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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

UA1 calls UA2, UA2 does not ring

  • If Brekeke SIP Server, UA1, and UA2 are on the same network, please check the followings:
    • Check that UA1’s codec and UA2’s codec are the same.
    • Is UA2 displayed on the Registered screen?
    • Is the Dial Plan properly set up?
    • If INVITE authentication is turned on at Brekeke SIP Server, is the authentication information correctly set on UA?
    • Please check your devices manual for the appropriate settings.
  • When Brekeke SIP Server and UA2 are on different networks from UA1.
    • When UA2 is on a local network, [NAT traversal] > [Keep address/port mapping] should be set to ON in the [Configuration] menu to resolve NAT traversal on UA2 side. The setting doesn’t matter if UA2 can use STUN, and STUN supports the router(NAT)’s type on UA2’s side.
  • When UA1 and UA2 talk through Brekeke SIP Server and another SIP server:
    • Please check that your Dial Plan is set correctly.
    • Additionally, Brekeke SIP Server does not support multi-staged authentication
      so concerning communication through multiple SIP servers, please ensure that authentication will be done through only one server. You can disable authentication for specific devices in the Dial Plan ($auth=off).
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« Neither a busy tone nor a ring back toneOne way call »
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