asterisk*CLI> asterisk*CLI> asterisk*CLI> Destroying call 'h5XQokbconBWWbCI@192.168.8.22' asterisk== New H.323 Connection created. -- Setting up Call -- Call token: [ip$192.168.8.4:2088/31521] -- Calling party name: [4FXS] -- Calling party number: [20001] -- Called party name: [300] -- Called party number: [300] --Received SETUP message Allowed Codecs: Table: G.711-uLaw-64k <1> UserInput/hookflash <2> UserInput/RFC2833 <3> Set: 0: 0: G.711-uLaw-64k <1> 1: UserInput/hookflash <2> 2: UserInput/RFC2833 <3> asterisk=-= In OnAnswerCall for call 31521 - Progress Indicator: 0 - Inserting PI of 0 into ALERTING message -- Started logical channel: sending G.711-uLaw-64k -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.8.4 -- remotePort: 16384 -- ExternalIpAddress: 192.168.8.1 -- ExternalPort: 15138 -- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: asterisk-- Transmitting RFC2833 on payload 101 -- Executing Macro("H323/ip$192.168.8.4:2088/31521", "localextension|SIP/petew") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2088/31521", ""4FXS" <20001>") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2088/31521", "inhouse") in new stack -- Executing Dial("H323/ip$192.168.8.4:2088/31521", "SIP/petew|30|t") in new stack We're at 192.168.8.1 port 13460 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.0.70:5061: INVITE sip:petew@192.168.0.70:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5846cea3 From: "4FXS" ;tag=as097d9083 To: Contact: Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 08:43:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 13838 13838 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 13460 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called petew asterisk*CLI> <-- SIP read from 192.168.0.70:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5846cea3 From: "4FXS" ;tag=as097d9083 To: ;tag=2761691638 Contact: Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.70:5061: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5846cea3 From: "4FXS" ;tag=as097d9083 To: ;tag=2761691638 Contact: Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/petew-af5f is ringing Sending alerting asterisk*CLI> <-- SIP read from 192.168.0.44:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 f: ;tag=94709219808 CSeq: 464 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a0181f00005e3600000573 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a0181f00005e3600000573 From: ;tag=94709219808 To: ;tag=as65bac3f9 Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 464 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44' asterisk-- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 3 External RTP Session Starting RTP channel id 32 parameters: asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> <-- SIP read from 192.168.0.70:5061: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5846cea3 From: "4FXS" ;tag=as097d9083 To: ;tag=2761691638 Contact: Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1103m Content-Length: 196 v=0 o=petew 188307593 188310390 IN IP4 192.168.0.70 s=X-Lite c=IN IP4 192.168.0.70 t=0 0 m=audio 8000 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.70:8000 Found description format pcmu Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.70, port 5061 Transmitting (no NAT) to 192.168.0.70:5061: ACK sip:petew@192.168.0.70:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK3c5f6693 From: "4FXS" ;tag=as097d9083 To: ;tag=2761691638 Contact: Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/petew-af5f answered H323/ip$192.168.8.4:2088/31521 Answering call ip$192.168.8.4:2088/31521 =-= In OnConnectionEstablished for call 31521 -- Connection Established with "4FXS (20001, 4FXS-01e6f3) [192.168.8.4]" asterisk*CLI> <-- SIP read from 192.168.0.23:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 f: ;tag=832893712278 CSeq: 398 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a0180c00006962000004ce --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a0180c00006962000004ce From: ;tag=832893712278 To: ;tag=as77682957 Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 398 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23' asterisk*CLI> <-- SIP read from 192.168.0.41:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 f: ;tag=18047412510166 CSeq: 8875 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a0182300000cc5000047a3 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.41;branch=z9hG4bKc0a800290131c9b142a0182300000cc5000047a3 From: ;tag=18047412510166 To: ;tag=as7a78c98f Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8875 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41' Jun 3 11:43:39 WARNING[13838]: chan_h323.c:659 oh323_indicate: Don't know how to indicate condition 16 on ip$192.168.8.4:2088/31521 -- Started music on hold, class 'default', on H323/ip$192.168.8.4:2088/31521 -- Playing 'pbx-transfer' (language 'en') asterisk*CLI> <-- SIP read from 192.168.0.71:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 f: ;tag=980475842734 CSeq: 4740 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.71;rport;branch=z9hG4bKc0a800470131c9b142a0183600001ce90000251f --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.71:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.71;branch=z9hG4bKc0a800470131c9b142a0183600001ce90000251f From: ;tag=980475842734 To: ;tag=as4f98ab2e Call-ID: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 CSeq: 4740 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71' asterisk*CLI> <-- SIP read from 192.168.0.45:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 f: ;tag=913978128865 CSeq: 447 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a0182b00007c7f0000054a --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a0182b00007c7f0000054a From: ;tag=913978128865 To: ;tag=as59ddbc5c Call-ID: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 CSeq: 447 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45' -- Executing Macro("Local/201@inhouse-0203,2", "localextension|SIP/plamen") in new stack -- Executing NoOp("Local/201@inhouse-0203,2", "300") in new stack -- Executing NoOp("Local/201@inhouse-0203,2", "inhouse") in new stack -- Executing Dial("Local/201@inhouse-0203,2", "SIP/plamen|30|t") in new stack We're at 192.168.8.1 port 13500 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.0.23:5060: INVITE sip:plamen@192.168.0.23:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK10945b61 From: "300" ;tag=as38b2a857 To: Contact: Call-ID: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 08:43:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 13841 13841 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 13500 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called plamen asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 100 Trying l: 0 i: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 CSeq: 102 INVITE f: ;tag=as38b2a857 t: "Plamen";tag=833715724508 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK10945b61 --- (8 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 180 Ringing l: 0 m: i: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 CSeq: 102 INVITE f: ;tag=as38b2a857 t: "Plamen";tag=833715724508 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK10945b61 --- (9 headers 0 lines)--- -- SIP/plamen-9930 is ringing asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 200 OK l: 219 m: i: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 c: application/sdp CSeq: 102 INVITE f: ;tag=as38b2a857 t: "Plamen";tag=833715724508 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK10945b61 v=0 o=- 3326776980 3326776980 IN IP4 192.168.0.23 s=SJphone c=IN IP4 192.168.0.23 t=0 0 a=direction:active m=audio 49162 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (10 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.23:49162 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.23, port 5060 Transmitting (no NAT) to 192.168.0.23:5060: ACK sip:plamen@192.168.0.23:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK2c003625 From: "300" ;tag=as38b2a857 To: ;tag=833715724508 Contact: Call-ID: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/plamen-9930 answered Local/201@inhouse-0203,2 asterisk*CLI> <-- SIP read from 192.168.0.70:5061: BYE sip:20001@192.168.8.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.70:5061;rport;branch=z9hG4bKBB71A3B006D849A7B2E49EC9A75867AA From: ;tag=2761691638 To: "4FXS" ;tag=as097d9083 Contact: Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 43240 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.0.70 : 5061 (non-NAT) Transmitting (no NAT) to 192.168.0.70:5061: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.70:5061;branch=z9hG4bKBB71A3B006D849A7B2E49EC9A75867AA From: ;tag=2761691638 To: "4FXS" ;tag=as097d9083 Call-ID: 3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1 CSeq: 43240 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Stopped music on hold on H323/ip$192.168.8.4:2088/31521 -- Playing 'beep' (language 'en') asterisk*CLI> <-- SIP read from 192.168.0.44:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 f: ;tag=949092123764 CSeq: 465 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a01833000028bc00000575 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a01833000028bc00000575 From: ;tag=949092123764 To: ;tag=as326ae6ce Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 465 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44' Destroying call '3f524f1f6b2e607b2fbd28d0341f20e5@192.168.8.1' asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 f: ;tag=834895918125 CSeq: 399 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a0182000002012000004d2 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a0182000002012000004d2 From: ;tag=834895918125 To: ;tag=as6baf3a6c Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 399 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23' asterisk*CLI> <-- SIP read from 192.168.0.23:5060: BYE sip:300@192.168.8.1 SIP/2.0 l: 0 m: i: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 Max-Forwards: 70 CSeq: 1 BYE f: ;tag=833715724508 t: ;tag=as38b2a857 User-Agent: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a018220000754a000004d4 --- (10 headers 0 lines)--- Sending to 192.168.0.23 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a018220000754a000004d4 From: ;tag=833715724508 To: ;tag=as38b2a857 Call-ID: 6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.8.22:1720: REGISTER sip:192.168.8.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKkQhu4itK9QSWZ0vb Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=0VxhXnJ5mB1upJeC To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9631 REGISTER Contact: Expires: 60 Content-Length: 0 --- (11 headers 0 lines)--- Using latest request as basis request Sending to 192.168.8.22 : 1720 (non-NAT) Transmitting (no NAT) to 192.168.8.22:1720: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKkQhu4itK9QSWZ0vb From: "ted" ;tag=0VxhXnJ5mB1upJeC To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9631 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.8.22:1720: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKkQhu4itK9QSWZ0vb From: "ted" ;tag=0VxhXnJ5mB1upJeC To: "ted" ;tag=as3af62709 Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9631 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="68a6cc83" Content-Length: 0 --- Scheduling destruction of call 'h5XQokbconBWWbCI@192.168.8.22' in 15000 ms Destroying call '6fdb45b31d764a1106d6d6f30cd74473@192.168.8.1' asterisk*CLI> <-- SIP read from 192.168.8.22:1720: REGISTER sip:192.168.8.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKacyA1jepQzvQDxLV Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=2CwENy6OICLtYt1V To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9632 REGISTER Contact: Expires: 60 Authorization: Digest username="ted", realm="asterisk", nonce="68a6cc83", uri="sip:192.168.8.1", response="8bcddcae20e3595f1e2fe12f1318236f", algorithm=MD5 Content-Length: 0 --- (12 headers 0 lines)--- Using latest request as basis request Sending to 192.168.8.22 : 1720 (non-NAT) Transmitting (no NAT) to 192.168.8.22:1720: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKacyA1jepQzvQDxLV From: "ted" ;tag=2CwENy6OICLtYt1V To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9632 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.8.22:1720: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKacyA1jepQzvQDxLV From: "ted" ;tag=2CwENy6OICLtYt1V To: "ted" ;tag=as3af62709 Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9632 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 60 Contact: ;expires=60 Date: Fri, 03 Jun 2005 08:43:58 GMT Content-Length: 0 --- Scheduling destruction of call 'h5XQokbconBWWbCI@192.168.8.22' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.41:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 f: ;tag=1804941257892 CSeq: 8876 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a01837000007e2000047a5 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.41;branch=z9hG4bKc0a800290131c9b142a01837000007e2000047a5 From: ;tag=1804941257892 To: ;tag=as5f6102c6 Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8876 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41' asterisk-- ClearCall: Request to clear call with token ip$192.168.8.4:2088/31521, cause EndedByRemoteUser asterisk-- Sending RELEASE COMPLETE asterisk*CLI> channelsOpen = 2 asterisk*CLI> channelsOpen = 1 asterisk*CLI> channelsOpen = 0 asteriskExternalRTPChannel Destroyed asteriskExternalRTPChannel Destroyed asteriskExternalRTPChannel Destroyed asterisk-- ClearCall: Request to clear call with token ip$192.168.8.4:2088/31521, cause EndedByTransportFail -- 4FXS (20001, 4FXS-01e6f3) [192.168.8.4] has cleared the call asterisk== H.323 Connection deleted. asterisk*CLI> <-- SIP read from 192.168.0.71:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 f: ;tag=9806759320806 CSeq: 4741 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.71;rport;branch=z9hG4bKc0a800470131c9b142a0184a00001df500002521 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.71:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.71;branch=z9hG4bKc0a800470131c9b142a0184a00001df500002521 From: ;tag=9806759320806 To: ;tag=as482857c5 Call-ID: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 CSeq: 4741 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71' asterisk*CLI> <-- SIP read from 192.168.0.45:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 f: ;tag=91597966248 CSeq: 448 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a0183f000004f20000054c --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a0183f000004f20000054c From: ;tag=91597966248 To: ;tag=as704c6c94 Call-ID: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 CSeq: 448 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45' asterisk*CLI> se <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> set debug 0 Core debug is now OFF asterisk*CLI> set verbose 0 Verbosity is now OFF asterisk*CLI> set verbose <-- SIP read from 192.168.0.44:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 f: ;tag=951092132169 CSeq: 466 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a018470000652000000577 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a018470000652000000577 From: ;tag=951092132169 To: ;tag=as3049aff7 Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 466 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44' asterisk*CLI> set verbose 3 Verbosity was 0 and is now 3 Destroying call 'h5XQokbconBWWbCI@192.168.8.22' asterisk*CLI> <-- SIP read from 192.168.0.45:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45 Max-Forwards: 70 f: ;tag=917029617781 CSeq: 153 REGISTER t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a0184900005cb90000054e User-Agent: SJphone/1.50.271d (SJ Labs) --- (10 headers 0 lines)--- Using latest request as basis request Sending to 192.168.0.45 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a0184900005cb90000054e From: ;tag=917029617781 To: Call-ID: 71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45 CSeq: 153 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a0184900005cb90000054e From: ;tag=917029617781 To: ;tag=as5ab8bc78 Call-ID: 71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45 CSeq: 153 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="29c74c91" Content-Length: 0 --- Scheduling destruction of call '71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.45:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45 Max-Forwards: 70 f: ;tag=91702962805 CSeq: 154 REGISTER t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a01849000068d700000551 User-Agent: SJphone/1.50.271d (SJ Labs) Authorization: Digest username="silver",realm="asterisk",nonce="29c74c91",uri="sip:192.168.8.1:5060",response="a34465a191244b2e12121645e93f21 e2" --- (11 headers 0 lines)--- Using latest request as basis request Sending to 192.168.0.45 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a01849000068d700000551 From: ;tag=91702962805 To: Call-ID: 71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45 CSeq: 154 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a01849000068d700000551 From: ;tag=91702962805 To: ;tag=as5ab8bc78 Call-ID: 71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45 CSeq: 154 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 120 Contact: ;expires=120 Date: Fri, 03 Jun 2005 08:44:13 GMT Content-Length: 0 --- Scheduling destruction of call '71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45' in 15000 ms asterisk*CLI> se <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> s <-- SIP read from 192.168.0.23:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 f: ;tag=836897827964 CSeq: 400 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a0183400000e84000004d6 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a0183400000e84000004d6 From: ;tag=836897827964 To: ;tag=as557a4406 Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 400 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23' asterisk*CLI> sip.de <-- SIP read from 192.168.0.44:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44 Max-Forwards: 70 f: ;tag=95147652933 CSeq: 159 REGISTER t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a0184b000013fc00000579 User-Agent: SJphone/1.50.271d (SJ Labs) --- (10 headers 0 lines)--- Using latest request as basis request Sending to 192.168.0.44 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a0184b000013fc00000579 From: ;tag=95147652933 To: Call-ID: 5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44 CSeq: 159 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a0184b000013fc00000579 From: ;tag=95147652933 To: ;tag=as55ef39aa Call-ID: 5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44 CSeq: 159 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="72fa116a" Content-Length: 0 --- Scheduling destruction of call '5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44' in 15000 ms asterisk*CLI> sip.de <-- SIP read from 192.168.0.44:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44 Max-Forwards: 70 f: ;tag=95147654804 CSeq: 160 REGISTER t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a0184b00006c830000057c User-Agent: SJphone/1.50.271d (SJ Labs) Authorization: Digest username="koce",realm="asterisk",nonce="72fa116a",uri="sip:192.168.8.1:5060",response="65c7ae0370068bfa6d472e26e75d1289 " --- (11 headers 0 lines)--- Using latest request as basis request Sending to 192.168.0.44 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a0184b00006c830000057c From: ;tag=95147654804 To: Call-ID: 5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44 CSeq: 160 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.44:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.44;branch=z9hG4bKc0a8002c0131c9b142a0184b00006c830000057c From: ;tag=95147654804 To: ;tag=as55ef39aa Call-ID: 5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44 CSeq: 160 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 120 Contact: ;expires=120 Date: Fri, 03 Jun 2005 08:44:16 GMT ontent-Length: 0p.de --- Scheduling destruction of call '5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44' in 15000 ms asterisk*CLI> sip.de <-- SIP read from 192.168.0.41:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 f: ;tag=1805141251590 CSeq: 8877 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a0184b00005935000047a7 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.41:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.41;branch=z9hG4bKc0a800290131c9b142a0184b00005935000047a7 From: ;tag=1805141251590 To: ;tag=as45ca7f5d Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8877 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41' asterisk*CLI> sip debug no <-- SIP read from 192.168.0.71:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 f: ;tag=9808760214788 CSeq: 4742 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.71;rport;branch=z9hG4bKc0a800470131c9b142a0185e0000632f00002523 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.71:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.71;branch=z9hG4bKc0a800470131c9b142a0185e0000632f00002523 From: ;tag=9808760214788 To: ;tag=as6ffd5a14 Call-ID: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 CSeq: 4742 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71' asterisk*CLI> sip debug <-- SIP read from 192.168.0.45:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 f: ;tag=91797961893 CSeq: 449 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a0185300006d4b00000554 --- (8 headers 0 lines)--- Looking for 192.168.8.1:5060 in local Transmitting (no NAT) to 192.168.0.45:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.45;branch=z9hG4bKc0a8002d0131c9b142a0185300006d4b00000554 From: ;tag=91797961893 To: ;tag=as6d38f4ab Call-ID: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 CSeq: 449 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45' asterisk*CLI> sip ndebug <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> sip no debug SIP Debugging Disabled asterisk*CLI> h.323 no debug H323 Debug disabled asterisk*CLI>