Press 1 Survey Routing To Custom DID Hanging Up Calls

[vili]

Single Server Setup
GoAutoDial CE 3.0-1369195200
Kernel Version 2.6.18-348.6.1.el5 (SMP)



Hello! Ive been reading so many posts on this forum and trying to fix an error that I have been consistently getting with my outbound survey that directs to inbound group when 1 is pressed. The issue is that once the call is picked up and recording plays and 1 is pressed, it plays another recording and than drops the call right after. I would like the call to go to an agent that is part of an inbound group. I use a remote agent to get the campaign running and placing outbound calls which play the survey, and the live agents who are in the same inbound group are in other campaigns dialing. The inbound group is able to receive the call when you dial into it, but not when 1 is pressed. Here is the information from my extensions.conf file, cli debug reporting and my carrier stats. If you could help or point me in the right direction that would be great.

From Carrier:
[vitel-inbound]
type=friend
dtmfmode=auto
host=64.2.142.90
context=trunkinbound
allow=all
insecure=port,invite
canreinvite=no

Dial Plan
exten => xxx.xxx.xxxx,1,Answer

FROM ext-vici.conf:

; VICIDIAL Carrier: -vInbound - vitality inbound
; inbound
exten => 209XXXXXX,1,Answer


FROM extensions.conf

; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,Dial(${TRUNKloop}/9${EXTEN},,to)
exten => _90009.,3,Hangup
exten => _990009.,1,Answer ; Answer the line, Sometimes needs to be removed
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
exten => _990009.,3,Hangup
; DID forwarded calls
exten => _99909*.,1,Answer
exten => _99909*.,2,AGI(agi-VDAD_ALL_inbound.agi)
exten => _99909*.,3,Hangup

From Asterisk CLI :

dialer1*CLI> sip set debug peer vitel-inbound
SIP Debugging Enabled for IP: xxx.xxx.xxxx
[Aug 3 15:38:07] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from 'xxx.xxx.xxxx'
[Aug 3 15:38:09] NOTICE[5202]: chan_sip.c:9330 check_auth: Correct auth, but based on stale nonce received from '"1183"<sip:1183@xxx.xxx.xxxx>;tag=b03da63d'
[Aug 3 15:38:12] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from 'xxx.xxx.xxxx'
[Aug 3 15:38:16] == Parsing '/etc/asterisk/manager.conf': [Aug 3 15:38:16] Found
[Aug 3 15:38:16] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 3 15:38:16] -- Executing [91xxx.xxx.xxxx@default:1] AGI("Local/91xxx.xxx.xxxx@default-8b06,2", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 3 15:38:16] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 3 15:38:16] -- Executing [91xxx.xxx.xxxx@default:2] Dial("Local/91xxx.xxx.xxxx@default-8b06,2", "SIP/1xxx.xxx.xxxx@vitel-outbound||To") in new stack
[Aug 3 15:38:16] -- Called 1xxx.xxx.xxxx@vitel-outbound
[Aug 3 15:38:16] -- SIP/vitel-outbound-00000a9e answered Local/91xxx.xxx.xxxx@default-8b06,2
[Aug 3 15:38:16] > Channel Local/91xxx.xxx.xxxx@default-8b06,1 was answered.
[Aug 3 15:38:16] -- Executing [8366@default:1] Playback("Local/91xxx.xxx.xxxx@default-8b06,1", "sip-silence") in new stack
[Aug 3 15:38:16] -- <Local/91xxx.xxx.xxxx@default-8b06,1> Playing 'sip-silence' (language 'en')
[Aug 3 15:38:16] WARNING[20218]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Aug 3 15:38:16] -- Executing [h@default:1] DeadAGI("Local/91xxx.xxx.xxxx@default-8b06,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0") in new stack
[Aug 3 15:38:16] -- Executing [8366@default:2] AGI("SIP/vitel-outbound-00000a9e", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 3 15:38:16] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 3 15:38:16] -- Executing [8366@default:3] AGI("SIP/vitel-outbound-00000a9e", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Aug 3 15:38:16] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Aug 3 15:38:16] NOTICE[20218]: channel.c:2616 __ast_read: Dropping incompatible voice frame on SIP/vitel-outbound-00000a9e of format gsm since our native format has changed to 0x4 (ulaw)
[Aug 3 15:38:17] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20160803-153817_xxx.xxx.xxxx)
[Aug 3 15:38:17] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Aug 3 15:38:17] -- Playing '85100008' (escape_digits=12) (sample_offset 0)
[Aug 3 15:38:17] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from 'xxx.xxx.xxxx'
[Aug 3 15:38:17] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---0-----0 completed, returning 0
[Aug 3 15:38:17] == Spawn extension (default, 91xxx.xxx.xxxx', 2) exited non-zero on 'Local/91xxx.xxx.xxxx@default-8b06,2'
[Aug 3 15:38:18] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 3 15:38:22] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.2.232'
[Aug 3 15:38:27] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.2.232'
[Aug 3 15:38:32] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.2.232'
[Aug 3 15:38:37] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.2.232'
[Aug 3 15:38:39] -- Playing 'US_pol_survey_transfer' (escape_digits=) (sample_offset 0)
[Aug 3 15:38:42] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from '192.168.2.232'
[Aug 3 15:38:46] ERROR[20218]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Aug 3 15:38:46] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Aug 3 15:38:46] -- Executing [xxx.xxx.xxxx@default:1] Answer("SIP/vitel-outbound-00000a9e", "") in new stack
[Aug 3 15:38:46] == Auto fallthrough, channel 'SIP/vitel-outbound-00000a9e' status is 'UNKNOWN'
[Aug 3 15:38:46] -- Executing [h@default:1] DeadAGI("SIP/vitel-outbound-00000a9e", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 3 15:38:46] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 3 15:38:47] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from 'xxx.xxx.x.xxxx''
[Aug 3 15:38:51] Reliably Transmitting (NAT) to 64.2.142.90:5060:
OPTIONS sip:xxx.xxx.x.xxxx';cpd=on SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.x.xxxx';branch=z9hG4bK37b7cb38;rport
From: "asterisk" <sip:asterisk@xxx.xxx.x.xxxx'>;tag=as047559e1
To: <sip:64.2.142.90;cpd=on>
Contact: <sip:asterisk@xxx.xxx.xxxx>
Call-ID: 522dc40e51f4c6af3055a46d47549657@xxx.xxx.xxxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 03 Aug 2016 22:38:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Aug 3 15:38:52]
<--- SIP read from xxx.xxx.xxxx --->
SIP/2.0 200 Options, there are none.
Via: SIP/2.0/UDP xxx.xxx.x.xxxx';branch=z9hG4bK37b7cb38;rport=5060
From: "asterisk" <sip:asterisk@xxx.xxx.xxxx>;tag=as047559e1
To: <sip:64.2.142.90;cpd=on>;tag=85b5bce8d0418e4b3793b5cf6301ea1c.5752
Call-ID: 522dc40e51f4c6af3055a46d47549657@xxx.xxx.xxxx
CSeq: 102 OPTIONS
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0


<------------->
[Aug 3 15:38:52] --- (8 headers 0 lines) ---
[Aug 3 15:38:52] Really destroying SIP dialog '522dc40e51f4c6af3055a46d47549657@xxx.xxx.x.xxxx' Method: OPTIONS
[Aug 3 15:38:52] NOTICE[7574]: rtp.c:1378 ast_rtp_read: Unknown RTP codec 126 received from 'xxx.xxx.x.xxxx'
dialer1*CLI> logger mute
Console is muted.



If I missed any information or need to check anything, let me know, I would love any feed back thanks.

[mflorell]

Just a starting point, if the vicidial version is over 4 years old like yours is, we won't even try to troubleshoot it. We will recommend an upgrade to the latest version, then we'll see if the issue is still happening.

[williamconley]

1) Was this working but stopped recently? If it's a new configuration, attempting a transfer to an ingroup, there are several errors (or non-standard items, all likely to cause failure as described.)
2) As Matt points out, very old code and you have some non-standard conf files in use. Your Carrier dialplan does not have agi/dial/hangup as it should, or you just listed it incorrectly leaving out any useful details. LOL
3) Newer version of Survey campaign allows the Survey Method to be ... Call Menu. And the Call Menu can immediately drop the call to an Ingroup. Thus, NO custom code or modified configuration files to accomplish this task.

[vili]

The campaign is able to call out send the survey recording, but when press 1 occurs the call drops. You can reach the inbound group by directly calling from a phone not on the network.

My extensions-vicidial.conf looks like this:

[vicidial-auto-external]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Local Server: 192.168.2.100
exten => _192*168*002*100*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: -vInbound - vitality inbound
; inbound
exten => 209xxxxxxx,1,Answer
exten => 209xxxxxxx,1,Answer
exten => 209xxxxxxx,1,Answer
; VICIDIAL Carrier: _vitelity - Vitelity
; Prepaid carrier used as primary on PBX
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@vitel-outbound,,To)
exten => _91NXXNXXXXXX,3,Hangup

exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@vitel-outbound,,To)
exten => _1NXXNXXXXXX,3,Hangup

exten => _NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _NXXXXXX,2,Dial(SIP/1209${EXTEN}@vitel-outbound,,To)
exten => _NXXXXXX,3,Hangup


I've tried the call menu option as well but still having issues with the call dropping, im thinking that it has something to do with the dialplan/routing/extensions
Heres my dial plan for the call menu:
; custom dialplan entries
exten => _X.,1,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => _X.,n,Goto(default,${EXTEN},1)

; Inbound Call Menu
[lexingtonCallMenu]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,Lexington_Law-----YES-----lexingtonCallMenu--------------------)
exten => s,n,Set(INVCOUNT=0)
exten => s,n,Background()
exten => s,n,WaitExten(1)

exten => t,1,AGI(agi-VDAD_ALL_inbound.agi,CIDLOOKUPRC-----LB-----Lexington_Law-----lexingtonCallMenu--------------------998-----1-----1485------------------------------)
exten => i,1,Goto(s,4)
; hangup
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

[vicidial-auto-external]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})



And this is my inbound extensions.conf

[trunkinbound]
; DID call routing process
; exten => _XXXXXXXXXX,1,AGI(agi-DID_route.agi) ; use this one instead of the one below if you are having delay issues, and match to number of received digits
exten => _X.,1,AGI(agi-DID_route.agi)



; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,Dial(${TRUNKloop}/9${EXTEN},,to)
exten => _90009.,3,Hangup
exten => _990009.,1,Answer ; Answer the line, Sometimes needs to be removed
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
exten => _990009.,3,Hangup
; DID forwarded calls
exten => _99909*.,1,Answer
exten => _99909*.,2,AGI(agi-VDAD_ALL_inbound.agi)
exten => _99909*.,3,Hangup


Im just not to sure how to fix this, ive been trying to look around on these forums but i finally had to post something.
Thanks for your help, im still trying to learn this. Were actually going to be setting up a server cluster in the next few weeks, so I think thats we are going to up grade.

[williamconley]

My advice: build a new system that's Stock and use the Vicidial Manager's Manual to configure it.

since you didn't answer my earlier question, that's the only opinion I can offer.