192.168.254.102 = Asterisk 192.168.254.120 = SipXecs Call was initiated from the asterisk box X201 dialing the SipXecs extension 2010
192.168.254.102 192.168.254.120 | | 1: |U------------INVITE----------->| 2: |<------100 Trying/INVITE------U| 3: |<-----404 Not Found/INVITE----U| 4: |U-------------ACK------------->| Generated SIP Workbench by BreakPoint Software, Inc. (www.sipworkbench.com) <<<<< Msg #1 / Packet #36: 192.168.254.102:5060 --> 192.168.254.120:5060 >>>>> INVITE sip:2...@192.168.254.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK73879b9d;rport From: "Mark Wood" <sip:2...@192.168.254.102>;tag=as16de1754 To: <sip:2...@192.168.254.120> Contact: <sip:2...@192.168.254.102> Call-ID: 546261573c578496015ad7bb1742f...@192.168.254.102 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 05 May 2009 23:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 268 v=0 o=root 2707 2707 IN IP4 192.168.254.102 s=session c=IN IP4 192.168.254.102 t=0 0 m=audio 19334 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <<<<< Msg #2 / Packet #37: 192.168.254.120:5060 --> 192.168.254.102:5060 >>>>> SIP/2.0 100 Trying From: "Mark Wood" <sip:2...@192.168.254.102>;tag=as16de1754 To: <sip:2...@192.168.254.120> Call-Id: 546261573c578496015ad7bb1742f...@192.168.254.102 Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK73879b9d;rport=5060 Content-Length: 0 <<<<< Msg #3 / Packet #38: 192.168.254.120:5060 --> 192.168.254.102:5060 >>>>> SIP/2.0 404 Not Found From: "Mark Wood" <sip:2...@192.168.254.102>;tag=as16de1754 To: <sip:2...@192.168.254.120>;tag=48b99501 Call-Id: 546261573c578496015ad7bb1742f...@192.168.254.102 Cseq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK73879b9d;rport=5060 Record-Route: <sip:192.168.254.120:5060;lr> User-Agent: sipXecs/4.0.0 sipXecs/registry (Linux) Date: Tue, 05 May 2009 23:54:19 GMT Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE Accept-Language: en Supported: gruu, path Content-Length: 0 <<<<< Msg #4 / Packet #39: 192.168.254.102:5060 --> 192.168.254.120:5060 >>>>> ACK sip:2...@192.168.254.120 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.102:5060;branch=z9hG4bK73879b9d;rport From: "Mark Wood" <sip:2...@192.168.254.102>;tag=as16de1754 To: <sip:2...@192.168.254.120>;tag=48b99501 Contact: <sip:2...@192.168.254.102> Call-ID: 546261573c578496015ad7bb1742f...@192.168.254.102 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -----Original Message----- From: Scott Lawrence [mailto:scott.lawre...@nortel.com] Sent: Tuesday, May 05, 2009 1:42 PM To: Mark Wood Cc: 'sipx-users@list.sipfoundry.org' Subject: Re: [sipx-users] Unmanged Gateways On Tue, 2009-05-05 at 13:28 -0700, Mark Wood wrote: > OK, went thru the dialplan and it is identical to the 3.10 version. > Ran the tail command and got the log info: Don't paste logs into email - line wrapping and other mail-client nonsense makes it hard to analyze them. To debug this, you'll need to trace the message flow and see what's going wrong across multiple components. See: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_Messages_to_display when you get the trace data, take a look at it using sipviewer and/or post the trace with a description of your configuration (identify components by IP address), what you were doing, and which call in the trace you're talking about (by call-id or frame number in the trace, preferably). _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
