vicidial.org

Posted: **Wed Oct 03, 2018 2:55 pm**

Vicibox 8.1.2 - Vicidial VERSION: 2.14-692a - BUILD: 180927-0018 - vicibox-express - asterisk 13.21.1-vici - No add-on hardware - No extra software

Inbound campaign with Blended enabled and selected on login.

Dial Method: Ratio
Allow Inbound and Blended: Yes

On hook phone rings, but asterisk sends a BYE almost right away after picking

agi out -- Getting a lot of this

Code: Select all: 2018-10-03 15:44:58|15:45:42|agi-VDAD_ALL_inbound.agi|NNNNN No available balance agent found 2018-10-03 15:44:58|15:45:43|agi-VDAD_ALL_inbound.agi|0|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and campaign_id = 'Inbound' and call_time < "2018-10-03 15:44:58" and lead_id != '23' and queue_priority >= '0' and agent_only='';| 2018-10-03 15:44:58|15:45:43|agi-VDAD_ALL_inbound.agi|0|SELECT count(*) FROM vicidial_inbound_callback_queue where icbq_status IN('LIVE','SENDING') and group_id='Inbound' and call_date < "2018-10-03 15:44:58" and lead_id != '23' and queue_priority >= '0';| 2018-10-03 15:44:58|15:45:43|agi-VDAD_ALL_inbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '23' and agent_only='' and ( (queue_priority > '0') or (queue_priority = '0' and call_time < "2018-10-03 15:44:58") );| 2018-10-03 15:44:58|15:45:43|agi-VDAD_ALL_inbound.agi|0|SELECT count(*) FROM vicidial_inbound_callback_queue where icbq_status IN('LIVE','SENDING') and lead_id != '23' and ( (queue_priority > '0') or (queue_priority = '0' and call_date < "2018-10-03 15:44:58") );| 2018-10-03 15:44:58|15:45:43|agi-VDAD_ALL_inbound.agi|RING-AGENT TIMER: 8|15|15 SIP/voip140-00000044|Y0031544580000000023 2018-10-03 15:44:58|15:45:43|agi-VDAD_ALL_inbound.agi|-- VDAD get agent: |360|45|34|60|0|0|0|0|update of vla table: Inbound|

core/ SIP debug

Code: Select all: [Oct 3 15:50:06] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 3 15:50:06] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 3 15:50:13] == Using SIP RTP CoS mark 5 [Oct 3 15:50:13] > 0x7ffab801eb20 -- Strict RTP learning after remote address set to: 60.60.60.151:16310 [Oct 3 15:50:13] -- Executing [6136561806@trunkinbound:1] AGI("SIP/voip140-00000049", "agi-DID_route.agi") in new stack [Oct 3 15:50:13] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi [Oct 3 15:50:13] -- <SIP/voip140-00000049>AGI Script agi-DID_route.agi completed, returning 0 [Oct 3 15:50:13] -- Executing [99909*2***DID@default:1] Answer("SIP/voip140-00000049", "") in new stack [Oct 3 15:50:14] -- Executing [99909*2***DID@default:2] AGI("SIP/voip140-00000049", "agi-VDAD_ALL_inbound.agi") in new stack [Oct 3 15:50:14] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi [Oct 3 15:50:14] > 0x7ffab801eb20 -- Strict RTP switching to RTP target address 60.60.60.151:16310 as source [Oct 3 15:50:14] -- <SIP/voip140-00000049> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en') [Oct 3 15:50:14] -- <SIP/voip140-00000049> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en') [Oct 3 15:50:14] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 3 15:50:14] -- Called 060*060*060*201*6666@default [Oct 3 15:50:14] -- Executing [060*060*060*201*6666@default:1] Goto("Local/060*060*060*201*6666@default-0000003e;2", "default,6666,1") in new stack [Oct 3 15:50:14] -- Goto (default,6666,1) [Oct 3 15:50:14] -- Executing [6666@default:1] Dial("Local/060*060*060*201*6666@default-0000003e;2", "SIP/6666,60,") in new stack [Oct 3 15:50:14] == Using SIP RTP CoS mark 5 [Oct 3 15:50:14] Audio is at 19220 [Oct 3 15:50:14] Adding codec ulaw to SDP [Oct 3 15:50:14] Adding codec gsm to SDP [Oct 3 15:50:14] Adding non-codec 0x1 (telephone-event) to SDP [Oct 3 15:50:14] Reliably Transmitting (NAT) to 60.80.90.218:11913: [Oct 3 15:50:14] INVITE sip:6666@192.168.1.105:5060;transport=udp SIP/2.0 [Oct 3 15:50:14] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3ce46b3a;rport [Oct 3 15:50:14] Max-Forwards: 70 [Oct 3 15:50:14] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:14] To: <sip:6666@192.168.1.105:5060;transport=udp> [Oct 3 15:50:14] Contact: <sip:asterisk@60.60.60.201:5060> [Oct 3 15:50:14] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:14] CSeq: 102 INVITE [Oct 3 15:50:14] User-Agent: Asterisk PBX 13.21.1-vici [Oct 3 15:50:14] Date: Wed, 03 Oct 2018 19:50:14 GMT [Oct 3 15:50:14] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 3 15:50:14] Supported: replaces, timer [Oct 3 15:50:14] Content-Type: application/sdp [Oct 3 15:50:14] Content-Length: 282 [Oct 3 15:50:14] [Oct 3 15:50:14] v=0 [Oct 3 15:50:14] o=root 1462870511 1462870511 IN IP4 60.60.60.201 [Oct 3 15:50:14] s=Asterisk PBX 13.21.1-vici [Oct 3 15:50:14] c=IN IP4 60.60.60.201 [Oct 3 15:50:14] t=0 0 [Oct 3 15:50:14] m=audio 19220 RTP/AVP 0 3 101 [Oct 3 15:50:14] a=rtpmap:0 PCMU/8000 [Oct 3 15:50:14] a=rtpmap:3 GSM/8000 [Oct 3 15:50:14] a=rtpmap:101 telephone-event/8000 [Oct 3 15:50:14] a=fmtp:101 0-16 [Oct 3 15:50:14] a=ptime:20 [Oct 3 15:50:14] a=maxptime:060 [Oct 3 15:50:14] a=sendrecv [Oct 3 15:50:14] [Oct 3 15:50:14] --- [Oct 3 15:50:14] -- Called SIP/6666 [Oct 3 15:50:14] Retransmitting #1 (NAT) to 60.80.90.218:11913: [Oct 3 15:50:14] INVITE sip:6666@192.168.1.105:5060;transport=udp SIP/2.0 [Oct 3 15:50:14] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3ce46b3a;rport [Oct 3 15:50:14] Max-Forwards: 70 [Oct 3 15:50:14] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:14] To: <sip:6666@192.168.1.105:5060;transport=udp> [Oct 3 15:50:14] Contact: <sip:asterisk@60.60.60.201:5060> [Oct 3 15:50:14] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:14] CSeq: 102 INVITE [Oct 3 15:50:14] User-Agent: Asterisk PBX 13.21.1-vici [Oct 3 15:50:14] Date: Wed, 03 Oct 2018 19:50:14 GMT [Oct 3 15:50:14] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Oct 3 15:50:14] Supported: replaces, timer [Oct 3 15:50:14] Content-Type: application/sdp [Oct 3 15:50:14] Content-Length: 282 [Oct 3 15:50:14] [Oct 3 15:50:14] v=0 [Oct 3 15:50:14] o=root 1462870511 1462870511 IN IP4 60.60.60.201 [Oct 3 15:50:14] s=Asterisk PBX 13.21.1-vici [Oct 3 15:50:14] c=IN IP4 60.60.60.201 [Oct 3 15:50:14] t=0 0 [Oct 3 15:50:14] m=audio 19220 RTP/AVP 0 3 101 [Oct 3 15:50:14] a=rtpmap:0 PCMU/8000 [Oct 3 15:50:14] a=rtpmap:3 GSM/8000 [Oct 3 15:50:14] a=rtpmap:101 telephone-event/8000 [Oct 3 15:50:14] a=fmtp:101 0-16 [Oct 3 15:50:14] a=ptime:20 [Oct 3 15:50:14] a=maxptime:060 [Oct 3 15:50:14] a=sendrecv [Oct 3 15:50:14] [Oct 3 15:50:14] --- [Oct 3 15:50:14] [Oct 3 15:50:14] <--- SIP read from UDP:60.80.90.218:11913 ---> [Oct 3 15:50:14] SIP/2.0 180 Ringing [Oct 3 15:50:14] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3ce46b3a;rport=5060;received=60.60.60.201 [Oct 3 15:50:14] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:14] To: <sip:6666@192.168.1.105:5060;transport=udp>;tag=3119923050 [Oct 3 15:50:14] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:14] CSeq: 102 INVITE [Oct 3 15:50:14] Accept-Language: en [Oct 3 15:50:14] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH [Oct 3 15:50:14] Allow-Events: talk, hold, conference, LocalModeStatus [Oct 3 15:50:14] Contact: "6666" <sip:6666@192.168.1.105:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D47FCB4>" [Oct 3 15:50:14] Server: Aastra 6869i/4.1.0.128 [Oct 3 15:50:14] Supported: path [Oct 3 15:50:14] Content-Length: 0 [Oct 3 15:50:14] [Oct 3 15:50:14] <-------------> [Oct 3 15:50:14] --- (13 headers 0 lines) --- [Oct 3 15:50:14] sip_route_dump: route/path hop: <sip:6666@192.168.1.105:5060;transport=udp> [Oct 3 15:50:14] -- SIP/6666-0000004a is ringing [Oct 3 15:50:14] -- Local/060*060*060*201*6666@default-0000003e;1 is ringing [Oct 3 15:50:14] [Oct 3 15:50:14] <--- SIP read from UDP:60.80.90.218:11913 ---> [Oct 3 15:50:14] SIP/2.0 180 Ringing [Oct 3 15:50:14] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3ce46b3a;rport=5060;received=60.60.60.201 [Oct 3 15:50:14] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:14] To: <sip:6666@192.168.1.105:5060;transport=udp>;tag=3119923050 [Oct 3 15:50:14] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:14] CSeq: 102 INVITE [Oct 3 15:50:14] Accept-Language: en [Oct 3 15:50:14] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH [Oct 3 15:50:14] Allow-Events: talk, hold, conference, LocalModeStatus [Oct 3 15:50:14] Contact: "6666" <sip:6666@192.168.1.105:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D47FCB4>" [Oct 3 15:50:14] Server: Aastra 6869i/4.1.0.128 [Oct 3 15:50:14] Supported: path [Oct 3 15:50:14] Content-Length: 0 [Oct 3 15:50:14] [Oct 3 15:50:14] <-------------> [Oct 3 15:50:14] --- (13 headers 0 lines) --- [Oct 3 15:50:14] sip_route_dump: route/path hop: <sip:6666@192.168.1.105:5060;transport=udp> [Oct 3 15:50:14] -- SIP/6666-0000004a is ringing [Oct 3 15:50:16] -- Started music on hold, class 'default', on channel 'SIP/voip140-00000049' [Oct 3 15:50:17] [Oct 3 15:50:17] <--- SIP read from UDP:60.80.90.218:11913 ---> [Oct 3 15:50:17] SIP/2.0 200 OK [Oct 3 15:50:17] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3ce46b3a;rport=5060;received=60.60.60.201 [Oct 3 15:50:17] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:17] To: <sip:6666@192.168.1.105:5060;transport=udp>;tag=3119923050 [Oct 3 15:50:17] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:17] CSeq: 102 INVITE [Oct 3 15:50:17] Accept-Language: en [Oct 3 15:50:17] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH [Oct 3 15:50:17] Allow-Events: talk, hold, conference, LocalModeStatus [Oct 3 15:50:17] Contact: "6666" <sip:6666@192.168.1.105:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D47FCB4>" [Oct 3 15:50:17] Server: Aastra 6869i/4.1.0.128 [Oct 3 15:50:17] Supported: path, replaces [Oct 3 15:50:17] Content-Type: application/sdp [Oct 3 15:50:17] Content-Length: 235 [Oct 3 15:50:17] [Oct 3 15:50:17] v=0 [Oct 3 15:50:17] o=MxSIP 0 1 IN IP4 192.168.1.105 [Oct 3 15:50:17] s=SIP Call [Oct 3 15:50:17] c=IN IP4 192.168.1.105 [Oct 3 15:50:17] t=0 0 [Oct 3 15:50:17] m=audio 3000 RTP/AVP 0 101 [Oct 3 15:50:17] a=rtpmap:0 PCMU/8000 [Oct 3 15:50:17] a=rtpmap:101 telephone-event/8000 [Oct 3 15:50:17] a=silenceSupp:off - - - - [Oct 3 15:50:17] a=fmtp:101 0-15 [Oct 3 15:50:17] a=ptime:20 [Oct 3 15:50:17] a=sendrecv [Oct 3 15:50:17] <-------------> [Oct 3 15:50:17] --- (14 headers 12 lines) --- [Oct 3 15:50:17] Found RTP audio format 0 [Oct 3 15:50:17] Found RTP audio format 101 [Oct 3 15:50:17] Found audio description format PCMU for ID 0 [Oct 3 15:50:17] Found audio description format telephone-event for ID 101 [Oct 3 15:50:17] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Oct 3 15:50:17] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Oct 3 15:50:17] > 0x7ffab8011af0 -- Strict RTP learning after remote address set to: 192.168.1.105:3000 [Oct 3 15:50:17] Peer audio RTP is at port 192.168.1.105:3000 [Oct 3 15:50:17] sip_route_dump: route/path hop: <sip:6666@192.168.1.105:5060;transport=udp> [Oct 3 15:50:17] Transmitting (NAT) to 60.80.90.218:11913: [Oct 3 15:50:17] ACK sip:6666@192.168.1.105:5060;transport=udp SIP/2.0 [Oct 3 15:50:17] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK7106eb54;rport [Oct 3 15:50:17] Max-Forwards: 70 [Oct 3 15:50:17] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:17] To: <sip:6666@192.168.1.105:5060;transport=udp>;tag=3119923050 [Oct 3 15:50:17] Contact: <sip:asterisk@60.60.60.201:5060> [Oct 3 15:50:17] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:17] CSeq: 102 ACK [Oct 3 15:50:17] User-Agent: Asterisk PBX 13.21.1-vici [Oct 3 15:50:17] Content-Length: 0 [Oct 3 15:50:17] [Oct 3 15:50:17] [Oct 3 15:50:17] --- [Oct 3 15:50:17] -- SIP/6666-0000004a answered Local/060*060*060*201*6666@default-0000003e;2 [Oct 3 15:50:17] -- Local/060*060*060*201*6666@default-0000003e;1 answered [Oct 3 15:50:17] -- Executing [138331*27*Y0031550140000000024*6666*6666@default:1] AGI("Local/060*060*060*201*6666@default-0000003e;1", "agi-VDAD_local_optimize.agi,RINGAGENT00000000027") in new stack [Oct 3 15:50:17] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi [Oct 3 15:50:17] -- Channel SIP/6666-0000004a joined 'simple_bridge' basic-bridge <62771257-7bb9-4afd-aaa3-8ea19100b8c4> [Oct 3 15:50:17] -- Channel Local/060*060*060*201*6666@default-0000003e;2 joined 'simple_bridge' basic-bridge <62771257-7bb9-4afd-aaa3-8ea19100b8c4> [Oct 3 15:50:17] > 0x7ffab8011af0 -- Strict RTP qualifying stream type: audio [Oct 3 15:50:17] -- <Local/060*060*060*201*6666@default-0000003e;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0 [Oct 3 15:50:17] -- Executing [138331*27*Y0031550140000000024*6666*6666@default:2] Wait("Local/060*060*060*201*6666@default-0000003e;1", "2") in new stack [Oct 3 15:50:17] > 0x7ffab8011af0 -- Strict RTP switching source address to 60.80.90.218:22181 [Oct 3 15:50:18] == Manager 'sendcron' logged off from 127.0.0.1 [Oct 3 15:50:18] > 0x7ffab801eb20 -- Strict RTP learning complete - Locking on source address 60.60.60.151:16310 [Oct 3 15:50:19] -- Executing [138331*27*Y0031550140000000024*6666*6666@default:3] Hangup("Local/060*060*060*201*6666@default-0000003e;1", "") in new stack [Oct 3 15:50:19] == Spawn extension (default, 138331*27*Y0031550140000000024*6666*6666, 3) exited non-zero on 'Local/060*060*060*201*6666@default-0000003e;1' [Oct 3 15:50:19] WARNING[29692][C-00000091]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Oct 3 15:50:19] -- Executing [h@default:1] AGI("Local/060*060*060*201*6666@default-0000003e;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [Oct 3 15:50:19] -- <Local/060*060*060*201*6666@default-0000003e;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [Oct 3 15:50:19] -- Channel Local/060*060*060*201*6666@default-0000003e;2 left 'simple_bridge' basic-bridge <62771257-7bb9-4afd-aaa3-8ea19100b8c4> [Oct 3 15:50:19] == Spawn extension (default, 6666, 1) exited non-zero on 'Local/060*060*060*201*6666@default-0000003e;2' [Oct 3 15:50:19] -- Executing [h@default:1] AGI("Local/060*060*060*201*6666@default-0000003e;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----2-----SIP 200 OK)") in new stack [Oct 3 15:50:19] -- Channel SIP/6666-0000004a left 'simple_bridge' basic-bridge <62771257-7bb9-4afd-aaa3-8ea19100b8c4> [Oct 3 15:50:19] Scheduling destruction of SIP dialog '33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060' in 6400 ms (Method: INVITE) [Oct 3 15:50:19] Reliably Transmitting (NAT) to 60.80.90.218:11913: [Oct 3 15:50:19] BYE sip:6666@192.168.1.105:5060;transport=udp SIP/2.0 [Oct 3 15:50:19] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3bf69608;rport [Oct 3 15:50:19] Max-Forwards: 70 [Oct 3 15:50:19] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:19] To: <sip:6666@192.168.1.105:5060;transport=udp>;tag=3119923050 [Oct 3 15:50:19] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:19] CSeq: 103 BYE [Oct 3 15:50:19] User-Agent: Asterisk PBX 13.21.1-vici [Oct 3 15:50:19] X-Asterisk-HangupCause: Normal Clearing [Oct 3 15:50:19] X-Asterisk-HangupCauseCode: 16 [Oct 3 15:50:19] Content-Length: 0 [Oct 3 15:50:19] [Oct 3 15:50:19] [Oct 3 15:50:19] --- [Oct 3 15:50:19] -- Stopped music on hold on SIP/voip140-00000049 [Oct 3 15:50:19] -- <SIP/voip140-00000049> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en') [Oct 3 15:50:19] -- <Local/060*060*060*201*6666@default-0000003e;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----2-----SIP 200 OK) completed, returning 0 [Oct 3 15:50:19] -- <SIP/voip140-00000049> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en') [Oct 3 15:50:19] [Oct 3 15:50:19] <--- SIP read from UDP:60.80.90.218:11913 ---> [Oct 3 15:50:19] SIP/2.0 200 OK [Oct 3 15:50:19] Via: SIP/2.0/UDP 60.60.60.201:5060;branch=z9hG4bK3bf69608;rport=5060;received=60.60.60.201 [Oct 3 15:50:19] From: "RINGAGENT00000000027" <sip:asterisk@60.60.60.201>;tag=as1663132e [Oct 3 15:50:19] To: <sip:6666@192.168.1.105:5060;transport=udp>;tag=3119923050 [Oct 3 15:50:19] Call-ID: 33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060 [Oct 3 15:50:19] CSeq: 103 BYE [Oct 3 15:50:19] Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH [Oct 3 15:50:19] Allow-Events: talk, hold, conference, LocalModeStatus [Oct 3 15:50:19] Server: Aastra 6869i/4.1.0.128 [Oct 3 15:50:19] Supported: path [Oct 3 15:50:19] Content-Length: 0 [Oct 3 15:50:19] [Oct 3 15:50:19] <-------------> [Oct 3 15:50:19] --- (11 headers 0 lines) --- [Oct 3 15:50:19] Really destroying SIP dialog '33e84fac62dc5a200acf39005cf13922@60.60.60.201:5060' Method: INVITE [Oct 3 15:50:19] -- <SIP/voip140-00000049> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en') [Oct 3 15:50:19] -- <SIP/voip140-00000049> Playing 'generic_hold.gsm' (escape_digits=) (sample_offset 0) (language 'en') vici-7Dental*CLI> Disconnected from Asterisk server [Oct 3 15:50:22] Asterisk cleanly ending (0). [Oct 3 15:50:22] Executing last minute cleanups

If the same phone is set to have On Hook = No things seem to work just find, audio is good and agent screen works as it should

Posted: **Thu Oct 04, 2018 5:15 am**

I did fix a rare issue with Asterisk 13 for some users with On-Hook agent phones last night. Try upgrading to the most recent svn/trunk, reboot your dialers and see if that fixes your issues.

Posted: **Thu Oct 04, 2018 11:51 am**

Cool, thank you. I'll take a backup and try it out tonight as we're in production now with on-hook = no and it's working just fine for that part

Posted: **Mon Nov 05, 2018 11:04 am**

Yesterday I have installed ViciBox v.8.1.2 181002 and having same issue when extension is on Hook. Off Hook its working fine.

Error:-

Code: Select all: [Nov 5 18:35:48] > 0x26ab850 -- Strict RTP learning after remote address set to: 192.168.1.247:5062 [Nov 5 18:35:48] -- SIP/115-0000005b answered Local/192*168*001*250*115@default-00000031;2 [Nov 5 18:35:48] -- Local/192*168*001*250*115@default-00000031;1 answered [Nov 5 18:35:48] -- Channel SIP/115-0000005b joined 'simple_bridge' basic-bridge <78bb3d75-94bc-481c-8055-a7a10c9fd44b> [Nov 5 18:35:48] -- Executing [138331*42*Y1051833320000000021*haneen*115@default:1] AGI("Local/192*168*001*250*115@default-00000031;1", "agi-VDAD_local_optimize.agi,RINGAGENT00000000042") in new stack [Nov 5 18:35:48] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi [Nov 5 18:35:48] -- Channel Local/192*168*001*250*115@default-00000031;2 joined 'simple_bridge' basic-bridge <78bb3d75-94bc-481c-8055-a7a10c9fd44b> vicibox81*CLI> vicibox81*CLI> [Nov 5 18:35:48] > 0x26ab850 -- Strict RTP switching to RTP target address 192.168.1.247:5062 as source [Nov 5 18:35:48] -- <Local/192*168*001*250*115@default-00000031;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0 [Nov 5 18:35:48] -- Executing [138331*42*Y1051833320000000021*haneen*115@default:2] Wait("Local/192*168*001*250*115@default-00000031;1", "2") in new stack vicibox81*CLI> [Nov 5 18:35:49] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 5 18:35:50] -- Executing [138331*42*Y1051833320000000021*haneen*115@default:3] Hangup("Local/192*168*001*250*115@default-00000031;1", "") in new stack [Nov 5 18:35:50] == Spawn extension (default, 138331*42*Y1051833320000000021*haneen*115, 3) exited non-zero on 'Local/192*168*001*250*115@default-00000031;1' [Nov 5 18:35:50] WARNING[29314][C-000000a9]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Nov 5 18:35:50] -- Executing [h@default:1] AGI("Local/192*168*001*250*115@default-00000031;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack [Nov 5 18:35:50] -- <Local/192*168*001*250*115@default-00000031;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0 [Nov 5 18:35:50] -- Channel Local/192*168*001*250*115@default-00000031;2 left 'simple_bridge' basic-bridge <78bb3d75-94bc-481c-8055-a7a10c9fd44b> [Nov 5 18:35:50] == Spawn extension (default, 115, 1) exited non-zero on 'Local/192*168*001*250*115@default-00000031;2' [Nov 5 18:35:50] -- Executing [h@default:1] AGI("Local/192*168*001*250*115@default-00000031;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----2-----SIP 200 OK)") in new stack [Nov 5 18:35:50] -- Channel SIP/115-0000005b left 'simple_bridge' basic-bridge <78bb3d75-94bc-481c-8055-a7a10c9fd44b> [Nov 5 18:35:50] -- <Local/192*168*001*250*115@default-00000031;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----6-----2-----SIP 200 OK) completed, returning 0 vicibox81*CLI>

Here is my SVN details:-

Code: Select all: /usr/src/astguiclient/trunk Path: . Working Copy Root Path: /usr/src/astguiclient/trunk URL: svn://svn.eflo.net/agc_2-X/trunk Relative URL: ^/agc_2-X/trunk Repository Root: svn://svn.eflo.net Repository UUID: 3d104415-ff17-0410-8863-d5cf3c621b8a Revision: 3052 Node Kind: directory Schedule: normal Last Changed Author: mattf Last Changed Rev: 3052 Last Changed Date: 2018-10-28 14:53:01 -0400 (Sun, 28 Oct 2018)

Server Configuration:-
4 Core 3.10GHz with 16GB RAM.

vicidial.org

On Hook Agent Unable to Answer Call

On Hook Agent Unable to Answer Call

Re: On Hook Agent Unable to Answer Call

Re: On Hook Agent Unable to Answer Call

Re: On Hook Agent Unable to Answer Call