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Thread: firmware 5.1.17 missed call issue

  1. #1

    Default firmware 5.1.17 missed call issue

    hi everyone, I have three snom 300 phone running firmware 7.3.14 + PBX 2x running firmware 5.1.17 and i am not able to call via the missed call menu. When a call is coming from the outside the phone show the number of the caller in this way 2222111111. If i don't answer the call the number is saved to the missed call menu. If i am trying to go to the missed call menu and i am trying to show the details of the last missed call I see:

    the_caller_number@ip_sip_server_ITSP <sip: the_caller_number@ip_sip_server_ITSP>.

    If I am trying to call that number the PBX answer "user_not_found". I have tried to roll back to firmware 5.0.19 and everything is working. So i think the problem is in the latest firmware (5.1.17).
    I thought the problem it was in the sip address sent by the PBX to the phone so I have tried to edit this record saved by the phone:

    the_caller_number@sip_server_ITSP <sip:the_caller_number@ip_sip_server_ITSP>


    the_caller_number@sip_server_ITSP <sip:the_caller_number@ip_epygi_PBX>

    and everything is working.

    The question is:

    Is it possible to fix this "bug"?
    Is it possible to change the sip address of the caller via the call routing?

    I would like to edit and replace the ip_sip_server_ITSP with ip_epygi_PBX.

    thanks in advance

  2. #2
    Quadro Architect
    Join Date
    Jun 2006
    Around myself


    nitrox, I am not 100% sure, but I think this is not a bug, but a change in functionality, which is not properly reflected in release notes...

    This problem could be connected with the intruduction of "@"-based routing rules to allow direct calls for SIP addresses.
    To keep the story short - try to do the following:
    add in the Call Routing Table the following record:
    pattern=*@sip_server_ITSP:5060, call type=SIP, UES=<the extension used for regular calls to that ITSP>, keep all the rest values unchanged.

    This is the new feature of the 5.1 - ability to use full SIP address in the pattern filed of call routing enties. That feature is the reason of your problem, and the same feature could help you to solve that problem...

    Please test that solution and tell me if that works...

    Best regards,

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