Unable to link carrier and DID

[Mysdradon]

Hi everyone.

Happy New Year

I'm building a solution based on Vicidial / Goautodial for my call center department.
Version GoAutodial CE 2.1

The idea is to be able to call from Vicidial by Campaign (with Leads and script) and to answer call by Vicidial too (another team).
Firstly, I want to pass and answer for and from France (phone number is like : +33 1 23 45 67 89 for international calls, 01 23 45 67 89 for internal call)

I've configured the Carrier like this :

Name : T3312456789

Description : blablabla

Registration String : register => xxxxxx:yyyyyyy@78.153.253.76:5060/xxxxxxx

Account ENTRY

[CARRIEREOS]
disallow=all
allow=ulaw
allow=alaw
type=peer
username=xxxxxxx
secret=yyyyyyy
host=78.153.253.76
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
fromuser=xxxxxxx
canreinvite=no
reinvite=no

GLOBAL STRING : NULL

DIALPLAN ENTRY

exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _.,2,Dial(SIP/${EXTEN:1}@CARRIEREOS,,tTo)
exten => _.,3,Hangup

And I've configured the DIDs like this : (I've tried a lot of configuration)

DID : 0033123456789 / 33123456789 / xxxxxxx / 0123456789 / 123456789 /
Description : blablabla
ACTIVE : Y
DID ROUTE : PHONE
PHONE EXTENSION : 8001 (Vicidial Standard phone)
SERVER IP : 192.168.1.2 (Vicidial Standard config)

When I'm calling the phone number, I've got this message :

[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- Executing [33182880165@default:1] AGI("SIP/78.153.253.76-00000003", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- Executing [33182880165@default:2] Dial("SIP/78.153.253.76-00000003", "SIP/3182880165@CARRIEREOS||tTo") in new stack
[Jan 14 16:41:22] WARNING[12595] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] == Everyone is busy/congested at this time (1:0/0/1)
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- Executing [33182880165@default:3] Hangup("SIP/78.153.253.76-00000003", "") in new stack
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] == Spawn extension (default, 33182880165, 3) exited non-zero on 'SIP/78.153.253.76-00000003'
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- Executing [h@default:1] DeadAGI("SIP/78.153.253.76-00000003", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- Executing [h@default:2] Dial("SIP/78.153.253.76-00000003", "SIP/@CARRIEREOS||tTo") in new stack
[Jan 14 16:41:22] WARNING[12595] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] == Everyone is busy/congested at this time (1:0/0/1)
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] -- Executing [h@default:3] Hangup("SIP/78.153.253.76-00000003", "") in new stack
[Jan 14 16:41:22] VERBOSE[12595] logger.c: [Jan 14 16:41:22] == Spawn extension (default, h, 3) exited non-zero on 'SIP/78.153.253.76-00000003'

I don't want to use Vicidial like a Freepbx, but I tell myself that if I can configure this, I would be able to understand how it's working and I'll go further.

In the extensions.vicidial.conf there is :

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569




; logging of all outbound calls from agent phones
[defaultlog]
exten => s,1,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----defaultlog--------------------)
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,2)
; hangup
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${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]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Local Server: 192.168.1.2
exten => _192*168*001*002*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: T33123456789 - CARRIER EOS
; Connextion IPBX Selfone pour EOS
exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _.,2,Dial(SIP/${EXTEN:1}@CARRIEREOS,,tTo)
exten => _.,3,Hangup



; Voicemail Extensions:
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,2,Voicemail(${EXTEN:14}|u)
exten => _85026666666666.,3,Hangup
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
exten => 8501,1,VoicemailMain(s${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


; Phones direct dial extensions:
exten => 7001,1,Dial(IAX2/7001|60|)
exten => 7001,2,Goto(default,850266666666667001,1)
exten => 7002,1,Dial(IAX2/7002|60|)
exten => 7002,2,Goto(default,850266666666667002,1)
exten => 7003,1,Dial(IAX2/7003|60|)
exten => 7003,2,Goto(default,850266666666667003,1)
exten => 7004,1,Dial(IAX2/7004|60|)
exten => 7004,2,Goto(default,850266666666667004,1)
exten => 7005,1,Dial(IAX2/7005|60|)
exten => 7005,2,Goto(default,850266666666667005,1)
exten => 7006,1,Dial(IAX2/7006|60|)
exten => 7006,2,Goto(default,850266666666667006,1)
exten => 7007,1,Dial(IAX2/7007|60|)
exten => 7007,2,Goto(default,850266666666667007,1)
exten => 7008,1,Dial(IAX2/7008|60|)
exten => 7008,2,Goto(default,850266666666667008,1)
exten => 7009,1,Dial(IAX2/7009|60|)
exten => 7009,2,Goto(default,850266666666667009,1)
exten => 7010,1,Dial(IAX2/7010|60|)
exten => 7010,2,Goto(default,850266666666667010,1)
exten => 7011,1,Dial(IAX2/7011|60|)
exten => 7011,2,Goto(default,850266666666667011,1)
exten => 7012,1,Dial(IAX2/7012|60|)
exten => 7012,2,Goto(default,850266666666667012,1)
exten => 7013,1,Dial(IAX2/7013|60|)
exten => 7013,2,Goto(default,850266666666667013,1)
exten => 7014,1,Dial(IAX2/7014|60|)
exten => 7014,2,Goto(default,850266666666667014,1)
exten => 7015,1,Dial(IAX2/7015|60|)
exten => 7015,2,Goto(default,850266666666667015,1)
exten => 7016,1,Dial(IAX2/7016|60|)
exten => 7016,2,Goto(default,850266666666667016,1)
exten => 7017,1,Dial(IAX2/7017|60|)
exten => 7017,2,Goto(default,850266666666667017,1)
exten => 7018,1,Dial(IAX2/7018|60|)
exten => 7018,2,Goto(default,850266666666667018,1)
exten => 7019,1,Dial(IAX2/7019|60|)
exten => 7019,2,Goto(default,850266666666667019,1)
exten => 7020,1,Dial(IAX2/7020|60|)
exten => 7020,2,Goto(default,850266666666667020,1)
exten => 8001,1,Dial(SIP/8001|60|)
exten => 8001,2,Goto(default,850266666666668001,1)
exten => 8002,1,Dial(SIP/8002|60|)
exten => 8002,2,Goto(default,850266666666668002,1)
exten => 8003,1,Dial(SIP/8003|60|)
exten => 8003,2,Goto(default,850266666666668003,1)
exten => 8004,1,Dial(SIP/8004|60|)
exten => 8004,2,Goto(default,850266666666668004,1)
exten => 8005,1,Dial(SIP/8005|60|)
exten => 8005,2,Goto(default,850266666666668005,1)
exten => 8006,1,Dial(SIP/8006|60|)
exten => 8006,2,Goto(default,850266666666668006,1)
exten => 8007,1,Dial(SIP/8007|60|)
exten => 8007,2,Goto(default,850266666666668007,1)
exten => 8008,1,Dial(SIP/8008|60|)
exten => 8008,2,Goto(default,850266666666668008,1)
exten => 8009,1,Dial(SIP/8009|60|)
exten => 8009,2,Goto(default,850266666666668009,1)
exten => 8010,1,Dial(SIP/8010|60|)
exten => 8010,2,Goto(default,850266666666668010,1)
exten => 8011,1,Dial(SIP/8011|60|)
exten => 8011,2,Goto(default,850266666666668011,1)
exten => 8012,1,Dial(SIP/8012|60|)
exten => 8012,2,Goto(default,850266666666668012,1)
exten => 8013,1,Dial(SIP/8013|60|)
exten => 8013,2,Goto(default,850266666666668013,1)
exten => 8014,1,Dial(SIP/8014|60|)
exten => 8014,2,Goto(default,850266666666668014,1)
exten => 8015,1,Dial(SIP/8015|60|)
exten => 8015,2,Goto(default,850266666666668015,1)
exten => 8016,1,Dial(SIP/8016|60|)
exten => 8016,2,Goto(default,850266666666668016,1)
exten => 8017,1,Dial(SIP/8017|60|)
exten => 8017,2,Goto(default,850266666666668017,1)
exten => 8018,1,Dial(SIP/8018|60|)
exten => 8018,2,Goto(default,850266666666668018,1)
exten => 8019,1,Dial(SIP/8019|60|)
exten => 8019,2,Goto(default,850266666666668019,1)
exten => 8020,1,Dial(SIP/8020|60|)
exten => 8020,2,Goto(default,850266666666668020,1)

; END OF FILE

Need something else ??
Ho yeah ... the phone is online ...

[ZoVoS]

exten => _.,2,Dial(SIP/${EXTEN:1}@CARRIEREOS,,tTo)???

ill just test if that works however I would use

exten => _.,2,Dial(SIP/CARRIEROS/${EXTEN:1},tTo)

If that's not the problem ill give it a proper read through


For testing purposes


exten => _55555X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _55555X.,n,Dial(SIP/44${EXTEN:6}@***MY CARRIER (ME)***,,tTo)
exten => _55555X.,n,Hangup

(dropping 6 because I like to dial with a 0 but I only accept 44)

just set my carrier settings to that,




and the outcome was the same as you recived

[Jan 14 20:37:29] -- Executing [555550**********@default:1] AGI("SIP/gs102-0001f080", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 14 20:37:29] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 14 20:37:29] -- Executing [555550**********@default:2] Dial("SIP/gs102-0001f080", "SIP/44**********@***MY CARRIER (ME)***||tTo") in new stack
[Jan 14 20:37:29] WARNING[29694]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 14 20:37:29] == Everyone is busy/congested at this time (1:0/0/1)
[Jan 14 20:37:29] -- Executing [555550**********@default:3] Hangup("SIP/gs102-0001f080", "") in new stack
[Jan 14 20:37:29] == Spawn extension (default, 555550**********, 3) exited non-zero on 'SIP/gs102-0001f080'
[Jan 14 20:37:29] -- Executing [h@default:1] DeadAGI("SIP/gs102-0001f080", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jan 14 20:37:29] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jan 14 20:37:31] -- Got SIP response 405 "Method Not Allowed" back from *********

[Mysdradon]

i've just made a test with your configuration ... same problem.

[ZoVoS]

can you paste the asterisk command line output now? or is it exactly the same problem

[Mysdradon]

Here it is :

Code: Select all

[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- Executing [33182880165@default:1] AGI("SIP/78.153.253.76-00000004", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- Executing [33182880165@default:2] Dial("SIP/78.153.253.76-00000004", "SIP/CARRIEREOS/3182880165|tTo") in new stack
[Jan 15 07:41:10] WARNING[20236] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 15 07:41:10] WARNING[20236] app_dial.c: Invalid timeout specified: 'tTo'
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- Executing [33182880165@default:3] Hangup("SIP/78.153.253.76-00000004", "") in new stack
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]   == Spawn extension (default, 33182880165, 3) exited non-zero on 'SIP/78.153.253.76-00000004'
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- Executing [h@default:1] DeadAGI("SIP/78.153.253.76-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- Executing [h@default:2] Dial("SIP/78.153.253.76-00000004", "SIP/CARRIEREOS/|tTo") in new stack
[Jan 15 07:41:10] WARNING[20236] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 15 07:41:10] WARNING[20236] app_dial.c: Invalid timeout specified: 'tTo'
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]     -- Executing [h@default:3] Hangup("SIP/78.153.253.76-00000004", "") in new stack
[Jan 15 07:41:10] VERBOSE[20236] logger.c: [Jan 15 07:41:10]   == Spawn extension (default, h, 3) exited non-zero on 'SIP/78.153.253.76-00000004'
[Jan 15 07:42:01] VERBOSE[20456] logger.c: [Jan 15 07:42:01]   == Parsing '/etc/asterisk/manager.conf': [Jan 15 07:42:01] VERBOSE[20456] logger.c: [Jan 15 07:42:01] Found
[Jan 15 07:42:01] VERBOSE[20456] logger.c: [Jan 15 07:42:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 07:42:01] VERBOSE[20460] logger.c: [Jan 15 07:42:01]   == Parsing '/etc/asterisk/manager.conf': [Jan 15 07:42:01] VERBOSE[20460] logger.c: [Jan 15 07:42:01] Found
[Jan 15 07:42:01] VERBOSE[20460] logger.c: [Jan 15 07:42:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 07:42:01] VERBOSE[20460] logger.c: [Jan 15 07:42:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 15 07:42:03] VERBOSE[20456] logger.c: [Jan 15 07:42:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 15 07:42:06] VERBOSE[20823] logger.c: [Jan 15 07:42:06]   == Parsing '/etc/asterisk/manager.conf': [Jan 15 07:42:06] VERBOSE[20823] logger.c: [Jan 15 07:42:06] Found
[Jan 15 07:42:06] VERBOSE[20823] logger.c: [Jan 15 07:42:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 07:42:06] VERBOSE[20823] logger.c: [Jan 15 07:42:06]   == Manager 'sendcron' logged off from 127.0.0.1
                                                                                                       



[ZoVoS]

may not matter but I think it will, I missed a , in my explanation which would be why yours is
SIP/CARRIEREOS/3182880165|tTo")

and mines

"SIP/***/***********||tTor")

[Jan 15 04:33:52] WARNING[11246]: app_dial.c:1424 dial_exec_full: Invalid timeout specified: 'tTor'
Well that caused an error again. lets iron them all out =P

exten => _.,2,Dial(SIP/CARRIEREOS/${EXTEN:1},,tTo)

[Mysdradon]

Here is the new one :

Code: Select all

[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- Executing [33182880165@default:1] AGI("SIP/78.153.253.76-00000005", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- Executing [33182880165@default:2] Dial("SIP/78.153.253.76-00000005", "SIP/CARRIEREOS/3182880165||tTo") in new stack
[Jan 15 09:03:37] WARNING[2629] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]   == Everyone is busy/congested at this time (1:0/0/1)
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- Executing [33182880165@default:3] Hangup("SIP/78.153.253.76-00000005", "") in new stack
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]   == Spawn extension (default, 33182880165, 3) exited non-zero on 'SIP/78.153.253.76-00000005'
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- Executing [h@default:1] DeadAGI("SIP/78.153.253.76-00000005", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- Executing [h@default:2] Dial("SIP/78.153.253.76-00000005", "SIP/CARRIEREOS/||tTo") in new stack
[Jan 15 09:03:37] WARNING[2629] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]   == Everyone is busy/congested at this time (1:0/0/1)
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]     -- Executing [h@default:3] Hangup("SIP/78.153.253.76-00000005", "") in new stack
[Jan 15 09:03:37] VERBOSE[2629] logger.c: [Jan 15 09:03:37]   == Spawn extension (default, h, 3) exited non-zero on 'SIP/78.153.253.76-00000005'
                                                                                                       



I don't understand why it is not working.
If someone have a manual for the Account Entries, maybe he can explain to me because I don't see where the problem lies

[ashis103]

Dear Team . Could you please help to set proper dial plan .Trunk setup done but call not heat carrier VOS .Please help me
My Trunk Confg :
[ZTELVOS]
disallow=all
allow=alaw
allow=ulaw
allow=g729
type=friend
username=ZZZZ
secret=YYYYY
host=188.94.240.96
port=5060
dtmfmode=rfc2833
qualify=yes
insecure=very
nat=no
-----------------------
Dial Plan :
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,n,Dial(${87649879:T456537UyhWg@188.94.240.96:5060}/${EXTEN},,tTor)
exten => _X.,n,Hangup

Asteric output :

[Jan 15 06:44:18] -- Executing [4420070964293@default:1] AGI("SIP/105-0000003b", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 15 06:44:18] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 15 06:44:18] -- Executing [4420070964293@default:2] Dial("SIP/105-0000003b", "/4420070964293||tTor") in new stack
[Jan 15 06:44:18] WARNING[3485]: channel.c:3776 ast_request: No channel type registered for ''
[Jan 15 06:44:18] WARNING[3485]: app_dial.c:1310 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
[Jan 15 06:44:18] == Everyone is busy/congested at this time (1:0/0/1)
[Jan 15 06:44:18] -- Executing [4420070964293@default:3] Hangup("SIP/105-0000003b", "") in new stack
[Jan 15 06:44:18] == Spawn extension (default, 4420070964293, 3) exited non-zero on 'SIP/105-0000003b'
[Jan 15 06:44:18] -- Executing [h@default:1] DeadAGI("SIP/105-0000003b", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
[Jan 15 06:44:18] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
SUCI-SIMPLECALL*CLI>

[DomeDan]

I don't understand why it is not working.
If someone have a manual for the Account Entries, maybe he can explain to me because I don't see where the problem lies

Good job posting your specs and cli-output in the first post!

I'm not a goautodial guy but heres some debugging help, use this command in the asterisk cli to view what part of the dialplan is executed when your dialing the specified number:
dialplan show 33182880165@default

because of how your dialplan is set the number that is being called at CARRIEREOS is 3182880165 (your dialplan removes the first digit
this is because under campaign settings you should have a dial prefix set, I use 9 as dial prefix
and you should set the leads phone_code to 0 or 33 and phone_number to something like 123456789 or 182880165

your dialplan matches every number, it don't seam very wise but I'm not familiar with how goautodial solves this...

but your dialplan would be something like this instead (when using dial prefix = 9):
exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(SIP/${EXTEN:1}@CARRIEREOS,,tTo)
exten => _9.,3,Hangup

but I would recommend to only setup dialplans for countries your are gonna call to.




ashis103: start a new thread

[Mysdradon]

Thanks for help again and sorry to answer only today. I was in meeting all the day yesterday.

Here is the dial plan

Code: Select all

Asterisk Call Manager/1.0
Response: Success
Message: Authentication accepted
Response: Follows
Privilege: Command
ActionID: 600d5e9aa7b60012d7b9cf57b3f91bda
[ Included context 'vicidial-auto' created by 'pbx_config' ]
  '_.' =>           1. AGI(agi://127.0.0.1:4577/call_log)         [pbx_config]
                    2. Dial(SIP/xxxxx:yyyyy@78.153.253.76:5060/${EXTEN:2}||tTor) [pbx_config]
                    3. Hangup()                                   [pbx_config]
-= 1 extension (3 priorities) in 1 context. =-


If I put 9 in the dial plan. It tells me that It cannot find the extension. So I delete it and I've got this message :

Code: Select all

[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- Executing [33182880165@default:1] AGI("SIP/78.153.253.76-0000000b", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- Executing [33182880165@default:2] Dial("SIP/78.153.253.76-0000000b", "SIP/3182880165@CARRIEREOS||tTo") in new stack
[Jan 17 07:05:27] WARNING[6905] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]   == Everyone is busy/congested at this time (1:0/0/1)
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- Executing [33182880165@default:3] Hangup("SIP/78.153.253.76-0000000b", "") in new stack
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]   == Spawn extension (default, 33182880165, 3) exited non-zero on 'SIP/78.153.253.76-0000000b'
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- Executing [h@default:1] DeadAGI("SIP/78.153.253.76-0000000b", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- Executing [h@default:2] Dial("SIP/78.153.253.76-0000000b", "SIP/@CARRIEREOS||tTo") in new stack
[Jan 17 07:05:27] WARNING[6905] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]   == Everyone is busy/congested at this time (1:0/0/1)
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]     -- Executing [h@default:3] Hangup("SIP/78.153.253.76-0000000b", "") in new stack
[Jan 17 07:05:27] VERBOSE[6905] logger.c: [Jan 17 07:05:27]   == Spawn extension (default, h, 3) exited non-zero on 'SIP/78.153.253.76-0000000b'


I need that everyone from world would be able to call our number. That's why we have this special dial plan (that's what I've been told)

[Mysdradon]

I've also tried this config

Code: Select all

exten => _33XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _33XXXXXXXXX,2,Dial(SIP/${EXTEN:1}@CARRIEREOS,,tTo)
exten => _33XXXXXXXXX,3,Hangup


Same problem ....

[DomeDan]

Im pretty sure its only outbound

so if you set 9 as "dial prefix" (as i told you to) in "Vicidial admin interface" -> "campaigns" -> "the campaings you use" -> "detail view"
and then change the dialplan to:
exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(SIP/${EXTEN:1}@CARRIEREOS,,tTo)
exten => _9.,3,Hangup

and try again you will probably get a different result

[Mysdradon]

Yes, it tells me : No extension found

[DomeDan]

it should find the extension. have you done a reload in asterisk?

alright the next option is to remove dial prefix and set dialplan to:
exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _.,2,Dial(SIP/${EXTEN}@CARRIEREOS,,tTo)
exten => _.,3,Hangup

I personally would not do that though, I would try to make the dialplan work with a dial prefix

[Mysdradon]

Here is the result

Code: Select all


[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]     -- Executing [33182880165@default:1] AGI("SIP/78.153.253.76-0000000d", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]     -- Executing [33182880165@default:2] Dial("SIP/78.153.253.76-0000000d", "SIP/33182880165@CARRIEREOS||tTo") in new stack
[Jan 21 23:29:14] WARNING[14853] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]   == Everyone is busy/congested at this time (1:0/0/1)
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]     -- Executing [33182880165@default:3] Hangup("SIP/78.153.253.76-0000000d", "") in new stack
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]   == Spawn extension (default, 33182880165, 3) exited non-zero on 'SIP/78.153.253.76-0000000d'
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]     -- Executing [h@default:1] DeadAGI("SIP/78.153.253.76-0000000d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Jan 21 23:29:14] VERBOSE[14853] logger.c: [Jan 21 23:29:14]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
                                                                                                       


[williamconley]

it tries to dial "SIP/33182880165@CARRIEREOS||tTo" carriereos ...
but it cannot because there is a problem creating a channel to that sip account: [Jan 21 23:29:14] WARNING[14853] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)

sip show peers
sip show registry

is carriereos registered? is it unreachable?