vicidial.org

by **noobdialer** » Tue Jun 05, 2018 7:32 pm

Hello Guys,
I'm new with vicidial, I hope to help me to solve my problem.
I install GoautoDial and add carrier then I log in with an agent and try to "manual dial", work perfectly, hear sound from two sides and no problem with that.

then I try to test "auto dial" but after start auto dialer and answer my phone (outside) after 8-10 seconds call hangup and I don't know why,
after check cli, I found, AMD hangup call after I answered, then I decided to turn off AMD from campaign setting and try another auto dial but again call drop
when I answer the phone and after I check cli just saw sip-silence ...

I post all my logs in this below, please check my log and help me to track and find the problem.

CLI OUTOUT MANUAL DIAL ( no problem )

Code: Select all: [Jun 6 08:24:07] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:24:16] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:24:16] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000001;2", "8600051,F") in new stack [Jun 6 08:24:16] > Channel Local/8600051@default-00000001;1 was answered. [Jun 6 08:24:16] -- Executing [4280625217989355645263@default:1] AGI("Local/8600051@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack [Jun 6 08:24:16] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=93073185)) [Jun 6 08:24:16] -- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jun 6 08:24:16] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:24:16] -- Executing [4280625217989355645263@default:2] Dial("Local/8600051@default-00000001;1", "SIP/989355645263@999,,tTo") in new stack [Jun 6 08:24:16] == Using SIP RTP CoS mark 5 [Jun 6 08:24:16] -- Called SIP/989355645263@999 [Jun 6 08:24:17] -- SIP/999-00000002 is making progress passing it to Local/8600051@default-00000001;1 [Jun 6 08:24:23] -- SIP/999-00000002 answered Local/8600051@default-00000001;1 [Jun 6 08:24:45] -- Executing [h@default:1] AGI("Local/8600051@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----29-----22") in new stack [Jun 6 08:24:45] -- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----29-----22 completed, returning 0 [Jun 6 08:24:45] == Spawn extension (default, 4280625217989355645263, 2) exited non-zero on 'Local/8600051@default-00000001;1' [Jun 6 08:24:45] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000001;2' [Jun 6 08:24:45] -- Executing [h@default:1] AGI("Local/8600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jun 6 08:24:45] -- <Local/8600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jun 6 08:24:50] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:24:50] == Manager 'sendcron' logged off from 127.0.0.1 go*CLI>

CLI OUTPUT AUTO DIAL ( problem )

Code: Select all: [Jun 6 08:20:31] -- Executing [4280625217989355645263@default:1] AGI("Local/4280625217989355645263@default-00000000;2", "agi://127.0.0.1:4577/call_log") in new stack [Jun 6 08:20:31] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=93073185)) [Jun 6 08:20:31] -- <Local/4280625217989355645263@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jun 6 08:20:31] -- Executing [4280625217989355645263@default:2] Dial("Local/4280625217989355645263@default-00000000;2", "SIP/989355645263@999,,tTo") in new stack [Jun 6 08:20:31] == Using SIP RTP CoS mark 5 [Jun 6 08:20:31] -- Called SIP/989355645263@999 [Jun 6 08:20:31] -- SIP/999-00000001 is making progress passing it to Local/4280625217989355645263@default-00000000;2 [Jun 6 08:20:40] -- SIP/999-00000001 answered Local/4280625217989355645263@default-00000000;2 [Jun 6 08:20:40] > Channel Local/4280625217989355645263@default-00000000;1 was answered. [Jun 6 08:20:40] -- Executing [8369@default:1] Playback("Local/4280625217989355645263@default-00000000;1", "sip-silence") in new stack [Jun 6 08:20:40] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:20:40] -- <Local/4280625217989355645263@default-00000000;1> Playing 'sip-silence.gsm' (language 'en') [Jun 6 08:20:40] -- Executing [8369@default:2] AGI("Local/4280625217989355645263@default-00000000;1", "agi://127.0.0.1:4577/call_log") in new stack [Jun 6 08:20:40] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=93073185)) [Jun 6 08:20:40] -- <Local/4280625217989355645263@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jun 6 08:20:40] -- Executing [8369@default:3] AMD("Local/4280625217989355645263@default-00000000;1", "2000,2000,1000,5000,120,50,4,256") in new stack [Jun 6 08:20:40] -- AMD: Local/4280625217989355645263@default-00000000;1 5164536886 (N/A) (Fmt: slin) [Jun 6 08:20:40] -- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256] maximumWordLength [5000] [Jun 6 08:20:41] -- AMD: Channel [Local/4280625217989355645263@default-00000000;1]. Changed state to STATE_IN_SILENCE [Jun 6 08:20:42] -- AMD: Channel [Local/4280625217989355645263@default-00000000;1]. ANSWERING MACHINE: silenceDuration:2000 initialSilence:2000 [Jun 6 08:20:42] -- Executing [8369@default:4] AGI("Local/4280625217989355645263@default-00000000;1", "VD_amd.agi,8369") in new stack [Jun 6 08:20:43] -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_amd.agi [Jun 6 08:20:44] -- <Local/4280625217989355645263@default-00000000;1>AGI Script VD_amd.agi completed, returning 0 [Jun 6 08:20:44] -- Executing [8369@default:5] AGI("Local/4280625217989355645263@default-00000000;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jun 6 08:20:44] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jun 6 08:20:45] -- <Local/4280625217989355645263@default-00000000;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jun 6 08:20:45] -- Executing [8369@default:6] AGI("Local/4280625217989355645263@default-00000000;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jun 6 08:20:45] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jun 6 08:20:46] -- <Local/4280625217989355645263@default-00000000;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jun 6 08:20:46] -- Executing [8369@default:7] Hangup("Local/4280625217989355645263@default-00000000;1", "") in new stack [Jun 6 08:20:46] == Spawn extension (default, 8369, 7) exited non-zero on 'Local/4280625217989355645263@default-00000000;1' [Jun 6 08:20:46] -- Executing [h@default:1] AGI("Local/4280625217989355645263@default-00000000;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jun 6 08:20:47] -- <Local/4280625217989355645263@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jun 6 08:20:47] -- Executing [h@default:1] AGI("Local/4280625217989355645263@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----7") in new stack [Jun 6 08:20:48] -- <Local/4280625217989355645263@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----7 completed, returning 0 [Jun 6 08:20:48] == Spawn extension (default, 4280625217989355645263, 2) exited non-zero on 'Local/4280625217989355645263@default-00000000;2' [Jun 6 08:21:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:21:01] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:21:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:21:02] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:21:06] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:21:06] == Manager 'sendcron' logged off from 127.0.0.1 go*CLI>

CLI OUTPUT AUTO DIAL WITH SIP DEBUG ON 999

Code: Select all: go*CLI> [Jun 6 08:26:18] Really destroying SIP dialog '1711d1fa7bcdf0d11fd066290fde1744@10.10.10.253:5060' Method: OPTIONS [Jun 6 08:26:18] Really destroying SIP dialog '2ad159a57efb5a8f12a198bf56fd126e@127.0.0.1' Method: REGISTER [Jun 6 08:26:18] Really destroying SIP dialog '294107f46c8ed99703c87a151c973196@10.10.10.253:5060' Method: NOTIFY [Jun 6 08:26:46] <--- SIP read from UDP:10.10.10.253:5060 ---> OPTIONS sip:999@10.10.10.252:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.253:5060;branch=z9hG4bK59e4a8c8 Max-Forwards: 70 From: "Unknown" <sip:Unknown@10.10.10.253>;tag=as75c24cc9 To: <sip:999@10.10.10.252:5060> Contact: <sip:Unknown@10.10.10.253:5060> Call-ID: 0204611b7972707530b89789129bd978@10.10.10.253:5060 CSeq: 102 OPTIONS User-Agent: FPBX-2.11.0(11.20.0) Date: Tue, 05 Jun 2018 23:28:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jun 6 08:26:46] --- (13 headers 0 lines) --- [Jun 6 08:26:46] Looking for 999 in trunkinbound (domain 10.10.10.252) [Jun 6 08:26:46] <--- Transmitting (NAT) to 10.10.10.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.253:5060;branch=z9hG4bK59e4a8c8;received=10.10.10.253;rport=5060 From: "Unknown" <sip:Unknown@10.10.10.253>;tag=as75c24cc9 To: <sip:999@10.10.10.252:5060>;tag=as197d2897 Call-ID: 0204611b7972707530b89789129bd978@10.10.10.253:5060 CSeq: 102 OPTIONS Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:10.10.10.252:5060> Accept: application/sdp Content-Length: 0 <------------> [Jun 6 08:26:46] Scheduling destruction of SIP dialog '0204611b7972707530b89789129bd978@10.10.10.253:5060' in 32000 ms (Method: OPTIONS) [Jun 6 08:26:46] Reliably Transmitting (NAT) to 10.10.10.253:5060: OPTIONS sip:10.10.10.253 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK7ae29b00;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.10.10.252>;tag=as669b5559 To: <sip:10.10.10.253> Contact: <sip:asterisk@10.10.10.252:5060> Call-ID: 738962f84d01d6ea7421df7810e18fdf@10.10.10.252:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Date: Wed, 06 Jun 2018 03:56:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jun 6 08:26:46] <--- SIP read from UDP:10.10.10.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK7ae29b00;received=10.10.10.252;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.252>;tag=as669b5559 To: <sip:10.10.10.253>;tag=as41fb612b Call-ID: 738962f84d01d6ea7421df7810e18fdf@10.10.10.252:5060 CSeq: 102 OPTIONS Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:10.10.10.253:5060> Accept: application/sdp Content-Length: 0 <-------------> [Jun 6 08:26:46] --- (12 headers 0 lines) --- [Jun 6 08:26:46] Really destroying SIP dialog '738962f84d01d6ea7421df7810e18fdf@10.10.10.252:5060' Method: OPTIONS [Jun 6 08:27:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:27:01] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:27:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:27:02] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:27:02] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:27:02] -- Executing [4280625217989355645263@default:1] AGI("Local/4280625217989355645263@default-00000002;2", "agi://127.0.0.1:4577/call_log") in new stack [Jun 6 08:27:02] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=93073185)) [Jun 6 08:27:02] -- <Local/4280625217989355645263@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jun 6 08:27:02] -- Executing [4280625217989355645263@default:2] Dial("Local/4280625217989355645263@default-00000002;2", "SIP/989355645263@999,,tTo") in new stack [Jun 6 08:27:02] == Using SIP RTP CoS mark 5 [Jun 6 08:27:02] Audio is at 19486 [Jun 6 08:27:02] Adding codec 0x2 (gsm) to SDP [Jun 6 08:27:02] Adding codec 0x4 (ulaw) to SDP [Jun 6 08:27:02] Adding codec 0x8 (alaw) to SDP [Jun 6 08:27:02] Adding non-codec 0x1 (telephone-event) to SDP [Jun 6 08:27:02] Reliably Transmitting (NAT) to 10.10.10.253:5060: INVITE sip:989355645263@10.10.10.253 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK5fe74844;rport Max-Forwards: 70 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253> Contact: <sip:5164536886@10.10.10.252:5060> Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Date: Wed, 06 Jun 2018 03:57:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "V6060827020000000016" <sip:5164536886@10.10.10.252>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 326 v=0 o=root 1991928378 1991928378 IN IP4 10.10.10.252 s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com c=IN IP4 10.10.10.252 t=0 0 m=audio 19486 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 6 08:27:02] -- Called SIP/989355645263@999 [Jun 6 08:27:02] <--- SIP read from UDP:10.10.10.253:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK5fe74844;received=10.10.10.252;rport=5060 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as104648f3 Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 102 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1dd81846" Content-Length: 0 <-------------> [Jun 6 08:27:02] --- (11 headers 0 lines) --- [Jun 6 08:27:02] Transmitting (NAT) to 10.10.10.253:5060: ACK sip:989355645263@10.10.10.253 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK5fe74844;rport Max-Forwards: 70 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as104648f3 Contact: <sip:5164536886@10.10.10.252:5060> Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Content-Length: 0 --- [Jun 6 08:27:02] Audio is at 19486 [Jun 6 08:27:02] Adding codec 0x2 (gsm) to SDP [Jun 6 08:27:02] Adding codec 0x4 (ulaw) to SDP [Jun 6 08:27:02] Adding codec 0x8 (alaw) to SDP [Jun 6 08:27:02] Adding non-codec 0x1 (telephone-event) to SDP [Jun 6 08:27:02] Reliably Transmitting (NAT) to 10.10.10.253:5060: INVITE sip:989355645263@10.10.10.253 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK79980ffe;rport Max-Forwards: 70 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253> Contact: <sip:5164536886@10.10.10.252:5060> Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Authorization: Digest username="999", realm="asterisk", algorithm=MD5, uri="sip:989355645263@10.10.10.253", nonce="1dd81846", response="2baf1291060b733a731656f9da62e7e9" Date: Wed, 06 Jun 2018 03:57:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "V6060827020000000016" <sip:5164536886@10.10.10.252>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 326 v=0 o=root 1991928378 1991928379 IN IP4 10.10.10.252 s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com c=IN IP4 10.10.10.252 t=0 0 m=audio 19486 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jun 6 08:27:02] <--- SIP read from UDP:10.10.10.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK79980ffe;received=10.10.10.252;rport=5060 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253> Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 103 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:989355645263@10.10.10.253:5060> Content-Length: 0 <-------------> [Jun 6 08:27:02] --- (12 headers 0 lines) --- [Jun 6 08:27:02] <--- SIP read from UDP:10.10.10.253:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK79980ffe;received=10.10.10.252;rport=5060 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as4b555288 Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 103 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:989355645263@10.10.10.253:5060> Content-Type: application/sdp Require: timer Content-Length: 283 v=0 o=root 1473031772 1473031772 IN IP4 10.10.10.253 s=Asterisk PBX 11.20.0 c=IN IP4 10.10.10.253 t=0 0 m=audio 18774 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 6 08:27:02] --- (14 headers 13 lines) --- Jun 6 08:27:02] list_route: hop: <sip:989355645263@10.10.10.253:5060> [Jun 6 08:27:02] Found RTP audio format 8 [Jun 6 08:27:02] Found RTP audio format 3 [Jun 6 08:27:02] Found RTP audio format 0 [Jun 6 08:27:02] Found RTP audio format 101 [Jun 6 08:27:02] Found audio description format PCMA for ID 8 [Jun 6 08:27:02] Found audio description format GSM for ID 3 [Jun 6 08:27:02] Found audio description format PCMU for ID 0 [Jun 6 08:27:02] Found audio description format telephone-event for ID 101 [Jun 6 08:27:02] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jun 6 08:27:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jun 6 08:27:02] Peer audio RTP is at port 10.10.10.253:18774 [Jun 6 08:27:02] -- SIP/999-00000003 is making progress passing it to Local/4280625217989355645263@default-00000002;2 [Jun 6 08:27:06] == Manager 'sendcron' logged on from 127.0.0.1 [Jun 6 08:27:06] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:27:13] <--- SIP read from UDP:10.10.10.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK79980ffe;received=10.10.10.252;rport=5060 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as4b555288 Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 103 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:989355645263@10.10.10.253:5060> Content-Type: application/sdp Require: timer Content-Length: 283 v=0 o=root 1473031772 1473031772 IN IP4 10.10.10.253 s=Asterisk PBX 11.20.0 c=IN IP4 10.10.10.253 t=0 0 m=audio 18774 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jun 6 08:27:13] --- (14 headers 13 lines) --- [Jun 6 08:27:13] list_route: hop: <sip:989355645263@10.10.10.253:5060> [Jun 6 08:27:13] set_destination: Parsing <sip:989355645263@10.10.10.253:5060> for address/port to send to [Jun 6 08:27:13] set_destination: set destination to 10.10.10.253:5060 [Jun 6 08:27:13] Transmitting (NAT) to 10.10.10.253:5060: ACK sip:989355645263@10.10.10.253:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK239536a2;rport Max-Forwards: 70 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as4b555288 Contact: <sip:5164536886@10.10.10.252:5060> Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Content-Length: 0 --- [Jun 6 08:27:13] -- SIP/999-00000003 answered Local/4280625217989355645263@default-00000002;2 [Jun 6 08:27:13] > Channel Local/4280625217989355645263@default-00000002;1 was answered. [Jun 6 08:27:13] -- Executing [8369@default:1] Playback("Local/4280625217989355645263@default-00000002;1", "sip-silence") in new stack [Jun 6 08:27:13] -- <Local/4280625217989355645263@default-00000002;1> Playing 'sip-silence.gsm' (language 'en') [Jun 6 08:27:13] == Manager 'sendcron' logged off from 127.0.0.1 [Jun 6 08:27:13] -- Executing [8369@default:2] AGI("Local/4280625217989355645263@default-00000002;1", "agi://127.0.0.1:4577/call_log") in new stack [Jun 6 08:27:13] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=93073185)) [Jun 6 08:27:13] -- <Local/4280625217989355645263@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jun 6 08:27:13] -- Executing [8369@default:3] AMD("Local/4280625217989355645263@default-00000002;1", "2000,2000,1000,5000,120,50,4,256") in new stack [Jun 6 08:27:13] -- AMD: Local/4280625217989355645263@default-00000002;1 5164536886 (N/A) (Fmt: slin) [Jun 6 08:27:13] -- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256] maximumWordLength [5000] [Jun 6 08:27:13] -- AMD: Channel [Local/4280625217989355645263@default-00000002;1]. Changed state to STATE_IN_SILENCE [Jun 6 08:27:15] -- AMD: Channel [Local/4280625217989355645263@default-00000002;1]. ANSWERING MACHINE: silenceDuration:2000 initialSilence:2000 [Jun 6 08:27:15] -- Executing [8369@default:4] AGI("Local/4280625217989355645263@default-00000002;1", "VD_amd.agi,8369") in new stack [Jun 6 08:27:15] -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_amd.agi [Jun 6 08:27:16] -- <Local/4280625217989355645263@default-00000002;1>AGI Script VD_amd.agi completed, returning 0 [Jun 6 08:27:16] -- Executing [8369@default:5] AGI("Local/4280625217989355645263@default-00000002;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jun 6 08:27:16] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jun 6 08:27:17] -- <Local/4280625217989355645263@default-00000002;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jun 6 08:27:17] -- Executing [8369@default:6] AGI("Local/4280625217989355645263@default-00000002;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jun 6 08:27:17] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jun 6 08:27:18] -- <Local/4280625217989355645263@default-00000002;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jun 6 08:27:18] -- Executing [8369@default:7] Hangup("Local/4280625217989355645263@default-00000002;1", "") in new stack [Jun 6 08:27:18] == Spawn extension (default, 8369, 7) exited non-zero on 'Local/4280625217989355645263@default-00000002;1' [Jun 6 08:27:18] -- Executing [h@default:1] AGI("Local/4280625217989355645263@default-00000002;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jun 6 08:27:18] Really destroying SIP dialog '0204611b7972707530b89789129bd978@10.10.10.253:5060' Method: OPTIONS [Jun 6 08:27:19] -- <Local/4280625217989355645263@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jun 6 08:27:19] -- Executing [h@default:1] AGI("Local/4280625217989355645263@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----6") in new stack [Jun 6 08:27:20] -- <Local/4280625217989355645263@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----6 completed, returning 0 [Jun 6 08:27:20] Scheduling destruction of SIP dialog '00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060' in 6400 ms (Method: INVITE) [Jun 6 08:27:20] set_destination: Parsing <sip:989355645263@10.10.10.253:5060> for address/port to send to [Jun 6 08:27:20] set_destination: set destination to 10.10.10.253:5060 [Jun 6 08:27:20] Reliably Transmitting (NAT) to 10.10.10.253:5060: BYE sip:989355645263@10.10.10.253:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK59b4a91f;rport Max-Forwards: 70 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as4b555288 Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com Authorization: Digest username="999", realm="asterisk", algorithm=MD5, uri="sip:989355645263@10.10.10.253:5060", nonce="1dd81846", response="4d4c8deeba86b17071086d1b341584cb" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jun 6 08:27:20] == Spawn extension (default, 4280625217989355645263, 2) exited non-zero on 'Local/4280625217989355645263@default-00000002;2' [Jun 6 08:27:20] <--- SIP read from UDP:10.10.10.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.252:5060;branch=z9hG4bK59b4a91f;received=10.10.10.252;rport=5060 From: "V6060827020000000016" <sip:5164536886@10.10.10.252>;tag=as7d300929 To: <sip:989355645263@10.10.10.253>;tag=as4b555288 Call-ID: 00ad60ca14d70368007d49f22c87d0e9@10.10.10.252:5060 CSeq: 104 BYE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------->

CARRIER CONFIGURATION ( SIP TRUNK )

Code: Select all: register => 999:exten999@10.10.10.253:5060/999 . . [999] disallow=all allow=gsm allow=ulaw type=friend dtmfmode=rfc2833 context=trunkinbound qualify=yes nat=yes host=10.10.10.253 username=999 secret=exten999 allow=alaw allow=g729 . . exten => _4280625217.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _4280625217.,2,Dial(SIP/${EXTEN:10}@999,,tTo) exten => _4280625217.,3,Hangup

Hardware :
All Test in VMware / Esxi / Bare Metal
With 4G RAM + 4 Core 3.1 Intel CPU ( Corei3 / Xeon )

Software :
goautodial-64bit-ce-3.3-final.iso ( Fresh Install )
Asterisk 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
ViciDial VERSION: 2.9-441a BUILD: 140612-1628

Carrier :
SIP Extention on Elastix
Elastix-4.0.74-Stable-x86_64-bin-10Feb2016.iso
Asterisk 11.20.0

thanks for attention

I'm waiting for your reply.

by **blackbird2306** » Wed Jun 06, 2018 4:57 am

How did you turn off AMD? All these calls you provided go through AMD extension and are stated as "answering machine". Try to change in campaign settings the Routing Extension to "8368".
Your vicidial version is pretty old (2014). My advice: Don't use goautodial (even if design seems to be better). Right way is Vicibox installation from here: http://vicibox.com/server/index.html

by **noobdialer** » Wed Jun 06, 2018 5:15 am

First, thank you for the reply.

How did you turn off AMD?

I try all type of this but not work and call hangup after trying with autodial

Code: Select all: Routing Extension - This field allows for a custom outbound routing extension. This allows you to use different call handling methods depending upon how you want to route calls through your outbound campaign. Formerly called Campaign VDAD extension. - 8364 - same as 8368 - 8365 - Will send the call only to an agent on the same server as the call is placed on - 8366 - Used for press-1, broadcast and survey campaigns - 8367 - Will try to first send the call to an agent on the local server, then it will look on other servers - 8368 - DEFAULT - Will send the call to the next available agent no matter what server they are on - 8369 - Used for Answering Machine Detection after that, same behavior as 8368 - 8373 - Used for Answering Machine Detection after that same behavior as 8366 - 8374 - Used for press-1, broadcast and survey campaigns with Cepstral Text-to-speech - 8375 - Used for Answering Machine Detection then press-1, broadcast and survey campaigns with Cepstral Text-to-speech

My advice: Don't use goautodial (even if design seems to be better). Right way is Vicibox installation from here: http://vicibox.com/server/index.html

dear, I install vicibox too but too much problem with that, for example when fresh install vicibox express mode and after the first login, I cant anything to do, because say you don't have permission and I don't know why fresh install doesn't have permission user 6666 ...

ViciBox_v8.x86_64-8.0.1.iso

Code: Select all: vicibox8:/usr/src/astguiclient/trunk/extras # asterisk -rtvvv [Jun 6 14:32:41] Asterisk 11.25.3-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. [Jun 6 14:32:41] Created by Mark Spencer <markster@digium.com> [Jun 6 14:32:41] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. [Jun 6 14:32:41] This is free software, with components licensed under the GNU General Public [Jun 6 14:32:41] License version 2 and other licenses; you are welcome to redistribute it under [Jun 6 14:32:41] certain conditions. Type 'core show license' for details. [Jun 6 14:32:41] ========================================================================= [Jun 6 14:32:41] Connected to Asterisk 11.25.3-vici currently running on vicibox8 (pid = 1190)

ADMIN USER

Code: Select all: 6666 Admin 9 ADMIN Y

EXAMPLE ERROR

Code: Select all: You are not allowed to view reports: |6666|GOOD| OR You do not have permission to view this page

and I many searches on google and forum but all post for many years ago and all solution are expired and not work like this post

http://www.vicidial.org/VICIDIALforum/viewtopic.php?f=4&t=34824

by **blackbird2306** » Wed Jun 06, 2018 5:39 am

Users --> Show Users --> select user "6666" --> make sure that in "ADMIN INTERFACE OPTIONS" section all settings are "1" (e.g. View Reports --> "1")

Vicibox installation is gold standard!

by **striker** » Wed Jun 06, 2018 6:27 am

Try the below settings in Ur carrier settings

Change
nat=no
insecure=invite
fromuser=999

Then SSH to Ur server and open sip.conf
vi /etc/asterisk/sip.conf
Search for line localnet=

Check whether 10.10.0.0/255.255.0.0

Added or not if not add it

by **noobdialer** » Wed Jun 06, 2018 3:01 pm

blackbird2306 wrote:Users --> Show Users --> select user "6666" --> make sure that in "ADMIN INTERFACE OPTIONS" section all settings are "1" (e.g. View Reports --> "1")

Vicibox installation is gold standard!

this solution solved the problem with the admin interface, but after I can add camp and other options and start a test, just like goautodial

when I make a call manual, all things ok,

but when I try to autodial and I answer my phone call hangup after 4 seconds.

and all cli log and other output just like my first post.

by **noobdialer** » Wed Jun 06, 2018 3:03 pm

striker wrote:Try the below settings in Ur carrier settings

Change
nat=no
insecure=invite
fromuser=999

Then SSH to Ur server and open sip.conf
vi /etc/asterisk/sip.conf
Search for line localnet=

Check whether 10.10.0.0/255.255.0.0

Added or not if not add it

thanks for the reply, I try this option but all action like before and the problem with autodial still there.

by **noobdialer** » Fri Jun 08, 2018 11:33 pm

Thanks to all, I found the problem

when I use DMZ between GoAutoDialer and Carrier problem solved.

I hope this solution works for others.

vicidial.org

Auto Dial Hangup Call After 10 Sec

Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

Re: Auto Dial Hangup Call After 10 Sec

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