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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

Disconnection occurs during a call

Either caller or callee can not hear anything before call gets disconnected.
Or sudden disconnection

  • When you use a machine which implement SIP protocol improperly, SIP session fails and may get disconnected after about 1 minutes.
  • When sounds do not go through properly, there are cases that either the UA (phone) or Brekeke SIP Server disconnect the session. Please check timeout setting.
    • Brekeke SIP Server : Go to [Config] > [RTP]> [RTP Session Timeout (ms)]
    • Brekeke PBX : Go to [Options] > [Media Server system settings] > [RTP Session Timeout (ms)] Please refer to one way call for more details.
  • The call may have reached to [Talking timeout]. Please check [Talking timeout] setting on both Brekeke SIP Server and Brekeke PBX.
    • Brekeke SIP Server > [Config] > [SIP(Advanced)] > [Timeout] > [Talking Timeout (ms)]
    • Brekeke PBX > [Options] and check [Talking timeout ] for both [PBX system setting]and [Media Server system setting]
  • Since ITSP tends to use old version of SIP standard, there are some cases that they can not process lr parameters. Please set [B2B-UA mode] on from SIP Server >[Configuration]>[SIP], or set $b2bua = true in the applicable deploy pattern in the dial plan.
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