MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> MJLM-Hou-AsteriVOX-00*CLI> sip set debug on SIP Debugging enabled*CLI> 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=z9hG4bK2a87bcc2DDEE33BB From: "Jeff" ;tag=999A693F-68D2B440 To: CSeq: 471 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="132ff7e2", uri="sip:192.168.100.180:5060", response="22c106829d505800648c6ef92387abf7", 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=z9hG4bK77ba7407;rport Max-Forwards: 70 From: "asterisk" ;tag=as0f82efad To: Contact: Call-ID: 092890f9694807d6028ffc2e544995df@192.168.100.180 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Tue, 07 Apr 2009 21:40:04 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=z9hG4bK2a87bcc2DDEE33BB;received=192.168.101.250 From: "Jeff" ;tag=999A693F-68D2B440 To: ;tag=as63da68e5 Call-ID: 1b97ffac-c6830435-66db1ca6@192.168.101.250 CSeq: 471 REGISTER Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Tue, 07 Apr 2009 21:40:04 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=z9hG4bK77ba7407;rport From: "asterisk" ;tag=as0f82efad To: ;tag=AA51A625-DF2B6416 CSeq: 102 OPTIONS Call-ID: 092890f9694807d6028ffc2e544995df@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 '092890f9694807d6028ffc2e544995df@192.168.100.180' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:192.168.200.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK24836fd0;rport Max-Forwards: 70 From: "asterisk" ;tag=as33a82e5b To: Contact: Call-ID: 1b58b1f87c6b6c7e19d96b2f60793013@192.168.100.180 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Tue, 07 Apr 2009 21:40:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.11:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK24836fd0;rport From: "asterisk" ;tag=as33a82e5b To: Call-ID: 1b58b1f87c6b6c7e19d96b2f60793013@192.168.100.180 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '1b58b1f87c6b6c7e19d96b2f60793013@192.168.100.180' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.10:5060: OPTIONS sip:192.168.200.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK7b55a896;rport Max-Forwards: 70 From: "asterisk" ;tag=as200823fe To: Contact: Call-ID: 7017df686fd805ef1a1e5a2a31996fa6@192.168.100.180 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Tue, 07 Apr 2009 21:40:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- MJLM-Hou-AsteriVOX-00*CLI> <--- SIP read from UDP://192.168.200.10:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 192.168.100.180:5060;branch=z9hG4bK7b55a896;rport From: "asterisk" ;tag=as200823fe To: Call-ID: 7017df686fd805ef1a1e5a2a31996fa6@192.168.100.180 CSeq: 102 OPTIONS Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog '7017df686fd805ef1a1e5a2a31996fa6@192.168.100.180' Method: OPTIONS 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=z9hG4bKe7e1cfc From: "Kevin Bailey" ;tag=33635667 To: Date: Tue, 07 Apr 2009 22:45:34 GMT Call-ID: ccd24080-1df141c8-6831-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: "Kevin Bailey" ;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 30896 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 - ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 Found peer 'callman02' for '1681' from 192.168.200.11:5060 Found RTP audio format 0I> Found RTP audio format 101 Peer audio RTP is at port 192.168.200.10:30896 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 192.168.200.10:30896 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=z9hG4bKe7e1cfc;received=192.168.200.11 From: "Kevin Bailey" ;tag=33635667 To: Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc4 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-b7a71198", "0?HangUp()") in new stack -- Executing [1688@incoming:2] Goto("SIP/192.168.200.11-b7a71198", "did,1688,1") in new stack -- Goto (did,1688,1)I> [Apr 7 16:40:05] NOTICE[25508]: /home/murf/asterisk/1.6.1/main/ast_expr2.y:760 compose_func_args: argbuf allocated 5 bytes; [Apr 7 16:40:05] NOTICE[25508]: /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-b7a71198", "CALLERID(name)=Kevin Bailey") in new stack -- Executing [1688@did:2] NoOp("SIP/192.168.200.11-b7a71198", "Kevin Bailey is calling 1688 from 1681") in new stack -- Executing [1688@did:3] NoOp("SIP/192.168.200.11-b7a71198", "1688 is currently NOT_INUSE") in new stack -- Executing [1688@did:4] Ringing("SIP/192.168.200.11-b7a71198", "") 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=z9hG4bKe7e1cfc;received=192.168.200.11 From: "Kevin Bailey" ;tag=33635667 To: ;tag=as68906636 Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc4 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-b7a71198", "3") in new stack -- Executing [1688@did:6] GotoIf("SIP/192.168.200.11-b7a71198", "1?30:") in new stack -- Goto (did,1688,30)> sip set debug off -- Executing [1688@did:30] Dial("SIP/192.168.200.11-b7a71198", "SIP/sip1688&SIP/688,17") in new stack == Using SIP RTP CoS mark 5p set debug off Audio is at 192.168.100.180 port 11426ug off Adding codec 0x4 (ulaw) to SDP set debug off 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=z9hG4bK1b50358d;rport Max-Forwards: 70 From: "Kevin Bailey" ;tag=as1200a382 To: Contact: Call-ID: 02da66c21f56a9911fa1bee2584b377e@192.168.100.180 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Tue, 07 Apr 2009 21:40:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 320 v=0 o=root 1783001294 1783001294 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.180 t=0 0 m=audio 11426 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 '6649aaef0267f49e680b5e4b0fb9c389@127.0.1.1' Method: INVITE [Apr 7 16:40:08] WARNING[25508]: app_dial.c:1518 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) MJLM-Hou-AsteriVOX-00*CLI> sip set debug off <--- 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=z9hG4bK1b50358d;rport From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 CSeq: 102 INVITE Call-ID: 02da66c21f56a9911fa1bee2584b377e@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) ---ip set debug off MJLM-Hou-AsteriVOX-00*CLI> sip set debug off <--- 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=z9hG4bK1b50358d;rport From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 CSeq: 102 INVITE Call-ID: 02da66c21f56a9911fa1bee2584b377e@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) ---ip set debug off -- SIP/sip1688-08ba0160 is ringing MJLM-Hou-AsteriVOX-00*CLI> sip set debug off <--- 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=z9hG4bKe7e1cfc;received=192.168.200.11 From: "Kevin Bailey" ;tag=33635667 To: ;tag=as68906636 Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc4 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 set debug off <--- 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=z9hG4bK1b50358d;rport From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 CSeq: 102 INVITE Call-ID: 02da66c21f56a9911fa1bee2584b377e@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=- 1239140388 1239140388 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 m=audio 2262 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> sip set debug off Found RTP audio format 101 Peer audio RTP is at port 192.168.101.250:2262 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 192.168.101.250:2262 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=z9hG4bK33b0ded0;rport Max-Forwards: 70 From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 Contact: Call-ID: 02da66c21f56a9911fa1bee2584b377e@192.168.100.180 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- -- SIP/sip1688-08ba0160 answered SIP/192.168.200.11-b7a71198 Audio is at 192.168.100.180 port 14076ug off Adding codec 0x4 (ulaw) to SDP set debug off Adding non-codec 0x1 (telephone-event) to SDP MJLM-Hou-AsteriVOX-00*CLI> sip set debug off <--- 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=z9hG4bKe7e1cfc;received=192.168.200.11 From: "Kevin Bailey" ;tag=33635667 To: ;tag=as68906636 Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.0-rc4 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 1717363595 1717363595 IN IP4 192.168.100.180 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.180 t=0 0 m=audio 14076 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 <------------> -- Native bridging SIP/192.168.200.11-b7a71198 and SIP/sip1688-08ba0160 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 11426ug off 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=z9hG4bK1b50358d;rport Max-Forwards: 70 From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 Contact: Call-ID: 02da66c21f56a9911fa1bee2584b377e@192.168.100.180 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 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 1783001294 1783001295 IN IP4 192.168.200.10 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.200.10 t=0 0 m=audio 30896 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 --- <--- 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=z9hG4bK743be876 From: "Kevin Bailey" ;tag=33635667 To: ;tag=as68906636 Date: Tue, 07 Apr 2009 22:45:34 GMT Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 <-------------> --- (9 headers 0 lines) ---sip set debug off 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 14076ug off 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:1681@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=as68906636 To: "Kevin Bailey" ;tag=33635667 Contact: Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 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 1717363595 1717363596 IN IP4 192.168.101.250 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.101.250 t=0 0 m=audio 2262 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 set debug off <--- 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=as68906636 To: "Kevin Bailey" ;tag=33635667 Date: Tue, 07 Apr 2009 22:45:39 GMT Call-ID: ccd24080-1df141c8-6831-bc8a8c0@192.168.200.11 CSeq: 102 INVITE Allow-Events: telephone-event Remote-Party-ID: "Kevin Bailey" ;party=called;screen=no;privacy=off Content-Length: 0 <-------------> --- (10 headers 0 lines) --- MJLM-Hou-AsteriVOX-00*CLI> sip set debug off <--- 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=z9hG4bK1b50358d;rport From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 CSeq: 103 INVITE Call-ID: 02da66c21f56a9911fa1bee2584b377e@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=- 1239140388 1239140389 IN IP4 192.168.101.250 s=Polycom IP Phone c=IN IP4 192.168.101.250 t=0 0 m=audio 2262 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (12 headers 9 lines) ---ip set debug off Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.101.250:2262 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 192.168.101.250:2262 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=z9hG4bK44ef7618;rport Max-Forwards: 70 From: "Kevin Bailey" ;tag=as1200a382 To: ;tag=97DF7AB0-577C61D9 Contact: Call-ID: 02da66c21f56a9911fa1bee2584b377e@192.168.100.180 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- MJLM-Hou-AsteriVOX-00*CLI> sip set debug off SIP Debugging DisabledCLI> == Spawn extension (did, 1688, 30) exited non-zero on 'SIP/192.168.200.11-b7a71198' MJLM-Hou-AsteriVOX-00*CLI>