asterisk*CLI> asterisk*CLI> set verbose 255 Verbosity was 3 and is now 255 asterisk*CLI> set debug 255 Core debug was 0 and is now 255 asterisk*CLI> h.323 debug H323 debug enabled asterisk*CLI> sip debug SIP Debugging enabled 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=z9hG4bKToC3W374l7NFURk1 Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=vhcd8u46EXn0TYdx To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9651 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=z9hG4bKToC3W374l7NFURk1 From: "ted" ;tag=vhcd8u46EXn0TYdx To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9651 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=z9hG4bKToC3W374l7NFURk1 From: "ted" ;tag=vhcd8u46EXn0TYdx To: "ted" ;tag=as4a7eaa17 Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9651 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="628564d6" Content-Length: 0 --- Scheduling destruction of call 'h5XQokbconBWWbCI@192.168.8.22' in 15000 ms 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=z9hG4bKb9I3ckMaGjDq1eoD Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=13d8C8ooDjytP3vp To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9652 REGISTER Contact: Expires: 60 Authorization: Digest username="ted", realm="asterisk", nonce="628564d6", uri="sip:192.168.8.1", response="41345c2f4f44c1bc907f34beac5cca4e", 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=z9hG4bKb9I3ckMaGjDq1eoD From: "ted" ;tag=13d8C8ooDjytP3vp To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9652 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=z9hG4bKb9I3ckMaGjDq1eoD From: "ted" ;tag=13d8C8ooDjytP3vp To: "ted" ;tag=as4a7eaa17 Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9652 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 60 Contact: ;expires=60 Date: Fri, 03 Jun 2005 08:51:28 GMT Content-Length: 0 --- Scheduling destruction of call 'h5XQokbconBWWbCI@192.168.8.22' in 15000 ms 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=995039061 CSeq: 487 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a019fe00001a1e000005b9 --- (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=z9hG4bKc0a8002c0131c9b142a019fe00001a1e000005b9 From: ;tag=995039061 To: ;tag=as53655f7e Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 487 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== New H.323 Connection created. -- Setting up Call -- Call token: [ip$192.168.8.4:2090/31522] -- 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 31522 - 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: 14944 -- 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$192.168.8.4:2090/31522", "localextension|SIP/petew") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2090/31522", ""4FXS" <20001>") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2090/31522", "inhouse") in new stack -- Executing Dial("H323/ip$192.168.8.4:2090/31522", "SIP/petew|30|t") in new stack We're at 192.168.8.1 port 12620 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=z9hG4bK4a9bbe3f From: "4FXS" ;tag=as13e8abce To: Contact: Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 08:51:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 13957 13957 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 12620 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=z9hG4bK4a9bbe3f From: "4FXS" ;tag=as13e8abce To: ;tag=549106810 Contact: Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@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=z9hG4bK4a9bbe3f From: "4FXS" ;tag=as13e8abce To: ;tag=549106810 Contact: Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/petew-fa0b is ringing Sending alerting asterisk-- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 3 External RTP Session Starting RTP channel id 0 parameters: 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=88082111703 CSeq: 420 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a019eb000045b900000516 --- (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=z9hG4bKc0a800170131c9b142a019eb000045b900000516 From: ;tag=88082111703 To: ;tag=as185112d8 Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 420 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 192.168.0.70:5061: --- (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=z9hG4bK4a9bbe3f From: "4FXS" ;tag=as13e8abce To: ;tag=549106810 Contact: Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1103m Content-Length: 196 v=0 o=petew 188788578 188791968 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=z9hG4bK776efc9a From: "4FXS" ;tag=as13e8abce To: ;tag=549106810 Contact: Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/petew-fa0b answered H323/ip$192.168.8.4:2090/31522 Answering call ip$192.168.8.4:2090/31522 =-= In OnConnectionEstablished for call 31522 -- Connection Established with "4FXS (20001, 4FXS-01e6f3) [192.168.8.4]" 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=18095357810133 CSeq: 8898 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a01a03000004f6000047d1 --- (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=z9hG4bKc0a800290131c9b142a01a03000004f6000047d1 From: ;tag=18095357810133 To: ;tag=as0fda80b2 Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8898 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 11:51:41 WARNING[13957]: chan_h323.c:659 oh323_indicate: Don't know how to indicate condition 16 on ip$192.168.8.4:2090/31522 -- Started music on hold, class 'default', on H323/ip$192.168.8.4:2090/31522 -- 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=985277952235 CSeq: 4764 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.71;rport;branch=z9hG4bKc0a800470131c9b142a01a16000050580000254f --- (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=z9hG4bKc0a800470131c9b142a01a16000050580000254f From: ;tag=985277952235 To: ;tag=as4d6c0025 Call-ID: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 CSeq: 4764 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=96197966267 CSeq: 471 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a01a0b00001fd000000592 --- (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=z9hG4bKc0a8002d0131c9b142a01a0b00001fd000000592 From: ;tag=96197966267 To: ;tag=as3688fd9f Call-ID: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 CSeq: 471 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' Destroying call 'h5XQokbconBWWbCI@192.168.8.22' -- Executing Macro("Local/201@inhouse-aa3c,2", "localextension|SIP/plamen") in new stack -- Executing NoOp("Local/201@inhouse-aa3c,2", "300") in new stack -- Executing NoOp("Local/201@inhouse-aa3c,2", "inhouse") in new stack -- Executing Dial("Local/201@inhouse-aa3c,2", "SIP/plamen|30|t") in new stack We're at 192.168.8.1 port 15634 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=z9hG4bK0f3ce081 From: "300" ;tag=as4f5f0275 To: Contact: Call-ID: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 08:51:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 13961 13961 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 15634 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: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 CSeq: 102 INVITE f: ;tag=as4f5f0275 t: "Plamen";tag=881804029640 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK0f3ce081 --- (8 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 180 Ringing l: 0 m: i: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 CSeq: 102 INVITE f: ;tag=as4f5f0275 t: "Plamen";tag=881804029640 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK0f3ce081 --- (9 headers 0 lines)--- -- SIP/plamen-60b8 is ringing 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/2.0 200 OK l: 219 m: i: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 c: application/sdp CSeq: 102 INVITE f: ;tag=as4f5f0275 t: "Plamen";tag=881804029640 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK0f3ce081 v=0 o=- 3326777461 3326777461 IN IP4 192.168.0.23 s=SJphone c=IN IP4 192.168.0.23 t=0 0 a=direction:active m=audio 49164 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 192.168.0.23:49164 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=z9hG4bK6f78a427 From: "300" ;tag=as4f5f0275 To: ;tag=881804029640 Contact: Call-ID: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/plamen-60b8 answered Local/201@inhouse-aa3c,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=z9hG4bK84A3BB861E5246EA99984414AFF70503 From: ;tag=549106810 To: "4FXS" ;tag=as13e8abce Contact: Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1 CSeq: 43021 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=z9hG4bK84A3BB861E5246EA99984414AFF70503 From: ;tag=549106810 To: "4FXS" ;tag=as13e8abce Call-ID: 73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1 CSeq: 43021 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Stopped music on hold on H323/ip$192.168.8.4:2090/31522 -- 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=997039021105 CSeq: 488 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a01a12000038c4000005bb --- (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=z9hG4bKc0a8002c0131c9b142a01a12000038c4000005bb From: ;tag=997039021105 To: ;tag=as46f7e794 Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 488 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 '73fe1b7917ff7737071e1c2c397f8f7e@192.168.8.1' 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=8828230528 CSeq: 421 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a019ff0000187c0000051a --- (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=z9hG4bKc0a800170131c9b142a019ff0000187c0000051a From: ;tag=8828230528 To: ;tag=as14ebd1fa Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 421 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 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: BYE sip:300@192.168.8.1 SIP/2.0 l: 0 m: i: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 Max-Forwards: 70 CSeq: 1 BYE f: ;tag=881804029640 t: ;tag=as4f5f0275 User-Agent: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a01a0200001c200000051c --- (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=z9hG4bKc0a800170131c9b142a01a0200001c200000051c From: ;tag=881804029640 To: ;tag=as4f5f0275 Call-ID: 3e6136cd4366b1ef592e4f106c46292d@192.168.8.1 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- asterisk-- ClearCall: Request to clear call with token ip$192.168.8.4:2090/31522, 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$192.168.8.4:2090/31522, cause EndedByTransportFail -- 4FXS (20001, 4FXS-01e6f3) [192.168.8.4] has cleared the call asterisk== H.323 Connection deleted. asterisk*CLI> <-- SIP read from 192.168.0.41:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 f: ;tag=18097357827831 CSeq: 8899 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a01a1700002138000047d3 --- (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=z9hG4bKc0a800290131c9b142a01a1700002138000047d3 From: ;tag=18097357827831 To: ;tag=as2d3ce4d2 Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8899 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' Destroying call '3e6136cd4366b1ef592e4f106c46292d@192.168.8.1' asterisk*CLI> sip no debug SIP Debugging Disabled asterisk*CLI> h.323 no debug H323 Debug disabled asterisk*CLI> set debug 3 Core debug was 255 and is now 3 asterisk*CLI> set verbose 3 Verbosity was 255 and is now 3 asterisk*CLI>