SIP Phones connection issues

In case your SIP phone connection hasn't been successful, please check the following:

  1. Possibility to add a long server address

    When you connect SIP phone on the Telephony Users page, you need to use the data provided in this section for phone configuration:

    • server address (has the form ip.b24-хххх-хххххххххх - 38-symbols);

    • login (has the form phoneID, ID is equal to the employee's ID in Bitrix24);

    • password (is generated automatically, but can be edited).

    If your SIP phone does not allow you to add min. 38-symbols server address, you won't be able to connect it.
  2. It is preferable to use SIP phone with its default settings during connection. Several SIP phones may not be working if their default settings have been changed.

  3. It is preferable to have the last update installed on your SIP phones. Some SIP phones may not be working because they haven't been updated.

  4. Your SIP phone should support either TCP or UDP protocol, as server is using only these protocols.

    If your SIP phone supports both protocols, then UDP is preferable.

  5. Make sure you have STUN-server disabled in your SIP phone, as it may cause problems with traffic going through NAT (it results in conversation interruption\call break when the call is made).

  6. Make sure you have SIP - ALG option disabled in your office internet router settings. If the option is active, it may result in conversation interruption\call break).

  7. If you use SIP phones and To everybody type of call distribution, incoming calls cannot be transferred to more than 10 phones (devices) at the same time. This is Voximplant's limitation.

Refer to these articles for the list of verified devices and connection instructions:
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