root@simlabvoip asterisk]# cd /usr/sbin [root@simlabvoip sbin]# ./asterisk -r Asterisk CVS-HEAD-01/19/05-22:31:00, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-01/19/05-22:31:00 currently running on simlabvoip (pid = 9903) Verbosity is at least 3 -- Executing AgentCallbackLogin("SIP/1299-55b4", "|1299@voicepulse-outgoing") in new stack -- Playing 'agent-user' (language 'en') -- Playing 'agent-pass' (language 'en') -- Setting global variable 'AGENTBYCALLERID_1299' to '226' -- Playing 'agent-loginok' (language 'en') == Callback Agent '226' logged in on 1299@voicepulse-outgoing -- Executing Hangup("SIP/1299-55b4", "") in new stack == Spawn extension (voicepulse-outgoing, 55, 2) exited non-zero on 'SIP/1299-55b4' -- Executing Hangup("SIP/1299-55b4", "") in new stack == Spawn extension (voicepulse-outgoing, h, 1) exited non-zero on 'SIP/1299-55b4' Jan 19 22:36:35 NOTICE[9903]: chan_sip.c:7020 handle_response: Peer '4004' is now TOO LAGGED! (1144ms / 1000ms) -- Executing Answer("SIP/69.60.198.131-081915c0", "") in new stack -- Executing Goto("SIP/69.60.198.131-081915c0", "techsupport|s|1") in new stack -- Goto (techsupport,s,1) -- Executing Wait("SIP/69.60.198.131-081915c0", "1") in new stack Jan 19 22:36:46 NOTICE[9903]: chan_sip.c:7014 handle_response: Peer '4004' is now REACHABLE! (134ms / 1000ms) -- Executing DigitTimeout("SIP/69.60.198.131-081915c0", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("SIP/69.60.198.131-081915c0", "20") in new stack -- Set Response Timeout to 20 -- Executing BackGround("SIP/69.60.198.131-081915c0", "techmain") in new stack -- Playing 'techmain' (language 'en') == CDR updated on SIP/69.60.198.131-081915c0 -- Executing Answer("SIP/69.60.198.131-081915c0", "") in new stack -- Executing Wait("SIP/69.60.198.131-081915c0", "1") in new stack -- Executing Queue("SIP/69.60.198.131-081915c0", "techsupportq1") in new stack -- Started music on hold, class 'random', on SIP/69.60.198.131-081915c0 -- outgoing agentcall, to agent '226', on 'Local/1299@voicepulse-outgoing-d70c,1' -- Executing Dial("Local/1299@voicepulse-outgoing-d70c,2", "SIP/1299") in new stack -- Called 1299 -- Called Agent/226 -- SIP/1299-15b4 is ringing -- Agent/226 is ringing -- Nobody picked up in 20000 ms == Spawn extension (voicepulse-outgoing, 1299, 1) exited non-zero on 'Local/1299@voicepulse-outgoing-d70c,2' -- Executing Hangup("Local/1299@voicepulse-outgoing-d70c,2", "") in new stack == Spawn extension (voicepulse-outgoing, h, 1) exited non-zero on 'Local/1299@voicepulse-outgoing-d70c,2' -- Stopped music on hold on SIP/69.60.198.131-081915c0 -- Playing 'queue-youarenext' (language 'en') -- Told SIP/69.60.198.131-081915c0 in techsupportq1 their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') -- Started music on hold, class 'random', on SIP/69.60.198.131-081915c0 -- outgoing agentcall, to agent '226', on 'Local/1299@voicepulse-outgoing-9217,1' -- Executing Dial("Local/1299@voicepulse-outgoing-9217,2", "SIP/1299") in new stack -- Called 1299 -- Called Agent/226 -- SIP/1299-8448 is ringing -- Agent/226 is ringing -- SIP/1299-8448 answered Local/1299@voicepulse-outgoing-9217,2 -- Agent/226 answered SIP/69.60.198.131-081915c0 -- Stopped music on hold on SIP/69.60.198.131-081915c0 == Spawn extension (voicepulse-outgoing, 1299, 1) exited non-zero on 'Local/1299@voicepulse-outgoing-9217,2' -- Executing Hangup("Local/1299@voicepulse-outgoing-9217,2", "") in new stack == Spawn extension (voicepulse-outgoing, h, 1) exited non-zero on 'Local/1299@voicepulse-outgoing-9217,2' -- SIP/1299-8448 acknowledged simlabvoip*CLI> sip debug peer 1299 SIP Debugging Enabled for IP: 216.86.54.250:33968 simlabvoip*CLI> Sip read: INVITE sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK3868999b From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:21 GMT CSeq: 108 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 249 Remote-Party-ID: "1299" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 17997 13747 IN IP4 216.86.54.250 s=SIP Call c=IN IP4 216.86.54.250 t=0 0 m=audio 24866 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 216.86.54.250 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.86.54.250:24866 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 69.60.198.130 port 16186 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK3868999b;received=216.86.54.250;rport=33968 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 CSeq: 108 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 9903 9911 IN IP4 69.60.198.130 s=session c=IN IP4 69.60.198.130 t=0 0 m=audio 16186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 216.86.54.250:33968 simlabvoip*CLI> Sip read: ACK sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK50587ba5 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:22 GMT CSeq: 108 ACK User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines Destroying call '000ded9c-1ae90002-23b09a93-1d6092f9@192.168.100.244' simlabvoip*CLI> Sip read: INVITE sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK264c5125 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:24 GMT CSeq: 109 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 241 Remote-Party-ID: "1299" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 8067 8562 IN IP4 216.86.54.250 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 24866 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 216.86.54.250 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:24866 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 69.60.198.130 port 16186 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK264c5125;received=216.86.54.250;rport=33968 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 CSeq: 109 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 9903 9912 IN IP4 69.60.198.130 s=session c=IN IP4 69.60.198.130 t=0 0 m=audio 16186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 216.86.54.250:33968 simlabvoip*CLI> Sip read: ACK sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK5b0bec06 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:25 GMT CSeq: 109 ACK User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines simlabvoip*CLI> Sip read: INVITE sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK6e0bc238 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:27 GMT CSeq: 110 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 249 Remote-Party-ID: "1299" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 17157 25474 IN IP4 216.86.54.250 s=SIP Call c=IN IP4 216.86.54.250 t=0 0 m=audio 24866 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 216.86.54.250 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.86.54.250:24866 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 69.60.198.130 port 16186 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK6e0bc238;received=216.86.54.250;rport=33968 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 CSeq: 110 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 9903 9913 IN IP4 69.60.198.130 s=session c=IN IP4 69.60.198.130 t=0 0 m=audio 16186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 216.86.54.250:33968 simlabvoip*CLI> Sip read: ACK sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK2f985907 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:27 GMT CSeq: 110 ACK User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines simlabvoip*CLI> Sip read: INVITE sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK23eda762 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:33 GMT CSeq: 111 INVITE User-Agent: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 243 Remote-Party-ID: "1299" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 27423 11523 IN IP4 216.86.54.250 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 24866 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to 216.86.54.250 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:24866 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 69.60.198.130 port 16186 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK23eda762;received=216.86.54.250;rport=33968 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 CSeq: 111 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 9903 9914 IN IP4 69.60.198.130 s=session c=IN IP4 69.60.198.130 t=0 0 m=audio 16186 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 216.86.54.250:33968 simlabvoip*CLI> Sip read: ACK sip:9733901090@69.60.198.130:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.54.250:5060;branch=z9hG4bK383b92b0 From: ;tag=000ded9c1ae94bad7ef49671-53c11830 To: "9733901090" ;tag=as6c547f2b Call-ID: 197d0ab715c5ba2570f89a066512f341@69.60.198.130 Date: Thu, 20 Jan 2005 03:39:33 GMT CSeq: 111 ACK User-Agent: CSCO/7 Content-Length: 0 9 headers, 0 lines simlabvoip*CLI> sip no debug SIP Debugging Disabled == Spawn extension (techsupport, 1, 3) exited non-zero on 'SIP/69.60.198.131-081915c0' simlabvoip*CLI> show agents 226 (johnb) available at '1299@voicepulse-outgoing' (musiconhold is 'random')