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