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From Adam, 1 Week ago, written in Plain Text, viewed 3 times.
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  1. Connected to Asterisk 16.13.0 currently running on freepbx (pid = 2254)
  2. Core debug is still 3.
  3. freepbx*CLI> pjsip set logger on
  4. PJSIP Logging enabled
  5. <--- Transmitting SIP request (411 bytes) to UDP:192.168.5.9:5060 --->
  6. OPTIONS sip:192.168.5.9:5060 SIP/2.0
  7. Via: SIP/2.0/UDP 192.168.5.7:5180;rport;branch=z9hG4bKPjb35b417e-5e92-4488-be33-202936f785a0
  8. From: <sip:GSMGateway@192.168.5.9>;tag=471a96ee-40b2-4af3-96e0-a0d5d58dc69a
  9. To: <sip:192.168.5.9>
  10. Contact: <sip:GSMGateway@192.168.5.7:5180>
  11. Call-ID: b8c5455d-aba3-472d-8196-4f5bb28abedc
  12. CSeq: 20764 OPTIONS
  13. Max-Forwards: 70
  14. User-Agent: FPBX-15.0.16.77(16.13.0)
  15. Content-Length:  0
  16.  
  17. <--- Received SIP response (482 bytes) from UDP:192.168.5.9:5060 --->
  18. SIP/2.0 404 Not Found
  19. Via: SIP/2.0/UDP 192.168.5.7:5180;branch=z9hG4bKPjb35b417e-5e92-4488-be33-202936f785a0;received=192.168.5.7;rport=5180
  20. From: <sip:GSMGateway@192.168.5.9>;tag=471a96ee-40b2-4af3-96e0-a0d5d58dc69a
  21. To: <sip:192.168.5.9>;tag=as0ad99f3b
  22. Call-ID: b8c5455d-aba3-472d-8196-4f5bb28abedc
  23. CSeq: 20764 OPTIONS
  24. Server: TG100
  25. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  26. Supported: replaces, timer
  27. Accept: application/sdp
  28. Content-Length: 0

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