest  : default
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Max forwards : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : invite
  Force rport  : Yes
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: 4294967295
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : Yes
  TrustIDOutbnd: Legacy
  Subscriptions: Yes
  Overlap dial : No
  Outb. proxy  : nat5.babytel.ca
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : nat5.babytel.ca
  Addr->IP     : 198.38.7.34:5065
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 19876543210
  SIP Options  : (none)
  Codecs       : 0x6 (gsm|ulaw)
  Codec Order  : (ulaw:20,gsm:20)
  Auto-Framing : No
  Status       : OK (82 ms)
  Useragent    :
  Reg. Contact :
  Qualify Freq : 60000 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

vici:~ #



What should the context be?


channels:

Code: Select all
vici:~ #
vici:~ # cat /etc/asterisk/sip-vicidial.conf
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
register => 19876543210@sip.babytel.ca:fjdkl543543jgsdfklfsjd:19876543210@nat5.babytel.ca:5065/19876543210

; VICIDIAL Carrier: SIPEXAMPLE - TEST SIP carrier example
[testcarrier]
type=peer
username=19876543210
host=nat5.babytel.ca
outboundproxy=nat5.babytel.ca:5065
secret=fjdkl543543jgsdfklfsjd
canreinvite=no
insecure=invite
qualify=yes




[300]
username=300
secret=password
accountcode=300
callerid="300phone" <19876543210>
mailbox=300
context=default
type=friend
host=dynamic

[301]
username=301
secret=password
accountcode=301
callerid="301phone" <19876543210>
mailbox=301
context=default
type=friend
host=dynamic

[302]
username=302
secret=password
accountcode=302
callerid="302phone" <19876543210>
mailbox=302
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: 2015-04-11 22:19:10
vici:~ #




and extensions:

Code: Select all
vici:~ #
vici:~ #
vici:~ # 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
TESTSIPTRUNK = SIP/testcarrier



; 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: 192.168.0.31
exten => _192*168*000*031*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*000*031*.,2,Hangup()
exten => _**192*168*000*031*.,1,Goto(default,${EXTEN:18},1)
exten => _**192*168*000*031*.,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: SIPEXAMPLE - TEST SIP carrier example
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,To)
exten => _91999NXXXXXX,3,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 => 300,1,Dial(SIP/300,60,)
exten => 300,2,Goto(default,85026666666666300,1)
exten => 300,3,Hangup()
exten => 301,1,Dial(SIP/301,60,)
exten => 301,2,Goto(default,85026666666666301,1)
exten => 301,3,Hangup()
exten => 302,1,Dial(SIP/302,60,)
exten => 302,2,Goto(default,85026666666666302,1)
exten => 302,3,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: 2015-04-11 22:19:10
vici:~ #



which results in calls to external numbers failing:
Code: Select all
[Apr 12 18:41:07] VERBOSE[29781] manager.c: [Apr 12 18:41:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:41:38] VERBOSE[29818] manager.c: [Apr 12 18:41:38]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:41:38] VERBOSE[29818] netsock2.c: [Apr 12 18:41:38]   == Using SIP RTP CoS mark 5
[Apr 12 18:41:39] VERBOSE[29818] pbx.c: [Apr 12 18:41:39]        > Channel SIP/300-00000003 was answered.
[Apr 12 18:41:39] VERBOSE[29821] pbx.c: [Apr 12 18:41:39]     -- Executing [8600051@default:1] MeetMe("SIP/300-00000003", "8600051,F") in new stack
[Apr 12 18:41:39] VERBOSE[29821] config.c: [Apr 12 18:41:39]   == Parsing '/etc/asterisk/meetme.conf': [Apr 12 18:41:39] VERBOSE[29821] config.c: [Apr 12 18:41:39]   == Found
[Apr 12 18:41:39] VERBOSE[29821] config.c: [Apr 12 18:41:39]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 12 18:41:39] VERBOSE[29821] config.c: [Apr 12 18:41:39]   == Found
[Apr 12 18:41:39] VERBOSE[29821] app_meetme.c: [Apr 12 18:41:39]     -- Created MeetMe conference 1022 for conference '8600051'
[Apr 12 18:41:39] VERBOSE[29821] file.c: [Apr 12 18:41:39]     -- <SIP/300-00000003> Playing 'conf-onlyperson.gsm' (language 'en')
[Apr 12 18:41:39] WARNING[29821] res_rtp_asterisk.c: RTP Read too short
[Apr 12 18:41:40] VERBOSE[29818] manager.c: [Apr 12 18:41:40]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:41:46] VERBOSE[29835] manager.c: [Apr 12 18:41:46]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:41:46] VERBOSE[29836] pbx.c: [Apr 12 18:41:46]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000002;2", "8600051,F") in new stack
[Apr 12 18:41:46] VERBOSE[29835] pbx.c: [Apr 12 18:41:46]        > Channel Local/8600051@default-00000002;1 was answered.
[Apr 12 18:41:46] VERBOSE[29837] pbx.c: [Apr 12 18:41:46]   == Starting Local/8600051@default-00000002;1 at default,919876543210,1 failed so falling back to exten 's'
[Apr 12 18:41:46] VERBOSE[29837] pbx.c: [Apr 12 18:41:46]   == Starting Local/8600051@default-00000002;1 at default,s,1 still failed so falling back to context 'default'
[Apr 12 18:41:46] VERBOSE[29837] pbx.c: [Apr 12 18:41:46]     -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-00000002;1
[Apr 12 18:41:46] VERBOSE[29837] pbx.c: [Apr 12 18:41:46]     -- Executing [i@default:1] Playback("Local/8600051@default-00000002;1", "invalid") in new stack
[Apr 12 18:41:46] VERBOSE[29837] file.c: [Apr 12 18:41:46]     -- <Local/8600051@default-00000002;1> Playing 'invalid.gsm' (language 'en')
[Apr 12 18:41:47] VERBOSE[29835] manager.c: [Apr 12 18:41:47]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:41:50] VERBOSE[29837] pbx.c: [Apr 12 18:41:50]     -- Executing [i@default:2] Hangup("Local/8600051@default-00000002;1", "") in new stack
[Apr 12 18:41:50] VERBOSE[29837] pbx.c: [Apr 12 18:41:50]   == Spawn extension (default, i, 2) exited non-zero on 'Local/8600051@default-00000002;1'
[Apr 12 18:41:50] VERBOSE[29837] pbx.c: [Apr 12 18:41:50]     -- Executing [h@default:1] AGI("Local/8600051@default-00000002;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 12 18:41:50] VERBOSE[29837] res_agi.c: [Apr 12 18:41:50]     -- <Local/8600051@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 12 18:41:50] VERBOSE[29836] pbx.c: [Apr 12 18:41:50]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000002;2'
[Apr 12 18:41:50] VERBOSE[29836] pbx.c: [Apr 12 18:41:50]     -- Executing [h@default:1] AGI("Local/8600051@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 12 18:41:50] VERBOSE[29836] res_agi.c: [Apr 12 18:41:50]     -- <Local/8600051@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 12 18:42:02] VERBOSE[29889] manager.c: [Apr 12 18:42:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:42:02] VERBOSE[29890] manager.c: [Apr 12 18:42:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:42:03] VERBOSE[29890] manager.c: [Apr 12 18:42:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:42:03] VERBOSE[29889] manager.c: [Apr 12 18:42:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:42:06] VERBOSE[3281] chan_sip.c: [Apr 12 18:42:06]     -- Registered SIP '300' at 192.168.0.24:56259
[Apr 12 18:42:08] VERBOSE[29905] manager.c: [Apr 12 18:42:08]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:42:08] VERBOSE[29905] manager.c: [Apr 12 18:42:08]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:42:22] VERBOSE[3281] chan_sip.c: [Apr 12 18:42:22]     -- Registered SIP '302' at 192.168.0.24:56259
[Apr 12 18:42:38] VERBOSE[3281] chan_sip.c: [Apr 12 18:42:38]     -- Registered SIP '302' at 192.168.0.24:51150
[Apr 12 18:43:00] VERBOSE[29972] manager.c: [Apr 12 18:43:00]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:43:00] VERBOSE[29972] netsock2.c: [Apr 12 18:43:00]   == Using SIP RTP CoS mark 5
[Apr 12 18:43:02] VERBOSE[29972] pbx.c: [Apr 12 18:43:02]        > Channel SIP/302-00000004 was answered.
[Apr 12 18:43:02] VERBOSE[29998] pbx.c: [Apr 12 18:43:02]     -- Executing [8600052@default:1] MeetMe("SIP/302-00000004", "8600052,F") in new stack
[Apr 12 18:43:02] WARNING[29998] res_rtp_asterisk.c: RTP Read too short
[Apr 12 18:43:02] VERBOSE[30001] manager.c: [Apr 12 18:43:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:43:02] VERBOSE[30002] manager.c: [Apr 12 18:43:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:43:02] VERBOSE[30002] manager.c: [Apr 12 18:43:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:43:03] VERBOSE[30001] manager.c: [Apr 12 18:43:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:43:03] VERBOSE[29972] manager.c: [Apr 12 18:43:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:43:07] VERBOSE[30015] manager.c: [Apr 12 18:43:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:43:07] VERBOSE[30015] manager.c: [Apr 12 18:43:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:43:09] VERBOSE[30022] manager.c: [Apr 12 18:43:09]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:43:09] VERBOSE[30023] pbx.c: [Apr 12 18:43:09]     -- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000003;2", "8600052,F") in new stack
[Apr 12 18:43:09] VERBOSE[30022] pbx.c: [Apr 12 18:43:09]        > Channel Local/8600052@default-00000003;1 was answered.
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]   == Starting Local/8600052@default-00000003;1 at default,919876543210,1 failed so falling back to exten 's'
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]   == Starting Local/8600052@default-00000003;1 at default,s,1 still failed so falling back to context 'default'
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]     -- Sent into invalid extension 's' in context 'default' on Local/8600052@default-00000003;1
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]     -- Executing [i@default:1] Playback("Local/8600052@default-00000003;1", "invalid") in new stack
[Apr 12 18:43:09] VERBOSE[30024] file.c: [Apr 12 18:43:09]     -- <Local/8600052@default-00000003;1> Playing 'invalid.gsm' (language 'en')
[Apr 12 18:43:10] VERBOSE[30022] manager.c: [Apr 12 18:43:10]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:43:11] VERBOSE[3281] chan_sip.c: [Apr 12 18:43:11]     -- Registered SIP '300' at 192.168.0.24:51934
[Apr 12 18:43:13] VERBOSE[30024] pbx.c: [Apr 12 18:43:13]     -- Executing [i@default:2] Hangup("Local/8600052@default-00000003;1", "") in new stack
[Apr 12 18:43:13] VERBOSE[30024] pbx.c: [Apr 12 18:43:13]   == Spawn extension (default, i, 2) exited non-zero on 'Local/8600052@default-00000003;1'
[Apr 12 18:43:13] VERBOSE[30024] pbx.c: [Apr 12 18:43:13]     -- Executing [h@default:1] AGI("Local/8600052@default-00000003;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 12 18:43:13] VERBOSE[30024] res_agi.c: [Apr 12 18:43:13]     -- <Local/8600052@default-00000003;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 12 18:43:13] VERBOSE[30023] pbx.c: [Apr 12 18:43:13]   == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-00000003;2'
[Apr 12 18:43:13] VERBOSE[30023] pbx.c: [Apr 12 18:43:13]     -- Executing [h@default:1] AGI("Local/8600052@default-00000003;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 12 18:43:13] VERBOSE[30023] res_agi.c: [Apr 12 18:43:13]     -- <Local/8600052@default-00000003;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Apr 12 18:43:43] NOTICE[3281] chan_sip.c: Peer '302' is now UNREACHABLE!  Last qualify: 18
[Apr 12 18:44:02] VERBOSE[30117] manager.c: [Apr 12 18:44:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:44:02] VERBOSE[30118] manager.c: [Apr 12 18:44:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:44:02] VERBOSE[30118] manager.c: [Apr 12 18:44:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:44:03] VERBOSE[30117] manager.c: [Apr 12 18:44:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:44:07] VERBOSE[30130] manager.c: [Apr 12 18:44:07]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:44:07] VERBOSE[30130] manager.c: [Apr 12 18:44:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Apr 12 18:44:11] VERBOSE[3219] dnsmgr.c: [Apr 12 18:44:11]        > Refreshing DNS lookups.
[Apr 12 18:44:11] VERBOSE[3219] srv.c: [Apr 12 18:44:11]        > ast_get_srv: SRV lookup for '_sip._udp.nat5.babytel.ca' mapped to host nat5.babytel.ca, port 5065
[Apr 12 18:44:11] VERBOSE[3219] srv.c: [Apr 12 18:44:11]        > ast_get_srv: SRV lookup for '_sip._udp.nat5.babytel.ca' mapped to host nat5.babytel.ca, port 5065
[Apr 12 18:44:16] NOTICE[3281] chan_sip.c: Peer '300' is now UNREACHABLE!  Last qualify: 19
[Apr 12 18:44:20] NOTICE[3281] chan_sip.c: Disconnecting call 'SIP/300-00000003' for lack of RTP activity in 61 seconds
[Apr 12 18:44:20] NOTICE[3281] chan_sip.c: Disconnecting call 'SIP/302-00000004' for lack of RTP activity in 61 seconds
[Apr 12 18:44:20] NOTICE[3281] chan_sip.c: Disconnecting call 'SIP/302-00000000' for lack of RTP activity in 61 seconds
[Apr 12 18:44:20] VERBOSE[13989] pbx.c: [Apr 12 18:44:20]   == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/302-00000000'
[Apr 12 18:44:20] VERBOSE[13989] pbx.c: [Apr 12 18:44:20]     -- Executing [h@default:1] AGI("SIP/302-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 12 18:44:20] VERBOSE[29998] chan_dahdi.c: [Apr 12 18:44:20]     -- Hungup 'DAHDI/pseudo-175926657'
[Apr 12 18:44:20] VERBOSE[29821] chan_dahdi.c: [Apr 12 18:44:20]     -- Hungup 'DAHDI/pseudo-103744993'
[Apr 12 18:44:20] VERBOSE[29998] pbx.c: [Apr 12 18:44:20]   == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/302-00000004'
[Apr 12 18:44:20] VERBOSE[29998] pbx.c: [Apr 12 18:44:20]     -- Executing [h@default:1] AGI("SIP/302-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 12 18:44:20] VERBOSE[29821] pbx.c: [Apr 12 18:44:20]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/300-00000003'
[Apr 12 18:44:20] VERBOSE[29821] pbx.c: [Apr 12 18:44:20]     -- Executing [h@default:1] AGI("SIP/300-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 12 18:44:20] VERBOSE[29998] res_agi.c: [Apr 12 18:44:20]     -- <SIP/302-00000004>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 12 18:44:20] VERBOSE[13989] res_agi.c: [Apr 12 18:44:20]     -- <SIP/302-00000000>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 12 18:44:20] VERBOSE[29821] res_agi.c: [Apr 12 18:44:20]     -- <SIP/300-00000003>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Apr 12 18:45:02] VERBOSE[30247] manager.c: [Apr 12 18:45:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Apr 12 18:45:02] VERBOSE[30248] manager.c: [Apr 12 18:45:02]   == Manager 'sendcron' logged on from 127.0.0.1



Specifically, playing invalid.gsm:
Code: Select all

[Apr 12 18:43:09] VERBOSE[30023] pbx.c: [Apr 12 18:43:09]     -- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000003;2", "8600052,F") in new stack
[Apr 12 18:43:09] VERBOSE[30022] pbx.c: [Apr 12 18:43:09]        > Channel Local/8600052@default-00000003;1 was answered.
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]   == Starting Local/8600052@default-00000003;1 at default,917782934001,1 failed so falling back to exten 's'
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]   == Starting Local/8600052@default-00000003;1 at default,s,1 still failed so falling back to context 'default'
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]     -- Sent into invalid extension 's' in context 'default' on Local/8600052@default-00000003;1
[Apr 12 18:43:09] VERBOSE[30024] pbx.c: [Apr 12 18:43:09]     -- Executing [i@default:1] Playback("Local/8600052@default-00000003;1", "invalid") in new stack
[Apr 12 18:43:09] VERBOSE[30024] file.c: [Apr 12 18:43:09]     -- <Local/8600052@default-00000003;1> Playing 'invalid.gsm' (language 'en')
[Apr 12 18:43:10] VERBOSE[30022] manager.c: [Apr 12 18:43:10]   == Manager 'sendcron' logged off from 127.0.0.1
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 TESTSIPTRUNK does what, exactly?

Postby striker » Tue Apr 14, 2015 6:17 am

as per the log you are dialling 91XXXXXXXXXX
but your dialpaln pattern is _91999NXXXXXX.

as per your dialplan the first 5 digits should be 91999 and the 6th digit should be 2-9 .

better you change your dialplan as below

Code: Select all
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/testcarrier/${EXTEN:2},,Tto)
exten => _91XXXXXXXXXX,3,Hangup()



for more details on asterisk dialplan pattern refer : http://www.voip-info.org/wiki/view/Aste ... n+Patterns
Last edited by striker on Tue Apr 21, 2015 1:11 am, edited 1 time in total.
www.striker24x7.com www.youtube.com/c/striker24x7 Telegram/skype id : striker24x7
striker
 
Posts: 962
Joined: Sun Jun 06, 2010 10:25 am

Re: Dial TESTSIPTRUNK does what, exactly?

Postby THUFIR » Mon Apr 20, 2015 2:44 pm

Thanks, that works perfectly.
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


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