*CLI> *CLI> *CLI> <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- == New H.323 Connection created. -- Setting up Call -- Call token: [ip$192.168.8.4:2064/31509] -- 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> =-= In OnAnswerCall for call 31509 - 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: 11690 -- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- Transmitting RFC2833 on payload 101 -- Executing Macro("H323/ip$192.168.8.4:2064/31509", "localextension|SIP/petew") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2064/31509", ""4FXS" <20001>") in new stack -- Executing NoOp("H323/ip$192.168.8.4:2064/31509", "inhouse") in new stack -- Executing Dial("H323/ip$192.168.8.4:2064/31509", "SIP/petew|30|t") in new stack We're at 192.168.8.1 port 10378 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=z9hG4bK6c972171 From: "4FXS" ;tag=as49f48bfa To: Contact: Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 01 Jun 2005 04:47:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 8555 8555 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 10378 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 <-- SIP read from 192.168.0.70:5061: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK6c972171 From: "4FXS" ;tag=as49f48bfa To: ;tag=2265462397 Contact: Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- <-- SIP read from 192.168.0.70:5061: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK6c972171 From: "4FXS" ;tag=as49f48bfa To: ;tag=2265462397 Contact: Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 102 INVITE Server: X-Lite release 1103m Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/petew-1842 is ringing Sending alerting -- Started logical channel: receiving G.711-uLaw-64k -- channelsOpen = 3 External RTP Session Starting RTP channel id 0 parameters: Destroying call '8177FC22-D26E-11D9-9D7B-00C026A003C6@192.168.0.23' <-- SIP read from 192.168.0.70:5061: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK6c972171 From: "4FXS" ;tag=as49f48bfa To: ;tag=2265462397 Contact: Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1103m Content-Length: 192 v=0 o=petew 1357156 1360015 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=z9hG4bK5a500896 From: "4FXS" ;tag=as49f48bfa To: ;tag=2265462397 Contact: Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/petew-1842 answered H323/ip$192.168.8.4:2064/31509 Answering call ip$192.168.8.4:2064/31509 =-= In OnConnectionEstablished for call 31509 -- Connection Established with "4FXS (20001, 4FXS-01e6f3) [192.168.8.4]" <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- <-- 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=4774842127570 CSeq: 2346 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b1429d3dea0000406100001ba4 --- (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=z9hG4bKc0a8002d0131c9b1429d3dea0000406100001ba4 From: ;tag=4774842127570 To: ;tag=as24033bba Call-ID: BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45 CSeq: 2346 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' Jun 1 07:47:38 WARNING[8555]: chan_h323.c:659 oh323_indicate: Don't know how to indicate condition 16 on ip$192.168.8.4:2064/31509 -- Started music on hold, class 'default', on H323/ip$192.168.8.4:2064/31509 -- Playing 'pbx-transfer' (language 'en') -- Executing Macro("Local/201@inhouse-ac27,2", "localextension|SIP/plamen") in new stack -- Executing NoOp("Local/201@inhouse-ac27,2", "300") in new stack -- Executing NoOp("Local/201@inhouse-ac27,2", "inhouse") in new stack -- Executing Dial("Local/201@inhouse-ac27,2", "SIP/plamen|30|t") in new stack We're at 192.168.8.1 port 15558 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=z9hG4bK1e673fa6 From: "300" ;tag=as333d330f To: Contact: Call-ID: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 01 Jun 2005 04:47:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 8555 8555 IN IP4 192.168.8.1 s=session c=IN IP4 192.168.8.1 t=0 0 m=audio 15558 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 <-- SIP read from 192.168.0.23:5060: SIP/2.0 100 Trying l: 0 i: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 CSeq: 102 INVITE f: ;tag=as333d330f t: "Plamen";tag=137001816781 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK1e673fa6 --- (8 headers 0 lines)--- <-- SIP read from 192.168.0.23:5060: SIP/2.0 180 Ringing l: 0 m: i: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 CSeq: 102 INVITE f: ;tag=as333d330f t: "Plamen";tag=137001816781 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK1e673fa6 --- (9 headers 0 lines)--- -- SIP/plamen-a4ad is ringing <-- 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=z9hG4bKKH1ZN3vVnrd1PBfj Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=KZAag9HFJEC6B1T6 To: "ted" Call-ID: M8LrOOmvbpJq3XtY@192.168.8.22 CSeq: 12762 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=z9hG4bKKH1ZN3vVnrd1PBfj From: "ted" ;tag=KZAag9HFJEC6B1T6 To: "ted" Call-ID: M8LrOOmvbpJq3XtY@192.168.8.22 CSeq: 12762 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=z9hG4bKKH1ZN3vVnrd1PBfj From: "ted" ;tag=KZAag9HFJEC6B1T6 To: "ted" ;tag=as16b56767 Call-ID: M8LrOOmvbpJq3XtY@192.168.8.22 CSeq: 12762 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="7d4b5161" Content-Length: 0 --- Scheduling destruction of call 'M8LrOOmvbpJq3XtY@192.168.8.22' in 15000 ms <-- 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=z9hG4bKxvP2wQJkRZvMHk8t Max-Forwards: 70 User-Agent: PA168S From: "ted" ;tag=rcK25wUflDeuLI1a To: "ted" Call-ID: M8LrOOmvbpJq3XtY@192.168.8.22 CSeq: 12763 REGISTER Contact: Expires: 60 Authorization: Digest username="ted", realm="asterisk", nonce="7d4b5161", uri="sip:192.168.8.1", response="c46c70b39d8b3ec464787ee4281b1aa1", 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=z9hG4bKxvP2wQJkRZvMHk8t From: "ted" ;tag=rcK25wUflDeuLI1a To: "ted" Call-ID: M8LrOOmvbpJq3XtY@192.168.8.22 CSeq: 12763 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=z9hG4bKxvP2wQJkRZvMHk8t From: "ted" ;tag=rcK25wUflDeuLI1a To: "ted" ;tag=as16b56767 Call-ID: M8LrOOmvbpJq3XtY@192.168.8.22 CSeq: 12763 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 60 Contact: ;expires=60 Date: Wed, 01 Jun 2005 04:47:42 GMT Content-Length: 0 --- Scheduling destruction of call 'M8LrOOmvbpJq3XtY@192.168.8.22' in 15000 ms <-- SIP read from 192.168.0.23:5060: SIP/2.0 200 OK l: 219 m: i: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 c: application/sdp CSeq: 102 INVITE f: ;tag=as333d330f t: "Plamen";tag=137001816781 Server: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.8.1:5060;branch=z9hG4bK1e673fa6 v=0 o=- 3326590032 3326590032 IN IP4 192.168.0.23 s=SJphone c=IN IP4 192.168.0.23 t=0 0 a=direction:active m=audio 49158 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 192.168.0.23:49158 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=z9hG4bK1e08bb0c From: "300" ;tag=as333d330f To: ;tag=137001816781 Contact: Call-ID: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/plamen-a4ad answered Local/201@inhouse-ac27,2 <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- <-- 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=z9hG4bKD23E3C7990FB441E88AAD7A0360928C0 From: ;tag=2265462397 To: "4FXS" ;tag=as49f48bfa Contact: Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 30235 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=z9hG4bKD23E3C7990FB441E88AAD7A0360928C0 From: ;tag=2265462397 To: "4FXS" ;tag=as49f48bfa Call-ID: 07741d364ce2804d17a18d1244d41197@192.168.8.1 CSeq: 30235 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Stopped music on hold on H323/ip$192.168.8.4:2064/31509 -- Playing 'beep' (language 'en') == Spawn extension (macro-localextension, s, 3) exited non-zero on 'Transfered/H323/ip$192.168.8.4:2064/31509' in macro 'localexten sion' == Spawn extension (inhouse, 300, 1) exited non-zero on 'Transfered/H323/ip$192.168.8.4:2064/31509' Destroying call '07741d364ce2804d17a18d1244d41197@192.168.8.1' <-- SIP read from 192.168.0.23:5060: BYE sip:300@192.168.8.1 SIP/2.0 l: 0 m: i: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 Max-Forwards: 70 CSeq: 1 BYE f: ;tag=137001816781 t: ;tag=as333d330f User-Agent: SJphone/1.50.271d (SJ Labs) v: SIP/2.0/UDP 192.168.0.23;rport;branch=z9hG4bKc0a800170131c9b1429d3dda0000546a000000c4 --- (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=z9hG4bKc0a800170131c9b1429d3dda0000546a000000c4 From: ;tag=137001816781 To: ;tag=as333d330f Call-ID: 198813cf47f918b2143fe2cf72998d2d@192.168.8.1 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (macro-localextension, s, 3) exited non-zero on 'Local/201@inhouse-ac27,2' in macro 'localextension' == Spawn extension (inhouse, 201, 1) exited non-zero on 'Local/201@inhouse-ac27,2' -- Sending RELEASE COMPLETE -- ClearCall: Request to clear call with token ip$192.168.8.4:2064/31509, cause EndedByRemoteUser channelsOpen = 2 channelsOpen = 1 channelsOpen = 0 ExternalRTPChannel Destroyed ExternalRTPChannel Destroyed ExternalRTPChannel Destroyed -- ClearCall: Request to clear call with token ip$192.168.8.4:2064/31509, cause EndedByTransportFail -- 4FXS (20001, 4FXS-01e6f3) [192.168.8.4] has cleared the call == H.323 Connection deleted. Destroying call '198813cf47f918b2143fe2cf72998d2d@192.168.8.1' <-- SIP read from 192.168.0.70:5061: --- (0 headers 0 lines) Nat keepalive --- <-- 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=4776842123075 CSeq: 2347 OPTIONS Max-Forwards: 70 t: v: SIP/2.0/UDP 192.168.0.45;rport;branch=z9hG4bKc0a8002d0131c9b1429d3dfe0000151d00001ba6 --- (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=z9hG4bKc0a8002d0131c9b1429d3dfe0000151d00001ba6 From: ;tag=4776842123075 To: ;tag=as64bc6aa2 Call-ID: BE89976F-9F70-4AC0-A5FF-88E840A9D661@192.168.0.45 CSeq: 2347 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' Destroying call 'M8LrOOmvbpJq3XtY@192.168.8.22' sip no debug SIP Debugging Disabled *CLI> h.323 no debug H323 Debug disabled *CLI>