In case your SIP phone connection hasn't been successful, please check the following:
- 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.bitrixphone.com
- 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. -
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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.
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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.
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Your SIP phone should support either TCP or UDP protocol, as
ip.b24-xxx-xxxxxxxxxx.bitrixphone.com
server is using only these protocols.If your SIP phone supports both protocols, then UDP is preferable.
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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).
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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).
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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.