pbx1*CLI> == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found unknown media description format X-NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.1.7:18118 Looking for 2155555757 in inbound (domain 192.168.1.10) list_route: hop: <--- Transmitting (no NAT) to 192.168.1.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK9138D;received=192.168.1.7 From: ;tag=1A38E9-1329 To: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 101 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [2155555757@inbound:1] Set("SIP/vgw1-000007bc", "DNIS=2155555757") in new stack -- Executing [2155555757@inbound:2] Set("SIP/vgw1-000007bc", "CALLERID(num)=9910") in new stack -- Executing [2155555757@inbound:3] GotoIf("SIP/vgw1-000007bc", "0?override") in new stack -- Executing [2155555757@inbound:4] Set("SIP/vgw1-000007bc", "CALLERID(name)=") in new stack -- Executing [2155555757@inbound:5] Goto("SIP/vgw1-000007bc", "start") in new stack -- Goto (inbound,2155555757,7) -- Executing [2155555757@inbound:7] Goto("SIP/vgw1-000007bc", "attendant,s,1") in new stack -- Goto (attendant,s,1) -- Executing [s@attendant:1] Answer("SIP/vgw1-000007bc", "") in new stack Audio is at 28040 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK9138D;received=192.168.1.7 From: ;tag=1A38E9-1329 To: ;tag=as2b22faf8 Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 101 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 235 v=0 o=root 554503106 554503106 IN IP4 192.168.1.10 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 28040 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> -- Executing [s@attendant:2] Set("SIP/vgw1-000007bc", "TIMEOUT(absolute)=14400") in new stack Channel will hangup at 2012-08-17 05:23:14.668 EDT. -- Executing [s@attendant:3] GotoIf("SIP/vgw1-000007bc", "0?blacklist,1") in new stack -- Executing [s@attendant:4] Playback("SIP/vgw1-000007bc", "local/silence") in new stack -- Playing 'local/silence.gsm' (language 'en') -- Executing [s@attendant:5] GotoIf("SIP/vgw1-000007bc", "0?open,1") in new stack -- Executing [s@attendant:6] Set("SIP/vgw1-000007bc", "HOLIDAY=") in new stack -- Executing [s@attendant:7] GotoIf("SIP/vgw1-000007bc", "0?holiday,1") in new stack -- Executing [s@attendant:8] GotoIfTime("SIP/vgw1-000007bc", "09:00-17:30,mon-wed,*,*?open,1") in new stack -- Executing [s@attendant:9] GotoIfTime("SIP/vgw1-000007bc", "09:00-20:00,thu-fri,*,*?open,1") in new stack -- Executing [s@attendant:10] GotoIfTime("SIP/vgw1-000007bc", "10:00-16:00,sat,*,*?open,1") in new stack -- Executing [s@attendant:11] Set("SIP/vgw1-000007bc", "MSG=open") in new stack -- Executing [s@attendant:12] Goto("SIP/vgw1-000007bc", "main,1") in new stack -- Goto (attendant,main,1) -- Executing [main@attendant:1] BackGround("SIP/vgw1-000007bc", "local/open") in new stack -- Playing 'local/open.ulaw' (language 'en') [Aug 17 01:23:17] DTMF[27479]: channel.c:4090 __ast_read: DTMF begin '1' received on SIP/vgw1-000007bc [Aug 17 01:23:17] DTMF[27479]: channel.c:4094 __ast_read: DTMF begin ignored '1' on SIP/vgw1-000007bc [Aug 17 01:23:17] DTMF[27479]: channel.c:4005 __ast_read: DTMF end '1' received on SIP/vgw1-000007bc, duration 65 ms [Aug 17 01:23:17] DTMF[27479]: channel.c:4074 __ast_read: DTMF end passthrough '1' on SIP/vgw1-000007bc [Aug 17 01:23:18] DTMF[27479]: channel.c:4090 __ast_read: DTMF begin '0' received on SIP/vgw1-000007bc [Aug 17 01:23:18] DTMF[27479]: channel.c:4094 __ast_read: DTMF begin ignored '0' on SIP/vgw1-000007bc [Aug 17 01:23:18] DTMF[27479]: channel.c:4005 __ast_read: DTMF end '0' received on SIP/vgw1-000007bc, duration 53 ms [Aug 17 01:23:18] DTMF[27479]: channel.c:4074 __ast_read: DTMF end passthrough '0' on SIP/vgw1-000007bc [Aug 17 01:23:19] DTMF[27479]: channel.c:4090 __ast_read: DTMF begin '1' received on SIP/vgw1-000007bc [Aug 17 01:23:19] DTMF[27479]: channel.c:4094 __ast_read: DTMF begin ignored '1' on SIP/vgw1-000007bc [Aug 17 01:23:19] DTMF[27479]: channel.c:4005 __ast_read: DTMF end '1' received on SIP/vgw1-000007bc, duration 55 ms [Aug 17 01:23:19] DTMF[27479]: channel.c:4074 __ast_read: DTMF end passthrough '1' on SIP/vgw1-000007bc == CDR updated on SIP/vgw1-000007bc -- Executing [101@attendant:1] Dial("SIP/vgw1-000007bc", "SIP/101,30,mk") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/101 -- Started music on hold, class 'default', on SIP/vgw1-000007bc set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Audio is at 28040 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.7:5060: INVITE sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7d00f62e Max-Forwards: 70 From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Contact: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "Kate" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 235 v=0 o=root 554503106 554503107 IN IP4 192.168.1.10 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 28040 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Reliably Transmitting (no NAT) to 192.168.1.7:5060: UPDATE sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK20e10b3e Max-Forwards: 70 From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Contact: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 103 UPDATE User-Agent: Asterisk PBX 1.8.15.0 Remote-Party-ID: "Kate" ;party=calling;privacy=off;screen=no X-Asterisk-rpid-update: Yes Content-Length: 0 --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7d00f62e From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Date: Fri, 17 Aug 2012 05:23:19 GMT Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK7d00f62e From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Date: Fri, 17 Aug 2012 05:23:19 GMT Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=CiscoSystemsSIP-GW-UserAgent 8578 5784 IN IP4 192.168.1.7 s=SIP Call c=IN IP4 192.168.1.7 t=0 0 m=audio 18118 RTP/AVP 0 101 c=IN IP4 192.168.1.7 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Transmitting (no NAT) to 192.168.1.7:5060: ACK sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK74a7cf17 Max-Forwards: 70 From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Contact: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- Retransmitting #1 (no NAT) to 192.168.1.7:5060: UPDATE sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK20e10b3e Max-Forwards: 70 From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Contact: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 103 UPDATE User-Agent: Asterisk PBX 1.8.15.0 Remote-Party-ID: "Kate" ;party=calling;privacy=off;screen=no X-Asterisk-rpid-update: Yes Content-Length: 0 --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK20e10b3e From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 103 UPDATE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- -- Got SIP response 500 "Internal Server Error" back from 192.168.1.7:5060 -- SIP/101-000007bd is ringing pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> == Using SIP RTP CoS mark 5 [Aug 17 01:23:28] NOTICE[27484]: features.c:7270 ast_pickup_call: pickup SIP/101-000007bd attempt by SIP/139-000007be -- SIP/139-000007be answered SIP/vgw1-000007bc -- Stopped music on hold on SIP/vgw1-000007bc set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Audio is at 28040 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.7:5060: INVITE sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4de56a8b Max-Forwards: 70 From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Contact: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "testing" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 235 v=0 o=root 554503106 554503108 IN IP4 192.168.1.10 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 28040 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4de56a8b From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Date: Fri, 17 Aug 2012 05:23:29 GMT Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4de56a8b From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Date: Fri, 17 Aug 2012 05:23:29 GMT Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=CiscoSystemsSIP-GW-UserAgent 8578 5784 IN IP4 192.168.1.7 s=SIP Call c=IN IP4 192.168.1.7 t=0 0 m=audio 18118 RTP/AVP 0 101 c=IN IP4 192.168.1.7 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK4de56a8b From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Date: Fri, 17 Aug 2012 05:23:29 GMT Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 Server: Cisco-SIPGateway/IOS-12.x CSeq: 104 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=CiscoSystemsSIP-GW-UserAgent 8578 5784 IN IP4 192.168.1.7 s=SIP Call c=IN IP4 192.168.1.7 t=0 0 m=audio 18118 RTP/AVP 0 101 c=IN IP4 192.168.1.7 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (15 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Transmitting (no NAT) to 192.168.1.7:5060: ACK sip:9910@192.168.1.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK33979b5a Max-Forwards: 70 From: ;tag=as2b22faf8 To: ;tag=1A38E9-1329 Contact: Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> # call is connected! No such command '# call is connected!' (type 'core show help # call' for other possible commands) pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> pbx1*CLI> Sending to 192.168.1.7:5060 (no NAT) Scheduling destruction of SIP dialog '7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bKB1DD8;received=192.168.1.7 From: ;tag=1A38E9-1329 To: ;tag=as2b22faf8 Call-ID: 7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7 CSeq: 102 BYE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> -- Executing [h@attendant:1] NoOp("SIP/vgw1-000007bc", """ <9910> hung up") in new stack == Spawn extension (attendant, 101, 1) exited non-zero on 'SIP/vgw1-000007bc' Really destroying SIP dialog '7AB21DFA-E76211E1-801AE0FD-8B13B9A1@192.168.1.7' Method: BYE Reliably Transmitting (no NAT) to 192.168.1.7:5060: OPTIONS sip:192.168.1.7 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3e0ba2ce Max-Forwards: 70 From: "asterisk" ;tag=as61e16c59 To: Contact: Call-ID: 38c0c52b760abb20318d719e1097c87d@192.168.1.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Fri, 17 Aug 2012 05:24:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 --- <--- SIP read from UDP:192.168.1.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3e0ba2ce From: "asterisk" ;tag=as61e16c59 To: ;tag=1B16BE-262C Date: Fri, 17 Aug 2012 05:24:11 GMT Call-ID: 38c0c52b760abb20318d719e1097c87d@192.168.1.10:5060 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS Supported: 100rel,replaces Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Accept: application/sdp Allow-Events: telephone-event Content-Length: 443 Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 6584 7727 IN IP4 192.168.1.7 s=SIP Call c=IN IP4 192.168.1.7 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 c=IN IP4 192.168.1.7 m=image 0 udptl t38 c=IN IP4 192.168.1.7 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (14 headers 18 lines) --- Really destroying SIP dialog '38c0c52b760abb20318d719e1097c87d@192.168.1.10:5060' Method: OPTIONS