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 192.168.1.7:18912 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=z9hG4bK16C53;received=192.168.1.7 From: ;tag=447C29-1B8B To: Call-ID: ED33B20D-E76811E1-803AE0FD-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/vgw1-0000000c", "DNIS=2155555757") in new stack -- Executing [2155555757@inbound:2] Set("SIP/vgw1-0000000c", "CALLERID(num)=9910") in new stack -- Executing [2155555757@inbound:3] GotoIf("SIP/vgw1-0000000c", "0?override") in new stack -- Executing [2155555757@inbound:4] Set("SIP/vgw1-0000000c", "CALLERID(name)=") in new stack -- Executing [2155555757@inbound:5] Goto("SIP/vgw1-0000000c", "start") in new stack -- Goto (inbound,2155555757,7) -- Executing [2155555757@inbound:7] Goto("SIP/vgw1-0000000c", "attendant,s,1") in new stack -- Goto (attendant,s,1) -- Executing [s@attendant:1] Answer("SIP/vgw1-0000000c", "") in new stack Audio is at 28008 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=z9hG4bK16C53;received=192.168.1.7 From: ;tag=447C29-1B8B To: ;tag=as49eb19d6 Call-ID: ED33B20D-E76811E1-803AE0FD-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: 237 v=0 o=root 1258888007 1258888007 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 28008 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/vgw1-0000000c", "TIMEOUT(absolute)=14400") in new stack Channel will hangup at 2012-08-17 06:09:26.075 EDT. -- Executing [s@attendant:3] GotoIf("SIP/vgw1-0000000c", "0?blacklist,1") in new stack -- Executing [s@attendant:4] Playback("SIP/vgw1-0000000c", "local/silence") in new stack -- Playing 'local/silence.gsm' (language 'en') -- Executing [s@attendant:5] GotoIf("SIP/vgw1-0000000c", "0?open,1") in new stack -- Executing [s@attendant:6] Set("SIP/vgw1-0000000c", "HOLIDAY=") in new stack -- Executing [s@attendant:7] GotoIf("SIP/vgw1-0000000c", "0?holiday,1") in new stack -- Executing [s@attendant:8] GotoIfTime("SIP/vgw1-0000000c", "09:00-17:30,mon-wed,*,*?open,1") in new stack -- Executing [s@attendant:9] GotoIfTime("SIP/vgw1-0000000c", "09:00-20:00,thu-fri,*,*?open,1") in new stack -- Executing [s@attendant:10] GotoIfTime("SIP/vgw1-0000000c", "10:00-16:00,sat,*,*?open,1") in new stack -- Executing [s@attendant:11] Set("SIP/vgw1-0000000c", "MSG=open") in new stack -- Executing [s@attendant:12] Goto("SIP/vgw1-0000000c", "main,1") in new stack -- Goto (attendant,main,1) -- Executing [main@attendant:1] BackGround("SIP/vgw1-0000000c", "local/open") in new stack -- Playing 'local/open.ulaw' (language 'en') [Aug 17 02:09:31] DTMF[27726]: channel.c:4090 __ast_read: DTMF begin '1' received on SIP/vgw1-0000000c [Aug 17 02:09:31] DTMF[27726]: channel.c:4094 __ast_read: DTMF begin ignored '1' on SIP/vgw1-0000000c [Aug 17 02:09:31] DTMF[27726]: channel.c:4005 __ast_read: DTMF end '1' received on SIP/vgw1-0000000c, duration 50 ms [Aug 17 02:09:31] DTMF[27726]: channel.c:4074 __ast_read: DTMF end passthrough '1' on SIP/vgw1-0000000c == CDR updated on SIP/vgw1-0000000c -- Executing [1@attendant:1] GotoIf("SIP/vgw1-0000000c", "0?vm") in new stack -- Executing [1@attendant:2] Dial("SIP/vgw1-0000000c", "SIP/111&SIP/112&SIP/113&SIP/114,11,mk") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/111 == Using SIP RTP CoS mark 5 -- Called SIP/112 == Using SIP RTP CoS mark 5 -- Called SIP/113 == Using SIP RTP CoS mark 5 -- Called SIP/114 -- Started music on hold, class 'default', on SIP/vgw1-0000000c -- SIP/111-0000000d connected line has changed. Saving it until answer for SIP/vgw1-0000000c -- SIP/112-0000000e connected line has changed. Saving it until answer for SIP/vgw1-0000000c -- SIP/113-0000000f connected line has changed. Saving it until answer for SIP/vgw1-0000000c -- SIP/114-00000010 connected line has changed. Saving it until answer for SIP/vgw1-0000000c -- SIP/112-0000000e is ringing -- SIP/111-0000000d is ringing -- SIP/114-00000010 is ringing -- SIP/113-0000000f is ringing pbx1*CLI> pbx1*CLI> == Using SIP RTP CoS mark 5 [Aug 17 02:09:45] NOTICE[27731]: features.c:7270 ast_pickup_call: pickup SIP/112-0000000e attempt by SIP/139-00000011 -- SIP/139-00000011 answered SIP/vgw1-0000000c set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.7:5060 Audio is at 28008 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=z9hG4bK07c0d0a0 Max-Forwards: 70 From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Contact: Call-ID: ED33B20D-E76811E1-803AE0FD-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: "Sales Desk 2" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 237 v=0 o=root 1258888007 1258888008 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 28008 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 --- -- Stopped music on hold on SIP/vgw1-0000000c 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=z9hG4bK71f2e71d Max-Forwards: 70 From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Contact: Call-ID: ED33B20D-E76811E1-803AE0FD-8B13B9A1@192.168.1.7 CSeq: 103 UPDATE User-Agent: Asterisk PBX 1.8.15.0 Remote-Party-ID: "testing" ;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=z9hG4bK07c0d0a0 From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Date: Fri, 17 Aug 2012 06:09:43 GMT Call-ID: ED33B20D-E76811E1-803AE0FD-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=z9hG4bK07c0d0a0 From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Date: Fri, 17 Aug 2012 06:09:43 GMT Call-ID: ED33B20D-E76811E1-803AE0FD-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 8493 4653 IN IP4 192.168.1.7 s=SIP Call c=IN IP4 192.168.1.7 t=0 0 m=audio 18912 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=z9hG4bK66eec0dc Max-Forwards: 70 From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Contact: Call-ID: ED33B20D-E76811E1-803AE0FD-8B13B9A1@192.168.1.7 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 --- -- Playing 'beep.ulaw' (language 'en') 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=z9hG4bK71f2e71d Max-Forwards: 70 From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Contact: Call-ID: ED33B20D-E76811E1-803AE0FD-8B13B9A1@192.168.1.7 CSeq: 103 UPDATE User-Agent: Asterisk PBX 1.8.15.0 Remote-Party-ID: "testing" ;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=z9hG4bK71f2e71d From: ;tag=as49eb19d6 To: ;tag=447C29-1B8B Call-ID: ED33B20D-E76811E1-803AE0FD-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 -- Executing [h@attendant:1] NoOp("SIP/vgw1-0000000c", """ <9910> hung up") in new stack == Spawn extension (attendant, 1, 2) exited non-zero on 'SIP/vgw1-0000000c' Really destroying SIP dialog 'ED33B20D-E76811E1-803AE0FD-8B13B9A1@192.168.1.7' Method: ACK