I have a fresh install of vicibox.
Agents can make manual calls but when in ratio the call rings on customer. customer answer the phone the agents are available and the call gets dropped.
DIAL PLAN IN THE CARRIER
exten => _9199900XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9199900XXXXXXXXXXXX,2,Dial(${GOTELECOM}/${EXTEN:5},,To)
exten => _9199900XXXXXXXXXXXX,3,Hangup
[Jul 8 16:45:32] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 8 16:45:32] -- Executing [9199900351932577999@default:1] AGI("Local/9199900351932577999@default-00000018;2", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 8 16:45:32] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Jul 8 16:45:32] -- <Local/9199900351932577999@default-00000018;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 8 16:45:32] -- Executing [9199900351932577999@default:2] Dial("Local/9199900351932577999@default-00000018;2", "SIP/GoTelecom/00351932577999,,To") in new stack
[Jul 8 16:45:32] == Using SIP RTP CoS mark 5
[Jul 8 16:45:32] -- Called SIP/GoTelecom/00351932577999
[Jul 8 16:45:36] -- SIP/GoTelecom-00000013 is making progress passing it to Local/9199900351932577999@default-00000018;2
[Jul 8 16:45:42] -- SIP/GoTelecom-00000013 answered Local/9199900351932577999@default-00000018;2
[Jul 8 16:45:42] > Channel Local/9199900351932577999@default-00000018;1 was answered
[Jul 8 16:45:42] -- Executing [8368@default:1] Playback("Local/9199900351932577999@default-00000018;1", "sip-silence") in new stack
[Jul 8 16:45:42] -- <Local/9199900351932577999@default-00000018;1> Playing 'sip-silence.gsm' (language 'en')
[Jul 8 16:45:42] -- Executing [8368@default:2] AGI("Local/9199900351932577999@default-00000018;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 8 16:45:42] -- <Local/9199900351932577999@default-00000018;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 8 16:45:42] -- Executing [8368@default:3] AGI("Local/9199900351932577999@default-00000018;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 8 16:45:42] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 8 16:45:43] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 8 16:45:43] -- <Local/9199900351932577999@default-00000018;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 8 16:45:43] -- Executing [8368@default:4] AGI("Local/9199900351932577999@default-00000018;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 8 16:45:43] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 8 16:45:44] -- <Local/9199900351932577999@default-00000018;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 8 16:45:44] -- Executing [8368@default:5] Hangup("Local/9199900351932577999@default-00000018;1", "") in new stack
[Jul 8 16:45:44] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/9199900351932577999@default-00000018;1'
[Jul 8 16:45:44] -- Executing [h@default:1] AGI("Local/9199900351932577999@default-00000018;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jul 8 16:45:44] -- <Local/9199900351932577999@default-00000018;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 8 16:45:44] -- Executing [h@default:1] AGI("Local/9199900351932577999@default-00000018;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----2") in new stack
[Jul 8 16:45:45] -- <Local/9199900351932577999@default-00000018;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --12-----2 completed, returning 0
[Jul 8 16:45:45] == Spawn extension (default, 9199900351932577999, 2) exited non-zero on 'Local/9199900351932577999@default-00000018;2'
VERSION: 2.12-561aBUILD: 160708-0745
Calls Not Passing to Agents
Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N
10 posts
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Calls Not Passing to Agents
by fbarrento » Fri Jul 08, 2016 10:54 am
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by Mequetref43 » Mon Jul 11, 2016 1:35 pm
Hi, activate "sip set debug peer xxxxx" and post it
- Mequetref43
- Posts: 19
- Joined: Mon Jul 11, 2016 9:04 am
Re: Calls Not Passing to Agents
by fbarrento » Tue Jul 12, 2016 4:17 am
SIP Debugging Enabled for IP: 213.58.153.220
[Jul 12 09:58:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:01] -- Executing [9351932577999@default:1] AGI("Local/9351932577999@default-0000000c;2", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 12 09:58:01] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Jul 12 09:58:01] -- <Local/9351932577999@default-0000000c;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 12 09:58:01] -- Executing [9351932577999@default:2] Dial("Local/9351932577999@default-0000000c;2", "SIP/GoTelecom/932577999,,tTor") in new stack
[Jul 12 09:58:01] == Using SIP RTP CoS mark 5
[Jul 12 09:58:01] Audio is at 10558
[Jul 12 09:58:01] Adding codec 100002 (gsm) to SDP
[Jul 12 09:58:01] Adding codec 100003 (ulaw) to SDP
[Jul 12 09:58:01] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 12 09:58:01] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:01] INVITE sip:932577999@cloud.govoice.pt SIP/2.0
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK1bd1eb45;rport
[Jul 12 09:58:01] Max-Forwards: 70
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>
[Jul 12 09:58:01] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 102 INVITE
[Jul 12 09:58:01] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] Date: Tue, 12 Jul 2016 08:58:01 GMT
[Jul 12 09:58:01] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:01] Supported: replaces, timer
[Jul 12 09:58:01] Remote-Party-ID: "V7120958010000003103" <sip:211163015@mivarennes.gw.gotelecom.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:01] Content-Type: application/sdp
[Jul 12 09:58:01] Content-Length: 268
[Jul 12 09:58:01]
[Jul 12 09:58:01] v=0
[Jul 12 09:58:01] o=root 1627031330 1627031330 IN IP4 85.245.106.211
[Jul 12 09:58:01] s=Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] c=IN IP4 85.245.106.211
[Jul 12 09:58:01] t=0 0
[Jul 12 09:58:01] m=audio 10558 RTP/AVP 3 0 101
[Jul 12 09:58:01] a=rtpmap:3 GSM/8000
[Jul 12 09:58:01] a=rtpmap:0 PCMU/8000
[Jul 12 09:58:01] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:01] a=fmtp:101 0-16
[Jul 12 09:58:01] a=ptime:20
[Jul 12 09:58:01] a=sendrecv
[Jul 12 09:58:01]
[Jul 12 09:58:01] ---
[Jul 12 09:58:01] -- Called SIP/GoTelecom/932577999
[Jul 12 09:58:01]
[Jul 12 09:58:01] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:01] SIP/2.0 407 Proxy Authentication Required
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK1bd1eb45;rport=62270;received=85.245.106.211
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.16ca
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 102 INVITE
[Jul 12 09:58:01] Proxy-Authenticate: Digest realm="mivarennes.gw.gotelecom.pt", nonce="V4SyRVeEsRlAcjfaG6alL4qIylQZv4ki"
[Jul 12 09:58:01] Content-Length: 0
[Jul 12 09:58:01]
[Jul 12 09:58:01] <------------->
[Jul 12 09:58:01] --- (8 headers 0 lines) ---
[Jul 12 09:58:01] Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:01] ACK sip:932577999@cloud.govoice.pt SIP/2.0
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK1bd1eb45;rport
[Jul 12 09:58:01] Max-Forwards: 70
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.16ca
[Jul 12 09:58:01] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 102 ACK
[Jul 12 09:58:01] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] Content-Length: 0
[Jul 12 09:58:01]
[Jul 12 09:58:01]
[Jul 12 09:58:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:01] -- Executing [9351932577999@default:1] AGI("Local/9351932577999@default-0000000c;2", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 12 09:58:01] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Jul 12 09:58:01] -- <Local/9351932577999@default-0000000c;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 12 09:58:01] -- Executing [9351932577999@default:2] Dial("Local/9351932577999@default-0000000c;2", "SIP/GoTelecom/932577999,,tTor") in new stack
[Jul 12 09:58:01] == Using SIP RTP CoS mark 5
[Jul 12 09:58:01] Audio is at 10558
[Jul 12 09:58:01] Adding codec 100002 (gsm) to SDP
[Jul 12 09:58:01] Adding codec 100003 (ulaw) to SDP
[Jul 12 09:58:01] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 12 09:58:01] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:01] INVITE sip:932577999@cloud.govoice.pt SIP/2.0
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK1bd1eb45;rport
[Jul 12 09:58:01] Max-Forwards: 70
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>
[Jul 12 09:58:01] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 102 INVITE
[Jul 12 09:58:01] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] Date: Tue, 12 Jul 2016 08:58:01 GMT
[Jul 12 09:58:01] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:01] Supported: replaces, timer
[Jul 12 09:58:01] Remote-Party-ID: "V7120958010000003103" <sip:211163015@mivarennes.gw.gotelecom.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:01] Content-Type: application/sdp
[Jul 12 09:58:01] Content-Length: 268
[Jul 12 09:58:01]
[Jul 12 09:58:01] v=0
[Jul 12 09:58:01] o=root 1627031330 1627031330 IN IP4 85.245.106.211
[Jul 12 09:58:01] s=Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] c=IN IP4 85.245.106.211
[Jul 12 09:58:01] t=0 0
[Jul 12 09:58:01] m=audio 10558 RTP/AVP 3 0 101
[Jul 12 09:58:01] a=rtpmap:3 GSM/8000
[Jul 12 09:58:01] a=rtpmap:0 PCMU/8000
[Jul 12 09:58:01] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:01] a=fmtp:101 0-16
[Jul 12 09:58:01] a=ptime:20
[Jul 12 09:58:01] a=sendrecv
[Jul 12 09:58:01]
[Jul 12 09:58:01] ---
[Jul 12 09:58:01] -- Called SIP/GoTelecom/932577999
[Jul 12 09:58:01]
[Jul 12 09:58:01] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:01] SIP/2.0 407 Proxy Authentication Required
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK1bd1eb45;rport=62270;received=85.245.106.211
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.16ca
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 102 INVITE
[Jul 12 09:58:01] Proxy-Authenticate: Digest realm="mivarennes.gw.gotelecom.pt", nonce="V4SyRVeEsRlAcjfaG6alL4qIylQZv4ki"
[Jul 12 09:58:01] Content-Length: 0
[Jul 12 09:58:01]
[Jul 12 09:58:01] <------------->
[Jul 12 09:58:01] --- (8 headers 0 lines) ---
[Jul 12 09:58:01] Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:01] ACK sip:932577999@cloud.govoice.pt SIP/2.0
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK1bd1eb45;rport
[Jul 12 09:58:01] Max-Forwards: 70
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.16ca
[Jul 12 09:58:01] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 102 ACK
[Jul 12 09:58:01] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] Content-Length: 0
[Jul 12 09:58:01]
[Jul 12 09:58:01]
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by fbarrento » Tue Jul 12, 2016 4:19 am
[Jul 12 09:58:01] ---
[Jul 12 09:58:01] Audio is at 10558
[Jul 12 09:58:01] Adding codec 100002 (gsm) to SDP
[Jul 12 09:58:01] Adding codec 100003 (ulaw) to SDP
[Jul 12 09:58:01] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 12 09:58:01] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:01] INVITE sip:932577999@cloud.govoice.pt SIP/2.0
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK3d8355cd;rport
[Jul 12 09:58:01] Max-Forwards: 70
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>
[Jul 12 09:58:01] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 103 INVITE
[Jul 12 09:58:01] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] Proxy-Authorization: Digest username="beslim", realm="mivarennes.gw.gotelecom.pt", algorithm=MD5, uri="sip:932577999@cloud.govoice.pt", nonce="V4SyRVeEsRlAcjfaG6alL4qIylQZv4ki", response="75f661c6b25dcea873b2f2633f40d744"
[Jul 12 09:58:01] Date: Tue, 12 Jul 2016 08:58:01 GMT
[Jul 12 09:58:01] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:01] Supported: replaces, timer
[Jul 12 09:58:01] Remote-Party-ID: "V7120958010000003103" <sip:211163015@mivarennes.gw.gotelecom.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:01] Content-Type: application/sdp
[Jul 12 09:58:01] Content-Length: 268
[Jul 12 09:58:01]
[Jul 12 09:58:01] v=0
[Jul 12 09:58:01] o=root 1627031330 1627031331 IN IP4 85.245.106.211
[Jul 12 09:58:01] s=Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] c=IN IP4 85.245.106.211
[Jul 12 09:58:01] t=0 0
[Jul 12 09:58:01] m=audio 10558 RTP/AVP 3 0 101
[Jul 12 09:58:01] a=rtpmap:3 GSM/8000
[Jul 12 09:58:01] a=rtpmap:0 PCMU/8000
[Jul 12 09:58:01] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:01] a=fmtp:101 0-16
[Jul 12 09:58:01] a=ptime:20
[Jul 12 09:58:01] a=sendrecv
[Jul 12 09:58:01]
[Jul 12 09:58:01] ---
[Jul 12 09:58:01]
[Jul 12 09:58:01] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:01] SIP/2.0 100 trying -- your call is important to us
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK3d8355cd;rport=62270;received=85.245.106.211
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 103 INVITE
[Jul 12 09:58:01] Content-Length: 0
[Jul 12 09:58:01]
[Jul 12 09:58:01] <------------->
[Jul 12 09:58:01] --- (7 headers 0 lines) ---
[Jul 12 09:58:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:04]
[Jul 12 09:58:04] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:04] SIP/2.0 183 Session Progress
[Jul 12 09:58:04] Via: SIP/2.0/UDP 85.245.106.211:5060;received=85.245.106.211;branch=z9hG4bK3d8355cd;rport=62270
[Jul 12 09:58:04] Record-Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:04] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:04] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:04] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:04] CSeq: 103 INVITE
[Jul 12 09:58:04] Contact: <sip:932577999@213.58.153.222:5070;transport=udp>
[Jul 12 09:58:04] User-Agent: FSCloudIPBX-sip_wavecom/1.4.23git699403f~64bit
[Jul 12 09:58:04] Accept: application/sdp
[Jul 12 09:58:04] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
[Jul 12 09:58:04] Supported: timer, path, replaces
[Jul 12 09:58:04] Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
[Jul 12 09:58:04] Content-Type: application/sdp
[Jul 12 09:58:04] Content-Disposition: session
[Jul 12 09:58:04] Content-Length: 225
[Jul 12 09:58:04] Remote-Party-ID: "932577999" <sip:932577999@cloud.govoice.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:04]
[Jul 12 09:58:04] v=0
[Jul 12 09:58:04] o=FSCloudIPBX 1468294928 1468294929 IN IP4 213.58.153.222
[Jul 12 09:58:04] s=FSCloudIPBX
[Jul 12 09:58:04] c=IN IP4 213.58.153.222
[Jul 12 09:58:04] t=0 0
[Jul 12 09:58:04] m=audio 18956 RTP/AVP 3 101
[Jul 12 09:58:04] a=rtpmap:3 GSM/8000
[Jul 12 09:58:04] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:04] a=fmtp:101 0-16
[Jul 12 09:58:04] a=ptime:20
[Jul 12 09:58:04] <------------->
[Jul 12 09:58:04] --- (17 headers 10 lines) ---
[Jul 12 09:58:04] list_route: hop: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:04] Found RTP audio format 3
[Jul 12 09:58:04] Found RTP audio format 101
[Jul 12 09:58:04] Found audio description format GSM for ID 3
[Jul 12 09:58:04] Found audio description format telephone-event for ID 101
[Jul 12 09:58:04] Capabilities: us - (gsm|ulaw), peer - audio=(gsm)/video=(nothing)/text=(nothing), combined - (gsm)
[Jul 12 09:58:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jul 12 09:58:04] Peer audio RTP is at port 213.58.153.222:18956
[Jul 12 09:58:04] -- SIP/GoTelecom-0000000c is making progress passing it to Local/9351932577999@default-0000000c;2
[Jul 12 09:58:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:12]
[Jul 12 09:58:12] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:12] SIP/2.0 200 OK
[Jul 12 09:58:12] Via: SIP/2.0/UDP 85.245.106.211:5060;received=85.245.106.211;branch=z9hG4bK3d8355cd;rport=62270
[Jul 12 09:58:12] Record-Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:12] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:12] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:12] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:12] CSeq: 103 INVITE
[Jul 12 09:58:12] Contact: <sip:932577999@213.58.153.222:5070;transport=udp>
[Jul 12 09:58:12] User-Agent: FSCloudIPBX-sip_wavecom/1.4.23git699403f~64bit
[Jul 12 09:58:12] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
[Jul 12 09:58:12] Supported: timer, path, replaces
[Jul 12 09:58:12] Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
[Jul 12 09:58:12] Content-Type: application/sdp
[Jul 12 09:58:12] Content-Disposition: session
[Jul 12 09:58:12] Content-Length: 225
[Jul 12 09:58:12] Remote-Party-ID: "932577999" <sip:932577999@cloud.govoice.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:12]
[Jul 12 09:58:12] v=0
[Jul 12 09:58:12] o=FSCloudIPBX 1468294928 1468294929 IN IP4 213.58.153.222
[Jul 12 09:58:12] s=FSCloudIPBX
[Jul 12 09:58:12] c=IN IP4 213.58.153.222
[Jul 12 09:58:12] t=0 0
[Jul 12 09:58:12] m=audio 18956 RTP/AVP 3 101
[Jul 12 09:58:12] a=rtpmap:3 GSM/8000
[Jul 12 09:58:12] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:12] a=fmtp:101 0-16
[Jul 12 09:58:12] a=ptime:20
[Jul 12 09:58:12] <------------->
[Jul 12 09:58:12] --- (16 headers 10 lines) ---
[Jul 12 09:58:12] list_route: hop: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:12] set_destination: Parsing <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes> for address/port to send to
[Jul 12 09:58:12] set_destination: set destination to 213.58.153.220:5060
[Jul 12 09:58:12] Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:12] ACK sip:932577999@213.58.153.222:5070;transport=udp SIP/2.0
[Jul 12 09:58:12] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK6e9464b5;rport
[Jul 12 09:58:12] Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:12] Max-Forwards: 70
[Jul 12 09:58:12] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:12] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:12] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:12] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:12] CSeq: 103 ACK
[Jul 12 09:58:12] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:12] Content-Length: 0
[Jul 12 09:58:12]
[Jul 12 09:58:01] Audio is at 10558
[Jul 12 09:58:01] Adding codec 100002 (gsm) to SDP
[Jul 12 09:58:01] Adding codec 100003 (ulaw) to SDP
[Jul 12 09:58:01] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 12 09:58:01] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:01] INVITE sip:932577999@cloud.govoice.pt SIP/2.0
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK3d8355cd;rport
[Jul 12 09:58:01] Max-Forwards: 70
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>
[Jul 12 09:58:01] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 103 INVITE
[Jul 12 09:58:01] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] Proxy-Authorization: Digest username="beslim", realm="mivarennes.gw.gotelecom.pt", algorithm=MD5, uri="sip:932577999@cloud.govoice.pt", nonce="V4SyRVeEsRlAcjfaG6alL4qIylQZv4ki", response="75f661c6b25dcea873b2f2633f40d744"
[Jul 12 09:58:01] Date: Tue, 12 Jul 2016 08:58:01 GMT
[Jul 12 09:58:01] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:01] Supported: replaces, timer
[Jul 12 09:58:01] Remote-Party-ID: "V7120958010000003103" <sip:211163015@mivarennes.gw.gotelecom.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:01] Content-Type: application/sdp
[Jul 12 09:58:01] Content-Length: 268
[Jul 12 09:58:01]
[Jul 12 09:58:01] v=0
[Jul 12 09:58:01] o=root 1627031330 1627031331 IN IP4 85.245.106.211
[Jul 12 09:58:01] s=Asterisk PBX 11.22.0-vici
[Jul 12 09:58:01] c=IN IP4 85.245.106.211
[Jul 12 09:58:01] t=0 0
[Jul 12 09:58:01] m=audio 10558 RTP/AVP 3 0 101
[Jul 12 09:58:01] a=rtpmap:3 GSM/8000
[Jul 12 09:58:01] a=rtpmap:0 PCMU/8000
[Jul 12 09:58:01] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:01] a=fmtp:101 0-16
[Jul 12 09:58:01] a=ptime:20
[Jul 12 09:58:01] a=sendrecv
[Jul 12 09:58:01]
[Jul 12 09:58:01] ---
[Jul 12 09:58:01]
[Jul 12 09:58:01] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:01] SIP/2.0 100 trying -- your call is important to us
[Jul 12 09:58:01] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK3d8355cd;rport=62270;received=85.245.106.211
[Jul 12 09:58:01] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:01] To: <sip:932577999@cloud.govoice.pt>
[Jul 12 09:58:01] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:01] CSeq: 103 INVITE
[Jul 12 09:58:01] Content-Length: 0
[Jul 12 09:58:01]
[Jul 12 09:58:01] <------------->
[Jul 12 09:58:01] --- (7 headers 0 lines) ---
[Jul 12 09:58:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:04]
[Jul 12 09:58:04] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:04] SIP/2.0 183 Session Progress
[Jul 12 09:58:04] Via: SIP/2.0/UDP 85.245.106.211:5060;received=85.245.106.211;branch=z9hG4bK3d8355cd;rport=62270
[Jul 12 09:58:04] Record-Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:04] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:04] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:04] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:04] CSeq: 103 INVITE
[Jul 12 09:58:04] Contact: <sip:932577999@213.58.153.222:5070;transport=udp>
[Jul 12 09:58:04] User-Agent: FSCloudIPBX-sip_wavecom/1.4.23git699403f~64bit
[Jul 12 09:58:04] Accept: application/sdp
[Jul 12 09:58:04] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
[Jul 12 09:58:04] Supported: timer, path, replaces
[Jul 12 09:58:04] Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
[Jul 12 09:58:04] Content-Type: application/sdp
[Jul 12 09:58:04] Content-Disposition: session
[Jul 12 09:58:04] Content-Length: 225
[Jul 12 09:58:04] Remote-Party-ID: "932577999" <sip:932577999@cloud.govoice.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:04]
[Jul 12 09:58:04] v=0
[Jul 12 09:58:04] o=FSCloudIPBX 1468294928 1468294929 IN IP4 213.58.153.222
[Jul 12 09:58:04] s=FSCloudIPBX
[Jul 12 09:58:04] c=IN IP4 213.58.153.222
[Jul 12 09:58:04] t=0 0
[Jul 12 09:58:04] m=audio 18956 RTP/AVP 3 101
[Jul 12 09:58:04] a=rtpmap:3 GSM/8000
[Jul 12 09:58:04] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:04] a=fmtp:101 0-16
[Jul 12 09:58:04] a=ptime:20
[Jul 12 09:58:04] <------------->
[Jul 12 09:58:04] --- (17 headers 10 lines) ---
[Jul 12 09:58:04] list_route: hop: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:04] Found RTP audio format 3
[Jul 12 09:58:04] Found RTP audio format 101
[Jul 12 09:58:04] Found audio description format GSM for ID 3
[Jul 12 09:58:04] Found audio description format telephone-event for ID 101
[Jul 12 09:58:04] Capabilities: us - (gsm|ulaw), peer - audio=(gsm)/video=(nothing)/text=(nothing), combined - (gsm)
[Jul 12 09:58:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jul 12 09:58:04] Peer audio RTP is at port 213.58.153.222:18956
[Jul 12 09:58:04] -- SIP/GoTelecom-0000000c is making progress passing it to Local/9351932577999@default-0000000c;2
[Jul 12 09:58:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 09:58:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:12]
[Jul 12 09:58:12] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:12] SIP/2.0 200 OK
[Jul 12 09:58:12] Via: SIP/2.0/UDP 85.245.106.211:5060;received=85.245.106.211;branch=z9hG4bK3d8355cd;rport=62270
[Jul 12 09:58:12] Record-Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:12] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:12] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:12] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:12] CSeq: 103 INVITE
[Jul 12 09:58:12] Contact: <sip:932577999@213.58.153.222:5070;transport=udp>
[Jul 12 09:58:12] User-Agent: FSCloudIPBX-sip_wavecom/1.4.23git699403f~64bit
[Jul 12 09:58:12] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
[Jul 12 09:58:12] Supported: timer, path, replaces
[Jul 12 09:58:12] Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
[Jul 12 09:58:12] Content-Type: application/sdp
[Jul 12 09:58:12] Content-Disposition: session
[Jul 12 09:58:12] Content-Length: 225
[Jul 12 09:58:12] Remote-Party-ID: "932577999" <sip:932577999@cloud.govoice.pt>;party=calling;privacy=off;screen=no
[Jul 12 09:58:12]
[Jul 12 09:58:12] v=0
[Jul 12 09:58:12] o=FSCloudIPBX 1468294928 1468294929 IN IP4 213.58.153.222
[Jul 12 09:58:12] s=FSCloudIPBX
[Jul 12 09:58:12] c=IN IP4 213.58.153.222
[Jul 12 09:58:12] t=0 0
[Jul 12 09:58:12] m=audio 18956 RTP/AVP 3 101
[Jul 12 09:58:12] a=rtpmap:3 GSM/8000
[Jul 12 09:58:12] a=rtpmap:101 telephone-event/8000
[Jul 12 09:58:12] a=fmtp:101 0-16
[Jul 12 09:58:12] a=ptime:20
[Jul 12 09:58:12] <------------->
[Jul 12 09:58:12] --- (16 headers 10 lines) ---
[Jul 12 09:58:12] list_route: hop: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:12] set_destination: Parsing <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes> for address/port to send to
[Jul 12 09:58:12] set_destination: set destination to 213.58.153.220:5060
[Jul 12 09:58:12] Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:12] ACK sip:932577999@213.58.153.222:5070;transport=udp SIP/2.0
[Jul 12 09:58:12] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK6e9464b5;rport
[Jul 12 09:58:12] Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:12] Max-Forwards: 70
[Jul 12 09:58:12] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:12] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:12] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:12] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:12] CSeq: 103 ACK
[Jul 12 09:58:12] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:12] Content-Length: 0
[Jul 12 09:58:12]
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by fbarrento » Tue Jul 12, 2016 4:20 am
[Jul 12 09:58:12]
[Jul 12 09:58:12] ---
[Jul 12 09:58:12] -- SIP/GoTelecom-0000000c answered Local/9351932577999@default-0000000c;2
[Jul 12 09:58:12] > Channel Local/9351932577999@default-0000000c;1 was answered
[Jul 12 09:58:12] -- Executing [8364@default:1] Playback("Local/9351932577999@default-0000000c;1", "sip-silence") in new stack
[Jul 12 09:58:12] -- <Local/9351932577999@default-0000000c;1> Playing 'sip-silence.gsm' (language 'en')
[Jul 12 09:58:12] -- Executing [8364@default:2] AGI("Local/9351932577999@default-0000000c;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 12 09:58:12] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Jul 12 09:58:12] -- <Local/9351932577999@default-0000000c;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 12 09:58:12] -- Executing [8364@default:3] AGI("Local/9351932577999@default-0000000c;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 12 09:58:12] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 12 09:58:13] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:13] -- <Local/9351932577999@default-0000000c;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 12 09:58:13] -- Executing [8364@default:4] AGI("Local/9351932577999@default-0000000c;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 12 09:58:13] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 12 09:58:14] -- <Local/9351932577999@default-0000000c;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 12 09:58:14] -- Executing [8364@default:5] Hangup("Local/9351932577999@default-0000000c;1", "") in new stack
[Jul 12 09:58:14] == Spawn extension (default, 8364, 5) exited non-zero on 'Local/9351932577999@default-0000000c;1'
[Jul 12 09:58:14] -- Executing [h@default:1] AGI("Local/9351932577999@default-0000000c;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jul 12 09:58:15] -- <Local/9351932577999@default-0000000c;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 12 09:58:15] -- Executing [h@default:1] AGI("Local/9351932577999@default-0000000c;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----14-----3") in new stack
[Jul 12 09:58:16] -- <Local/9351932577999@default-0000000c;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --14-----3 completed, returning 0
[Jul 12 09:58:12] ---
[Jul 12 09:58:12] -- SIP/GoTelecom-0000000c answered Local/9351932577999@default-0000000c;2
[Jul 12 09:58:12] > Channel Local/9351932577999@default-0000000c;1 was answered
[Jul 12 09:58:12] -- Executing [8364@default:1] Playback("Local/9351932577999@default-0000000c;1", "sip-silence") in new stack
[Jul 12 09:58:12] -- <Local/9351932577999@default-0000000c;1> Playing 'sip-silence.gsm' (language 'en')
[Jul 12 09:58:12] -- Executing [8364@default:2] AGI("Local/9351932577999@default-0000000c;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jul 12 09:58:12] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Jul 12 09:58:12] -- <Local/9351932577999@default-0000000c;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jul 12 09:58:12] -- Executing [8364@default:3] AGI("Local/9351932577999@default-0000000c;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 12 09:58:12] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 12 09:58:13] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 09:58:13] -- <Local/9351932577999@default-0000000c;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 12 09:58:13] -- Executing [8364@default:4] AGI("Local/9351932577999@default-0000000c;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Jul 12 09:58:13] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jul 12 09:58:14] -- <Local/9351932577999@default-0000000c;1>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jul 12 09:58:14] -- Executing [8364@default:5] Hangup("Local/9351932577999@default-0000000c;1", "") in new stack
[Jul 12 09:58:14] == Spawn extension (default, 8364, 5) exited non-zero on 'Local/9351932577999@default-0000000c;1'
[Jul 12 09:58:14] -- Executing [h@default:1] AGI("Local/9351932577999@default-0000000c;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jul 12 09:58:15] -- <Local/9351932577999@default-0000000c;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 12 09:58:15] -- Executing [h@default:1] AGI("Local/9351932577999@default-0000000c;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----14-----3") in new stack
[Jul 12 09:58:16] -- <Local/9351932577999@default-0000000c;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --14-----3 completed, returning 0
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by fbarrento » Tue Jul 12, 2016 4:21 am
[Jul 12 09:58:16] Scheduling destruction of SIP dialog '0622cc3e2ca293280ffd4b454dabf117@mivar ... telecom.pt' in 6400 ms (Method: INVITE)
[Jul 12 09:58:16] set_destination: Parsing <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes> for address/port to send to
[Jul 12 09:58:16] set_destination: set destination to 213.58.153.220:5060
[Jul 12 09:58:16] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:16] BYE sip:932577999@213.58.153.222:5070;transport=udp SIP/2.0
[Jul 12 09:58:16] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK13f51690;rport
[Jul 12 09:58:16] Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:16] Max-Forwards: 70
[Jul 12 09:58:16] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:16] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:16] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:16] CSeq: 104 BYE
[Jul 12 09:58:16] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:16] Proxy-Authorization: Digest username="beslim", realm="mivarennes.gw.gotelecom.pt", algorithm=MD5, uri="sip:932577999@213.58.153.222:5070", nonce="V4SyRVeEsRlAcjfaG6alL4qIylQZv4ki", response="dfc87013152068652c0a5d1190724313"
[Jul 12 09:58:16] X-Asterisk-HangupCause: Normal Clearing
[Jul 12 09:58:16] X-Asterisk-HangupCauseCode: 16
[Jul 12 09:58:16] Content-Length: 0
[Jul 12 09:58:16]
[Jul 12 09:58:16]
[Jul 12 09:58:16] ---
[Jul 12 09:58:16] == Spawn extension (default, 9351932577999, 2) exited non-zero on 'Local/9351932577999@default-0000000c;2'
[Jul 12 09:58:16]
[Jul 12 09:58:16] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:16] SIP/2.0 200 OK
[Jul 12 09:58:16] Via: SIP/2.0/UDP 85.245.106.211:5060;received=85.245.106.211;branch=z9hG4bK13f51690;rport=62270
[Jul 12 09:58:16] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:16] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:16] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:16] CSeq: 104 BYE
[Jul 12 09:58:16] User-Agent: FSCloudIPBX-sip_wavecom/1.4.23git699403f~64bit
[Jul 12 09:58:16] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
[Jul 12 09:58:16] Supported: timer, path, replaces
[Jul 12 09:58:16] Content-Length: 0
[Jul 12 09:58:16]
[Jul 12 09:58:16] <------------->
[Jul 12 09:58:16] --- (10 headers 0 lines) ---
[Jul 12 09:58:16] Really destroying SIP dialog '0622cc3e2ca293280ffd4b454dabf117@mivar ... telecom.pt' Method: INVITE
[Jul 12 09:58:19]
[Jul 12 09:58:19] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:19] OPTIONS sip:beslim@85.245.106.211:5060 SIP/2.0
[Jul 12 09:58:19] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0
[Jul 12 09:58:19] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-a270f936
[Jul 12 09:58:19] To: sip:beslim@85.245.106.211:5060
[Jul 12 09:58:19] Call-ID: 2a980ea4-35ef6023-ad30d8@213.58.153.220
[Jul 12 09:58:19] CSeq: 1 OPTIONS
[Jul 12 09:58:19] Content-Length: 0
[Jul 12 09:58:19]
[Jul 12 09:58:19] <------------->
[Jul 12 09:58:19] --- (7 headers 0 lines) ---
[Jul 12 09:58:19] Sending to 213.58.153.220:5060 (NAT)
[Jul 12 09:58:19] Looking for beslim in trunkinbound (domain 85.245.106.211)
[Jul 12 09:58:16] set_destination: Parsing <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes> for address/port to send to
[Jul 12 09:58:16] set_destination: set destination to 213.58.153.220:5060
[Jul 12 09:58:16] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:16] BYE sip:932577999@213.58.153.222:5070;transport=udp SIP/2.0
[Jul 12 09:58:16] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK13f51690;rport
[Jul 12 09:58:16] Route: <sip:213.58.153.220;lr=on;ftag=as3c864613;did=79c.92a;nat=yes>
[Jul 12 09:58:16] Max-Forwards: 70
[Jul 12 09:58:16] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:16] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:16] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:16] CSeq: 104 BYE
[Jul 12 09:58:16] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:16] Proxy-Authorization: Digest username="beslim", realm="mivarennes.gw.gotelecom.pt", algorithm=MD5, uri="sip:932577999@213.58.153.222:5070", nonce="V4SyRVeEsRlAcjfaG6alL4qIylQZv4ki", response="dfc87013152068652c0a5d1190724313"
[Jul 12 09:58:16] X-Asterisk-HangupCause: Normal Clearing
[Jul 12 09:58:16] X-Asterisk-HangupCauseCode: 16
[Jul 12 09:58:16] Content-Length: 0
[Jul 12 09:58:16]
[Jul 12 09:58:16]
[Jul 12 09:58:16] ---
[Jul 12 09:58:16] == Spawn extension (default, 9351932577999, 2) exited non-zero on 'Local/9351932577999@default-0000000c;2'
[Jul 12 09:58:16]
[Jul 12 09:58:16] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:16] SIP/2.0 200 OK
[Jul 12 09:58:16] Via: SIP/2.0/UDP 85.245.106.211:5060;received=85.245.106.211;branch=z9hG4bK13f51690;rport=62270
[Jul 12 09:58:16] From: "V7120958010000003103" <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as3c864613
[Jul 12 09:58:16] To: <sip:932577999@cloud.govoice.pt>;tag=X9ceBSX3reN4D
[Jul 12 09:58:16] Call-ID: 0622cc3e2ca293280ffd4b454dabf117@mivare ... telecom.pt
[Jul 12 09:58:16] CSeq: 104 BYE
[Jul 12 09:58:16] User-Agent: FSCloudIPBX-sip_wavecom/1.4.23git699403f~64bit
[Jul 12 09:58:16] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
[Jul 12 09:58:16] Supported: timer, path, replaces
[Jul 12 09:58:16] Content-Length: 0
[Jul 12 09:58:16]
[Jul 12 09:58:16] <------------->
[Jul 12 09:58:16] --- (10 headers 0 lines) ---
[Jul 12 09:58:16] Really destroying SIP dialog '0622cc3e2ca293280ffd4b454dabf117@mivar ... telecom.pt' Method: INVITE
[Jul 12 09:58:19]
[Jul 12 09:58:19] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:19] OPTIONS sip:beslim@85.245.106.211:5060 SIP/2.0
[Jul 12 09:58:19] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0
[Jul 12 09:58:19] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-a270f936
[Jul 12 09:58:19] To: sip:beslim@85.245.106.211:5060
[Jul 12 09:58:19] Call-ID: 2a980ea4-35ef6023-ad30d8@213.58.153.220
[Jul 12 09:58:19] CSeq: 1 OPTIONS
[Jul 12 09:58:19] Content-Length: 0
[Jul 12 09:58:19]
[Jul 12 09:58:19] <------------->
[Jul 12 09:58:19] --- (7 headers 0 lines) ---
[Jul 12 09:58:19] Sending to 213.58.153.220:5060 (NAT)
[Jul 12 09:58:19] Looking for beslim in trunkinbound (domain 85.245.106.211)
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by fbarrento » Tue Jul 12, 2016 4:22 am
[Jul 12 09:58:19]
[Jul 12 09:58:19] <--- Transmitting (NAT) to 213.58.153.220:5060 --->
[Jul 12 09:58:19] SIP/2.0 404 Not Found
[Jul 12 09:58:19] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0;received=213.58.153.220;rport=5060
[Jul 12 09:58:19] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-a270f936
[Jul 12 09:58:19] To: sip:beslim@85.245.106.211:5060;tag=as6b45d247
[Jul 12 09:58:19] Call-ID: 2a980ea4-35ef6023-ad30d8@213.58.153.220
[Jul 12 09:58:19] CSeq: 1 OPTIONS
[Jul 12 09:58:19] Server: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:19] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:19] Supported: replaces, timer
[Jul 12 09:58:19] Accept: application/sdp
[Jul 12 09:58:19] Content-Length: 0
[Jul 12 09:58:19]
[Jul 12 09:58:19]
[Jul 12 09:58:19] <------------>
[Jul 12 09:58:19] Scheduling destruction of SIP dialog '2a980ea4-35ef6023-ad30d8@213.58.153.220' in 32000 ms (Method: OPTIONS)
[Jul 12 09:58:21] Really destroying SIP dialog '2a980ea4-c5cf6023-cb30d8@213.58.153.220' Method: OPTIONS
[Jul 12 09:58:34] NOTICE[1793]: chan_sip.c:15209 sip_reregister: -- Re-registration for beslim@cloud.govoice.pt
[Jul 12 09:58:34] REGISTER 11 headers, 0 lines
[Jul 12 09:58:34] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:34] REGISTER sip:mivarennes.gw.gotelecom.pt SIP/2.0
[Jul 12 09:58:34] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK45fd6be1;rport
[Jul 12 09:58:34] Max-Forwards: 70
[Jul 12 09:58:34] From: <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as08300808
[Jul 12 09:58:34] To: <sip:beslim@mivarennes.gw.gotelecom.pt>
[Jul 12 09:58:34] Call-ID: 72222f0d4fd988180fb3efc362ed9ce6@192.168.10.231
[Jul 12 09:58:34] CSeq: 141 REGISTER
[Jul 12 09:58:34] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:34] Authorization: Digest username="beslim", realm="mivarennes.gw.gotelecom.pt", algorithm=MD5, uri="sip:mivarennes.gw.gotelecom.pt", nonce="V4SxlFeEsGieveEgNOWj+Gp2e3zqw8xd", response="28887aa03a074c98286e683ff3daa4d4"
[Jul 12 09:58:34] Expires: 120
[Jul 12 09:58:34] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:34] Content-Length: 0
[Jul 12 09:58:34]
[Jul 12 09:58:34]
[Jul 12 09:58:34] ---
[Jul 12 09:58:34]
[Jul 12 09:58:34] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:34] SIP/2.0 200 OK
[Jul 12 09:58:34] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK45fd6be1;rport=62270;received=85.245.106.211
[Jul 12 09:58:34] From: <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as08300808
[Jul 12 09:58:34] To: <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.769a
[Jul 12 09:58:34] Call-ID: 72222f0d4fd988180fb3efc362ed9ce6@192.168.10.231
[Jul 12 09:58:34] CSeq: 141 REGISTER
[Jul 12 09:58:34] Contact: <sip:beslim@85.245.106.211:5060>;expires=120;received="sip:85.245.106.211:62270"
[Jul 12 09:58:34] Content-Length: 0
[Jul 12 09:58:34]
[Jul 12 09:58:34] <------------->
[Jul 12 09:58:34] --- (8 headers 0 lines) ---
[Jul 12 09:58:34] NOTICE[1793]: chan_sip.c:23776 handle_response_register: Outbound Registration: Expiry for cloud.govoice.pt is 120 sec (Scheduling reregistration in 105 s)
[Jul 12 09:58:34] Really destroying SIP dialog '72222f0d4fd988180fb3efc362ed9ce6@192.168.10.231' Method: REGISTER
[Jul 12 09:58:37] NOTICE[1793]: chan_sip.c:15209 sip_reregister: -- Re-registration for fbarrento*experbiz@sip.voip.e-xper.pt
[Jul 12 09:58:38] NOTICE[1793]: chan_sip.c:23776 handle_response_register: Outbound Registration: Expiry for sip.voip.e-xper.pt is 120 sec (Scheduling reregistration in 105 s)
[Jul 12 09:58:48] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:48] OPTIONS sip:cloud.govoice.pt SIP/2.0
[Jul 12 09:58:48] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK0ba87bda;rport
[Jul 12 09:58:48] Max-Forwards: 70
[Jul 12 09:58:48] From: "asterisk" <sip:beslim@85.245.106.211>;tag=as22abf1e8
[Jul 12 09:58:48] To: <sip:cloud.govoice.pt>
[Jul 12 09:58:48] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:48] Call-ID: 597e48ec5c79974032d5eda109c5477e@85.245.106.211:5060
[Jul 12 09:58:48] CSeq: 102 OPTIONS
[Jul 12 09:58:48] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:48] Date: Tue, 12 Jul 2016 08:58:48 GMT
[Jul 12 09:58:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:48] Supported: replaces, timer
[Jul 12 09:58:48] Content-Length: 0
[Jul 12 09:58:48]
[Jul 12 09:58:48]
[Jul 12 09:58:48] ---
[Jul 12 09:58:48]
[Jul 12 09:58:48] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:48] SIP/2.0 200 OK
[Jul 12 09:58:48] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK0ba87bda;rport=62270;received=85.245.106.211
[Jul 12 09:58:48] From: "asterisk" <sip:beslim@85.245.106.211>;tag=as22abf1e8
[Jul 12 09:58:48] To: <sip:cloud.govoice.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b034
[Jul 12 09:58:48] Call-ID: 597e48ec5c79974032d5eda109c5477e@85.245.106.211:5060
[Jul 12 09:58:48] CSeq: 102 OPTIONS
[Jul 12 09:58:48] Accept: */*
[Jul 12 09:58:48] Accept-Encoding:
[Jul 12 09:58:48] Accept-Language: en
[Jul 12 09:58:48] Supported:
[Jul 12 09:58:48] Content-Length: 0
[Jul 12 09:58:19] <--- Transmitting (NAT) to 213.58.153.220:5060 --->
[Jul 12 09:58:19] SIP/2.0 404 Not Found
[Jul 12 09:58:19] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0;received=213.58.153.220;rport=5060
[Jul 12 09:58:19] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-a270f936
[Jul 12 09:58:19] To: sip:beslim@85.245.106.211:5060;tag=as6b45d247
[Jul 12 09:58:19] Call-ID: 2a980ea4-35ef6023-ad30d8@213.58.153.220
[Jul 12 09:58:19] CSeq: 1 OPTIONS
[Jul 12 09:58:19] Server: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:19] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:19] Supported: replaces, timer
[Jul 12 09:58:19] Accept: application/sdp
[Jul 12 09:58:19] Content-Length: 0
[Jul 12 09:58:19]
[Jul 12 09:58:19]
[Jul 12 09:58:19] <------------>
[Jul 12 09:58:19] Scheduling destruction of SIP dialog '2a980ea4-35ef6023-ad30d8@213.58.153.220' in 32000 ms (Method: OPTIONS)
[Jul 12 09:58:21] Really destroying SIP dialog '2a980ea4-c5cf6023-cb30d8@213.58.153.220' Method: OPTIONS
[Jul 12 09:58:34] NOTICE[1793]: chan_sip.c:15209 sip_reregister: -- Re-registration for beslim@cloud.govoice.pt
[Jul 12 09:58:34] REGISTER 11 headers, 0 lines
[Jul 12 09:58:34] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:34] REGISTER sip:mivarennes.gw.gotelecom.pt SIP/2.0
[Jul 12 09:58:34] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK45fd6be1;rport
[Jul 12 09:58:34] Max-Forwards: 70
[Jul 12 09:58:34] From: <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as08300808
[Jul 12 09:58:34] To: <sip:beslim@mivarennes.gw.gotelecom.pt>
[Jul 12 09:58:34] Call-ID: 72222f0d4fd988180fb3efc362ed9ce6@192.168.10.231
[Jul 12 09:58:34] CSeq: 141 REGISTER
[Jul 12 09:58:34] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:34] Authorization: Digest username="beslim", realm="mivarennes.gw.gotelecom.pt", algorithm=MD5, uri="sip:mivarennes.gw.gotelecom.pt", nonce="V4SxlFeEsGieveEgNOWj+Gp2e3zqw8xd", response="28887aa03a074c98286e683ff3daa4d4"
[Jul 12 09:58:34] Expires: 120
[Jul 12 09:58:34] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:34] Content-Length: 0
[Jul 12 09:58:34]
[Jul 12 09:58:34]
[Jul 12 09:58:34] ---
[Jul 12 09:58:34]
[Jul 12 09:58:34] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:34] SIP/2.0 200 OK
[Jul 12 09:58:34] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK45fd6be1;rport=62270;received=85.245.106.211
[Jul 12 09:58:34] From: <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=as08300808
[Jul 12 09:58:34] To: <sip:beslim@mivarennes.gw.gotelecom.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.769a
[Jul 12 09:58:34] Call-ID: 72222f0d4fd988180fb3efc362ed9ce6@192.168.10.231
[Jul 12 09:58:34] CSeq: 141 REGISTER
[Jul 12 09:58:34] Contact: <sip:beslim@85.245.106.211:5060>;expires=120;received="sip:85.245.106.211:62270"
[Jul 12 09:58:34] Content-Length: 0
[Jul 12 09:58:34]
[Jul 12 09:58:34] <------------->
[Jul 12 09:58:34] --- (8 headers 0 lines) ---
[Jul 12 09:58:34] NOTICE[1793]: chan_sip.c:23776 handle_response_register: Outbound Registration: Expiry for cloud.govoice.pt is 120 sec (Scheduling reregistration in 105 s)
[Jul 12 09:58:34] Really destroying SIP dialog '72222f0d4fd988180fb3efc362ed9ce6@192.168.10.231' Method: REGISTER
[Jul 12 09:58:37] NOTICE[1793]: chan_sip.c:15209 sip_reregister: -- Re-registration for fbarrento*experbiz@sip.voip.e-xper.pt
[Jul 12 09:58:38] NOTICE[1793]: chan_sip.c:23776 handle_response_register: Outbound Registration: Expiry for sip.voip.e-xper.pt is 120 sec (Scheduling reregistration in 105 s)
[Jul 12 09:58:48] Reliably Transmitting (NAT) to 213.58.153.220:5060:
[Jul 12 09:58:48] OPTIONS sip:cloud.govoice.pt SIP/2.0
[Jul 12 09:58:48] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK0ba87bda;rport
[Jul 12 09:58:48] Max-Forwards: 70
[Jul 12 09:58:48] From: "asterisk" <sip:beslim@85.245.106.211>;tag=as22abf1e8
[Jul 12 09:58:48] To: <sip:cloud.govoice.pt>
[Jul 12 09:58:48] Contact: <sip:beslim@85.245.106.211:5060>
[Jul 12 09:58:48] Call-ID: 597e48ec5c79974032d5eda109c5477e@85.245.106.211:5060
[Jul 12 09:58:48] CSeq: 102 OPTIONS
[Jul 12 09:58:48] User-Agent: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:48] Date: Tue, 12 Jul 2016 08:58:48 GMT
[Jul 12 09:58:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:48] Supported: replaces, timer
[Jul 12 09:58:48] Content-Length: 0
[Jul 12 09:58:48]
[Jul 12 09:58:48]
[Jul 12 09:58:48] ---
[Jul 12 09:58:48]
[Jul 12 09:58:48] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:48] SIP/2.0 200 OK
[Jul 12 09:58:48] Via: SIP/2.0/UDP 85.245.106.211:5060;branch=z9hG4bK0ba87bda;rport=62270;received=85.245.106.211
[Jul 12 09:58:48] From: "asterisk" <sip:beslim@85.245.106.211>;tag=as22abf1e8
[Jul 12 09:58:48] To: <sip:cloud.govoice.pt>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b034
[Jul 12 09:58:48] Call-ID: 597e48ec5c79974032d5eda109c5477e@85.245.106.211:5060
[Jul 12 09:58:48] CSeq: 102 OPTIONS
[Jul 12 09:58:48] Accept: */*
[Jul 12 09:58:48] Accept-Encoding:
[Jul 12 09:58:48] Accept-Language: en
[Jul 12 09:58:48] Supported:
[Jul 12 09:58:48] Content-Length: 0
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by fbarrento » Tue Jul 12, 2016 4:23 am
[Jul 12 09:58:48]
[Jul 12 09:58:48] <------------->
[Jul 12 09:58:48] --- (11 headers 0 lines) ---
[Jul 12 09:58:48] Really destroying SIP dialog '597e48ec5c79974032d5eda109c5477e@85.245.106.211:5060' Method: OPTIONS
[Jul 12 09:58:49]
[Jul 12 09:58:49] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:49] OPTIONS sip:beslim@85.245.106.211:5060 SIP/2.0
[Jul 12 09:58:49] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0
[Jul 12 09:58:49] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-1290f936
[Jul 12 09:58:49] To: sip:beslim@85.245.106.211:5060
[Jul 12 09:58:49] Call-ID: 2a980ea4-a4007023-8f30d8@213.58.153.220
[Jul 12 09:58:49] CSeq: 1 OPTIONS
[Jul 12 09:58:49] Content-Length: 0
[Jul 12 09:58:49]
[Jul 12 09:58:49] <------------->
[Jul 12 09:58:49] --- (7 headers 0 lines) ---
[Jul 12 09:58:49] Sending to 213.58.153.220:5060 (NAT)
[Jul 12 09:58:49] Looking for beslim in trunkinbound (domain 85.245.106.211)
[Jul 12 09:58:49]
[Jul 12 09:58:49] <--- Transmitting (NAT) to 213.58.153.220:5060 --->
[Jul 12 09:58:49] SIP/2.0 404 Not Found
[Jul 12 09:58:49] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0;received=213.58.153.220;rport=5060
[Jul 12 09:58:49] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-1290f936
[Jul 12 09:58:49] To: sip:beslim@85.245.106.211:5060;tag=as19411e26
[Jul 12 09:58:49] Call-ID: 2a980ea4-a4007023-8f30d8@213.58.153.220
[Jul 12 09:58:49] CSeq: 1 OPTIONS
[Jul 12 09:58:49] Server: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:49] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:49] Supported: replaces, timer
[Jul 12 09:58:49] Accept: application/sdp
[Jul 12 09:58:49] Content-Length: 0
[Jul 12 09:58:49]
[Jul 12 09:58:49]
[Jul 12 09:58:49] <------------>
[Jul 12 09:58:49] Scheduling destruction of SIP dialog '2a980ea4-a4007023-8f30d8@213.58.153.220' in 32000 ms (Method: OPTIONS)
[Jul 12 09:58:51] Really destroying SIP dialog '2a980ea4-35ef6023-ad30d8@213.58.153.220' Method: OPTIONS
[Jul 12 09:58:48] <------------->
[Jul 12 09:58:48] --- (11 headers 0 lines) ---
[Jul 12 09:58:48] Really destroying SIP dialog '597e48ec5c79974032d5eda109c5477e@85.245.106.211:5060' Method: OPTIONS
[Jul 12 09:58:49]
[Jul 12 09:58:49] <--- SIP read from UDP:213.58.153.220:5060 --->
[Jul 12 09:58:49] OPTIONS sip:beslim@85.245.106.211:5060 SIP/2.0
[Jul 12 09:58:49] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0
[Jul 12 09:58:49] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-1290f936
[Jul 12 09:58:49] To: sip:beslim@85.245.106.211:5060
[Jul 12 09:58:49] Call-ID: 2a980ea4-a4007023-8f30d8@213.58.153.220
[Jul 12 09:58:49] CSeq: 1 OPTIONS
[Jul 12 09:58:49] Content-Length: 0
[Jul 12 09:58:49]
[Jul 12 09:58:49] <------------->
[Jul 12 09:58:49] --- (7 headers 0 lines) ---
[Jul 12 09:58:49] Sending to 213.58.153.220:5060 (NAT)
[Jul 12 09:58:49] Looking for beslim in trunkinbound (domain 85.245.106.211)
[Jul 12 09:58:49]
[Jul 12 09:58:49] <--- Transmitting (NAT) to 213.58.153.220:5060 --->
[Jul 12 09:58:49] SIP/2.0 404 Not Found
[Jul 12 09:58:49] Via: SIP/2.0/UDP 213.58.153.220:5060;branch=0;received=213.58.153.220;rport=5060
[Jul 12 09:58:49] From: sip:fscloud@finesource.pt;tag=uloc-577cf37a-3ac3-b73d-7e024c51-1290f936
[Jul 12 09:58:49] To: sip:beslim@85.245.106.211:5060;tag=as19411e26
[Jul 12 09:58:49] Call-ID: 2a980ea4-a4007023-8f30d8@213.58.153.220
[Jul 12 09:58:49] CSeq: 1 OPTIONS
[Jul 12 09:58:49] Server: Asterisk PBX 11.22.0-vici
[Jul 12 09:58:49] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jul 12 09:58:49] Supported: replaces, timer
[Jul 12 09:58:49] Accept: application/sdp
[Jul 12 09:58:49] Content-Length: 0
[Jul 12 09:58:49]
[Jul 12 09:58:49]
[Jul 12 09:58:49] <------------>
[Jul 12 09:58:49] Scheduling destruction of SIP dialog '2a980ea4-a4007023-8f30d8@213.58.153.220' in 32000 ms (Method: OPTIONS)
[Jul 12 09:58:51] Really destroying SIP dialog '2a980ea4-35ef6023-ad30d8@213.58.153.220' Method: OPTIONS
- fbarrento
- Posts: 9
- Joined: Wed Jul 15, 2009 3:53 pm
- Location: Lisboa
Re: Calls Not Passing to Agents
by Mequetref43 » Wed Jul 13, 2016 12:54 am
Good morning (here),
as we can see the call is maked but you have the same "horrible" debug with "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack" as i have.
So my suggestions are:
1.- can you direct your calls to an extension or phone directly to check if the SIP configuration it´s right?
2.- if point 1 works, must be something relationed with the agi (agi-VDAD_ALL_outbound.agi) of vicidial or maybe some installation issue.
I have the same issue with inbound calls to the agent and still trying to found the problem.
Hope that helps.
as we can see the call is maked but you have the same "horrible" debug with "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack" as i have.
So my suggestions are:
1.- can you direct your calls to an extension or phone directly to check if the SIP configuration it´s right?
2.- if point 1 works, must be something relationed with the agi (agi-VDAD_ALL_outbound.agi) of vicidial or maybe some installation issue.
I have the same issue with inbound calls to the agent and still trying to found the problem.
Hope that helps.
- Mequetref43
- Posts: 19
- Joined: Mon Jul 11, 2016 9:04 am
Re: Calls Not Passing to Agents
by williamconley » Tue Jul 19, 2016 7:08 pm
1) Good job posting your vicidial version with build. Please remember it's best to also post your installer full version (vicibox has had seven major releases ... which all have several minor releases).
2) Manual calls and autodialed calls can take different paths. Your Dial Prefix and Manual Dial prefix do not generate calls through the same path, and they also do not use the same methodology.
3) Manual calls work: But manual calls from the agent phone? Or from the agent Vicidial Screen after pushing the Manual Dial link?
4) verify the IP address of the server has not been changed since building it, and/or that if you have changed it you did so through the use of the admin ip update script which is shown during ssh console login. If you changed it with any other method, run that script now (one time for each previous address).
5) sip.conf "externip" value should be the public IP of the server
6) Admin->Servers -> (choose your server) ->Asterisk Version (verify that this matches the version actually running in the vicidial server).
2) Manual calls and autodialed calls can take different paths. Your Dial Prefix and Manual Dial prefix do not generate calls through the same path, and they also do not use the same methodology.
3) Manual calls work: But manual calls from the agent phone? Or from the agent Vicidial Screen after pushing the Manual Dial link?
4) verify the IP address of the server has not been changed since building it, and/or that if you have changed it you did so through the use of the admin ip update script which is shown during ssh console login. If you changed it with any other method, run that script now (one time for each previous address).
5) sip.conf "externip" value should be the public IP of the server
6) Admin->Servers -> (choose your server) ->Asterisk Version (verify that this matches the version actually running in the vicidial server).
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
- williamconley
- Posts: 20258
- Joined: Wed Oct 31, 2007 4:17 pm
- Location: Davenport, FL (By Disney!)
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