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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

Cannot complete calls. Calls are one-way; or neither side can hear the other side voice.

In the following examples, the call was made by UA1, and received by UA2. When Brekeke SIP Server and either UA1 or UA2 are on a different network:

In the Config menu, ensure that [RTP exchanger > Port mapping] is set to source port When Brekeke SIP Server does Near-End NAT Traversal:

Ensure that port forwarding is set correctly at the NAT router. If sound doesn’t get through, the port range between the Minimum Port and Maximum Port in the Config menu might not be forwarded to the Brekeke SIP Server.

Also, check here for more information.

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