CUE no user found

Hi all,

 

Going on with cue lab. I'm integrating it with cucme. I have only sip phones in cucme. All working fine.

Integration went ok, i can access CUE GUI but going through the wizard it cannot find any user in cme.

Under voice register pool i defined username and password for the sip phones in order to register them.

Any suggestion?

Could this depend by the fact that cme is 7.1 and cue i 7.0.1 ?

 thanks

smaikol

Comments

  • Hi,

    Several ways to check what you are missing:

    1. Create a user from the GUI and check the CME in CLI to check if it's the same what you did in CLI

    2. Access CME GUI and check if the user exists there. If not, follow #1.

    3. Create any user for SCCP phones and then check #1 & #2 to check if you are missing anything.

    Hope you can solve it by yourself by any of the above troubleshooting steps.

    HTH.

  • Hi,

     

    Thanks for suggestion, i already did some steps.

    Basically i manually created user and mailbox corresponding to my user in cme.

    I don't have sccp phones, only sip so i guess if this is a limitation of the cue this version,maybe only the wizard part.

    Still the vm doesn't work. If i call the pilot from a phone it works, if i call a dn and wait for the transfer to vm doesn't.

    Going on, thanks again, cheers

    smaikol

  • For Skinny phones import works definitely, but for SIP phones i remember i had to do it manually. At least on version that you are running.

     

  • yes, true, i had to create them manually.

    Now i can access voicemail being matched on primary extension in sip phones.

    My problem is now that when i call from one phone to the other i cannot leave a vm.

    The call is  dropped. I can see the VM DP matched but then drops. The codec is the same g711ulaw.

    I had some problem with the dialplan pattern command. After deleting it i can correctly access voice mail per user.

    Maybe mwi are not working properly?

     

    Cheers, smaikol

  • do you enable sip to sip calling under voice service voip ? 

    allow connections sip to sip

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