Any VoIP handset which supports SIP will normally work with the FB2700. 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 FB2700'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 FB2700 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
.
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.
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.
The VoIP status page shows the active registrations from handsets.
log-register
and log-register-debug
settings can provide a lot
of information about registrations and help diagnose any problems.