asterisk*CLI> asterisk*CLI> asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- asterisk== New H.323 Connection created. -- Setting up Call -- Call token: [ip$192.168.8.4:2058/31506] -- 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 31506 - 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: 12462 -- 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$192.168.8.4:2058/31506", "localextension|SIP/petew") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2058/31506", ""4FXS" <20001>") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2058/31506", "inhouse") in new stack -- Executing Dial("H323/ip$192.168.8.4:2058/31506", "SIP/petew|30|t") in new stack We're at 192.168.8.1 port 10976 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=z9hG4bK5b534018 From: "4FXS" ;tag=as48653ac9 To: Contact: Call-ID: 3207e13b09924745555982c2679279b7@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 01 Jun 2005 04:35:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 8437 8437 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 10976 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=z9hG4bK5b534018 From: "4FXS" ;tag=as48653ac9 To: ;tag=1848303209 Contact: Call-ID: 3207e13b09924745555982c2679279b7@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=z9hG4bK5b534018 From: "4FXS" ;tag=as48653ac9 To: ;tag=1848303209 Contact: Call-ID: 3207e13b09924745555982c2679279b7@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/petew-afda 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 192.168.0.23:5060: REGISTER sip:192.168.8.1:5060 SIP/2.0 l: 0 m: ;events="message-summary" i: 8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 Max-Forwards: 70 f: ;tag=62354829893 CSeq: 9 REGISTER t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b1429d3ae5000044c000000045 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=z9hG4bKc0a800170131c9b1429d3ae5000044c000000045 From: ;tag=62354829893 To: Call-ID: 8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 9 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=z9hG4bKc0a800170131c9b1429d3ae5000044c000000045 From: ;tag=62354829893 To: ;tag=as4e4c2c5b Call-ID: 8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 9 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="739ea38f" Content-Length: 0 --- Scheduling destruction of call '8177FC22-D26E-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: 8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 Max-Forwards: 70 f: ;tag=62355328075 CSeq: 10 REGISTER t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b1429d3ae500003a4500000048 User-Agent: SJphone/1.50.271d (SJ Labs) Authorization: Digest username="plamen",realm="asterisk",nonce="739ea38f",uri="sip:192.168.8.1:5060",response="538356f6278a76014f2e2891a70c4e 3a" --- (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=z9hG4bKc0a800170131c9b1429d3ae500003a4500000048 From: ;tag=62355328075 To: Call-ID: 8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 10 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/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.23;branch=z9hG4bKc0a800170131c9b1429d3ae500003a4500000048 From: ;tag=62355328075 To: ;tag=as4e4c2c5b Call-ID: 8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 10 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 120 Contact: ;expires=120 Date: Wed, 01 Jun 2005 04:35:14 GMT Content-Length: 0 --- Scheduling destruction of call '8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.23:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: 8177FC28-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 f: ;tag=6240982194 CSeq: 23 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b1429d3ae6000022cf0000004b --- (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=z9hG4bKc0a800170131c9b1429d3ae6000022cf0000004b From: ;tag=6240982194 To: ;tag=as72b5cddf Call-ID: 8177FC28-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 23 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '8177FC28-D26E-11D9-9D7B-00C026A003C6@192.168.0.23' 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=z9hG4bK8bd0Yx6p58f3hPgi Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=JgIVVZIwSztXCq4p To: "ted" Call-ID: Trtl47320zrRfYBn@192.168.8.22 CSeq: 12180 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=z9hG4bK8bd0Yx6p58f3hPgi From: "ted" ;tag=JgIVVZIwSztXCq4p To: "ted" Call-ID: Trtl47320zrRfYBn@192.168.8.22 CSeq: 12180 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=z9hG4bK8bd0Yx6p58f3hPgi From: "ted" ;tag=JgIVVZIwSztXCq4p To: "ted" ;tag=as7223d5c2 Call-ID: Trtl47320zrRfYBn@192.168.8.22 CSeq: 12180 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="0e4546af" Content-Length: 0 --- Scheduling destruction of call 'Trtl47320zrRfYBn@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=z9hG4bKVqUXgyYoEcYbs93Q Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=nvjm8GcAXAVbBZdI To: "ted" Call-ID: Trtl47320zrRfYBn@192.168.8.22 CSeq: 12181 REGISTER Contact: Expires: 60 Authorization: Digest username="ted", realm="asterisk", nonce="0e4546af", uri="sip:192.168.8.1", response="338600b6dc23c6c9fc7fa66fd2e4b9c6", 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=z9hG4bKVqUXgyYoEcYbs93Q From: "ted" ;tag=nvjm8GcAXAVbBZdI To: "ted" Call-ID: Trtl47320zrRfYBn@192.168.8.22 CSeq: 12181 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=z9hG4bKVqUXgyYoEcYbs93Q From: "ted" ;tag=nvjm8GcAXAVbBZdI To: "ted" ;tag=as7223d5c2 Call-ID: Trtl47320zrRfYBn@192.168.8.22 CSeq: 12181 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 60 Contact: ;expires=60 Date: Wed, 01 Jun 2005 04:35:16 GMT Content-Length: 0 --- Scheduling destruction of call 'Trtl47320zrRfYBn@192.168.8.22' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.0.45:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45 f: ;tag=4700892126351 CSeq: 2310 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b1429d3b0700002f3600001b3a --- (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=z9hG4bKc0a8002d0131c9b1429d3b0700002f3600001b3a From: ;tag=4700892126351 To: ;tag=as3c14159b Call-ID: BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45 CSeq: 2310 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45' 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=z9hG4bK5b534018 From: "4FXS" ;tag=as48653ac9 To: ;tag=1848303209 Contact: Call-ID: 3207e13b09924745555982c2679279b7@192.168.8.1 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1103m Content-Length: 190 v=0 o=petew 621390 626125 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=z9hG4bK55ea4f13 From: "4FXS" ;tag=as48653ac9 To: ;tag=1848303209 Contact: Call-ID: 3207e13b09924745555982c2679279b7@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/petew-afda answered H323/ip$192.168.8.4:2058/31506 Answering call ip$192.168.8.4:2058/31506 =-= In OnConnectionEstablished for call 31506 -- Connection Established with "4FXS (20001, 4FXS-01e6f3) [192.168.8.4]" asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- Jun 1 07:35:23 WARNING[8437]: chan_h323.c:659 oh323_indicate: Don't know how to indicate condition 16 on ip$192.168.8.4:2058/31506 -- Started music on hold, class 'default', on H323/ip$192.168.8.4:2058/31506 -- Playing 'pbx-transfer' (language 'en') asterisk*CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- -- Executing Macro("Local/201@inhouse-692a,2", "localextension|SIP/plamen") in new stack -- Executing NoOp("Local/201@inhouse-692a,2", "300") in new stack -- Executing NoOp("Local/201@inhouse-692a,2", "inhouse") in new stack -- Executing Dial("Local/201@inhouse-692a,2", "SIP/plamen|30|t") in new stack We're at 192.168.8.1 port 19030 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=z9hG4bK7cf2c76d From: "300" ;tag=as3422bf37 To: Contact: Call-ID: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 01 Jun 2005 04:35:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 8443 8443 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 19030 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: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 CSeq: 102 INVITE f: ;tag=as3422bf37 t: "Plamen";tag=63728431887 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK7cf2c76d --- (8 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 180 Ringing l: 0 m: i: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 CSeq: 102 INVITE f: ;tag=as3422bf37 t: "Plamen";tag=63728431887 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK7cf2c76d --- (9 headers 0 lines)--- -- SIP/plamen-4c67 is ringing Destroying call '8177FC22-D26E-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: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 c: application/sdp CSeq: 102 INVITE f: ;tag=as3422bf37 t: "Plamen";tag=63728431887 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK7cf2c76d v=0 o=- 3326589299 3326589299 IN IP4 192.168.0.23 s=SJphone c=IN IP4 192.168.0.23 t=0 0 a=direction:active m=audio 49152 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 192.168.0.23:49152 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=z9hG4bK0ad3655d From: "300" ;tag=as3422bf37 To: ;tag=63728431887 Contact: Call-ID: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/plamen-4c67 answered Local/201@inhouse-692a,2 Destroying call 'Trtl47320zrRfYBn@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=z9hG4bKD9AD9CC9B7EA42628747BC4BF881970D From: ;tag=1848303209 To: "4FXS" ;tag=as48653ac9 Contact: Call-ID: 3207e13b09924745555982c2679279b7@192.168.8.1 CSeq: 14801 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=z9hG4bKD9AD9CC9B7EA42628747BC4BF881970D From: ;tag=1848303209 To: "4FXS" ;tag=as48653ac9 Call-ID: 3207e13b09924745555982c2679279b7@192.168.8.1 CSeq: 14801 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Stopped music on hold on H323/ip$192.168.8.4:2058/31506 -- Playing 'beep' (language 'en') == Spawn extension (macro-localextension, s, 3) exited non-zero on 'Transfered/H323/ip$192.168.8.4:2058/31506' in macro 'localexten sion' == Spawn extension (inhouse, 300, 1) exited non-zero on 'Transfered/H323/ip$192.168.8.4:2058/31506' Destroying call '3207e13b09924745555982c2679279b7@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: 8177FC28-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 f: ;tag=64411823299 CSeq: 24 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b1429d3afa0000706f0000004f --- (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=z9hG4bKc0a800170131c9b1429d3afa0000706f0000004f From: ;tag=64411823299 To: ;tag=as2a1dd301 Call-ID: 8177FC28-D26E-11D9-9D7B-00C026A003C6@192.168.0.23 CSeq: 24 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '8177FC28-D26E-11D9-9D7B-00C026A003C6@192.168.0.23' asterisk*CLI> <-- SIP read from 192.168.0.45:5060: OPTIONS sip:192.168.8.1:5060 SIP/2.0 l: 0 i: BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45 f: ;tag=4702892117912 CSeq: 2311 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b1429d3b1b00001aa000001b3c --- (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=z9hG4bKc0a8002d0131c9b1429d3b1b00001aa000001b3c From: ;tag=4702892117912 To: ;tag=as4b9fec03 Call-ID: BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45 CSeq: 2311 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call 'BE89976F-9F70-4AC0-A5FF-88E840A9D661@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$192.168.8.4:2058/31506, cause EndedByRemoteUser -- Sending RELEASE COMPLETE asterisk*CLI> channelsOpen = 2 channelsOpen = 1 channelsOpen = 0 asteriskExternalRTPChannel Destroyed asteriskExternalRTPChannel Destroyed asteriskExternalRTPChannel Destroyed asterisk-- ClearCall: Request to clear call with token ip$192.168.8.4:2058/31506, 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=z9hG4bK72ba9a01 From: "300" ;tag=as3422bf37 To: ;tag=63728431887 Contact: Call-ID: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- == Spawn extension (macro-localextension, s, 3) exited non-zero on 'Local/201@inhouse-692a,2' in macro 'localextension' == Spawn extension (inhouse, 201, 1) exited non-zero on 'Local/201@inhouse-692a,2' asterisk*CLI> <-- SIP read from 192.168.0.23:5060: SIP/2.0 200 OK l: 0 m: i: 77a44e8614216b2b457cdbe220c81f6e@192.168.8.1 f: "300";tag=as3422bf37 CSeq: 103 BYE Server: SJphone/1.50.271d (SJ Labs) t: "Plamen";tag=63728431887 v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK72ba9a01 --- (9 headers 0 lines)--- Destroying call '77a44e8614216b2b457cdbe220c81f6e@192.168.8.1' asterisk*CLI> sip no debug <-- 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>