MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*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=z9hG4bK320064 From: ;tag=33635516 To: Date: Tue, 07 Apr 2009 22:26:53 GMT Call-ID: 30a73a00-1df141c6-681a-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 30868 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 ending to 192.168.200.11 : 5060 (no NAT) Using INVITE request as basis request - 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 Found peer 'callman02' for '2819002620' from 192.168.200.11:5060 Found RTP audio format 0I> Found RTP audio format 101 Peer audio RTP is at port 192.168.200.10:30868 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 192.168.200.10:30868 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=z9hG4bK320064;received=192.168.200.11 From: ;tag=33635516 To: Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1-rc1 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-b3c501c8", "0?HangUp()") in new stack MJLM-Hou-AsteriVOX-00*CLI> -- Executing [1688@incoming:2] Goto("SIP/192.168.200.11-b3c501c8", "did,1688,1") in new stack -- Goto (did,1688,1) [Apr 7 16:21:24] NOTICE[24421]: ast_expr2.y:703 compose_func_args: argbuf allocated 5 bytes; [Apr 7 16:21:24] NOTICE[24421]: ast_expr2.y:722 compose_func_args: argbuf uses 4 bytes; -- Executing [1688@did:1] Set("SIP/192.168.200.11-b3c501c8", "CALLERID(name)=Unavailable") in new stack -- Executing [1688@did:2] NoOp("SIP/192.168.200.11-b3c501c8", "Unavailable is calling 1688 from 2819002620") in new stack -- Executing [1688@did:3] NoOp("SIP/192.168.200.11-b3c501c8", "1688 is currently NOT_INUSE") in new stack -- Executing [1688@did:4] Ringing("SIP/192.168.200.11-b3c501c8", "") 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=z9hG4bK320064;received=192.168.200.11 From: ;tag=33635516 To: ;tag=as40eeb04a Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1-rc1 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-b3c501c8", "3") in new stack MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> -- Executing [1688@did:6] GotoIf("SIP/192.168.200.11-b3c501c8", "1?30:") in new stack -- Goto (did,1688,30)> -- Executing [1688@did:30] Dial("SIP/192.168.200.11-b3c501c8", "SIP/sip1688&SIP/688,17") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.100.180 port 16442 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=z9hG4bK4ab71c3e;rport Max-Forwards: 70 From: "Unavailable" ;tag=as366caacd To: Contact: Call-ID: 2e508a99712b8f925579dbb46664ad85@192.168.100.180 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1-rc1 Date: Tue, 07 Apr 2009 21:21:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 318 v=0 o=root 2075921121 2075921121 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1-rc1 c=IN IP4 192.168.100.180 t=0 0 m=audio 16442 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 '28ed65780fd959675ac528de1c2d9709@127.0.1.1' Method: INVITE [Apr 7 16:21:27] WARNING[24421]: app_dial.c:1534 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) 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=z9hG4bK4ab71c3e;rport From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 CSeq: 102 INVITE Call-ID: 2e508a99712b8f925579dbb46664ad85@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/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK4ab71c3e;rport From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 CSeq: 102 INVITE Call-ID: 2e508a99712b8f925579dbb46664ad85@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) --- 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=z9hG4bK6fd33f73;rport Max-Forwards: 70 From: ;tag=as5b1fbcc0 To: "Jeff" ;tag=C47994A9-574013A Contact: Call-ID: df846566-a742efff-5de72900@192.168.101.250 CSeq: 103 NOTIFY User-Agent: Asterisk PBX 1.6.1-rc1 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 358
--- == Extension Changed 1688[did] new state Ringing for Notify User sip1688 -- SIP/sip1688-093efee0 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=z9hG4bK320064;received=192.168.200.11 From: ;tag=33635516 To: ;tag=as40eeb04a Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1-rc1 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=z9hG4bK6fd33f73;rport From: ;tag=as5b1fbcc0 To: "Jeff" ;tag=C47994A9-574013A CSeq: 103 NOTIFY Call-ID: df846566-a742efff-5de72900@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=z9hG4bK4ab71c3e;rport From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 CSeq: 102 INVITE Call-ID: 2e508a99712b8f925579dbb46664ad85@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=- 1239139267 1239139267 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=z9hG4bK4b77e11e;rport Max-Forwards: 70 From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 Contact: Call-ID: 2e508a99712b8f925579dbb46664ad85@192.168.100.180 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1-rc1 Content-Length: 0 --- 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=z9hG4bK090b09ed;rport Max-Forwards: 70 From: ;tag=as5b1fbcc0 To: "Jeff" ;tag=C47994A9-574013A Contact: Call-ID: df846566-a742efff-5de72900@192.168.101.250 CSeq: 104 NOTIFY User-Agent: Asterisk PBX 1.6.1-rc1 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 358
--- == Extension Changed 1688[did] new state InUse for Notify User sip1688 -- SIP/sip1688-093efee0 answered SIP/192.168.200.11-b3c501c8 Audio is at 192.168.100.180 port 14092 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=z9hG4bK320064;received=192.168.200.11 From: ;tag=33635516 To: ;tag=as40eeb04a Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1-rc1 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: 271 v=0 o=root 1879162011 1879162011 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1-rc1 c=IN IP4 192.168.100.180 t=0 0 m=audio 14092 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 ---> ACK sip:1688@192.168.100.180:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK19f833f0 From: ;tag=33635516 To: ;tag=as40eeb04a Date: Tue, 07 Apr 2009 22:26:53 GMT Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- Native bridging SIP/192.168.200.11-b3c501c8 and SIP/sip1688-093efee0 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 14092 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=as40eeb04a To: ;tag=33635516 Contact: Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1-rc1 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: 270 v=0 o=root 1879162011 1879162012 IN IP4 192.168.101.250 s=Asterisk PBX 1.6.1-rc1 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 --- 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 16442 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=z9hG4bK4ab71c3e;rport Max-Forwards: 70 From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 Contact: Call-ID: 2e508a99712b8f925579dbb46664ad85@192.168.100.180 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1-rc1 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 2075921121 2075921122 IN IP4 192.168.200.10 s=Asterisk PBX 1.6.1-rc1 c=IN IP4 192.168.200.10 t=0 0 m=audio 30868 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=as40eeb04a To: ;tag=33635516 Date: Tue, 07 Apr 2009 22:26:58 GMT Call-ID: 30a73a00-1df141c6-681a-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://192.168.200.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK00000000;rport From: ;tag=as40eeb04a To: ;tag=33635516 Date: Tue, 07 Apr 2009 22:26:58 GMT Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 102 INVITE Session-Expires: 1800;refresher=uas Require: timer Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: 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 30868 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 <-------------> --- (15 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.11, port 5060 Transmitting (no NAT) to 192.168.200.11:5060: ACK sip:2819002620@192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK228e3287;rport Max-Forwards: 70 From: ;tag=as40eeb04a To: ;tag=33635516 Contact: Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1-rc1 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=z9hG4bK090b09ed;rport From: ;tag=as5b1fbcc0 To: "Jeff" ;tag=C47994A9-574013A CSeq: 104 NOTIFY Call-ID: df846566-a742efff-5de72900@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=z9hG4bK4ab71c3e;rport From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 CSeq: 103 INVITE Call-ID: 2e508a99712b8f925579dbb46664ad85@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=- 1239139267 1239139268 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=z9hG4bK0a559344;rport Max-Forwards: 70 From: "Unavailable" ;tag=as366caacd To: ;tag=B7C4778E-63127C67 Contact: Call-ID: 2e508a99712b8f925579dbb46664ad85@192.168.100.180 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1-rc1 Content-Length: 0 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.101.250:5060 ---> REGISTER sip:192.168.100.180:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.250;branch=z9hG4bKc164e9a8F2B25391 From: "Jeff" ;tag=999A693F-68D2B440 To: CSeq: 433 REGISTER Call-ID: 1b97ffac-c6830435-66db1ca6@192.168.101.250 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.2.0392 Proxy-Require: 0 Accept-Language: en Authorization: Digest username="sip1688", realm="asterisk", nonce="5668c4fc", uri="sip:192.168.100.180:5060", response="289d5b67a82aa963bf8c5f9ac8289aa0", algorithm=MD5 Max-Forwards: 70 Expires: 60 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Sending to 192.168.101.250 : 5060 (no NAT) Reliably Transmitting (no NAT) to 192.168.101.250:5060: OPTIONS sip:sip1688@192.168.101.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK1d03f34c;rport Max-Forwards: 70 From: "asterisk" ;tag=as0eca3653 To: Contact: Call-ID: 53e2bdba7cebfdf375c85772190cfa4d@192.168.100.180 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1-rc1 Date: Tue, 07 Apr 2009 21:21:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- 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=z9hG4bKc164e9a8F2B25391;received=192.168.101.250 From: "Jeff" ;tag=999A693F-68D2B440 To: ;tag=as29b32302 Call-ID: 1b97ffac-c6830435-66db1ca6@192.168.101.250 CSeq: 433 REGISTER Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 07 Apr 2009 21:21:30 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1b97ffac-c6830435-66db1ca6@192.168.101.250' in 32000 ms (Method: REGISTER) 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=z9hG4bK1d03f34c;rport From: "asterisk" ;tag=as0eca3653 To: ;tag=F0B3F05B-7050FDBC CSeq: 102 OPTIONS Call-ID: 53e2bdba7cebfdf375c85772190cfa4d@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 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Supported: 100rel,timer,replaces Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Really destroying SIP dialog '53e2bdba7cebfdf375c85772190cfa4d@192.168.100.180' Method: OPTIONS 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=z9hG4bK63fb5cc5CB0F33B6 From: ;tag=B7C4778E-63127C67 To: "Unavailable" ;tag=as366caacd CSeq: 1 BYE Call-ID: 2e508a99712b8f925579dbb46664ad85@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=z9hG4bK63fb5cc5CB0F33B6;received=192.168.101.250 From: ;tag=B7C4778E-63127C67 To: "Unavailable" ;tag=as366caacd Call-ID: 2e508a99712b8f925579dbb46664ad85@192.168.100.180 CSeq: 1 BYE Server: Asterisk PBX 1.6.1-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> 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 14092 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=as40eeb04a To: ;tag=33635516 Contact: Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1-rc1 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: 271 v=0 o=root 1879162011 1879162013 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1-rc1 c=IN IP4 192.168.100.180 t=0 0 m=audio 14092 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 --- == Spawn extension (did, 1688, 30) exited non-zero on 'SIP/192.168.200.11-b3c501c8' 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=as40eeb04a To: ;tag=33635516 Date: Tue, 07 Apr 2009 22:27:04 GMT Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 103 INVITE Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- 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=z9hG4bK264ef826;rport Max-Forwards: 70 From: ;tag=as5b1fbcc0 To: "Jeff" ;tag=C47994A9-574013A Contact: Call-ID: df846566-a742efff-5de72900@192.168.101.250 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.6.1-rc1 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- == Extension Changed 1688[did] new state Idle for Notify User sip1688 Scheduling destruction of SIP dialog '30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11' in 6400 ms (Method: ACK) Really destroying SIP dialog '2e508a99712b8f925579dbb46664ad85@192.168.100.180' Method: BYE MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK00000000;rport From: ;tag=as40eeb04a To: ;tag=33635516 Date: Tue, 07 Apr 2009 22:27:04 GMT Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 103 INVITE Session-Expires: 1800;refresher=uas Require: timer Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK Allow-Events: telephone-event Remote-Party-ID: ;party=called;screen=no;privacy=off Contact: 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 30868 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 <-------------> --- (15 headers 11 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.11, port 5060 Transmitting (no NAT) to 192.168.200.11:5060: ACK sip:2819002620@192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK12d90066;rport Max-Forwards: 70 From: ;tag=as40eeb04a To: ;tag=33635516 Contact: Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1-rc1 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.11, port 5060 Reliably Transmitting (no NAT) to 192.168.200.11:5060: BYE sip:2819002620@192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK45d68b3b;rport Max-Forwards: 70 From: ;tag=as40eeb04a To: ;tag=33635516 Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 CSeq: 104 BYE User-Agent: Asterisk PBX 1.6.1-rc1 Content-Length: 0 --- Scheduling destruction of SIP dialog '30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11' in 6400 ms (Method: ACK) MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.11:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK45d68b3b;rport From: ;tag=as40eeb04a To: ;tag=33635516 Date: Tue, 07 Apr 2009 22:27:04 GMT Call-ID: 30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11 Content-Length: 0 CSeq: 104 BYE <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '30a73a00-1df141c6-681a-bc8a8c0@192.168.200.11' Method: ACK 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=z9hG4bK264ef826;rport From: ;tag=as5b1fbcc0 To: "Jeff" ;tag=C47994A9-574013A CSeq: 105 NOTIFY Call-ID: df846566-a742efff-5de72900@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 set debug of No such command 'sip set debug of' (type 'help sip set debug' for other possible commands) MJLM-Hou-AsteriVOX-00*CLI> sip set debug off SIP Debugging DisabledCLI> MJLM-Hou-AsteriVOX-00*CLI>