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oquidave

debug outup

May 5th, 2011
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  1. baraza*CLI> console dial 0800707020@to-warid
  2. Really destroying SIP dialog '0634e24b5e3f41214e0da8d002f1a278@41.222.6.236' Method: INVITE
  3. [May 5 11:46:03] WARNING[16668]: app_dial.c:1759 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  4. << Hangup on console >>
  5. [May 5 11:46:03] NOTICE[6034]: console_video.c:133 console_video_start: voice only, console video support not present
  6.  
  7. <--- SIP read from UDP:10.1.16.72:5060 --->
  8. SIP/2.0 504 Server Time-out
  9. Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK48c68106;rport=5060
  10. Call-ID: 03e775230861409921bce60b0463e30a@172.18.0.34
  11. From: "asterisk"<sip:asterisk@172.18.0.34>;tag=as4f689133
  12. To: <sip:10.1.16.72>;tag=d671624f
  13. CSeq: 102 OPTIONS
  14. Content-Length: 0
  15.  
  16.  
  17. <------------->
  18. --- (7 headers 0 lines) ---
  19. Reliably Transmitting (no NAT) to 10.1.16.72:5060:
  20. OPTIONS sip:10.1.16.72 SIP/2.0
  21. Via: SIP/2.0/UDP 172.18.0.34:5060;branch=z9hG4bK202f7414;rport
  22. Max-Forwards: 70
  23. From: "asterisk" <sip:asterisk@172.18.0.34>;tag=as58458a47
  24. To: <sip:10.1.16.72>
  25. Contact: <sip:asterisk@172.18.0.34>
  26. Call-ID: 30d0b1dc56e0a470688d1e6000768567@172.18.0.34
  27. CSeq: 102 OPTIONS
  28. User-Agent: Asterisk PBX 1.6.2.18
  29. Date: Thu, 05 May 2011 08:46:05 GMT
  30. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  31. Supported: replaces, timer
  32. Content-Length: 0
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