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Brekeke SIP Server Wiki

  1. Brekeke SIP Server Basic Info
  2. Security
  3. Update & Upgrade
  4. Configuration
  5. Troubleshooting
    1. How to access to the Embedded Database
    2. How to check dropped packets on Linux
    3. How to get debug logs 
    4. How to get jstack information
    5. Brekeke SIP Server License and Activation
    6. Starting up
    7. Registering problem
    8. Trouble with SIP UAs
    9. Trouble with Calling
      1. 401 Unauthorized, 407 Proxy Authentication Required
      2. 404 Not Found, 484 Address Incomplete, 603 Decline
      3. 415 Unsupported Media Type, 488 Not Acceptable Here
      4. 482 Loop Detected, 483 Too Many Hops
      5. 486 Busy Here
      6. 603 Packet-Too-Big
      7. Call drops after getting connected
      8. Call drops after getting connected and caller hears ring back tone
      9. Call drops; only one-side voice; no voice
      10. Callee not ring, caller hears ring tome
      11. Disconnected during a call
      12. Neither a busy tone nor a ring back tone
      13. UA1 calls UA2, UA2 does not ring
      14. One way call
      15. Other response codes
      16. Port Allocation failed
      17. Can not find a response code
      18. Poor sound quality
    10. Thread resources may not be released when it is integrated with Responder5
  6. Developer's Guide
  7. Version History

There is not a busy tone nor a ring back tone

  • There are some phones that require to press “#” or “send botton” followed by a phone number. To ensure how to use a phone properly, please refer to the user manual of the phone or call the customer service of the phone maker.
  • Please view on the Start/Shutdown page at the Admintool to check Brekeke SIP Server status, and see if it is in “active” status.
  • If you are using Brekeke PBX, view on the Start/Shutdown page on Brekeke PBX, too and see if it is in “active” status.
  • Have you changed the port number at Brekeke SIP Server using other port number than port 5060? There are possibilities that it may not work properly depending on the UA or the gateway you choose.
  • Check the setting and network status of the caller’s phone. SIP port may be blocked by the Router or Firewall. Please open the port and RTP range at the router to allow communication flow.
  • Is the callee number (SIP URI) correct?
  • Is Interface setting correct on Brekeke SIP Server? Please ensure if Interface is displayed correctly on the Status page. For more details about Interface, refer to “Brekeke SIP Server Administrator’s guide – Server Status”.
  • If you have changed the Dial Plansettings on Brekeke SIP Server, please make sure they are set correctly.
  • While the caller is trying to call the callee, is there active session displayed on the [Sessions] page on Brekeke SIP Server? If there are active sessions listed, click on [Session ID] and go to [Session Detail]. Please check the followings:
[to-ip] - callee’s IP address
[rule] - dial plan rule applied
[session-status] - for current phase (check [phase] if your Brekeke SIP Server is before v.1.5.1.3) When [session-status] is set as "Inviting", calls won’t be sent to the callee correctly nor callee won’t respond.

If possible, please use network-analyzer, such as Ethereal, to check SIP packet

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« Disconnected during a callUA1 calls UA2, UA2 does not ring »
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