Calls Are Hungup by System on Answer

[savagenoob]

Hello, I setup Goautodial 4 from scratch, everything seems to be working fine except when I make predictive dialing calls, the system hangs up on the call instead of transferring to logged in agent. I’ve done my best research on this and cannot find anything. I am assuming it has something to do with either my extensions.conf or my dial plan.

Dialplan:

exten => _91.,1,Set(callerid(num)=+1mynumber)
exten => _91.,2,Set(callerid(ani)=my company)
exten => _91.,3,AGI(agi://localhost:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@callcentric,55,o)
exten => _91,5,Hangup

Asterisk CLI output with verbose set at 21:

-- SIP/callcentric-0000015f answered Local/91NUMBERDIALED@default-0000025f;2
-- Local/91NUMBERDIALED@default-0000025f;1 answered
-- Executing [138369@default:1] AGI("Local/91NUMBERDIALED at default-0000025f;1", "agi-VDAD_local_optimize.agi,V4030331230000032090") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi
== Manager 'sendcron' logged off from localhost
-- Channel SIP/callcentric-0000015f joined 'simple_bridge' basic-bridge <4738081b-3633-404d-a499-302df08cb048>
-- Channel Local/91NUMBERDIALED at default-0000025f;2 joined 'simple_bridge' basic-bridge <4738081b-3633-404d-a499-302df08cb048>
== Manager 'sendcron' logged on from localhost
-- Channel SIP/callcentric-0000015f left 'simple_bridge' basic-bridge <4738081b-3633-404d-a499-302df08cb048>
-- Channel Local/91NUMBERDIALED at default-0000025f;2 left 'simple_bridge' basic-bridge <4738081b-3633-404d-a499-302df08cb048>
-- Executing [8369@default:1] AGI("SIP/callcentric-0000015f", "agi://127.0.0.1:4577/call_log") in new stack
== Spawn extension (default, 91NUMBERDIALED, 4) exited non-zero on 'Local/91NUMBERDIALED@default-0000025f;2'
-- Executing [h@default:1] AGI("Local/91NUMBERDIALED@default-0000025f;2", "agi://localhost :4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----0-----SIP 200 OK)") in new stack
-- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=18989146))
-- <SIP/callcentric-0000015f>AGI Script agi://localhost:4577/call_log completed, returning 0
-- Executing [8369@default:2] AMD("SIP/callcentric-0000015f", "2000,2000,1000,5000,120,50,4,256") in new stack
-- AMD: SIP/callcentric-0000015f (N/A) (N/A) (Fmt: slin)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256] maximumWordLength [5000]
-- AMD: Channel [SIP/callcentric-0000015f]. Changed state to STATE_IN_SILENCE
-- AMD: Channel [SIP/callcentric-00000160]. ANSWERING MACHINE: silenceDuration:2000 initialSilence:2000
-- Executing [8369@default:3] AGI("SIP/callcentric-00000160", "VD_amd.agi,8369") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/VD_amd.agi
-- <Local/91NUMBERDIALED@default-00000260;1>AGI Script agi://localhost:4577/call_log--HVcauses ... ----------) completed, returning 0
-- <SIP/callcentric-00000160> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
-- <SIP/callcentric-00000160> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
-- <SIP/callcentric-00000160>AGI Script VD_amd.agi completed, returning 4
== Spawn extension (default, 8369, 3) exited non-zero on 'SIP/callcentric-00000160'
-- Executing [h@default:1] AGI("SIP/callcentric-00000160", "agi://localhost:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
-- <SIP/callcentric-00000160>AGI Script agi://localhost:4577/call_log--HVcauses ... -------SIP 200 OK) completed, returning 0
Scheduling destruction of SIP dialog '627fb1b470cb2e196e78947b51845624@callcentric.com' in 6784 ms (Method: INVITE)
Reliably Transmitting (NAT) to callcentric:5080:
BYE sip:323ef9cbf6e9db4102e37f63daa1ebea at callcentric:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.233:5061;branch=z9hG4bK7b616856;rport
Max-Forwards: 70
From: "V4030331230000032091" <sip:1777USER at callcentric:5061>;tag=as25ac9fb3
To: <sip:1NUMBERDIALED at callcentric>;tag=12vQNF2rK4rFF
Call-ID: 627fb1b470cb2e196e78947b51845624@callcentric
CSeq: 104 BYE
User-Agent: Asterisk PBX 13.17.2-vici
Proxy-Authorization: Digest username="1777USER", realm="callcentric", algorithm=MD5, uri="sip:sip:callcentric", nonce="50129338849f25a58a45a992cb6aec44", response="d473b33059d2e0af99eb6660aa4cb772"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

It seems as though it tries to transfer to extension 8369 using the [default] context in extensions.conf. When predictive dialing, how do I just have it transfer to any agent that is available instead? I am not sure why it is pointing to that extension. Here is the part of extensions.conf with that extension listed:

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,AGI(agi:// localhost :4577/call_log)
exten => 8369,n,AMD(2000,2000,1000,5000,120,50,4,256)
exten => 8369,n,AGI(VD_amd.agi,${EXTEN})
exten => 8369,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,n,Hangup()

Thank you for your help in advance, please understand I am relatively new to this.

Centos 7 Kernal 3.10.0-1160.21.1.el7.x86_64
GoAutodial 4 from scratch
Asterisk 13.17.2-vici
10.1.36-MariaDB MariaDB Server
Urls removed, username and phone numbers are masked

[carpenox]

8369 is the AMD dialplan, and for goautodial question, point those to the subtopic for that as im sure Bill will tell you once he hits you with the "welcome to the forum" message and moves it accordingly, but as far as your question goes, irt seems the call was connected fine and then the logs after that where it disconnects once the agent gets it i do not see, you are missing the further logs......

[savagenoob]

OK, I have full log files I can send if this forum allows, maybe even through a DM if possible. Yeah the phone calls go out fine, its just when they are answered, instead of connecting the call to the live agent, it just pauses for a second and hangs up.

[carpenox]

sure, hit me up on skype ↓