SIP Debugging enabled*CLI> MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.11:5060 ---> INVITE sip:1688@192.168.100.180:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK7510ea2 From: ;tag=33662916 To: Date: Tue, 14 Apr 2009 21:50:33 GMT Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 Supported: timer Min-SE: 1800 User-Agent: Cisco-CCM4.1 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: ;party=calling;screen=no;privacy=off Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 231 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.200.11 s=SIP Call c=IN IP4 192.168.200.10 t=0 0 m=audio 25952 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (18 headers 11 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.200.11 : 5060 (no NAT) Using INVITE request as basis request - 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 Found peer 'callman02' for '2819002620' from 192.168.200.11:5060 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.200.10:25952 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), 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 192.168.200.10:25952 Looking for 1688 in incoming (domain 192.168.100.180) list_route: hop: MJLM-Hou-AsteriVOX-00*CLI> <--- Transmitting (no NAT) to 192.168.200.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK7510ea2;received=192.168.200.11 From: ;tag=33662916 To: Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> -- Executing [1688@incoming:1] ExecIf("SIP/192.168.200.11-08b00860", "0?HangUp()") in new stack -- Executing [1688@incoming:2] Goto("SIP/192.168.200.11-08b00860", "did,1688,1") in new stack -- Goto (did,1688,1)I> [Apr 14 15:45:03] NOTICE[6018]: /home/murf/asterisk/1.6.1/main/ast_expr2.y:760 compose_func_args: argbuf allocated 5 bytes; [Apr 14 15:45:03] NOTICE[6018]: /home/murf/asterisk/1.6.1/main/ast_expr2.y:779 compose_func_args: argbuf uses 4 bytes; -- Executing [1688@did:1] Set("SIP/192.168.200.11-08b00860", "CALLERID(name)=Unavailable-2819002620") in new stack -- Executing [1688@did:2] NoOp("SIP/192.168.200.11-08b00860", "Unavailable-2819002620 is calling 1688 from 2819002620") in new stack -- Executing [1688@did:3] NoOp("SIP/192.168.200.11-08b00860", "1688 is currently NOT_INUSE") in new stack -- Executing [1688@did:4] Ringing("SIP/192.168.200.11-08b00860", "") in new stack MJLM-Hou-AsteriVOX-00*CLI> <--- Transmitting (no NAT) to 192.168.200.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK7510ea2;received=192.168.200.11 From: ;tag=33662916 To: ;tag=as588ee7ab Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> -- Executing [1688@did:5] Wait("SIP/192.168.200.11-08b00860", "3") in new stack -- Executing [1688@did:6] GotoIf("SIP/192.168.200.11-08b00860", "1?30:") in new stack -- Goto (did,1688,30)> -- Executing [1688@did:30] Dial("SIP/192.168.200.11-08b00860", "SIP/sip1688&SIP/688,17") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.100.180 port 11776 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.101.250:5060: INVITE sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42d06162;rport Max-Forwards: 70 From: "Unavailable-2819002620" ;tag=as5df1a173 To: Contact: Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc3 Date: Tue, 14 Apr 2009 20:45:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 318 v=0 o=root 977184381 977184381 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc3 c=IN IP4 192.168.100.180 t=0 0 m=audio 11776 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called sip1688 == Using SIP RTP CoS mark 5 Really destroying SIP dialog '35a37fe04318449a47006d145e2d22e8@127.0.1.1' Method: INVITE [Apr 14 15:45:06] WARNING[6018]: app_dial.c:1497 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) Reliably Transmitting (no NAT) to 192.168.101.250:5060: NOTIFY sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK5c712008;rport Max-Forwards: 70 From: ;tag=as57bf279e To: "Jeff" ;tag=73EF64AE-DC1F7A17 Contact: Call-ID: 49866c63-56952d14-afaa946d@192.168.101.250 CSeq: 109 NOTIFY User-Agent: Asterisk PBX 1.6.1.0-rc3 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 358
--- == Extension Changed 1688[did] new state Ringing for Notify User sip1688 MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42d06162;rport From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B CSeq: 102 INVITE Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Length: 0 <-------------> --- (10 headers 0 lines) --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK5c712008;rport From: ;tag=as57bf279e To: "Jeff" ;tag=73EF64AE-DC1F7A17 CSeq: 109 NOTIFY Call-ID: 49866c63-56952d14-afaa946d@192.168.101.250 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42d06162;rport From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B CSeq: 102 INVITE Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- SIP/sip1688-08b25e08 is ringing MJLM-Hou-AsteriVOX-00*CLI> <--- Transmitting (no NAT) to 192.168.200.11:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK7510ea2;received=192.168.200.11 From: ;tag=33662916 To: ;tag=as588ee7ab Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42d06162;rport From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B CSeq: 102 INVITE Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Type: application/sdp Content-Length: 205 v=0 o=- 1239741902 1239741902 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 m=audio 2254 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0I> Found RTP audio format 101 Peer audio RTP is at port 192.168.101.250:2254 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), 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 192.168.101.250:2254 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.101.250, port 5060 Transmitting (no NAT) to 192.168.101.250:5060: ACK sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK263cec6f;rport Max-Forwards: 70 From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B Contact: Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0-rc3 Content-Length: 0 --- -- SIP/sip1688-08b25e08 answered SIP/192.168.200.11-08b00860 Audio is at 192.168.100.180 port 19370 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP MJLM-Hou-AsteriVOX-00*CLI> <--- Reliably Transmitting (no NAT) to 192.168.200.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK7510ea2;received=192.168.200.11 From: ;tag=33662916 To: ;tag=as588ee7ab Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Require: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 273 v=0 o=root 1503947663 1503947663 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc3 c=IN IP4 192.168.100.180 t=0 0 m=audio 19370 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Reliably Transmitting (no NAT) to 192.168.101.250:5060: NOTIFY sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK3fabf800;rport Max-Forwards: 70 From: ;tag=as57bf279e To: "Jeff" ;tag=73EF64AE-DC1F7A17 Contact: Call-ID: 49866c63-56952d14-afaa946d@192.168.101.250 CSeq: 110 NOTIFY User-Agent: Asterisk PBX 1.6.1.0-rc3 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 358
--- -- Native bridging SIP/192.168.200.11-08b00860 and SIP/sip1688-08b25e08 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.101.250, port 5060 Audio is at 192.168.100.180 port 11776 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.101.250:5060: INVITE sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42d06162;rport Max-Forwards: 70 From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B Contact: Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 269 v=0 o=root 977184381 977184382 IN IP4 192.168.200.10 s=Asterisk PBX 1.6.1.0-rc3 c=IN IP4 192.168.200.10 t=0 0 m=audio 25952 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- == Extension Changed 1688[did] new state InUse for Notify User sip1688 <--- SIP read from UDP://192.168.200.11:5060 ---> ACK sip:1688@192.168.100.180:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK66a9cd52 From: ;tag=33662916 To: ;tag=as588ee7ab Date: Tue, 14 Apr 2009 21:50:33 GMT Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.11, port 5060 Audio is at 192.168.100.180 port 19370 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.200.11:5060: INVITE sip:2819002620@192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK00000000;rport Max-Forwards: 70 From: ;tag=as588ee7ab To: ;tag=33662916 Contact: Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc3 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 272 v=0 o=root 1503947663 1503947664 IN IP4 192.168.101.250 s=Asterisk PBX 1.6.1.0-rc3 c=IN IP4 192.168.101.250 t=0 0 m=audio 2254 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.11:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK00000000;rport From: ;tag=as588ee7ab To: ;tag=33662916 Date: Tue, 14 Apr 2009 21:50:37 GMT Call-ID: 4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11 CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK3fabf800;rport From: ;tag=as57bf279e To: "Jeff" ;tag=73EF64AE-DC1F7A17 CSeq: 110 NOTIFY Call-ID: 49866c63-56952d14-afaa946d@192.168.101.250 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK42d06162;rport From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B CSeq: 103 INVITE Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Type: application/sdp Content-Length: 205 v=0 o=- 1239741902 1239741903 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 m=audio 2254 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0I> Found RTP audio format 101 Peer audio RTP is at port 192.168.101.250:2254 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xe (gsm|ulaw|alaw), 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 192.168.101.250:2254 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.101.250, port 5060 Transmitting (no NAT) to 192.168.101.250:5060: ACK sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK2a38d37c;rport Max-Forwards: 70 From: "Unavailable-2819002620" ;tag=as5df1a173 To: ;tag=BAEF982-F5962C0B Contact: Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1.0-rc3 Content-Length: 0 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> BYE sip:2819002620@192.168.100.180 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bKd1bc96fc46490295 From: ;tag=BAEF982-F5962C0B To: "Unavailable-2819002620" ;tag=as5df1a173 CSeq: 1 BYE Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.101.250 : 5060 (no NAT) MJLM-Hou-AsteriVOX-00*CLI> <--- Transmitting (no NAT) to 192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bKd1bc96fc46490295;received=192.168.101.250 From: ;tag=BAEF982-F5962C0B To: "Unavailable-2819002620" ;tag=as5df1a173 Call-ID: 7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180 CSeq: 1 BYE Server: Asterisk PBX 1.6.1.0-rc3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> == Spawn extension (did, 1688, 30) exited non-zero on 'SIP/192.168.200.11-08b00860' Scheduling destruction of SIP dialog '4629e000-1df14741-8fd4-bc8a8c0@192.168.200.11' in 6400 ms (Method: ACK) Reliably Transmitting (no NAT) to 192.168.101.250:5060: NOTIFY sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK40fdd646;rport Max-Forwards: 70 From: ;tag=as57bf279e To: "Jeff" ;tag=73EF64AE-DC1F7A17 Contact: Call-ID: 49866c63-56952d14-afaa946d@192.168.101.250 CSeq: 111 NOTIFY User-Agent: Asterisk PBX 1.6.1.0-rc3 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- == Extension Changed 1688[did] new state Idle for Notify User sip1688 MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK40fdd646;rport From: ;tag=as57bf279e To: "Jeff" ;tag=73EF64AE-DC1F7A17 CSeq: 111 NOTIFY Call-ID: 49866c63-56952d14-afaa946d@192.168.101.250 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Really destroying SIP dialog '7430d2cd12e6db1f59e4dd8f0d11adcc@192.168.100.180' Method: BYE MJLM-Hou-AsteriVOX-00*CLI> sip set debug off SIP Debugging DisabledCLI> MJLM-Hou-AsteriVOX-00*CLI>