asterisk*CLI> asterisk*CLI> asterisk*CLI> set debug 255 Core debug was 0 and is now 255 asterisk*CLI> set verbose 255 Verbosity was 3 and is now 255 asterisk*CLI> asterisk*CLI> h.323 debug H323 debug enabled asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> sip debug SIP Debugging enabled Destroying call '71D85885-6D13-4154-B15C-4BECA1FA01B5@192.168.0.45' Destroying call '5D0F1960-1EFE-4B4F-9BEF-AA8129226F76@192.168.0.44' asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- 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=1025039014742 CSeq: 502 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a01b2a00007ac6000005e9 --- (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=z9hG4bKc0a8002c0131c9b142a01b2a00007ac6000005e9 From: ;tag=1025039014742 To: ;tag=as3c5780ec Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 502 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> <-- 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=91085699839 CSeq: 435 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a01b18000062dc00000544 --- (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=z9hG4bKc0a800170131c9b142a01b18000062dc00000544 From: ;tag=91085699839 To: ;tag=as401cfd30 Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 435 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== New H.323 Connection created. -- Setting up Call -- Call token: [ip$192.168.8.4:2092/31523] -- 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 31523 - Progress Indicator: 0 - Inserting PI of 0 into ALERTING message asterisk-- 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: 15268 -- 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$192.168.8.4:2092/31523", "localextension|SIP/petew") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2092/31523", ""4FXS" <20001>") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2092/31523", "inhouse") in new stack -- Executing Dial("H323/ip$192.168.8.4:2092/31523", "SIP/petew|30|t") in new stack We're at 192.168.8.1 port 10942 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=z9hG4bK336e1e1a From: "4FXS" ;tag=as6dab3499 To: Contact: Call-ID: 450e092c46869f1552317f3228f85bdc@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 08:56:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 14050 14050 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 10942 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=z9hG4bK336e1e1a From: "4FXS" ;tag=as6dab3499 To: ;tag=879798348 Contact: Call-ID: 450e092c46869f1552317f3228f85bdc@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=z9hG4bK336e1e1a From: "4FXS" ;tag=as6dab3499 To: ;tag=879798348 Contact: Call-ID: 450e092c46869f1552317f3228f85bdc@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/petew-20f9 is ringing Sending alerting 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 192.168.0.41:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 f: ;tag=1812535781007 CSeq: 8913 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a01b2f000071f2000047ef --- (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=z9hG4bKc0a800290131c9b142a01b2f000071f2000047ef From: ;tag=1812535781007 To: ;tag=as4f36db8d Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8913 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 read from 192.168.0.70:5061: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK336e1e1a From: "4FXS" ;tag=as6dab3499 To: ;tag=879798348 Contact: Call-ID: 450e092c46869f1552317f3228f85bdc@192.168.8.1 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1103m Content-Length: 196 v=0 o=petew 189091093 189093765 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=z9hG4bK5a7b5973 From: "4FXS" ;tag=as6dab3499 To: ;tag=879798348 Contact: Call-ID: 450e092c46869f1552317f3228f85bdc@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/petew-20f9 answered H323/ip$192.168.8.4:2092/31523 Answering call ip$192.168.8.4:2092/31523 =-= In OnConnectionEstablished for call 31523 -- Connection Established with "4FXS (20001, 4FXS-01e6f3) [192.168.8.4]" asterisk*CLI> <-- SIP read from 192.168.0.23:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23 Max-Forwards: 70 f: ;tag=911500125783 CSeq: 151 REGISTER t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a01b1e000060f500000546 User-Agent: SJphone/1.50.271d (SJ Labs) --- (10 headers 0 lines)--- Using latest request as basis request Sending to 192.168.0.23 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a01b1e000060f500000546 From: ;tag=911500125783 To: Call-ID: 74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 151 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a01b1e000060f500000546 From: ;tag=911500125783 To: ;tag=as2975b1da Call-ID: 74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 151 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="4dbe2a90" Content-Length: 0 --- Scheduling destruction of call '74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.23:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23 Max-Forwards: 70 f: ;tag=911500512602 CSeq: 152 REGISTER t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a01b1e0000521b00000549 User-Agent: SJphone/1.50.271d (SJ Labs) Authorization: Digest username="plamen",realm="asterisk",nonce="4dbe2a90",uri="sip:192.168.8.1:5060",response="1fa7258cc13b902f70a286156828c7 d5" --- (11 headers 0 lines)--- Using latest request as basis request Sending to 192.168.0.23 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a01b1e0000521b00000549 From: ;tag=911500512602 To: Call-ID: 74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 152 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Saved useragent "SJphone/1.50.271d (SJ Labs)" for peer plamen Transmitting (no NAT) to 192.168.0.23:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b142a01b1e0000521b00000549 From: ;tag=911500512602 To: ;tag=as2975b1da Call-ID: 74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 152 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 120 Contact: ;expires=120 Date: Fri, 03 Jun 2005 08:56:41 GMT Content-Length: 0 --- Scheduling destruction of call '74E900E2-D411-11D9-9D7B-00C026A003C6@192.168.0.23' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- 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=99191564261 CSeq: 485 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a01b3600001a9c000005c0 --- (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=z9hG4bKc0a8002d0131c9b142a01b3600001a9c000005c0 From: ;tag=99191564261 To: ;tag=as0616dd02 Call-ID: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 CSeq: 485 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 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=9882792728277 CSeq: 4779 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.71;rport;branch=z9hG4bKc0a800470131c9b142a01b42000045900000256d --- (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=z9hG4bKc0a800470131c9b142a01b42000045900000256d From: ;tag=9882792728277 To: ;tag=as337c2ce8 Call-ID: F9476139-461B-4A15-BA15-D53C25F1C3CB@192.168.0.71 CSeq: 4779 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 192.168.8.22:1720: REGISTER sip:192.168.8.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.22:1720;branch=z9hG4bKLCgEvOMVYsP9PH7W Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=lRRm9sAjFUSYQhFz To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9665 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=z9hG4bKLCgEvOMVYsP9PH7W From: "ted" ;tag=lRRm9sAjFUSYQhFz To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9665 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=z9hG4bKLCgEvOMVYsP9PH7W From: "ted" ;tag=lRRm9sAjFUSYQhFz To: "ted" ;tag=as09392f03 Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9665 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="73992a18" 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=z9hG4bKXdMyVg8FZ6rHbxcB Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=ZHs32uWwhEl3sq7o To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9666 REGISTER Contact: Expires: 60 Authorization: Digest username="ted", realm="asterisk", nonce="73992a18", uri="sip:192.168.8.1", response="42ce5a103495a2acb16468f371724f9c", 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=z9hG4bKXdMyVg8FZ6rHbxcB From: "ted" ;tag=ZHs32uWwhEl3sq7o To: "ted" Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9666 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=z9hG4bKXdMyVg8FZ6rHbxcB From: "ted" ;tag=ZHs32uWwhEl3sq7o To: "ted" ;tag=as09392f03 Call-ID: h5XQokbconBWWbCI@192.168.8.22 CSeq: 9666 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 60 Contact: ;expires=60 Date: Fri, 03 Jun 2005 08:56:43 GMT Content-Length: 0 --- Scheduling destruction of call 'h5XQokbconBWWbCI@192.168.8.22' in 15000 ms Jun 3 11:56:44 WARNING[14050]: chan_h323.c:659 oh323_indicate: Don't know how to indicate condition 16 on ip$192.168.8.4:2092/31523 -- Started music on hold, class 'default', on H323/ip$192.168.8.4:2092/31523 -- Playing 'pbx-transfer' (language 'en') -- Executing Macro("Local/201@inhouse-487b,2", "localextension|SIP/plamen") in new stack -- Executing NoOp("Local/201@inhouse-487b,2", "300") in new stack -- Executing NoOp("Local/201@inhouse-487b,2", "inhouse") in new stack -- Executing Dial("Local/201@inhouse-487b,2", "SIP/plamen|30|t") in new stack We're at 192.168.8.1 port 14376 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=z9hG4bK20bb4636 From: "300" ;tag=as13ff9e0c To: Contact: Call-ID: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 03 Jun 2005 08:56:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 14056 14056 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 14376 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: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 CSeq: 102 INVITE f: ;tag=as13ff9e0c t: "Plamen";tag=912354626318 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK20bb4636 --- (8 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 180 Ringing l: 0 m: i: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 CSeq: 102 INVITE f: ;tag=as13ff9e0c t: "Plamen";tag=912354626318 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK20bb4636 --- (9 headers 0 lines)--- -- SIP/plamen-55ab is ringing 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=1027039021119 CSeq: 503 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a01b3e00007a28000005eb --- (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=z9hG4bKc0a8002c0131c9b142a01b3e00007a28000005eb From: ;tag=1027039021119 To: ;tag=as4bfc882c Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 503 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> <-- 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=912858923431 CSeq: 436 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a01b2c000017120000054d --- (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=z9hG4bKc0a800170131c9b142a01b2c000017120000054d From: ;tag=912858923431 To: ;tag=as119dd9fa Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 436 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: SIP/2.0 200 OK l: 219 m: i: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 c: application/sdp CSeq: 102 INVITE f: ;tag=as13ff9e0c t: "Plamen";tag=912354626318 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK20bb4636 v=0 o=- 3326777767 3326777767 IN IP4 192.168.0.23 s=SJphone c=IN IP4 192.168.0.23 t=0 0 a=direction:active m=audio 49166 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 192.168.0.23:49166 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=z9hG4bK732bc848 From: "300" ;tag=as13ff9e0c To: ;tag=912354626318 Contact: Call-ID: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/plamen-55ab answered Local/201@inhouse-487b,2 Destroying call '74E900E2-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=18127357829058 CSeq: 8914 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.41;rport;branch=z9hG4bKc0a800290131c9b142a01b43000039f9000047f1 --- (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=z9hG4bKc0a800290131c9b142a01b43000039f9000047f1 From: ;tag=18127357829058 To: ;tag=as02cffa26 Call-ID: CD3F296D-FD0E-4D54-BDCD-8371AFAA327C@192.168.0.41 CSeq: 8914 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 'h5XQokbconBWWbCI@192.168.8.22' 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=z9hG4bK7468F74624FB4D2983382959DC4D242A From: ;tag=879798348 To: "4FXS" ;tag=as6dab3499 Contact: Call-ID: 450e092c46869f1552317f3228f85bdc@192.168.8.1 CSeq: 19178 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=z9hG4bK7468F74624FB4D2983382959DC4D242A From: ;tag=879798348 To: "4FXS" ;tag=as6dab3499 Call-ID: 450e092c46869f1552317f3228f85bdc@192.168.8.1 CSeq: 19178 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Stopped music on hold on H323/ip$192.168.8.4:2092/31523 -- Playing 'beep' (language 'en') Destroying call '450e092c46869f1552317f3228f85bdc@192.168.8.1' 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=993915615149 CSeq: 486 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b142a01b4a000041cc000005c2 --- (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=z9hG4bKc0a8002d0131c9b142a01b4a000041cc000005c2 From: ;tag=993915615149 To: ;tag=as412a68d7 Call-ID: 0BBD731C-70C3-434F-AB04-43B29C8A3908@192.168.0.45 CSeq: 486 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 read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk-- ClearCall: Request to clear call with token ip$192.168.8.4:2092/31523, cause EndedByRemoteUser -- Sending RELEASE COMPLETE asterisk*CLI> channelsOpen = 2 channelsOpen = 1 asterisk*CLI> channelsOpen = 0 asteriskExternalRTPChannel Destroyed asteriskExternalRTPChannel Destroyed asteriskExternalRTPChannel Destroyed asterisk-- ClearCall: Request to clear call with token ip$192.168.8.4:2092/31523, cause EndedByTransportFail -- 4FXS (20001, 4FXS-01e6f3) [192.168.8.4] has cleared the call == H.323 Connection deleted. set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.23, port 5060 Reliably Transmitting (no NAT) to 192.168.0.23:5060: BYE sip:plamen@192.168.0.23:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5f03ed40 From: "300" ;tag=as13ff9e0c To: ;tag=912354626318 Contact: Call-ID: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 200 OK l: 0 m: i: 6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1 f: "300";tag=as13ff9e0c CSeq: 103 BYE Server: SJphone/1.50.271d (SJ Labs) t: "Plamen";tag=912354626318 v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK5f03ed40 --- (9 headers 0 lines)--- Destroying call '6b2a3e5f240ad67902b6920531b0dd74@192.168.8.1' asterisk*CLI> sip n <-- 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=1029039013465 CSeq: 504 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.44;rport;branch=z9hG4bKc0a8002c0131c9b142a01b5200007733000005ed --- (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=z9hG4bKc0a8002c0131c9b142a01b5200007733000005ed From: ;tag=1029039013465 To: ;tag=as0398c6c6 Call-ID: 9FD95229-0659-4F52-B296-4F818E498DD5@192.168.0.44 CSeq: 504 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> sip no <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk*CLI> sip no debug <-- 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=914865922465 CSeq: 437 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b142a01b4000003a7300000551 --- (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=z9hG4bKc0a800170131c9b142a01b4000003a7300000551 From: ;tag=914865922465 To: ;tag=as5f6a52df Call-ID: 74E900E8-D411-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 437 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 no debug SIP Debugging Disabled asterisk*CLI> h.323 no debug H323 Debug disabled asterisk*CLI> set debug 0 Core debug is now OFF asterisk*CLI> set verbose 3 Verbosity was 255 and is now 3 asterisk*CLI>