18.6. Telephone handsets

Any VoIP handset which supports SIP will normally work with the FB2900. Most makes of handset actually allow multiple identities on the handset so it can appear as multiple handsets to one or more phone systems, but a typical installation will not normally need more than one identity per handset.

On the handset you will need to set a registrar and/or proxy which is usually either a host name or an IP address. This will need to refer to the FB2900's address.

The handset will also have some form of login or username and a password. Typically you would use the extension number or DDI as the username, but in an office PABX you may want people's names as the user name.

On the FB2900 you add a telephone configuration object (VoIP users) for each telephone, specifying a username and password. If the handset will need to connect from somewhere that is not on the local network (LAN) then you need to set local-only to false.

Note

For certain endpoint configurations you may find that setting the global security-replies to false may be needed to make registrations work. If using Asterisk as a client it has been found that modifiying its configuration such that the qualify option is disabled can work around the issue without requiring the FireBrick to respond to potential scanning attempts.

The normal way to identify a telephone user is by the name in the config matching the local part of the From address. However, where a password is set, the Authorization will have to incude the matching username defined for the user.

It is also possible to match a user where the whole From address against the uri field. This is normally used for things like a call recording server referring a call to another number.

In any case, the telephone user configuration also has to match routing table, allowed IP addresses, and any local-only restrictions in order to match.

Note

It is possible to send an Authorization even before a challenge including a username. As a fallback, this can be used to match to a user even if name or uri do not match.

You also need to set the extn and ddi for the phone. In order to make external calls you need to select a carrier to use.

Tip

There are a number of other settings which can be useful. The display-name will show the caller name on internal calls. You can also limit the number of concurrent calls. Some other features are described in corresponding sections below.

The VoIP status page shows the active registrations from handsets.

Tip

The log-register and log-register-debug settings can provide a lot of information about registrations and help diagnose any problems.