Bizfon 2000 Guida Utente
Bizfon Manual II: Administrator's Guide
Administrator’s Menus
Bizfon2000 and Bizfon4000 (SW Version 4.1.x)
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The Use PSTN lines of the other device checkbox is used to
enable Bizfon to use the shared PSTN lines on a remote device.
This selection requires you to configure the Authorization
Parameters. Use the same named link to access the
Authorization Parameters table.
enable Bizfon to use the shared PSTN lines on a remote device.
This selection requires you to configure the Authorization
Parameters. Use the same named link to access the
Authorization Parameters table.
Fig. II-117: External PSTN Gateways page
The Authorization Parameters page is used to create
accounts for the remote Bizfons allowing them to connect the
Bizfon and share the available PSTN lines. The table on this
page lists all registered accounts and account information. It will
show the corresponding authentication parameters (username
and password) and date/time of the last registration.
accounts for the remote Bizfons allowing them to connect the
Bizfon and share the available PSTN lines. The table on this
page lists all registered accounts and account information. It will
show the corresponding authentication parameters (username
and password) and date/time of the last registration.
The Add functional button opens an Add Entry page where a
new account can be configured. A Username and a Password
is required for a new account on this page.
new account can be configured. A Username and a Password
is required for a new account on this page.
Fig. II-118: External PSTN Gateways – Authorization Parameters page
To use the shared remote PSTN lines
1.
Enable
the
Use PSTN lines of the other device checkbox.
2.
Press
Save to apply the selection.
3.
Enter
the
Authorization Parameters page.
4.
Create an account using a unique Username and a Password.
Gain Control
The Gain Control settings are used to define transmit and
receive gains. For FXS lines, Transmit Gain defines the phone
speaker volume and Receive Gain defines the volume of the
phone microphone. For FXO lines, Transmit Gain defines the
level of voice transmitted from Bizfon to the PSTN network and
Receive Gain defines the volume of voice received by Bizfon
from the PSTN network.
receive gains. For FXS lines, Transmit Gain defines the phone
speaker volume and Receive Gain defines the volume of the
phone microphone. For FXO lines, Transmit Gain defines the
level of voice transmitted from Bizfon to the PSTN network and
Receive Gain defines the volume of voice received by Bizfon
from the PSTN network.
The Gain Control page offers Transmit Gain and Receive
Gain drop down lists for each line that contains allowed gain
values, which can be set up by the administrator for every line.
Gain drop down lists for each line that contains allowed gain
values, which can be set up by the administrator for every line.
Please Note:
If the gain control has been configured incorrectly,
DTMF digits may not be properly recognized. Gain control
settings are strictly dependent on the location (country) of Bizfon
and the phone type. If a private PBX is attached to the FXO port
on the Bizfon, the voice level in the handset of the phone
connected to the Bizfon FXS port may be too loud (depending
on the PBX type and configuration). This can be adjusted by
decreasing the FXO Receive Gain to three or to zero.
settings are strictly dependent on the location (country) of Bizfon
and the phone type. If a private PBX is attached to the FXO port
on the Bizfon, the voice level in the handset of the phone
connected to the Bizfon FXS port may be too loud (depending
on the PBX type and configuration). This can be adjusted by
decreasing the FXO Receive Gain to three or to zero.
The Restore Default Gains button restores the default values.
Fig. II-119: Gain Control page
SIP Trunk Settings
The SIP Trunking service is used to build a tunnel between Bizfons and to use that tunnel for routing the SIP calls through the remote Bizfons.
When this service is enabled, slave Bizfons should be registered on the master Bizfon with the corresponding username/password. With the
appropriate configuration done on the master Bizfon, the master device can use the slave Bizfons for routing the SIP calls through them and
accessing peers located behind the slave Bizfon or recognized by it. This enables the master Bizfon to locate the slave, even when the network
settings, like IP address, SIP port and other settings are changed on the slave Bizfon.
When the SIP Trunking service is enabled, virtual tunnels between the master and its slaves are created. A possibility to use the created SIP trunks
will be automatically enabled in the
When this service is enabled, slave Bizfons should be registered on the master Bizfon with the corresponding username/password. With the
appropriate configuration done on the master Bizfon, the master device can use the slave Bizfons for routing the SIP calls through them and
accessing peers located behind the slave Bizfon or recognized by it. This enables the master Bizfon to locate the slave, even when the network
settings, like IP address, SIP port and other settings are changed on the slave Bizfon.
When the SIP Trunking service is enabled, virtual tunnels between the master and its slaves are created. A possibility to use the created SIP trunks
will be automatically enabled in the
table.
Optionally, a SIP trunk tunnel can be mutually established on two Bizfons allowing to route SIP calls back and forth. A Bizfon can be at the same
time configured both as a slave and as a master to the same remote device, i.e. the slave Bizfon can act as a master for the master device it is
registered on. For example, the Bizfon1 can act as a slave for the Bizfon2. In its turn, the Bizfon2 can act as a slave for the Bizfon1. With this
configuration and the corresponding routing rules added in the
time configured both as a slave and as a master to the same remote device, i.e. the slave Bizfon can act as a master for the master device it is
registered on. For example, the Bizfon1 can act as a slave for the Bizfon2. In its turn, the Bizfon2 can act as a slave for the Bizfon1. With this
configuration and the corresponding routing rules added in the
table on both devices, the SIP calls will be routed from Bizfon1 to Bizfon2
and vice versa.