bee*CLI>== New H.323 Connection created. bee*CLI>--Received SETUP message -- Setting up Call -- Call token: [ip$192.168.11.126:1798/4746] -- Calling party name: [214057 1] -- Calling party number: [] -- Called party name: [214058] -- Called party number: [214058] -- Calling party IP: [192.168.11.126] Setting capabilities to 0x8 (alaw) Capabilities in preference order is (alaw) Allowed Codecs: bee*CLI> Table: G.711-ALaw-64k <1> UserInput/hookflash <2> UserInput/RFC2833 <3> UserInput/dtmf <4> Set:LI> 0:LI> 0:> G.711-ALaw-64k <1> 1:> UserInput/hookflash <2> 2:> UserInput/RFC2833 <3> UserInput/dtmf <4> bee*CLI> bee*CLI>=-= In OnAnswerCall for call 4746 bee*CLI> - Progress Indicator: 0 bee*CLI> - Inserting PI of 0 into ALERTING message Audio is at 192.168.11.124 port 14516 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.11.125:5060: INVITE sip:214058@192.168.11.125:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK308ef9ed;rport From: "Unknown" ;tag=as22ac710e To: Contact: Call-ID: 6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 16 Jul 2007 06:42:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 244 v=0 o=root 15262 15262 IN IP4 192.168.11.124 s=session c=IN IP4 192.168.11.124 t=0 0 m=audio 14516 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> SIP/2.0 100 Trying Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK308ef9ed From:"Unknown" ;tag=as22ac710e To: Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq:102 INVITE Content-Type:application/sdp Content-Length:0 <-------------> --- (8 headers 0 lines) --- bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> SIP/2.0 180 Ringing Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK308ef9ed From:"Unknown" ;tag=as22ac710e To:;tag=64cde35d-918909 Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq:102 INVITE Contact: User-Agent:dlink 12-36-7924821 Content-Length:0 <-------------> --- (9 headers 0 lines) --- bee*CLI>Sending alerting bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> SIP/2.0 200 OK Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK308ef9ed From:"Unknown" ;tag=as22ac710e To:;tag=64cde35d-918909 Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq:102 INVITE Contact: User-Agent:dlink 12-36-7924821 Content-Type:application/sdp Content-Length:154 v=0 o=214058 2026106300 2026106300 IN IP4 192.168.11.125 s=Session SDP c=IN IP4 192.168.11.125 t=0 0 m=audio 9000 RTP/AVP 8 a=rtpmap:8 PCMA/8000/1 <-------------> --- (11 headers 7 lines) --- Found RTP audio format 8 Peer audio RTP is at port 192.168.11.125:9000 Found description format PCMA for ID 8 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.11.125:9000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.11.125, port 5060 Transmitting (no NAT) to 192.168.11.125:5060: ACK sip:214058@192.168.11.125:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK08efcfc5;rport From: "Unknown" ;tag=as22ac710e To: ;tag=64cde35d-918909 Contact: Call-ID: 6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- bee*CLI>Answering call ip$192.168.11.126:1798/4746 Peer capability is G.711-ALaw-64k <2> Found peer capability G.711-ALaw-64k <2>, Asterisk code is 8, frame size (in ms) is 20 Peer capability is G.711-uLaw-64k <3> Found peer capability G.711-uLaw-64k <3>, Asterisk code is 4, frame size (in ms) is 30 Peer capability is G.723.1 <4> Found peer capability G.723.1 <4>, Asterisk code is 1, frame size (in ms) is 120 Peer capabilities = 0xd (g723|ulaw|alaw), ordered list is (alaw|ulaw|g723) bee*CLI>-- Started logical channel: sending G.711-ALaw-64k bee*CLI> -- channelsOpen = 1 bee*CLI>=-= In OnConnectionEstablished for call 4746 bee*CLI> -- Connection Established with "214057 1 [192.168.11.126]" bee*CLI>-- Started logical channel: receiving G.711-ALaw-64k bee*CLI> -- channelsOpen = 2 bee*CLI> External RTP Session Starting bee*CLI> RTP channel id 1 parameters: bee*CLI> -- remoteIpAddress: 192.168.11.126 bee*CLI> -- remotePort: 49606 bee*CLI> -- ExternalIpAddress: 195.239.214.91 bee*CLI> -- ExternalPort: 18734 bee*CLI>MyH323_ExternalRTPChannel::OnReceivedAckPDU -- remoteIpAddress: 192.168.11.126 -- remotePort: 49606 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.11.126 -- remotePort: 49606 -- ExternalIpAddress: 195.239.214.91 -- ExternalPort: 18734 [Jul 16 10:42:23] WARNING[19810]: rtp.c:885 ast_rtcp_read: RTCP Read too short bee*CLI> <--- SIP read from 192.168.11.125:5060 ---> BYE sip:Unknown@192.168.11.124 SIP/2.0 Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bKa92ce9c8d826284a From:;tag=64cde35d-918909 To:"Unknown" ;tag=as22ac710e Call-ID:6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq:804 BYE Contact: Max-Forwards:70 User-Agent:dlink 12-36-7924821 Content-Length:0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.11.125 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.11.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bKa92ce9c8d826284a;received=192.168.11.125 From: ;tag=64cde35d-918909 To: "Unknown" ;tag=as22ac710e Call-ID: 6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124 CSeq: 804 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> bee*CLI>-- Sending RELEASE COMPLETE -- ClearCall: Request to clear call with token ip$192.168.11.126:1798/4746, cause EndedByRemoteUser bee*CLI> channelsOpen = 1 bee*CLI> channelsOpen = 0 bee*CLI>ExternalRTPChannel Destroyed bee*CLI>ExternalRTPChannel Destroyed bee*CLI>-- ClearCall: Request to clear call with token ip$192.168.11.126:1798/4746, cause EndedByTransportFail -- 214057 1 [192.168.11.126] has cleared the call bee*CLI>== H.323 Connection deleted. Really destroying SIP dialog '6fcae6215fb9037d6d6f42e80d8c5b14@192.168.11.124' Method: BYE