Bizfon 4000 User Guide

Bizfon Manual II: Administrator's Guide

Manual II:

Administrator’s Guide

Edition 1 SW Release 3.1.23 and higher, May 2006

Bizfon Manual II: Administrator's Guide

Table of Contents

Manual I: see Installation Guide

Step-by-step guide to install and configure Bizfon basically

Manual II: Administrator's Guide

About this Administrator's Guide ................................................................................................................ 4

Bizfon’s Graphical Interface....................................................................................................................... 5

Administrator’s Main Page ............................................................................................................................................ 5

Recurrent Buttons ....................................................................................................................................................... 6

Recurrent Functional Buttons........................................................................................................................................ 6

Entering a SIP Addresses correctly ................................................................................................................................ 6

Administrator’s Menus .............................................................................................................................. 7

System Menu ............................................................................................................................................................. 7

System Configuration Wizard .................................................................................................................................... 7

Internet Configuration Wizard ................................................................................................................................... 9

Status ...................................................................................................................................................................11

General Information ............................................................................................................................................11

Network Status ...................................................................................................................................................11

Lines Status .......................................................................................................................................................13

Memory Status ...................................................................................................................................................14

Hardware Status .................................................................................................................................................14

SIP Registration Status ........................................................................................................................................15

MGCP Registration Status.....................................................................................................................................15

IP Routing Configuration..........................................................................................................................................15

Configuration Management ......................................................................................................................................17

Events ..................................................................................................................................................................18

Time/Date Settings .................................................................................................................................................21

Mail Settings ..........................................................................................................................................................21

SMS Settings .........................................................................................................................................................22

Firmware Update ....................................................................................................................................................23

Networking Tools....................................................................................................................................................24

Diagnostics ............................................................................................................................................................25

Automatic Provisioning ............................................................................................................................................25

Features................................................................................................................................................................26

User Rights Management .........................................................................................................................................26

Users Menu............................................................................................................................................................29

Extensions Management ..........................................................................................................................................29

Extension Codecs ................................................................................................................................................38

Call Park Service .................................................................................................................................................39

Authorized Phones Database ................................................................................................................................39

Call Back Services ...............................................................................................................................................40

Upload Universal Extension Recordings ..................................................................................................................41

Receptionist Management ........................................................................................................................................41

Extensions Directory ...............................................................................................................................................44

Telephony Menu ........................................................................................................................................................46

Call Statistics .........................................................................................................................................................46

RTP Statistics .....................................................................................................................................................47

SIP Settings...........................................................................................................................................................48

RTP Settings ..........................................................................................................................................................49

NAT Traversal Settings ............................................................................................................................................50

Line Settings..........................................................................................................................................................52

Onboard Line Settings .........................................................................................................................................52

Bizfon Manual II: Administrator's Guide

IP Line Settings ..................................................................................................................................................54

Supported SIP Phones.....................................................................................................................................54

Loopback Settings ...............................................................................................................................................55

FXO Settings..........................................................................................................................................................55

Gain Control ..........................................................................................................................................................57

Call Routing ...........................................................................................................................................................57

Best Matching Algorithm ......................................................................................................................................63

VoIP Carrier Wizard ................................................................................................................................................66

RADIUS Client Settings ...........................................................................................................................................67

Voice Mail Common Settings ....................................................................................................................................69

Dial Plan Settings ...................................................................................................................................................70

Internet Uplink Menu ..................................................................................................................................................71

PPP/ PPTP Settings .................................................................................................................................................71

Advanced PPP Settings ........................................................................................................................................71

VPN Configuration ..................................................................................................................................................72

Dynamic DNS Settings ............................................................................................................................................80

Firewall and NAT ....................................................................................................................................................81

Advanced Firewall Settings ...................................................................................................................................81

Filtering Rules ........................................................................................................................................................82

Service Pool .......................................................................................................................................................84

IP Pool...............................................................................................................................................................84

IDS Log.................................................................................................................................................................86

LAN Services Menu.....................................................................................................................................................87

DNS Settings .........................................................................................................................................................87

DHCP Settings for the LAN Interface .........................................................................................................................87

Administrator’s Additional Features ..............................................................................................................................89

Incoming Call Blocking and Outgoing Call Blocking .....................................................................................................89

Logout......................................................................................................................................................................89

Appendix: Extension User's Welcome Page ................................................................................................ 90

Appendix: System Default Values............................................................................................................. 91

Administrator Settings ................................................................................................................................................91

Extension Settings .....................................................................................................................................................94

Appendix: Software License Agreement .................................................................................................... 95

Manual III: see Extension User's Guide

Describes detailed the menus available for extension users and includes further all call codes at a glance.

Bizfon Manual II: Administrator's Guide About this Administrator's Guide

About this Administrator's Guide

The Bizfon Manual is divided into three parts:

• Manual-I: Installation Guide

gives step-by-step instructions to provision the Bizfon IP PBX and configure the phone extensions with the Bizfon SIP Server. After successfully
configuring the Bizfon IP PBX, users will be able to make SIP phone calls to remote Bizfon devices, make local calls to the PSTN and access
the Internet from devices connected to the LAN.

• Manual-II: Administrator's Guide explains all Bizfon management menus available for administrators only. It includes a list of all System Default

Values.

• Manual-III: Extension User's Guide explains all Bizfon management menus available for extension users. A list of all call codes can be found

there, too.

This Administrator's Guide explains Bizfon4000, which is shown as the reference system.

This guide contains many example screen illustrations. Since Bizfon IP PBXs offer a wide variety of features and functionality, the example screens
shown may not appear exactly the same for your particular Bizfon IP PBX as they appear in this manual. The example screens are for illustrative and
explanatory purposes, and should not be construed to represent your own unique environment.

Bizfon’s Graphical Interface describes to the Bizfon's graphical user interface and explains all recurrent buttons.

Administrator’s Menus explains the Administrator's management pages according to the menu structure shown on the main page of the Bizfon

management.

Administrator’s Additional Features explains some input-options for administrators only, that may be selected from the extension user's main

page.

Appendix: Extension User's Welcome Page includes a preprinted MS-Word form that allows the administrator to inform his extension user with

all individually needed addresses and phone numbers.

Appendix: System Default Values lists all factory defaults.

Appendix: Software License Agreement includes the contract for using Bizfon's hardware and software.

Bizfon4000 (SW Version 3.1.x) 4

Bizfon Manual II: Administrator's Guide Bizfon's Graphical Interface

Bizfon’s Graphical Interface

Administrator’s Main Page

As the result of logging in as an administrator, the page Bizfon Management is displayed with a table of active calls (including information about call
peers, call duration and start time) at the startup. Here the administrator may access the following settings and perform the actions:

Fig. II-1: Bizfon4000 Management

System Menu

• System Configuration Wizard

• Internet Configuration Wizard

• Status

• IP Routing Configuration

• Configuration Management

• Events

• Time/Date Settings

• Mail Settings

• SMS Settings

• Firmware Update

• Networking Tools

• Diagnostics

• Automatic Provisioning

• Features

• User Rights Management

Telephony Menu

• Call Statistics

• SIP Settings

• RTP Settings

• NAT Traversal Settings

• Line Settings

• FXO Settings

• Gain Control

• Call Routing

• VoIP Carrier Wizard

• RADIUS Client Settings

• Voice Mail Common Settings

• Dial Plan Settings

Internet Uplink Menu

• PPP/ PPTP Settings

• VPN Configuration

• Dynamic DNS Settings

• Firewall and NAT

• Filtering Rules

• IDS Log

Users Menu

• Extensions Management

• Receptionist Management

• Extensions Directory

LAN Services Menu

• DNS Settings

• DHCP Settings for the LAN

Interface

2Logout

The functional button Renew Wan IP Address appears on the administrator’s main Bizfon Management page if the Bizfon device acts as a DHCP
client. The Renew WAN IP Address button is used to get a new WAN IP address in case, e.g., the Bizfon moves to another network.

The functional button Establish Your Internet Connection Now respectively Terminate Your Internet Connection Now occurs on the Bizfon
Management page if PPPoE is used as WAN interface protocol.

The link Please Check Your Pending Events will be displayed on the administrator Main Menu page if new system events exist. The link leads to
the Events page that can be also accessed from the System menu.

The list of Users currently logged into the system is seen in the lower right corner of the Administrator's Main Menu. Information about IP address
user accessed Bizfon GUI from, the username user is logged in and the time until the next automatically logout is provided herein. The idle session
timeout is set to 20 minutes. If no action is performed during that time, user will be automatically moved to the Login page and will be requested to
login again.

The link Refresh in occurs in the upper right corner beside the field displaying the number of seconds until the next refresh and is used to perform a
manual reload of the page. If a page with a Refresh counter is left opened, the session time-out counter will be updated periodically and the logout
timeout will never expire.

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Bizfon Manual II: Administrator's Guide Bizfon's Graphical Interface

Recurrent Buttons

Button Description Button Description

This button leads back to the previous
page of a fixed sequence of pages (used
mainly in wizards).

This button takes you to the next page of
a fixed sequence of pages (used mainly in
wizards).

This button discards the latest not yet
confirmed entries.

This is the last button of a fixed sequence
of pages that completes and saves the
entries of the whole sequence.

This button opens the help page
belonging to the currently active Bizfon
management page.

This button opens a window where the last
inserted IP addresses are listed. It is
basically a clipboard that helps the user to
make a quick selection of an IP address in
case it has been already used in the past,
thus avoiding typing it in again. The
clipboard can hold up to 10 IP addresses
and a new IP address will replace the
oldest one from the list.

This button leads back to the page you have
been on before.

This button confirms an operation you started
before.

This button confirms an operation you chose
before.

This button discards an operation you chose
before.

This button saves the settings modified on the
currently active management page.

This button opens a window where the last
inserted SIP addresses are listed. It is basically
a clipboard that helps the user to make a quick
selection of a SIP address in case it has been
already used in the past, thus avoiding typing it
in again. The clipboard can hold up to 10 SIP
addresses and a new SIP address will replace
the oldest one from the list.

Recurrent Functional Buttons

In connection with tables, the following buttons among others will appear:

Functional Button Description

Add Allows adding a new record to the displayed table. A new page will be displayed to enter any new settings.

Edit

Delete Deletes the selected entry(s) of a table. A warning message will demand a confirmation before deleting an existing entry.

Select All Selects all table entry(s)for example for further deletion.

Inverse Selection Inverses an existing selection of table entry(s). If no entries are selected, clicking the button will select all records.

Refresh in...

Most of the tables offer the option to sort the entries in ascending or descending order by clicking the headings of the columns. A small arrow next to
the column heading will show the direction of sorting - upward or downward. The entries of the table can be selected by using the corresponding
checkboxes in order to edit or delete them.

Allows modifying the settings of the record selected by its checkbox. Normally only one record may be selected. A new
page will be displayed to enter the modified settings.

May appear in the upper right corner of a page. It displays the number of seconds remaining until the next refresh of the
page and it may be used to reload the page manually.

Entering a SIP Addresses correctly

Calls over IP are implemented based on Session Initiating Protocol (SIP) on the Bizfon. When making a call to a destination that is somewhere on
the Internet, an SIP address must be used.

SIP addresses must be specified in one of the following formats:

“display name” <username@ipaddress:port>
“display name” <username@ipaddress>
username@ipaddress:port
username@ipaddress
username

The following combinations can be used for your convenience:

• *@ipaddress - any user from the specified SIP server

• username@* - a specified user from any SIP server

• *@* - any user from any SIP server

The display name and the port number are optional parameters in the
SIP address. If a port is not specified, 5060 will be set up as the default
one. The range of valid ports is between 1024 and 65536.

A flexible structure of wildcards is allowed. In comparison with a
wildcard, the “?” character stands for only one unknown digit and the “*”
character stands for any number of any digits.

Please Note: Wildcards are available for caller addresses only. No

wildcard characters are allowed for called party addresses. Exceptions
are addresses in the Supplementary Addresses table that is used by
Outgoing Call Blocking and Hiding Caller Information Settings
services. To use “*” and “?” alone (as non wildcard characters), use “\*”
and “\?” correspondingly.

Bizfon4000 (SW Version 3.1.x) 6

Bizfon Manual II: Administrator's Guide Administrator's Menus

Administrator’s Menus

System Menu

Fig. II-1: System Menu in Dynamo theme

Fig. II-2: System Menu in Plain theme

System Configuration Wizard

The System Configuration Wizard is the helpful tool for the administrator to define the Bizfon’s Local Area Network settings and to specify regional
configuration settings to make Bizfon operational in its LAN. The System Configuration Wizard MUST be run upon Bizfon's first startup to make
sure that it works properly in its network environment. The Wizard allows navigating through the following basic configuration parameters and
settings:

• System Configuration (see below)

• DHCP Settings for the LAN Interface

• Regional Settings and Preferences (see below)

• Emergency Codes and PSTN Access Codes Settings (see below)

DHCP Settings for LAN are described in the chapters
below while LAN configuration and regional settings will be
described in the current chapter.

Please Note: It is strongly recommended to leave the

factory default settings if their meanings are not fully
clear to the administrating person.

Fig. II-3: System Configuration Wizard - Start page

Bizfon4000 (SW Version 3.1.x) 7

Bizfon Manual II: Administrator's Guide Administrator's Menus

The System Configuration page contains the host name,
IP address and Subnet Mask information about the Bizfon
LAN interface. These settings make Bizfon available to the
internal network.

The System Configuration page offers the following input
options:

Host Name requires a host name for the Bizfon device.

IP Address requires the Bizfon host address for the LAN

interface.

Subnet Mask requires the Bizfon hosts’ Subnet Mask.

Fig. II-4: System Configuration Wizard - System Configuration page

The Regional Settings and Preferences are used to
select settings specific to the location of the Bizfon.

This is important for the functionality of the voice
subsystem.

The Regional Settings and Preferences page has two
drop down lists to select the Location (country) and a
corresponding Timezone and a manipulation radio button
group to choose:

• System Language – selection is available only when

custom Language Pack has been uploaded and is
used to enable custom language for system voice
messages or turn back to default (English).

• GUI Theme - selection used to select the GUI theme

style of the web based configuration pages.

Fig. II-5: System Configuration Wizard - Regional Settings page

The Emergency Codes and PSTN Access Codes
Settings are used to configure the dial plan parameters
used in the routing mode.

The Emergency Codes text field requires the PSTN
numbers of the emergency or lifeline services. Multiple
emergency codes, separated by commas, can be inserted
in this field. For each emergency code, a routing pattern
will be generated in the Call Routing Table, which will allow
to make fast and easy calls to emergency destinations.

The PSTN Access Code drop down list allows to select
the prefix code for accessing the PSTN line in the routing
mode. Dialing the digits inserted in this text field will
provide the PSTN dial tone, when acting in the routing
mode or making routing calls (for routing calls an additional
“0” will need to be dialed first.)

Fig. II-6: System Configuration Wizard - Emergency Codes and PSTN Codes Settings page

Bizfon4000 (SW Version 3.1.x) 8

Bizfon Manual II: Administrator's Guide Administrator's Menus

Internet Configuration Wizard

The Internet Configuration Wizard is the helpful tool for the administrator to configure the WAN interface settings and to adjust Bizfon’s
connectivity with an external network. The Internet Configuration Wizard MUST be run if it is desired for Bizfon to be connected to the
Internet.

All the settings of the Internet Configuration Wizard are
described in the chapters below except those for the IP
settings, which will be described in this chapter.

Please Note: It is strongly recommended to leave the

factory default settings if their meanings are not fully clear
to the administrating person.

The Wizard allows navigating through the following basic configuration parameters and settings:

Fig. II-7: Internet Configuration Wizard - Start page

• Uplink configuration (see below)

For WAN Interface protocol PPPoE:

• PPP/ PPTP Settings

• WAN Interface Configuration (see below)

• DNS Settings

The Switch to Auto Provisioning link moves you to the
provisioning mechanism.

The Uplink Configuration page allows to select the Bizfon‘s WAN
interface connection type and its bandwidth settings. These settings
make Bizfon available to the external network.

Depending on the Uplink Interface Protocol selection, the page
following the Uplink Configuration page is different. Thus if
PPPoE is selected, the next page will be PPP Configuration, while
selecting Ethernet will bring up the WAN IP Configuration page.

The Uplink Configuration page offers the following components:

The WAN Interface Protocol radio buttons are used to choose the
protocol depending on the requirements of the ISP (Internet Service
Provider):

PPPoE - turns on the PPP over Ethernet connection type.

PPTP – turns on the Point to Point Tunneling Protocol (PPTP)

interface used for the connection between Bizfon and ADSL
modem. Fixed IP address configuration is needed in this case.

Ethernet - turns on the Ethernet connection type.

The WAN Interface Bandwidth settings allow the specification of the upstream and downstream speeds in kbit/s, helping to assure the quality of IP
calls. An IP call looses the voice quality if there will be no available bandwidth. In case of reaching the borders of bandwidth, another IP call will be
declined.

The bandwidth provided by the ISP has to be specified in the text fields Upstream Speed and Downstream Speed. The default entry in both fields
is 10000, the maximum bandwidth of a 10 MB Ethernet. But most providers offer smaller bandwidth than 10000 kbit/s.

The bandwidth required by an IP call depends on the codecs used and is listed in the tables below:

For WAN Interface protocol PPTP:

• WAN IP Configuration (see below)

• PPP/ PPTP Settings

• WAN Interface Configuration (see below)

• DNS Settings

Automatic Provisioning page where Bizfon can be configured by the automatically

For WAN Interface protocol Ethernet:

• WAN IP Configuration

• WAN Interface Configuration (see below)

• DNS Settings

Fig. II-8: Internet Configuration Wizard - Uplink Configuration page

Bizfon4000 (SW Version 3.1.x) 9

Bizfon Manual II: Administrator's Guide Administrator's Menus

Required Bandwidth for Standard Packets:

Needed bandwidth in kbit/s using the Codecs: Packet

Size in

msec.

G.711u/G.711a G.726-16 G.726-24 G.726-32 G.726-40 G.729a G723-5.3 G723-6.3 iLBC-13.33

10 105 58 66 73 81 50 - - -

20 84 37 45 53 60 29 - - -

30 77 30 38 45 53 22 19 20 27

40 73 26 34 42 50 18 - - -

50 71 24 32 40 48 16 - - -

60 70 23 30 38 46 15 12 13 60

Required Bandwidth for Encrypted Packets (in the case a VPN is used):

Needed bandwidth in kbit/s using the Codecs: Packet Size in

msec.

G.711u/G.711a G.726-16 G.726-24 G.726-32 G.726-40 G.729a G723-5.3 G723-6.3 iLBC-13.33

10 148 98 105 117 123 92 - - -

20 105 59 65 74 80 49 - - -

30 91 43 52 60 66 35 33 34 41

40 84 37 45 53 61 29 - - -

50 80 33 41 48 56 25 - - -

60 77 30 37 46 53 22 19 20 27

The Min Data Rate text field requires the amount of upstream bandwidth that ought to remain for data applications even if voice applications use the
entire available upstream bandwidth. The value selected here needs to be smaller than the upstream bandwidth and is measured in kbit/s.

The WAN IP Configuration page only is displayed if Ethernet or
PPTP has been selected to be the uplink protocol. It offers the
following components:

The Assign automatically via DHCP radio-button selection
switches to automatic retrieval of the WAN IP address from a
DHCP server at the ISP/uplink.

Please Note: DHCP referred to here is the one running on the

provider’s side and not the Bizfon’s personal DHCP server.

The Assign Manually radio-button switches to the manual
adjustment of IP settings. This selection requests the following
parameters:

IP Address requires the IP address for the Bizfon WAN interface.

Subnet Mask requires the subnet mask for the Bizfon device WAN

interface.

Default Gateway requires the IP address of the router all packets
are sent to, for example, the router of the provider.

Fig. II-9: Internet Configuration Wizard - WAN IP Configuration page

The WAN Interface Configuration page may be used to modify
the MAC address of the Bizfon. This might be necessary, if the ISP
(Internet Service Provider) requires a certain MAC address, for
example, for authentication. The page offers the following
components:

MAC Address Assignment manipulation radio-buttons:

• This Device turns to the default MAC address of the Bizfon.

• User Defined requires user defined MAC Address.

MTU drop down list allows to select the maximum packet size on
the Ethernet (in bytes). MTU is used to fragment the packets before
transmitting them to the network. MTU preferred value is
dependent on the Ethernet connection type. The default MTU size
is 1500 Bytes for Ethernet and 1400 Bytes for PPPoE.

Bizfon4000 (SW Version 3.1.x) 10

Fig. II-10: Internet Configuration Wizard - WAN MAC Address Configuration page

Bizfon Manual II: Administrator's Guide Administrator's Menus

Status

The system status window displays non-editable tables providing extensive status information about Bizfon: General Information, Network Status,

Lines Status, Memory Status, Hardware Status, SIP Registration Status and MGCP Registration Status. The links on this page lead to device

Transfer Statistics, user mailboxes and supplementary services configuration pages.

The System Status page has several tables providing system information.

General Information

The General Information page includes the following
information:

• Uptime duration - Period Bizfon is on since last reboot.

• Device Hostname - Bizfon device host name.

• Bizfon Operating System - Bizfon operating system

version.

• Application Software - Software and file system versions

of the Bizfon.

• Boot Loader - Bizfon boot loader version.

•

DSP Software - Bizfon DSP software version and the

date of build.

Fig. II-11: Bizfon Status - General Information page

Network Status

The Network Status page includes the following information
about Interfaces:

Interface Name lists the Network interfaces available on the
Bizfon (LAN, WAN, IPSec and a number of PPPs, depending on
the number of active PPP connections).

IP Address lists the IP addresses corresponding to each
network interface.

Subnet Mask lists the subnet masks corresponding to each
network interface.

Properties lists either the MAC address corresponding to each
network interface on the Bizfon or PPTP, L2TP and IPSec peer
IP address if an active VPN (IPSec or PPP) interface exists.

Monitor includes links to survey LAN, WAN, IPSec and PPP
traffic correspondingly. The VPN traffic link will be displayed only
if a VPN has been configured. The selection of these links will
open a new window with a table of network traffic statistics on
the selected interface:

• Received Bytes

• Received Packets

• Received Errors

• Received Drop Errors

• Received Overrun Errors

• Received MultiCast Packets

When opening the corresponding interface statistics window, no traffic values are displayed at first. Then every one minute, traffic statistics will be
updated. The tables serve as a kind of counter.

DNS Server, Alternative DNS Server and Default Gateway - displays the Bizfon settings corresponding to what has been configured with the

System Configuration Wizard.

Services (NTP Server and Client, DHCP Server and Client, DNS, Firewall, NAT, PPP, IDS) statuses: stopped or running.

View VPN Status link refers to the

Transfer Statistics - link to the Transfer Statistics page.

• Transmitted Bytes

• Transmitted Packets

• Transmitted Errors

• Transmitted Drop Errors

• Transmitted Carrier Errors

• Transmitted Collisions

VPN Configuration page where all VPN (IPSec, PPTP and L2TP) connections can be viewed and edited.

Fig. II-12: Bizfon Status Network Status page

Bizfon4000 (SW Version 3.1.x) 11

Bizfon Manual II: Administrator's Guide Administrator's Menus

The Transfer Statistics page allows a user-defined statistic
table depending on the transmit/receive value (criteria),
interface type and time period. It contains the following
components:

Time Range of statistic table - the drop down list includes
the period(in days) statistics data is to be collected and the
corresponding diagram charts are to be built.

Interface - the drop-down list offer the values:

• WAN - Wide Area Network (WAN) events only

• LAN - Local Area Network (LAN) events only

When Show also as readable values checkbox is selected,
an additional table with statistics values will be displayed on
the next page.

Fig. II-13: Transfer Statistics page

The area Receive Values:

• Receive Bytes - number of received bytes

• Receive Packets - number of received Ethernet packets

• Receive Errors - number of received packets containing

errors

• Receive Drop Errors - number of received packets that

have been discarded

• Receive Overrun Errors - number of received overrun

errors that occur when the receive buffer is not large
enough to hold all incoming packets. This error mostly
appears because of a slow receiving system.

• Receive MultiCast Packets - number of received

broadcast packets

The area Transmit Values:

• Transmit Bytes - number of transmitted bytes

• Transmit Packets - number of transmitted Ethernet

packets

• Transmit Errors - number of transmitted packets

containing errors

• Transmit Drop Errors - number of transmitted packets

that have been discarded

• Transmit Carrier Errors - number of transmit carrier

errors that occur because of a defective or lost
connection on the Ethernet link

• Transmit Collisions - number of transfer errors that

occurred during a simultaneous packet transmission
from both sides

Fig. II-14: Transfer Statistics Diagram Chart

Bizfon4000 (SW Version 3.1.x) 12

Bizfon Manual II: Administrator's Guide Administrator's Menus

To show the Transfer Statistics Diagram Charts, select the desired criteria and click Save to generate the corresponding chart and the table with
transfer statistics values (if enabled). The letters M and K used in the legend of the displayed diagrams show the total number of specified criteria: K
means thousands and M millions. Reset Statistics button is used to reset the chart and the table (if enabled).

Lines Status

The page Bizfon Status - Lines Status shows the current status of each of the FXS, IP and FXO lines with all details of the attached extension.
Since only one line of information can be displayed at a time, the Line, IP Line and FXO functional buttons are used to navigate through the other
lines’ information.

The Lines Status table displayed for FXS and IP lines include a group of static and dynamic parameters. Static parameters are displayed, always,
while dynamic parameters only appear whenever an event takes place on the extension.

Static Parameters:

Extension shows the extension number of the selected telephone line.

Display Name shows the corresponding name.

Phone State may have the value on hook or off hook.

Number of Active Calls shows the number of calls that are currently

present on the phone.

Dynamic Parameters:

Call State shows the current state of the extension (in voice mail, in
call, waiting, busy, call out, ring in, etc.).

Caller Party appears whenever a call is received and indicates the
caller extension and the IP address or a phone number, depending on
the call type.

Called Party appears whenever a call is placed and indicates the
destination extension and the IP address or a phone number,
depending on the call type.

Call Type shows whether the call is Internal or External and whether it
is a PSTN, PBX or an IP Call.

Call Start Time shows the call start date and time.

Call Duration shows the current call duration.

RX Codec shows the codec used to encrypt the incoming packets. TX
Codec shows the codec used to encrypt the outgoing packets. If RX

and TX codecs are the same, one Codec field will be displayed instead.

Fig. II-15: Lines Status - Line Status page upon established call

There is a list of supplementary services with their statuses for each
telephone line: Enabled, Disabled or -in case of the services Incoming
and Outgoing Call Blocking, Speed Calling, Hiding Caller Info and
Voice Mailbox - the number of Entries in the corresponding service table.
Thus the administrator may follow and will be notified about services
running on Bizfon for every line. The services are designed as links that
guide the administrator to the corresponding service page of the selected
user.

Fig. II-16: Bizfon Status - Lines Status

Bizfon4000 (SW Version 3.1.x) 13

Bizfon Manual II: Administrator's Guide Administrator's Menus

The Lines Status table of each FXO Line gives information about the
Allowed Call Types, shows the extension number (attendant or routing
client), shows to whom the Incoming Call is Routed To and displays the
State of the line (Free or Busy).

Fig. II-17: Line Status - FXO Status page

Memory Status

The Memory Status page includes tables with the available User
Space information for each extension. These tables display the space

used by the voice mailbox and uploaded/recorded system greetings,
and the free and total space (counted in minutes/seconds) for every
extension. The page includes the following information:

Memory Size shows total memory space (counted in
minutes/seconds) available on the Bizfon and assigned to all
extensions.

The table’s links lead the administrator to the extension settings page
where User Space may be altered.

System Memory row indicates the space occupied by the universal
extension recordings. Link refers to the

Recordings

be uploaded.

Call Statistic shows the current number of call statistic entries.

page where universal extension system messages may

Upload Universal Extension

Fig. II-18: Memory Status page

Hardware Status

The Hardware Status table displays a list of the hardware devices
present and currently available on the Bizfon board. The hardware
device version number and additional comments about its state are
indicated here.

Bizfon4000 (SW Version 3.1.x) 14

Fig. II-19: Hardware Status page

Bizfon Manual II: Administrator's Guide Administrator's Menus

SIP Registration Status

The SIP registration Status is a table displaying the SIP registration
status of the Bizfon extensions.

The table contains a list of all the registered extensions of Bizfon,
information about SIP registration states for them, addresses of SIP
servers where they are registered (if so), registration date and time, as
well as SIP registration names. By clicking on the row heading, the table
will be sorted by the selected column. Upon sorting (ascending or
descending), arrows will be displayed next to the column heading.

The links inside the table link you to the

Extensions Management page

where the SIP registration settings may be altered.

The Detected Connection Type field displays the connection type
Bizfon currently is acting in (direct connection or behind NAT). If Bizfon
is acting behind NAT, the NAT machine IP address is also displayed.

Registered IP Lines table lists the IP lines and remote extensions
registered on the Bizfon. Table indicates the actual IP addresses of the
remote devices, the usernames by which the devices have been
registered on the Bizfon, as well as the registration status information.

Fig. II-20: SIP Registration Status page

MGCP Registration Status

The MGCP registration Status page is only present when the MGCP
IP phone is registered on the Bizfon (see

IP Line Settings).

The table on the page lists the MGCP IP lines and remote extensions
registered on the Bizfon. Table indicates the actual IP addresses of the
remote MGCP devices, the usernames by which the devices have been
registered on the Bizfon, as well as the registration status information.

Fig. II-21: MGCP Registration Status page

IP Routing Configuration

Routing is used to relay information across the Internet from a source to a destination. Along the way, at least one intermediate node is typically
encountered. Routing is often confused with bridging, which may seem to accomplish precisely the same thing to the casual observer.

Bizfon’s IP Routing service allows to route IP packets from one destination to another (or to a specified router) through Bizfon or a Bizfon VPN.

The IP Routing Configuration page is used to make IP Static, IP Policy and VPN routes for IP packets routing and has three tables. Entries in the
tables are colored according to the state of the route, i.e. yellow for disabled routes, green for successfully enabled routes and red for
enabled routes.

IP Static Routes are used to forward IP packets from the
Network, where the Bizfon is connected, to the specified
destination.

The IP Static Routes table displays all established IP static
routes with their parameters: Target State for the state of the
route (enabled or disabled), Actual State for the state of the route
connection (up, down or erroneous), Route To for the subnet
where the incoming packets should be routed and Via IP Address
for the router IP address incoming packets should be routed
through.

Add opens the Add IP Static Route page where a new static
route can be established.

Fig. II-22: IP Static Routing table

erroneously

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Enable/Disable are used to activate/deactivate selected route(s).
At least one route should be selected in order to use these
functions, otherwise the error message appears: “No record(s)
selected.”

The page Add IP Static Route offers the following components:

Route To requires the IP address and subnet mask of the
destination the IP packet ought to be forwarded to.

Via IP address requires the IP address of the subsequent router
for IP packet forwarding to the specified destination.

Attention: The rule with the longest subnet (smallest IP range)

will take effect when having two or more IP Static routing rules
with the coinciding subnets.

Fig. II-23: Add IP Static Routing page

IP Policy Routes allow IP packets forwarding to the specified
router depending on the source IP address as well as defining the
priority for the current routing rule.

The IP Policy Routes table displays all specified IP policy routes
with their parameters: Target State for the state of the route
(enabled or disabled), Actual State for the state of the route
connection (up, down or erroneous), Priority for the route priority,

Route From for the subnet, routed packets come from and Via IP
Address for the router IP address incoming packets should be

routed through.

Add opens the page Add IP Policy Route to establish a new
policy route.

Enable and Disable are used to activate or to deactivate the
selected route(s).

Raise and Lower Priority are used to increase or to decrease the
priority of the selected policy route(s) by one. At least one route
should be selected to use these functions, otherwise the error

Fig. II-24: IP Policy Routing table

message appears: “No record(s) selected.”

The page Add IP Policy Route offers the following input options:

Priority requires a numeric value (from 1 to 252) to define the
priority of the routing rule. The lower the number, the sooner the
routing rule will take effect (higher priority).

From requires the packet source IP address and subnet mask of
the specified destination to match with the rule.

Via IP address requires the IP address of the subsequent router
for IP packet forwarding.

Fig. II-25: Add IP Policy Route page

The VPN Routes allow IP packets forwarding through the PPTP
and L2TP tunnels of the Bizfon. If no PPTP/L2TP connections
exist on Bizfon, no VPN routes can be generated.

The VPN Routes table displays all generated VPN routes with
their parameters: Target State for the state of the route (enabled
or disabled), Actual State for the state of the route connection
(up, down or erroneous), Route To for the subnet where the
incoming packets should be routed, Via Tunnel for the VPN
tunnel incoming packets should be routed through and Tunnel
State for the actual state of the route tunnel (up or down).

The Add button opens the Add VPN Route page where a new
VPN route can be generated.

Bizfon4000 (SW Version 3.1.x) 16

Fig. II-26: VPN Routing table

Bizfon Manual II: Administrator's Guide Administrator's Menus

The Add VPN Route page offers the following components:

Route Via contains the available PPTP and L2TP connections on

the Bizfon. A connection selected from this list will be used to
route the IP packet from the Bizfon’s LAN to the peer behind the
PPTP/L2TP tunnel.

Route To requires the IP address range of the possible peers
behind the PPTP/L2TP tunnel whereto IP packets could be
routed.

Fig. II-27: Add VPN Route page

The Enable and Disable functional buttons are used to activate or to deactivate the selected route(s). At least one route should be selected to use
these functions, otherwise the error message appears: “No record(s) selected.”

To Add an IP Static Route

1. Select the IP Static Routes link on the Routing Configuration page.

2. Press the Add button on the IP Static Routes page. The Add Entry page will appear in the browser window.

3. Enter the destination IP address and subnet mask in the Route To text fields. Use the IP-Clip button to select a previously entered IP address.

4. Enter the router IP address into the Via IP Address text fields.

5. Press the Save button to make the static route with these settings.

To Add an IP Policy Route

1. Select the IP Policy Routes link on the Routing Configuration page.

2. Press the Add button on the IP Policy Routes page. The Add Entry page will appear in the browser window.

3. Specify the policy routing rule priority in the Priority text field.

4. Enter the packet source IP address and subnet mask in the From text fields. Use the IP-Clip button to select a previously entered IP address.

5. Enter the router IP address into the Via IP Address To text fields.

6. Press the Save button to make the policy route with these settings.

To Add a VPN Route

1. Select the VPN Routes link on the Routing Configuration page.

2. Press the Add button on the VPN Routes page. The Add Entry page will appear in the browser window.

3. Choose the VPN connection from the Route Via drop down list.

4. Enter the destination IP address and the subnet mask into the Route To text fields.

5. Press the Save button to make the VPN route with these settings.

Configuration Management

The Configuration Management assists the administrator to manage the system configuration settings and voice data, i.e., to backup and
download the settings to the PC and then to upload and restore them back to the Bizfon. Additionally this page gives a possibility to restore the
factory default configuration settings.

The Backup & Download all config & voice data link
generates a backup file with all configuration settings and
user uploaded greeting messages and opens a file chooser
window for immediate download to the user PC.

Attention: Configuration and voice data cannot be backed

up if the size of voice data is too large. In this case, to be
able to backup configuration and voice data on the Bizfon,
please remove some user defined system messages (by
restoring the default ones, see chapter Update System

Messages), or remove some extensions from the
Extensions Management table.

The Upload & Restore all config & voice data link opens a page with the Browse button, (which opens a file chooser to select a backed-up file)
and a Configuration to Upload field requiring the file path to upload and to restore it immediately. Pressing Save will restore the selected backup
file, and delete all current user defined greetings and replace configuration settings.

Attention: Restoring the configuration and voice data requires switching Memory Allocation (see chapter

Voice Mail Common Settings) to the state

that was selected when the configuration and voice data were backed up, otherwise an error message prevents uploading the backup file.

The Use Default functional button resets all configuration settings and restores the board’s factory default configuration. By restoring the default
configuration you will replace your current one, lose all voice mails and reboot the device. You will not be automatically redirected to the GUI start
page. After the successful reboot you will need to enter into the management and login again to access the Bizfon’s configuration. A warning
message will ask you to confirm your selection before restoring the default configuration.

Fig. II-28: Configuration Management page

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Please Note: Unlike the factory default settings restore procedure initialized from the Reset button on the Bizfon board, this link will keep the

following data:

• Call Statistics

• Transfer Statistics

• System Events

• Feature Keys

• Device Registration state

Events

The Events page shows two tables and displays all system events that have occurred in one table and event settings in the other.

The System Events page may be accessed with Events link
from the main menu. It lists information about system events that
have occurred on Bizfon. When a new event takes place a
record is added to the System Event table and for failure events
(priority 2 and 3, see below) additionally a warning “Please
check your pending events!” appears at the bottom of all
management pages.

The system events and the warning message are visible only for
the administrator. The warning link, (which leads directly to the
System Events page) will disappear from the management
pages if the administrator has marked all new events as read.

Fig. II-29: Event Warning on the Main Menu page

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Bizfon Manual II: Administrator's Guide Administrator's Menus

The System Events table is the list of new and read system
events. System events have the corresponding coloring
depending on the nature of the event: success (priority 1, color
green), low importance failure (priority 2, color yellow), critical
failure (priority 3, color red).

The table shows the Status of the event (new or read) as well as
the name of the application the event refers to, event description,
and the date when the event was received. For example, if the
event has been caused by the IDS service, the Check IDS link
appears in the reference row that will lead to the

IDS Log page,

or if the event has occurred due to incorrect mail sending or SIP
registration, corresponding links will be seen in the Reference
column of the table. There the administrator can view the
detailed log for the event that has occurred.

The System Events page offers the following components

:

Current System Time displays the local date and time on
Bizfon.

Mark all as read marks newly occurred events as read.

Disable LED switches off the LED flashing (if any do flash) on

the board. A LED notification may appear (depending on the
notification type given) in the page

Events settings whenever a

new event occurs.

Fig. II-30: System Events list

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Bizfon Manual II: Administrator's Guide Administrator's Menus

Numerous circumstances may cause a certain application on
Bizfon to flag an event.

The page Event Settings lists all possible events on the Bizfon
and allows controlling the way of notification (action), if one of
those events takes place.

Each entry in the events’ table has its checkbox assigned to the
row. By selecting the corresponding checkboxes, operations
such as Edit may be done for one or more events.

Edit opens the Edit Event Settings page to modify the event
action.

Display Notification - A notification link will be displayed on
the bottom of all pages and a record is added into the Events
table. The notification is executed as the link “Please Check
you pending events!” that leads to the page System Events.
This action also will take place if Flash LED or Send Mail has
been selected, even if not selected explicitly.
Flash LED - The second LED (yellow) will be blinking once a
second and a notification will be displayed on the bottom of all
pages. For some events the LED will start flashing after a

delay.
Send Mail - An e-mail with a notification about the new event
and an event description in the mail body will be sent to the
e-mail address specified in the

Mail Settings page.

Fig. II-31: Event Configuration Settings page

Actions that are not allowed for the selected event (like mail notification if the PPP link is down or the mail server has been misconfigured) are
hidden. For multiple events editing, actions that do not fit at least to one of the selected events will be hidden.

Please Note:

In case of an IDS (Intrusion Detection System) intrusion alert, only the first possible intrusion in each period of 10 minutes is initiating

an event. This method particularly helps to avoid the flooding of the System Events table, and the flooding of the user through various intrusion alerts
that result from each possible Denial of Service attack. As these events are displayed in the System Events table with a link to the IDS log list, the
user can get the detailed information about the intrusions there.

If Bizfon cannot get an IP address from the DHCP or PPP servers, or cannot register an extension on the SIP or Routing servers, or cannot reach an
NTP server, it raises only one event for the entire period the action has failed, but continues to try. When the required action is successful, Bizfon
raises an appropriate message.

The page Edit Event Settings offers the following input
options:

Application displays the application the event refers to.
Multiple is shown here, if more than one event has been

selected for the action assignment.

Name displays the name of the event. Multiple is shown here,
if more than one event has been selected for the action
assignment.

Description displays additional information about the event.
Multiple is shown here, if more than one event has been

selected for the action assignment. Action offers radio
buttons to choose one of the actions to notify the Bizfon
administrator whenever the selected event(s) takes place.

Fig. II-32: Edit Event Settings page

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Bizfon Manual II: Administrator's Guide Administrator's Menus

To Assign an Action to the Event

1. Select the checkbox of one or more events to assign an action to them.

2. Press the Edit button. The Edit Event Settings page appears.

3. Select an action type from the Action radio buttons to notify the administrator about the event in the desired way.

4. Press the Save button to submit the changes or use Back to abort the selected action.

Time/Date Settings

The Time and Date Settings provide information about the current system time and date. The settings may be updated through the international
time and date servers.

Time is used to set the local time (hour, minute).

Date is used to set the date (month, day, year).

Timezone provides a selection of world time zones and is used

to select the local country time zone. Timezones are specified
by GMT (Greenwich Mean Time) and by specific timezones for
the United States and Canada.

Enable Simple Network Time Protocol Server enables the
SNTP (Simple Network Time Protocol) server on Bizfon, thus
Bizfon becomes the timeserver for its LAN.

Enable Simple Network Time Protocol Client enables the
SNTP client on the Bizfon, thus Bizfon becomes a client to an
external timeserver. The checkbox disables Date and Time drop
down lists and enables the following parameters:

The SNTP Servers table lists all defined NTP Servers.

Add functional button opens an Add NTP Server page where a
new NTP server can be defined. This page offers the NTP
Server radio buttons that are used to choose between a manual
and a predefined NTP server.

Manual requires the NTP server’s FQDN (Full Qualified
Domain Name) or its IP address.

Predefined is used to select the NTP server’s host
address from the drop down list, where the most common
NTP servers are listed.

The Move Up and the Move Down functional buttons are used
to sort NTP servers in the order they need to be accessed. If the
NTP server on the first position in the SNTP Servers table does
not answer, NTP server on the next position will try to be
reached.

Please Note: Add another NTP server to the list if you feel

defined NTP servers are not functional, i.e., the Bizfon's
date/time is not being updated automatically.

Polling Interval indicates the time interval for the periodical
synchronization between timeserver and Bizfon. It counts in
hours.

Attention: Time and Date Settings will be reset if Bizfon has lost power.

Fig. II-33: Time and Date Settings page

Fig. II-34: Add NTP Server page

Mail Settings

The System Mail Settings page gives a possibility to send warnings automatically about the board status or problems to the administrator. System
events that require email notification are selected on the
configured for voice message transmission to the extension user’s mailing account.

Bizfon4000 (SW Version 3.1.x) 21

Events page. Besides, system mail has to be enabled and the SMTP server needs to be

Bizfon Manual II: Administrator's Guide Administrator's Menus

Enable enables the system mail sending possibility and voice
messages transmission to the extension user’s mailbox.

SMTP Host requires the SMTP host IP address or domain
name. The SMTP host needs to be configured to enable voice
message transmission.

Mail Sender Address requires the source address for the
Bizfon notification emails. The email address defined here
should be an existing valid email address registered on the
selected SMTP server or should have permission to use that
particular SMTP server for emails transmission.

Mail Recipient Address requires an active email address. The
e-mail recipient here can be a Bizfon administrator or someone
responsible for network and system problems.

Enable SMTP Authentication has to be selected if the
specified SMTP server requires an authentication. In this case,
authentication User Name and Password configured on the
SMTP server should be defined in the corresponding text fields.

Send Test Mail is used to initiate a test e-mail transmission.
This button will be enabled if correct values have been
submitted and saved on this page.

Fig. II-35: System Mail Settings page

To configure the System Mail

1. Enable the system mail sending by the Enable checkbox selection.

2. Update or set the SMTP host in the SMTP Host text field.

3. Update or set the e-mail sender address in the Mail Sender Address text field.

4. Update or set the e-mail address in the Mail Recipient Address text field.

5. Enable SMTP Authentication if it is required of the server.

6. Insert into the corresponding text fields an authentication User Name and a User Password defined by your SMTP server.

7. Press the Save button to submit these settings.

8. Use the Send Test Mail button to send a test e-mail with the configured settings.

SMS Settings

The SMS Settings are used to configure the SMS parameters that will allow Bizfon to send the voice mail notifications via SMS to the extension
user’s mobile phone. Every extension user is free to enable the voice mail notifications upon new voice mail arrival and to define own mobile
numbers from the Voice Mail Settings. However, to make Bizfon able to deliver SMS notifications, SMS service should be enabled and SMS settings
should be configured from this page.

Enable SMS Service enables the SMS service on the Bizfon.

User Name and Password text fields require the authentication

settings of the SMS server.

SMS Sender Address requires the source address for the
Bizfon notification SMS. The address defined in this field will be
seen in the “From” field of the SMS delivered to the mobile
phone.

SMS Recipient Address requires a destination mobile number
for a test SMS.

HTTP Parameters:

ID text field requires an identification number defined by

the SMS server.

Server text field requires an IP address or the host name
of the SMS server.

Port text field requires a HTTP port number of the SMS
server.

Use Secure HTTP checkbox enables access to SMS
server via HTTPS. Checkbox selection enables a Secure
Port text field that requires the port number for HTTPS
traffic.

Send Test Mail is used to send a test SMS to the defined SMS Recipient Address. This button will be enabled if correct values have been submitted
and saved on this page.

Fig. II-36: SMS Settings page

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Firmware Update

This window allows to update the software of Bizfon by installing new firmware called image. To learn more about new firmware available, please
contact Bizfon Technical Support.

Updating a new firmware requires a perfectly working power supply. Therefore Bizfon is provided with a battery (accumulator). If the battery is low or
simply absent the “There is no battery or voltage is low” warning is displayed

Please Note: Installing new firmware will take about 15

minutes. During this time, the Bizfon, telephony and Internet
access will be disabled.
The firmware update will cause the loss of the following
data:

• All internally stored voice mails and custom voice

messages

• DHCP leases

• Transfer statistics

• Call statistics

• All pending events

• User specific GUI states

The following main processes will be stopped during the
firmware update and will be restarted afterwards:

• Voice Software

• Network Time Protocol Daemon

• Network Interface Statistic Daemon

• Dynamic DNS Daemon

Please Note: If you consider the

the displayed tables to be important, it is recommended to
download them from the corresponding page prior to
starting the Firmware Update.

Next will move you to the second page of Firmware Update
where the image file should be selected.

Call Statistics entries in

Fig. II-37: Firmware Update page 1

The second page of Firmware update has the Browse
button used to browse the image file, and the Specify
Image text field that will display the selected image
filename.

Pressing Save will start uploading the image file to the
board. The next page will be displayed, showing the result
of a verification of the image being burned.

Bizfon4000 (SW Version 3.1.x) 23

Fig. II-38: Firmware Update page 2

Bizfon Manual II: Administrator's Guide Administrator's Menus

This page displays non-editable information about the
image validity. The Image Check will display invalid if the
image does not correspond to the hardware version.

The fields Current Software Version and New Software
Version show the old software version and the version of
the new image.

This page needs a confirmation to continue image updating.
If you are sure that the image version is appropriate for your
device press Save.

Fig. II-39: Firmware Check page

Networking Tools

The Networking Tools provide the possibility to check the Internet connection.

Ping sends four ICMP (Internet Control Message Protocol) requests with a default size of 64 bytes to the destination (IP address or host name)

specified in the text field Ping Target. The response times are logged, and the round trip time (the time required from being sent until being received
again) is measured. The results are displayed in the lower area of the page: The minimum and maximum round trip time and its average, the
percentage of lost and of received frames.

Traceroute checks the Internet connection by triggering the routers (hops) that are passed to reach the destination specified in the text field
Traceroute Target. Trace routing gives feedback on the routers passed by packets on the way toward the destination and the round trip delay of

packets to these routers.

Attention: No Traceroute is possible if a high priority

Firewall has been enabled (see chapter

For the purpose of tracerouting, several IP packets are sent
out. UDP (User Datagram Protocol) is used to send packets
and ICMP (Internet Control Message Protocol) is used to
receive information about the routers. In their headers, the
TTL (Time To Live) value increases from 1 to 30. When the
first IP frame is received by the first router, its IP address
will be returned in its acknowledgement.

The second frame delivers the IP address of the second
router and so on and so forth. The results of Traceroute are
displayed on the lower area of the page.

Ping Target requires the destination (IP address or host
name) for the ICMP request.

The Ping button starts pinging the specified ping target.

Traceroute Target is used to enter the IP address or host
name of the destination to be trace routed.

The Traceroute button is used to process the router
triggering to check the Internet connection.

In the field below the output of the Ping or Traceroute
procedure is shown.

To Check the Internet connection

1. Specify destination address for the ICMP request in the Ping Target text field.

2. Press the Ping button to process the ICMP request.

3. Specify the destination address to trace the route.

4. Press the Traceroute button to process the router triggering.

Firewall and NAT).

Fig. II-40: Networking Tools page

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Diagnostics

The System Diagnostic page gives a possibility to run Network and WAN protocol diagnostics, to verify Bizfon's connectivity.

The Start Detecting WAN Protocol button is used to initiate WAN diagnostics that will detect the WAN IP configuration: static or through DHCP and
PPP servers. For static WAN IP configuration, gateway availability is checked. When acting as a client, DHCP and PPP servers' accessibilities are
being verified.

The Start Network Diagnostics button is used to initiate network
diagnostics, i.e., to check the WAN link and IP configuration, to
verify gateway, DNS primary and secondary (if configured) servers'
accessibilities.

The field below will display the diagnostics results and the
connectivity conditions. The system should be reconfigured if
problems occur during the diagnostics.

The Reboot this Device button is used to reboot the Bizfon.
Please note that the session with the Bizfon will be closed, i.e., the
Bizfon GUI should be newly opened and a new login will be
required afterward.

Fig. II-41: System Diagnostic page

Automatic Provisioning

Automatic Provisioning gives a possibility to automatically
configure the WAN network settings of Bizfon. This is very useful,
when the administrator is not actually aware about the Bizfon’s
network settings. Automatic Provisioning automatically detects
the matching network configuration settings and by applying them
on the Bizfon, it connects the device to the internet through the
available ISP connection.

Please Note: Automatic Provisioning can be run only from the

LAN side of the Bizfon, i.e. from the PC connected to the Bizfon’s
LAN.

Automatic Provisioning automatically detects and configures the
following settings on the Bizfon:

• WAN interface type (PPPoE or Ethernet)

• WAN IP settings

• PPP settings

• ISP settings

• DHCP settings

• DNS settings

• NAT Traversal settings

Fig. II-42: Auto Provisioning page

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Features

This page lists all features, that may be activated by a software
key, characterized by a Feature Description and provided with its

Status:

• No Key Found - the feature is currently not available

• Reboot Needed - the feature key has been entered and

Bizfon has to be rebooted

• Activated - the feature is available

Following features may be activated via the software key:

• IP Phone support - enables additional LAN-sided IP

phones on the Bizfon. For Bizfon4000, 16 more IP phones
could be connected to the Bizfon.

Fig. II-43: Features page

• Debug – enabled Telnet connection towards the Bizfon for

debugging purposes.

To enter a Feature Key click Add. A page with the text field
Feature Key is opened. Enter the key and press Save. The status
of the selected feature entry will change to Reboot needed.
Reboot the Bizfon and the feature will get the status Activated.

To get a Feature Key register the Bizfon device and send a
corresponding request to Bizfon's Technical Support. This request
must include the Unique ID that is displayed in the Features page
above the features list.

Fig. II-44: Features Add page

User Rights Management

User Rights Management service is used to set restrictions on the GUI access for various users, to permit or deny the access to certain Web GUI
configuration pages and to create a multilevel user management of the Bizfon. Feature is mainly useful to the ISPs to set the restrictions for certain
customers to manage the Bizfon’s configuration.

Two levels of Bizfon GUI administration are available:

• Administrator – this is a main (super) administrator’s account. It is a matter of configuration whether to have password factory reset safe

or not, i.e. whether default password will be retrieved after the factory reset or not. Administrator has an access to all Web GUI pages, no
configuration permissions can be adjusted for this account. Administrator is responsible for granting access to all other user groups.

• Local Administrator – this is a common (sub-) administrator’s account. Password is not factory reset safe. Local Administrator’s

permissions can be adjusted per each GUI page.

• Extension – account refers to all extensions created on the Bizfon. The password for default extensions is not factory reset safe but is

contained in the backed up configuration. Extension’s permissions can be adjusted per each GUI page.

User Rights Management page consists of two pages: Users, no manage the available users on the Bizfon, and Roles, to assign the
corresponding permissions to the users.

Users page contains a table where Administrator and Local
Administrator users are listed. Page allows to modify the
passwords of available users in the table and to manage the
Local Administrator’s account. Following functional buttons
are available on the page:

Change Password functional button is used to change the
password of the Administrator and Local Administrator user’s
account. Select one of the available users in the table by
toggling the corresponding checkbox and press Change
Password to open the corresponding page.

Fig. II-45: Users page at User Rights Management

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Bizfon Manual II: Administrator's Guide Administrator's Menus

The Change Password page is used to change the user’s
password. It offers the following components:

Old Password text field is only present when modifying the
Administrator account password and requires the current
password of Administrator. An error message prevents
from entering a wrong one.

New Password requires the new password for
Administrator or Local Administrator, which has to be
confirmed in Confirm New Password.

The password may consist of numerical values only, up to
20 digits are allowed. A corresponding warning appears if
any other symbols are inserted.

Store password in persistent area (Factory reset save)

checkbox is only present when modifying the Administrator
account password and is used to save the Administrator’s
in the factory reset safe place.

Attention: Be EXTREMELY careful when enabling this

checkbox – if it is done, Administrator’s password won’t be
retrieved even after factory reset. In this case, if
Administrator’s password has been forgotten, the Bizfon
will be considered as broken. Please contact Bizfon
Technical Support Center for device replacing.

Fig. II-46: Change Password page

Enable User and Disabled User functional buttons are used to enable/disable the Local Administrator’s account.

Please Note: Administrator’s account cannot be disabled.

Roles page contains a table where Local Administrator and

Extensions users are listed. Page allows to set the
permissions to the GUI pages for each user in the table.

Edit functional button leads to a Change Access Rights
page where a list of user specific GUI pages is displayed.
Select on of the users in the table and press Edit to manage
the permission for the corresponding user.

Fig. II-47: Roles page at User Rights Management

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On the Change Access Rights page, Grant Access/Deny
Access functional buttons are used to grant/deny the access

to the certain GUI page(s) for the selected user.

When the access to the certain GUI page is denied for a user,
“You are not authorized to access this page!” warning
message will be displayed when user attempts to open the
corresponding GUI page.

Fig. II-48: Edit Roles page at User Rights Management

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Users Menu

F Fig. II-50: Telephone Users Menu in Plain Theme

Fig. II-49: Telephone Users Menu in Dynamo Theme

Extensions Management

The Extensions Management is used to create user extensions and auto attendants on the Bizfon. From this page, by clicking on the user
extension, administrator can get the extension settings pages.

Two types of user extensions can be created on the Bizfon: active and inactive extensions. Active extensions are those that are attached to a line,
can place and receive calls and use available telephony services. Inactive extensions are those that are not attached to the line, they can use some
available telephony services but cannot place and receive calls; instead, they have a voice mailbox available to keep the brief messages from the
callers.

Bizfon4000 has four available lines and up to four active extensions can be established.

Attendant extensions are dedicated to the IVR system on the Bizfon, which is used by the callers to reach Bizfon’s users, use remote access and call
relay services. It is possible to create Auto Attendants with the custom scenarios. By default, Bizfon has one Auto Attendant extension (00) which is
undeletable.

The Extensions table is a list of all extensions and their parameters.

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Fig. II-51: Extensions Management page

The following columns are present in the table:

• Extension - lists the 2-digit user or attendant extensions on the Bizfon. This number is used for internal PBX calls.

• Display Name - indicates an optional display name to identify the caller.

• Attached Line - indicates the FXS or IP line corresponding extension is attached to. “R” is displayed in this column when SIP Remote

Extension (see below) functionality is enabled on the extension.

• SIP Address - displays the SIP address of the corresponding extension. Column displays the full SIP address, (i.e.,

username@sipserver:port) when the Registration on SIP Server checkbox is selected. Else, if registration is disabled, the SIP address will
be displayed in the following format: “username, Proxy: sipserver:port”. If no SIP registration server or SIP server port is defined,
corresponding information will be skipped in this column. If no username is defined, the extension number will be displayed instead.

• Percentage of System Memory - indicates the user space (in percent) configured for each extension. The actual available duration (in

minutes) for the extension voice mails, uploaded/recorded greetings and blocking messages is also displayed herein. The available minutes
corresponding to the selected user space are dependent on the Voice Recording codec selected from the
page, for example, for the same amount of marked out user space, selection of the G726 voice recording codec will provide more space for
voice mails and user defined voice greetings than the G711 codec selection.

Voice Mail Common Settings

• Call Relay - indicates whether or not Call Relay option is enabled on the extension.

• Codecs – column lists the short information (full information is seen in the tool tip) about extension specific voice Codecs. Extension

codec’s can be accessed and modified by clicking on the link of the corresponding extension’s Codecs. The link leads to the

Codecs

page.

Clicking on each user extension in the Extensions table will open the extension specific Extension Settings menu. When Call Park service is
enabled on the extension, it is displayed without a link in the Extensions Management table and extension pages. Additionally, the supplementary
services configuration pages will not be accessible.

Extension

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Add opens the Add Entry page where the type and the
number of new extension should be defined. Page consists of
the following components:

Extension text field is used to enter a new extension number.
The extension number is a two-digit number. If non-digit
symbols have been entered, the error “Incorrect Extension: no
symbol characters allowed” occurs. If the extension length is
shorter than 2 digits, the error “Incorrect Extensions length” will
prevent the creation of the extension. If an extension with the
same number already exists in the Extensions Management
table, the error “Extension already exists” will appear.

Please Note: Extension number cannot start with digits 0, 8 or

9.

Type drop down list is used to select the type of the extension
(user, attendant or pickup group) to be created (for details see
below).

Fig. II-52: Extensions Management - Add Entry page

Edit opens the Edit Entry page where a newly created user or attendant extension settings might be adjusted. To operate with Edit, one or more
record(s) have to be selected, otherwise an error will occur: “No records selected”.

The Edit Entry page consists of two frames. In the left frame settings groups are listed. Clicking on the corresponding settings group, its
configuration options will be displayed in the right frame.

Please Note: Pay attention to save changes before moving among settings groups.

1. General Settings

This group requires extension‘s personal information and has
the following components:

Display Name is an optional parameter used to recognize the
caller. Usually the display name appears on the called party’s
phone display whenever a call is performed or a voice mail is
sent.

Password requires a password for the new extension.

The extension password may only contain digits. If non-numeric
symbols are entered an “Incorrect Password: no symbol
characters allowed” error will prevent making the extension.

Confirm Password requires a password confirmation. If the
input is not corresponding to the one in the Extension
Password field, the error will appear: “Incorrect Password
confirm”.

Attached Line lists all free lines to where an extension may be
attached.

Please Note: Extension cannot be detached from the line if SIP

Remote Extension service is enabled on it. To detach the

extension from the line, disable the SIP Remote Extension
service on the extension first.

Fig. II-53: Extensions Management - Edit Entry – General Settings page

Allow Call Relay enables the current extension to be used to access the Call Relay service in the Bizfon’s Auto Attendant. It is recommended to
define a proper and non-empty password when enabling this feature in order to protect the Call Relay service from an unauthenticated access.

Use for Call Park allows to use the extension for the

Call Park Service. It is recommended to use virtual extensions for this service and to configure

all available codecs, so parked call won’t be lost in case if a caller, who picks up the parked call, doesn’t support some specific codecs. When Call
Park service is enabled on the extension, it is displayed without link in the Extensions Management table and extension pages and additionally
supplementary services configuration pages will not be accessible.

When External Call Policy checkbox is enabled, all incoming IP calls to the corresponding extension will be handled by the external Policy Server.

The Percentage of System Memory drop down list allows to select the space for the extension’s voice mails and uploaded/recorded greetings and
blocking messages. The maximum value in the drop down list is equal to the maximal available space for voice messages on Bizfon. While editing
an existing extension and decreasing the voice mailbox size, the system will check the present amount of voice mails in the mailbox of the extension.
If the memory required for these voice mails exceeds the size entered, the system will suggest either to remove all voice messages from the
extension’s voice mailbox or to select a larger size so that the existing voice messages can be stored in the mailbox.

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2. SIP Settings
This group is used to configure the extension’s SIP registration
settings and consists of the following components:

Registration User Name requires a user name for the
extension registration on the SIP server. The registration user
name needs to be unique on the SIP server and is being
displayed on the called phone whenever performing an IP call.

Registration Password indicates the password for the
extension registration on a SIP server.

Confirm Registration Password is used to confirm the
password. If the entered password does not correspond to the
one given in the Registration Password field, the error will
appear: “The passwords do not match. Please try again”.

Registration SIP Server indicates the host address of the SIP
server. The field is not limited regarding symbol usage and
length as it can be either an IP address, e.g., 192.168.0.26 or a
host address, e.g., sip.bizfon.com.

Fig. II-54: Extensions Management - Edit Entry – SIP Settings page

Registration SIP Port indicates the host port number to connect to the SIP server. The SIP server port may only contain digit values, otherwise the
error message “SIP Server Port is incorrect” will be displayed when applying the extension settings. If the SIP server port is not specified, Bizfon will
access the SIP server through the default port 5060.

Registration on SIP Server enables the SIP server registration option. If the extension has already been registered at some SIP server, its IP
address will be displayed in brackets.

Attention: By default, SIP registration settings defined are pre-configured for all extensions and the SIP calls will not be successful if these settings

are modified. However, if the SIP registration settings are somehow changed, only a factory reset will restore the default values.

3. Advanced SIP Settings

This group is used to configure advanced SIP settings (Outbound Proxy, Secondary SIP Server and Outbound Proxy for the Secondary SIP Server
settings and to define other SIP server specific settings).

SIP Outbound proxy is such a SIP server, where all the SIP requests and other SIP messages are transferred. Some SIP servers use an outbound
proxy server to escape restrictions of NAT, e.g., Free World Dialup service uses an Outbound Proxy server. If an Outbound proxy is specified for an
extension, all SIP calls originating from that extension are made through that outbound proxy, i.e., all requests are sent to that outbound proxy, even
those call by Speed Calling.

The Secondary SIP Server acts as an alternative SIP registration server when the primary SIP Registration Server is inaccessible. If the connection
with the primary SIP server fails, Bizfon will automatically start sending SIP messages to the Secondary SIP Server, and will switch back to the
primary SIP server, as soon as its connection is reestablished.

UserID requires an identification parameter to reach the SIP
server. It should have been provided by the SIP service provider
and can be requested for some SIP servers only, for others, the
field should be left empty.

Send Keep-alive Messages to Proxy enables the SIP
registration server accessibility verification mechanism. Timeout
indicates the timeout between two attempts of SIP registration
server accessibility verification. If no reply is received from the
primary SIP server within this timeout, the Secondary SIP server
will be contacted. When the primary SIP server recovers, SIP
packets will be sent to it once again.

A group of Host address and Port text fields respectively
require the host address (IP address or the host name), the port
number of the Outbound Proxy, Secondary SIP Server and
the Outbound Proxy for the Secondary SIP Server. These
settings are provided by the SIP servers’ providers and are used
by Bizfon to reach the selected SIP servers.

RTP Priority Level drop down list is used to select the priority
(low, medium or high) of RTP packets sent from corresponding
extension. RTP packets with higher priority will be preceded first
in case of heavy traffic.

Fig. II-55: Extensions Management - Edit Entry – Advanced SIP Settings page

4. Remote Settings

This group is used to configure SIP Remote Extension functionality which is an advanced telephony feature that allows Bizfon users to remotely
operate on Bizfon when being away. User needs to register a hardware or software SIP phone on the Bizfon, by defining the Bizfon’s global IP
address and an appropriate Username/Password. Registered SIP Remote phone can fully act as a phone connected locally to Bizfon, i.e. use
Bizfon’s PBX features, place and receive calls, access voice mails, etc.

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Enable checkbox activates the SIP Remote Extension’s functionality.

Please Note: SIP Remote Extension functionality may be enabled only for active (attached to an onboard FXS or IP line) extensions.

Identification parameters used by the remote SIP device for registration on the Bizfon should be defined in the Username and Password text fields.

When Enable RTP Proxy checkbox is selected, incoming and
outgoing RTP streams to and from the remote SIP phone will be
routed through Bizfon, otherwise, when checkbox is not
selected, RTP packets will be moving directly between peers.

When Use Only When Registered checkbox is selected,
incoming calls towards the corresponding extension on the
Bizfon will be forwarded to the remote SIP phone, only in case it
is registered. Otherwise, when remote SIP phone is
unregistered, incoming calls will be routed to the line extension
is attached to. When this checkbox is not selected, all incoming
calls will be routed to remote SIP phone regardless on its actual
registration.

The Symmetric RTP checkbox should be selected when remote
extension is located behind the symmetrical NAT.

Fig. II-56: Extensions Management - Edit Entry – Remote Settings page

5. Call Queue Settings

This group is used to configure Call Queue service that allows to keep multiple incoming calls in the queue when being on the line and to answer
calls in the order they have been received. Feature can be also used within

Receptionist Management (see below for more details).

Enable checkbox activates the Call Queue functionality on the
extension.

Call Queue Size text field requires the length of the call queue,
i.e. the number of calls that can be held during extension being
in call. If a maximal number of calls is already held in the call
queue, next incoming call will be disconnected.

Max Call Queue Appearance text field requires the maximal
number of active calls on the line, i.e. if 1 is configured in this
field and extension is in call, next incoming call will go to the call
queue. If 2 is configured in this field and extension is in call, next
incoming call alert will be heard in the background (if Call
Waiting service is enabled on the corresponding extension) and
extension can hold the first call to answer the second one, either
he can join the two calls into the call conference. However, the
next incoming call will again go to the call queue.

Fig. II-57: Extensions Management - Edit Entry – Call Queue Settings page

Upload new call queue welcome message allows updating the active Call Queue welcome message (played when caller joins the extension’s call
queue), downloading it to the PC, or restoring the default one.

The Remove call queue welcome message functional link appears only when custom call queue welcome message is already uploaded and is
used to remove it and restore the default call queue welcome message.

The Download call queue welcome message functional link appears only when custom call queue welcome message is already uploaded and is
used to download it to PC and opens the file chooser window where the saving location can be specified.

Upload new call queue message allows updating the active call queue message (played upon caller being in the queue), downloading it to the PC,
or restoring the default one.

The Remove call queue message functional link appears only when custom call queue message is already uploaded and is used to remove it and
restore the default call queue welcome message.

The Download call queue message functional link appears only when custom call queue message is already uploaded and is used to download it
to PC and opens the file chooser window where the saving location can be specified.

Browse buttons open the file chooser window to browse for a new Call Queue welcome message file. The uploaded files should to be in PCMU
wave format, otherwise the system will prevent uploading it with the “Invalid audio file, or format is not supported” warning message. The system also
prevents uploading if there is not enough memory available for the corresponding extension, which will cause the “You do not have enough space”
warning.

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6. Voice Mailbox Settings

This group is used to configure the voice mailbox storage and
consists of a group of manipulation radio buttons used to define
the location where voice mails will be collected.

• Disable Voice Mail – disables the Voice Mail service for

the corresponding extension. With this selection, extension
user will be unable to reach his Voice Mail Settings, but will
be able to access his Voice Mailbox and manage with the
existing voice mails.

• Use Internal Voice Mail – enables the Voice Mail service

for the corresponding extension and defines the Bizfon’s
internal storage as a location for the Voice Mails.

• Use External Voice Mail – enables the Voice Mail service

for the corresponding extension and is used to define a
remote Voice Mail Server as a location for the Voice Mails.
With this selection, it is requires to enter the SIP URI of the
Voice Mail Server where voice mails of the corresponding
extension will be collected.

Pickup Group & Access List

The Pickup Group service is used to monitor the calls addressed to a certain list of extensions and to pick up the calls ringing on the listed
extensions. Service may be particularly used when a group of extensions are located in the same area, so the ringing on one of extensions can be
heard by the persons sitting nearby. Feature allows to pick up the call ringing on the certain extension by dialing the number of the pickup extension.

Pickup Group list is used to define the extensions that can be monitored by calling the certain pickup extension.

Access List is used to define PBX, SIP or PSTN users that are allowed/forbidden to intercept the calls ringing on the extensions in the Pickup

Group.

When user dials the pickup extension having several extensions of the pickup group ringing, the first (oldest in time) call will be picked up. When user
dials the pickup extension having no ringing extensions of the pickup group, “No call is available to pickup” message will be played to the user. When
user that is not listed in the Access List dials the pickup extension, password authorization (of the pickup extension) will be required to pick up the
call. When a denied user dials the pickup extension, “Party does not accept your call” message will be played to the user.

For Pickup Group extensions, Extensions Management - Edit Entry page consists of General Settings, SIP Settings and Advanced SIP

Settings pages. The SIP Settings and Advanced SIP Settings pages are the same as for the regular extensions and are described above, while
General Settings page has a different content:

1. General Settings (for pickup group extension)

This group requires personal extension information and has the
following components:

Display Name is an optional parameter used to recognize the
caller. Usually the display name appears on the called party’s
phone display whenever a call is performed or a voice mail is
sent.

Password requires a password for the new extension.

The extension password may only contain digits. If non-numeric
symbols are entered an “Incorrect Password: no symbol
characters allowed” error will prevent making the extension.

Confirm Password requires a password confirmation. If the
input is not corresponding to the one in the Extension
Password field, the error will appear: “Incorrect Password
confirm”.

The Edit Pickup Group link leads to the page where a list of monitored extensions can be defined.

Fig. II-59: Extensions Management - Edit Entry – General Settings for pickup extension page

Fig. II-58: Extensions Management - Edit Entry – Voice Mailbox Settings page

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The Pickup Group of Extension page lists all extensions in
the pickup group, i.e. those that can be monitored and the
calls addressed to which may be picked up by calling the
corresponding pickup extension.

Add functional button opens an Add Entry page with an only
drop down list containing all available extensions on the
Bizfon.

Fig. II-60: Pickup Group of Extension page

The Edit Access List link leads to the page where users permissions to use the pickup service can be defined.

The Access List of Extension page lists all users (or a group of
users, if wildcard is used) and the appropriate permissions to
pickup the calls ringing on the extensions from the Pickup Group.

Fig. II-61: Access List of Extension page

Add functional button opens an Add Entry page where new user
with corresponding permissions might be created. Page consists
of the following components:

Call Type lists the available call types:

• PBX - local calls from Bizfon’s extensions

• SIP – calls through a SIP server

• PSTN – calls from global telephone network

• Auto – used for undefined call types. Destination

(independent on whether it is a PBX number, SIP address
or PSTN number) will be parsed through Call Routing
Table.

Address text field is used to define the address to be included in the Access List table. The value in this field is strictly dependent on the Call Type
defined in the same named drop down list. If PBX call type is selected, the Bizfon extension number should be defined in this field. For the SIP call
type, the SIP address should be defined, for the PSTN call type, the PSTN user number should be defined here.

Action drop down list is used to select the defined user’s permissions (allow or deny) to use the pickup service for the extensions included in the
Pickup Group.

For Attendant extensions, Extensions Management - Edit Entry page consists of General Settings, Attendant Scenario, SIP Settings and
Advanced SIP Settings pages. The SIP Settings and Advanced SIP Settings pages are the same as for the regular extensions and are described
above, while General Settings and Attendant Scenario pages’ content is described below:

1. General Settings (for attendant extension)
This group requires personal extension information and has the
following components:

Display Name is an optional parameter used to define the Auto
Attendant’s description. Usually the display name appears on
the called party’s phone display whenever a call is performed or
a voice mail is sent.

With the Enable FAX Forwarding checkbox enabled, the
system moves the incoming FAX to the selected extension if a
FAX tone is detected on the Auto Attendant.

The Extension to forward drop down list is used to choose the
extension where the incoming FAX addressed to the Bizfon’s
Auto Attendant will be forwarded. The list contains only those
extensions that have FAX support enabled. FAX support can be
enabled from the

Extension Codecs page.

Fig. II-63: Extensions Management - Edit Entry – General Settings for Auto Attendant page

Fig. II-62: Access List of Extension –Add Entry page

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Please Note: FAX forwarding is applicable only for incoming

calls from PSTN and IP networks, it is not valid for PBX calls.

The Percentage of System Memory drop down list is used to defined the space for the Auto Attendant’s system messages. The maximum value in
the drop down list is equal to the maximal available space for voice messages on Bizfon.

2. Attendant Scenario

This group is used to select between default and custom
attendant functionality scenarios. When Default scenario is
selected, a group of settings should be adjusted, user defined
Auto Attendant welcome messages can be uploaded and the list
of Friendly Phones can be configured. For Custom scenario,
scenario script file (in XML coding, the coding standard can be
found at Bizfon Technical Support) should be defined and
custom voice messages can be uploaded.

The Default manipulation radio button selection enables following components:

Fig. II-64: Extensions Management - Edit Entry – Attendant Scenario page

• Send AA Digits to Routing Table checkbox selection switches the Auto Attendant to the routing mode. Any inserted digits in the Connection

menu will be parsed through the Routing Table on the Bizfon.

Please Note: This checkbox affects ONLY Connections Menu (see Auto Attendant Services). In Call Relay Menu, the routing prefix needs

to be dialed (see Feature Codes) to parse the dialed number through the Routing Table.

• Attendant Welcome Message - this group allows updating the active Auto Attendant welcome message (played only once when entering

Auto Attendant), downloading it to the PC, or restoring the default one. The group offers the following components:

The Restore Default Welcome Message checkbox allows restoring the Auto Attendant default welcome message file if another one has
been previously selected. If the checkbox is selected, the file upload will be disabled.

Upload new welcome message indicates the file name used to upload a new welcome message. The uploaded file needs to be in PCMU
wave format, otherwise the system will prevent uploading it with the “Invalid audio file, or format is not supported” warning message. The
system also prevents uploading if there is not enough memory available for the corresponding extension, which will cause the “You do not
have enough space” warning.

Browse opens the file chooser window to browse for a new welcome message file.

The Download Welcome Message File link appears only if a file has been previously uploaded. The link is used to download the audio
file to the PC and opens the file chooser window where the saving location can be specified.

• Attendant Menu Message - this group allows updating the active Auto Attendant menu message (played after the Attendant Welcome

Message and then periodically repeated while being in the Auto Attendant), downloading it to the PC, or restoring the default one. The
group offers the following components:

The Restore Default Menu Message checkbox allows restoring the Attendant Menu Message file if another one has been previously
selected. If the checkbox is selected, the file upload will be disabled.

Upload new menu message indicates the file name used to upload a new menu message. The uploaded file needs to be in PCMU wave
format, otherwise the system will prevent uploading it with the “Invalid audio file, or format is not supported” warning message. The system
prevents uploading also if there is not enough memory available for the corresponding extension. This will cause the “You do not have
enough space” warning.

Browse opens the file chooser window to browse for a new menu message file.

The Download Menu Message File link appears only if a menu message has been previously uploaded. The link is used to download the
audio file to the PC and opens the file chooser window where the saving location can be specified.

• Friendly Phones - the Edit Authorized Phones Database link refers to the

• Authorized Phones Database page where a list of trusted external phones can be created.

The Custom manipulation radio button selection allows to upload Attendant’s custom scenario file and voice messages:

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• Upload Scenario File indicates the file name used to upload a new scenario file. The uploaded file needs to be in XML format (the coding

standard can be found at Bizfon Technical Support) and is restricted to 20KB file size. Browse opens the file chooser window to browse for
a custom scenario file.

• The View/Download Scenario link appears only when a custom scenario file has been previously uploaded and is used to view or

download the scenario file. Remove Scenario link is used to remove a custom scenario file and to turn to default Auto Attendant scenario.

• Upload Custom Voice Messages link refers to the same named page where voice messages used in the uploaded custom scenario

should be managed.

This page provides a possibility to upload voice messages to be
played in the custom Auto Attendant scenario, as well as to remove
and to download the uploaded files to PC.

Upload Custom Voice Messages page contains a table where
uploaded custom voice messages are listed. Use Download
functional button to download and Remove to delete the
corresponding custom voice message. Browse opens a file
chooser window to browse for a custom voice message.

The Edit functional button provides a possibility of editing multiple
extensions at the time. In this case, fields that cannot be edited for
multiple records have Multiple values in the Edit Entry page.
When editing user and attendant extensions together, Edit Entry
page displayed only those fields that are general for both user
extension and attendant settings. Additionally, for the fields that
need to be modified, a Select to modify fields checkbox
alongside the corresponding field needs to be selected to submit
changes, otherwise the fields will not be updated.

Delete removes the selected extensions. If no records are selected
an error message occurs. Deleting an extension from the
Extensions Table will automatically remove the Name attached to
the deleted extension from the

Upload Universal Extension Recordings link leads to the page

where universal default voice messages for all extensions are
being defined.

To Configure an Extension

1. Press the Add button on the Extensions Management page. The Add Entry page will appear in the browser window.

2. Enter the desired extension number in the Extension text field and select the extension type from the Type drop down list.

3. Press Save to create a extension with the defined number.

4. Select the checkbox of the newly created extension in the Extensions Management table and press Edit button. The Edit Entry page will

appear in the browser window.

5. Move through extension’s configuration pages and fill the fields with desired information.

6. To apply extension settings, press Save.

To Delete an Extension

1. To remove an extension with all its settings select one or more checkboxes of the corresponding extensions that ought to be deleted from the

Extensions Management table. Press Select all if all extensions ought to be deleted.

2. Click on the Delete button on the Extensions Management page.

3. Confirm the deletion with Yes. The extension will be deleted. To abort the deletion and keep the extension in the list, click No.

To Add an Authorized phone to the database

1. Enter the desired Auto Attendant Settings page.

2. Select Edit Authorized Phones Database to enter the Authorized Phones Database page.

3. Press the Add button on the Authorized Phones Database page. The Add Entry page will appear in the browser window.

4. Choose the call type and enter a caller address in the corresponding text field.

5. Select a Login Extension and the Automatically Enter Call Relay Menu checkbox (if needed).

6. Enable Call Back service if needed and define a Call Back Destination in the same named field.

7. Fill in an optional Description in the appropriate field, if needed.

8. Press Save to submit the settings.

To Delete an Authorized phone from the database

Extensions Directory.

Fig. II-66: Extensions Management - Edit Entry page for multiple edit operation

Fig. II-65: Upload Custom Voice Messages page

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1. Enter the desired Auto Attendant Settings page.

2. Select Edit Authorized Phones Database to enter the Authorized Phones Database page.

3. To remove an authorized phone(s), select one or more checkboxes of the corresponding records that ought to be deleted from the Authorized

Phones Database table. Press Select all if all records ought to be deleted.

4. Press the Delete button on the Authorized Phones Database page.

5. Confirm the deletion with Yes or cancel with No.

Extension Codecs

To establish IP voice communication, both partners have to use the same codec. During establishing the communication line, this codec is
negotiated. If the caller does not find a fitting codec, the communication cannot take place. So, if you want to be reachable by preferably all IP calls, it
is helpful to support as many codecs as possible. In this case, all the codecs that Bizfon offers should be added to the Active Codecs table. On the
other hand, some codecs require a high transfer rate - up to 64 kbit/s. If you are certain you do not want to use these codecs, you have to make sure
they are not listed in the table Active Codecs.

The Extension Codecs page displays a list of Active

Codecs and the state of the Out of Band DTMF and FAX
Support features for Bizfon extensions (also Auto

Attendant).

Please Note: Use caution when configuring Auto Attendant

Codecs as they are used by virtual extensions for
redirecting the incoming calls.

The table Active Codecs lists active voice codecs for the
selected line that are supported by Bizfon. The order of
records in the Active Codecs table is important for
transmitting and receiving. A codec placed at the top of the
table will be used as the preferred codec. If the remote
party does not support the preferred codec, the following
codecs will be tried out in a top-down order in the Active
Codecs table.

Each record in the table has an assigned checkbox. It is
used to select the record to be deleted or moved up or
down.

An error occurs if no records are selected and the user
activates the delete button: “No records selected”. At least
one codec must be attached to the line. When attempting
to delete the last codec, this error will occur: “At least one
codec should stay in the codec list”.

Fig. II-67: Extension Codecs list

Add opens the Add Entry page where the user may add
codecs supported by Bizfon. The voice codec defines the
voice compression algorithm for the incoming and outgoing
DSP packages.

Codecs lists all codecs supported by Bizfon. If no more
codecs are available (all available codecs have already
been transferred to the Active Codecs table), the Add
Entry page will display the message “No Available Codecs”
instead of the drop down menu.

Fig. II-68: Extension Codecs - Add Codec page

The Move Up/Move Down buttons are used to move the selected codec one level up/down in the table.

The Out of Band DTMF Transport checkbox enables DTMF code transmission in parallel with the voice stream. The destination receiving the
DTMF code will play it locally if it supports the feature. This is helpful to avoid DTMF’s loss upon bad traffic. This feature is valuable for all codecs but
it is especially recommended to enable it in case low bit rate codecs (G729, G723, G726/16, etc.) are selected.

Enable T.38 FAX checkbox enables the FAX tone detection and the T.38 codec support for the FAX transmission from/to the Fax Machine/Fax
modem attached to the line. Enable Pass Through FAX checkbox enables the FAX tone detection and the G711 codec support for the FAX
transmission from/to the Fax Machine/Fax modem attached to the line.

If both of these checkboxes are enabled, T.38 codec will be used as preferred codec for FAX transmit/receive and if not acceptable by the peer,
G711 codec will be used instead.

Please Note: If both of these checkboxes are disabled, no FAX transmission to the peer’s voice mailbox will be possible. Checkboxes are applicable

for FAX transmission/receipt over IP network only.

Enable Pass Through Modem checkbox enables the modem tone detection and the G711 codec support for the data transmission from/to the
modem attached to the line. During data transmission, Silence Suppression (see

RTP Settings) and Echo Cancellation are being disabled on the

line.

Force Self Codecs Preference for Inbound Calls checkbox enables the usage of the own preferred codecs (if available on both peers) for the IP

connection establishment on the extension.

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Call Park Service

Call Park service is used to store a call on a specific number so that any other user on the system can retrieve it. For example, a user receives a call
but wants to take it in a conference room where it is possible to speak privately. Transferring the call to the conference room is not an option,
because the conference room it is transferred to might be in use, or the user is unable to walk to the conference room in time to answer the call. The
user can use Call Park to place the call at a specific number and then retrieve the call on reaching the conference room.

To use the Call Park feature, the call parking service should be enabled for one or more extensions on the Bizfon from the
page.

To activate the Call Park service, the Bizfon user should dial the appropriate digit combination (see Feature Codes) during the call. The destination
party will be placed on hold, while the SIP username of the first available extension configured for the call parking (if the extension is registered on
the SIP server) and the extension’s PBX number, will be played to the Bizfon user. The Call Parking is valid for 15 minutes, during this time hold
music (if configured) will be played to the parked destination party. When the Call Park timeout expires, the phone initiating call parking will start to
ring, and if nobody picks up the parked call, or if the phone is off hook, the parked destination party will be disconnected automatically.

The pickup user will be able to pick up the parked call from any destination by simply calling the extension where the call has been parked. Either
PBX or IP calls are allowed. For PBX calls, the extension number should be dialed; for IP calls, the - SIP address played by the Bizfon when
activating the Call Park service, if it is routed to the corresponding extension. The pickup user will be prompted to pass the authentication by
inserting the password of the Bizfon user (to which call has been parked) in order to retrieve the parked call.

Example: Call Park service is enabled for extension 23, which has been registered on the SIP Server under the 892220 registration username.
Being in a call with user A, the Bizfon user dials the appropriate calling code. As a reply, Bizfon will play the 892220 to the Bizfon user, while user A
will go on hold. The Bizfon user then moves to the different location and makes a SIP call to the 892220 number. When this SIP call is established to
the 892220 number, user A will be then be connected to the Bizfon user and the conversation will resume.

Please Note: Any PBX or IP calls addressed to the extension where the call has been parked, will require to pass the authentication to reconnect the

Destination party being parked. The parked Destination party will be disconnected if an incorrect password has been inserted and authentication has
been rejected. This is why, to avoid fortuitous calls receipt on the extension used for the call parking, it is recommended to use virtual extensions for
the Call Park service.

Extensions Management

Authorized Phones Database

Authorized Phones Database page is used to create a list of trusted external phones. If they are part of the Bizfon Authorized Phones database,
external SIP or PSTN users are free to access the Bizfon Auto Attendant services without passing the authentication. When adding a friendly phone
to the list, an existing extension has to be chosen whose parameters (extension number and password, as well as SIP and Speed Calling Settings)
will be used automatically for the trusted caller access of the Bizfon Auto Attendant. A direct connection to the Call Relay menu can be provided
optionally.

The Authorized Phones Database page displays the
Authorized Phones Database table where the trusted phones
are listed. Only SIP and PSTN users can be added to the
Authorized Phones Database.

The Authorized Phones Database table displays all trusted
callers with their settings, e.g., call type, caller address,
extension they automatically login with, information if they have
automatic access to Auto Attendant (Call Relay Menu, etc.).

Each record in the table has an assigned checkbox. The checkbox is used to edit or delete the corresponding record. The “No records selected”
error occurs if the user activates the edit or delete button having no records selected. The error “One record should be selected” appears if the user
tries to edit more than one record. Each column heading in the table is created as a link. By clicking on the column heading, the table will be sorted
by the selected column. Upon sorting (ascending or descending), arrows will be displayed close to the column heading.

The Add functional button refers to the Authorized Phones Database- Add Entry page where new trusted users may be entered.

The Authorized Phones Database- Add Entry page offers two group of input options:

Fig. II-69: Authorized Phones Database

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Caller Settings

The Call Type drop down list includes possible incoming
call types (PSTN, SIP or Auto). In SIP, the caller connects
Bizfon through a SIP server and PSTN means the caller is
a PSTN user. Auto is used for undefined call types:
destination (independent on whether it is a PBX number,
SIP address or PSTN number) will be reached through
Routing.

The Caller Address text field requires the caller’s SIP
address (see chapter

Entering a SIP Addresses correctly)

or PSTN number to be added to the trusted phones’ list.
The PSTN number length depends on the area code and
phone number. The wildcard is supported in this field. If
the caller address already exists in the Authorized
Phones Database, the error “The record already exists”
appears when selecting the Save button.

Fig. II-70: Authorized Phones Database - Add Entry page

The Login Extension drop down list provides all existing extensions on the Bizfon. When calling the Bizfon Auto Attendant, a trusted user will
automatically login with the selected extension, i.e., extension number and its password will be automatically submitted by the Bizfon system. The
trusted user will directly access the Bizfon Auto Attendant services. The SIP settings of login extension will be used while making IP calls.

The Automatically Enter Call Relay Menu checkbox enables direct access for the trusted user to the Bizfon Auto Attendant Call Relay menu. If the
checkbox is not selected, a trusted caller will be directed to the Auto Attendant's main menu, but still will be able to reach Remote Access (Voice
Mailbox of the specified extension) and Call Relay services (see Feature Codes) with no authentication.

The Description text field allows entering an optional comment.

Callback Settings

The Enable Callback checkbox selection gives a possibility for specified trusted caller to use the Instant Call Back service (see chapter

Services

).

Call Back

The Callback Destination text field requires the destination PSTN number where Bizfon should Instantly Call Back. If this field is empty, caller
address will be implied as a callback destination.

Please Note: The Call Back service is functional and can be enabled only for PSTN callers and is valid for the PSTN callback destinations only.

Call Back Services

With the Call Back service the PSTN callers can save the call charge when calling to/through Bizfon. Bizfon gives a possibility to create a list of
those trusted PSTN callers that are allowed to make free of charge calls to Bizfon’s Auto Attendant or through its Call Relay menu to the third party
IP or PSTN destination.

Two types of Call Back are available on the Bizfon: Instant Call Back and Roaming Call Back.

Instant Call Back

For Instant Call Back service a list of trusted PSTN callers must be pre-configured in the Authorized Phones Database on the Bizfon. Call Back
service should be enabled and a valid callback PSTN destination should be specified for the corresponding PSTN caller.

To use Instant Call Back, PSTN caller registered in the Authorized Phones Database should simply call to Bizfon’s PSTN number (that should be
previously routed to the Auto Attendant or Routing Manager from the

FXO

Settings page) from the global PSTN network, let the call ring twice and then hang up. Call Back will get instantly activated, i.e. Bizfon will call
back to the defined Call Back destination and by answering the incoming call PSTN party will be automatically connected either to Auto Attendant or
Routing Manager depending on the configuration of the corresponding FXO line on the Bizfon.

Roaming Call Back

The Roaming Call Back allows to configure the call back by callers registered in Authorized Phones Database on the Bizfon when calling from a
PSTN number. Roaming Call Back is divided into two modes accessible from the Bizfon’s Auto Attendant: Non Permanent Call Back and

Permanent Call Back.

Non Permanent Call Back can be used from the corresponding menu of the Bizfon’s Auto Attendant (see Call Codes). PSTN caller should pass the

authorization by dialing existing extension number and an appropriate password. Normally, the PSTN caller’s address should be detected
automatically and then system will simply ask for the confirmation (in particular cases when caller is configuring Non Permanent Call Back service for
him (or for anyone else) calling from the other number, other caller number should be defined here, so Instant Call Back will get activated only when
calling from the defined caller number). If PSTN caller’s address is not detected automatically, caller will be required to insert it manually (in this case
Instant Call Back service will get activated immediately after hanging up). Call Back destination, where Bizfon should call to, will be requested

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afterwards. It can be the same as the caller’s address or can be different. When system accepts the call back settings, PSTN caller will be
disconnected from the Bizfon’s Auto Attendant.

If the Non Permanent Call Back has been configured for the other caller address, system will wait till the incoming call will arrive from that other
caller number, and after caller will let the call ring twice and hang up, Bizfon will send a call to the defined PSTN destination in the next 45 seconds (if
FXO line is available on the Bizfon, network connectivity is fine and destination is reachable). Answering the incoming call, PSTN caller will be
connected to the Bizfon’s Auto Attendant.

Next time, when PSTN caller reaches Bizfon from the same number, he needs to pass the described procedure again since this was the one-time
Call Back only and no entry was stored in the Authorized Phones Database on the Bizfon.

Permanent Call Back service offers a convenience of registering new trusted PSTN Callers and to edit the Call Back destination of an existing
PSTN Caller in the Authorized Phones Database. By calling Bizfon’s PSTN number (that is previously routed to the Auto Attendant) caller enters the
Bizfon’s Auto Attendant and by Permanent Call Back menu (see Call Codes) he is able to register himself (or anyone else) as a trusted PSTN caller
that is allowed to place free of charge calls to Bizfon or through its Call Relay menu to the third party IP or PSTN destination as well as to modify the
Call Back destination of an already registered Caller in the Authorized Phones Database.

Entering the Permanent Call Back menu, system will ask to login by dialing existing extension number and an appropriate password. PSTN caller’s
address confirmation will be required, or, if not detected automatically, it should be defined manually (in particular cases when caller is configuring
Permanent Call Back service for him (or for anyone else) calling from the other number, other caller number should be defined here, so next time
calling from that number, Instant Call Back will get automatically activated). Call Back destination, where Bizfon should call to, will be requested
afterwards. It can be the same as the caller’s address or can be different. When system accepts the call back settings, the corresponding entry will
be logged to the Authorized Phones Database.

PSTN caller will be disconnected from the Bizfon’s Auto Attendant and the defined Call Back destination will receive a call from the Bizfon in the next
45 seconds (if FXO line is available on the Bizfon, network connectivity is fine and destination is reachable) if the detected PSTN caller address
corresponds to the one applied by the caller (i.e. caller hasn’t changed the detected caller address) or if caller address is not detected at all (due to
system configuration problems or CO peculiarity). Otherwise, system will send a call back to the specified callback destination only if call arrives from
the address logged in the Authorized Phones Database. Answering the incoming call, PSTN caller will be connected to the Bizfon’s Auto Attendant.

Being registered in the Authorized Phones Database once (by means of Permanent Call Back service or from the Bizfon’s Web Management),
PSTN caller is able to use Instant Call Back service, i.e. next time when calling from the same PSTN number to the Bizfon and hanging up after the
second ring, the system will call the defined Call Back destination since the number is already registered in the Authorized Phones Database on the
Bizfon.

Upload Universal Extension Recordings

The Upload Universal Extension Recordings are to be defined by the Bizfon administrator, will stand instead of the default voice messages for all
extensions on the Bizfon and will be used when no custom messages has been uploaded or recorded.

Following system messages can be uploaded from this page:

• Hold Music – played to the held user

• Voice Mail Regular Greeting – played when caller reaches the extension’s voice mailbox

• Voice Mail Out-of-Office Greeting – played when caller reaches the extension’s voice mailbox if Out-of-office greeting is enabled

• Incoming call blocking - played when a blocked user calls the extension

• Outgoing call blocking – played when extension dials a blocked destination

The Upload Universal Extension Recordings page consists of
a table where universal voice messages are listed.

An Upload functional link is present for each not uploaded voice
message in the table and is used to upload the custom system
message. When a message is uploaded, Upload functional link
is replaced by Download and Remove functional links
respectively used to download to the PC and to remove the
uploaded system message.

Memory Allocation group includes a drop down list used to
specify the Percentage of System Memory for the universal
extension recordings. The maximum value in the drop down list
is equal to the maximal available space for voice messages on
Bizfon.

Please Note: Changing the Percentage of System Memory on this page will stop any recordings of universal extension voice messages from the

handset.

Fig. II-71: Upload Universal Extension Recordings page

Receptionist Management

Receptionist feature on the Bizfon offers a bunch of services to manipulate with multiple calls, to keep the calls in the queue with the perspective to
be answered by the receptionist and finally to be forwarded to the corresponding destination, if needed.

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Please Note: It is recommended to have the Snom360 IP phone for the receptionist in order to be able to use the services below.

Following services are available to the receptionist:

• Call Queue

• Extension Status

• Call Interception

• Voicemail Transfer

• Multi-Company Receptionist

Call Queue

Feature allows to keep multiple incoming calls in the queue when being on the line and to answer calls in the order they have been received. The
usage of this service is not limited to receptionist only and can be used also by the extension user, if configured correspondingly.

The configuration of Call Queue feature is done from the

Extensions Management – Edit Entry page, where the length of the call queue and the call

queue appearance can be particularly defined. When Call Queue service is enabled, the second arriving call to the receptionist/extension user will be
either set into the queue (if call queue appearance is 1) or will be ringing in the background of the active call (if call waiting is enabled for the user
and the call queue appearance value is greater than 1). If the call ringing in the background won’t be answered, it will be transferred to the user’s
voice mailbox or, if no answer forwarding is enabled, will be forwarded to the corresponding destination.

If call is set into the queue, the caller will hear a message asking to wait until the call will be answered. Once receptionist/extension user terminates
the call, the next call in the queue will be ringing to the user.

Most of IP phone provide a possibility to the receptionist/extension user to monitor the call queue and the members in it even while being in call.
Some IP phones have a lamp indication when there will be at least one caller in the call queue (for Snom360 IP Phone, it is the Messages lamp).
Additionally, IP phones have an appropriate button (for Snom360 IP Phone, it is the Retrieve button) which allows the receptionist/extension user to
get information about the total number of callers in the queue and the name/phone number of the last caller.

For the regular FXS users, indication about the callers in the queue might be got by means of Call Waiting service (see ManualIII-Extension Users
Guide). When new caller arrives to the call queue, the phone display (if available) of the phone connected to the FXS will display the total number of
callers in the queue along with the name/phone number of the last caller.

Extension Status

Bizfon provides a possibility to control and determine the actual state of the manager phones watched to the receptionist’s IP phone (configuration of
the IP phone is done automatically by Bizfon through Receptionist Phone Configuration Wizard). Hence, the programmable key assigned to the
corresponding manager will blink if incoming call is received and manager’s phone is currently ringing, key lamp will be ON when manager is in call
and will be OFF if manager’s phone is in the idle state. Extension status can be used by the receptionist to get the actual information about the
available managers for incoming call transfer.

Call Interception

The functionality of this service is limited to the capabilities of the Snom360 IP phone used as an official hardware for the Receptionist Management
on the Bizfon. To use the service of Call Interception, managers’ phones watch option should be enabled and each manager should have a
programmable key assigned on the receptionist’s IP phone. This is performed automatically by Bizfon through Receptionist Phone Configuration
Wizard.

When incoming call addressed to the certain manager comes in, receptionist can see blinking of the corresponding programmable key and the
caller’s ID (for Snom360 only) on the phone’s display. Receptionist is able to intercept the incoming call by pressing the blinking key. Caller will be
connected to the receptionist then. If receptionist does not answer the call addressed to the manager, and if manager does not answer it either, call
will be directed to the manager’s voice mailbox, if enabled, otherwise disconnected.

Voicemail Transfer

Bizfon allows receptionist/extension user to forward incoming calls directly to the voicemail of the other attached extension. To do so, an appropriate
routing pattern should be added to the Call Routing table. Hence, when transferring a call to the assigned extension, incoming call will directly go to
the extension’s voice mailbox.

Multi-Company Receptionist

Bizfon provides a possibility to use the single IP phone (Snom 360) to manage the receptionist features for multiple companies at once. To do so,
incoming line appearance on the phone should be created, attached to the IP line of the IP phone and be labeled to the corresponding company
name. Being busy with a call related to one company, receptionist is able to receive also the calls related to other companies,

calls will be ringing in
the background, and receptionist can switch between the incoming calls. However, if receptionist does not answer the incoming calls, and if Call
Queue service is enabled on the extensions, incoming calls will be stored into the queue specific for each company line.

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Receptionist Management page allows to configure IP phones
to be used as a receptionist on the Bizfon. Page contains the list
of configured receptionists with information about the attached
IP lines and watched extensions.

Fig. II-72: Receptionist Management page

Add opens the Receptionist Phone Configuration Wizard where the new receptionist can be created and configured. Wizard consists of several
pages.

The Receptionist Phone Configuration Wizard - Page 1 has
the following components:

Description text field requires the description of the receptionist
to be configured.

Phone Model drop down list is used to select the IP phone
model to be used by the receptionist. Snom selection in the list
enables the MAC address text fields used to insert the MAC
Address of the corresponding Snom IP phone.

Based on the selected IP phone model and the inserted MAC
Address, the SIP phone can be automatically configuration by
simple reset/reboot (for more information about IP phone
configuration, refer to the corresponding IP phone’s users
manual).

Attached IP Lines text field requires the numbers of Bizfon’s IP
lines used by the receptionist. IP lines should be separated by
commas.

Fig. II-73: Receptionist Phone Configuration Wizard – Page 1

The Receptionist Phone Configuration Wizard - Page 2 is
available only for Snom selection in the Phone Model drop
down list on the previous page and for multiple Attached IP
Lines. Page is used to set the correspondence between the
selected IP lines and the available Programmable keys on the
IP Phone.

To do so, select the IP lines corresponding to each
programmable key from the drop down list on the page.

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Fig. II-74: Receptionist Phone Configuration Wizard – Page 2

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The Receptionist Phone Configuration Wizard - Page 3 is
available only for Snom selection in the Phone Model drop
down list on the previous page. Page is used to set the watched
extensions to the Programmable keys on the IP Phone.

To do so, select the extension from the corresponding drop
down lists in order to associate the corresponding extension
with the certain programmable key.

Please Note: A Programmable Key can be either assigned to

an IP line or to a watched extension.

Fig. II-75: Receptionist Phone Configuration Wizard – Page 3

Please Note: Once a new receptionist is created, Call Queue feature will be automatically enabled with the corresponding Call Queue Size and

Max Call Queue Appearance settings on all extensions attached to the IP lines defined in the Attached IP Lines text field.

Extensions Directory

The Extensions Directory is a useful tool for callers to get direct access to the Bizfon extensions by spelling the username with the help of the
phone keypad. The Extensions Directory can be accessed through Bizfon’s Auto Attendant and has its own manipulation buttons to browse the
directory.

The Extensions Directory Settings page allows to make a list of names assigned to the extensions on the Bizfon. If the name spelled by the caller
matched the one(s) listed in the Extensions Directory, the corresponding extension user name(s) will be played to the caller, for verifying the input
and selecting the user to connect. Each extension’s user should record their name with the help of the handset (see chapter Update System

Messages), or can simply upload a wave file from the Account Settings page.

The Custom Greeting column in the Extensions
Directory table displays whether or not a custom
greeting (user’s name) is recorded or uploaded. Users
cannot be accessed through the Extensions Directory
and it is implied as being an inactive entry in the event
a custom greeting is not recorded or uploaded.
Warnings will be seen in the Extensions Directory
table for inactive entries. Extension numbers in the
Extensions Directory table are made as a link to move
to the corresponding extension's Account Settings
page. This can help the administrator access the
extension's settings page where a custom greeting
can be manually uploaded.

Fig. II-76: Extension Directory table

Move Up and Move Down are used to move the
selected record one level up or down in the
Extensions Directory table. The consecution of the
entries in the Extensions Directory is important if
several records match the spelled name. The
Extensions Directory table is being parsed from the
top down and the matched entries will be played
according to their position in the table.

Add opens the Add Entry page where a new name
may be assigned to the extension. An error message
appears and prevents form adding a new entry to the
Extensions Directory if no extensions are available in
the

Extensions Management table.

Fig. II-77: Extensions Directory - Add Entry page

The Add Entry page offers the following components:

Name requires the name of the extension owner. Several extensions can have the same name and a single extension may have several names.
User’s Name is the identification parameter being searched within the Extensions Directory. It is desirable to use uppercases in this field, otherwise
name will be automatically changed to uppercase when saving it to the Extensions Directory table.

Call to includes a list of all extensions on the Bizfon that should ring when selecting the specified Name.

Description f can be used for any optional information requiring entry in the Extensions Directory.

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Please Note: The entries in the Extensions Directory can automatically be deleted if the extensions assigned to the entries are removed from the

Extensions Management table.

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Telephony Menu

Fig. II-78: Telephony Menu in Dynamo Theme

Fig. II-79: Telephony Menu in Plain Theme

Call Statistics

The Call Statistics page displays four tables and provides information on successful, unsuccessful and missed incoming and outgoing calls on the
first three tables, and statistics settings on the fourth page. Call statistics allows the collecting of call events on the Bizfon with their parameters and
to search them by various criteria.

The Statistics Settings page offers the following input options:

The Enable Call Reporting checkbox enables Call Statistics
reporting. The selected number of statistics entries will be
displayed in the Call Statistics tables.

The Maximal Number of Displayed Call Records drop down
lists are used to select the number of Successful, Missed and
Nonsuccessful statistics entries to be displayed in the
corresponding Call Statistics tables. If the record numbers
exceed the numbers specified in these drop down lists, the
oldest record will be removed.

The Download Call Statistics link is used to download all
displayed statistics in a file that can be viewed with a simple text
editor.

The Clear all Records button is used to clear all statistics
records.

The Number of Records displays the current number of
statistics entries in the table. For the successful calls, Total

Duration, Maximum Duration, Average Duration and
Minimum Duration statistics are displayed on top of the table.

The Call Statistics: Successful Calls, Missed Calls and
Nonsuccessful Calls pages consist of the general information

on successful, missed and unsuccessful calls, search fields and
the calls table. The search components are as follows:

The From and To text fields are used to search by date and
time. The data has to be entered in either of the following
formats: dd-mm-yyyy hh:mm:ss or dd-Mon-yyyy hh:mm:ss. The
time criteria is optional. From requires an earlier date and time
than the To field. If the entered data does not meet this
condition, the error message “Minimal date should be less than
maximal date” prevents statistics filtering.

The From and To drop down lists are used to search by
duration. The duration has to be selected from the list values.
The From field has to indicate a shorter duration than the To
field. If the inserted data does not meet this condition, the error
message “Minimal duration should be less than maximal
duration” prevents statistics filtering.

Calling Phone and Called Phone respectively require the caller
and called party’s SIP address (see chapter

Entering a SIP

Addresses correctly), extension or PSTN number as search

Fig. II-80: Call Statistics Settings page

Fig. II-81: Call Statistics page

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criteria. Wildcard symbols are allowed here.

The Call Statistics - Successful Calls, Call Statistics -
Missed Calls and Call Statistics - NonSuccessful Calls tables
are lists of successful, missed and unsuccessful incoming and
outgoing calls and their parameters (Call Start Time, Call
Duration, Call destinations). Each column heading in the tables
is a link. By clicking on the column heading, the table will be
sorted by the selected column. Upon sorting (ascending or
descending), arrows will be displayed close to the column
heading.

Network Details column is present in Successful Calls table only and provides brief information about the call quality and codecs used for receive
and transmit packets. Clicking on the successful call details will open

RTP Statistics page where detailed information about the established call is

provided. Call Detail column is present in Non Successful Calls table only and indicates the reason of the call being unsuccessful.

Filter performs a search procedure by the selected criteria. The search may be done with several criteria at once.

To Enable/Disable the Statistics

1. Enter the Call Statistics Settings page.

2. Select/deselect the Enable Call Reporting checkbox to enable/disable statistics recording.

3. If enabling the statistics, the maximum number of records to be stored in the statistics table should be selected from the corresponding drop

down lists.

4. Press Save to apply the new configuration.

To Filter the Statistics

1. Enter the desired criteria fields.

2. Press the Filter button to search the call reports within the Call Statistics table.
Please Note: To return to the complete Statistics Table clear all search criteria and press Filter.

To Reset the Statistics

1. Press the Clear All Records button in the Call Statistics Settings page.

2. Confirm the deletion with Yes. The call statistics will be deleted. To abort the deletion and keep the statistics information, click No.

RTP Statistics

The RTP Statistics page provides detailed information about the
established call is provided.

Quality - estimated call quality, which depends on RTP statistic.
Below is the legend for Call Quality definitions on the displayed RTP
Statistics:

excellent – RX Lost Packets < 1% & RX Jitter < 20
good - RX Lost Packets < 5% & RX Jitter < 80
satisfactory - RX Lost Packets < 10% & RX Jitter < 150
bad - RX Lost Packets < 20% & RX Jitter < 200

Fig. II-82: RTP Statistics page

very bad - RX Lost Packets > 20% or RX Jitter > 200

Rx/Tx Codec - codec for received and transmitted RTP stream respectively.

Rx/Tx Packets - number of RTP packets received and transmitted respectively.

Rx/Tx Packet Size - size of RTP packet (payload) received and transmitted respectively.

Rx Lost Packets - number of lost RTP packets for received stream.

Rx Jitter - inter-arrival jitter, an estimate of the statistical variance of the RTP data packet inter-arrival time, measured in timestamp units.

The inter-arrival jitter is defined to be the mean deviation (smoothed absolute value) of the difference D in packet spacing at the receiver compared
to the sender for a pair of packets. If Si is the RTP timestamp from packet i, and Ri is the time of arrival in RTP timestamp units for packet i, then for
two packets i and j, D may be expressed as:

D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si)

J(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16, where J(i) is Rx Jitter for packet i.

For more details about Jitter calculations, please refer to the RFC1889.

Rx Maximum Delay - maximum variance (absolute value) of actual arrival time of the RTP data packet compared to estimated arrival time,
measured in milliseconds.

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If Si is the RTP timestamp from packet i, and Ri is the time of arrival in RTP timestamp units for packet i, then variance for packet i may be
expressed as following: V(i) = |(Ri - R1) - (Si - S1)| = |(Ri - Si) - (R1 - S1)|

Rx Maximum Delay = max V(i) / 8

Please Note: RTP Statistics is logged only when at least one of the call endpoints is located on the Bizfon, e.g. it will not be logged when:

• calls incoming from or addressed to the IP lines or remote extension,

• calls from an external user are routed to another external user through Bizfon’s routing rules.

In the first case, RTP statistics will be logged if remote extension or IP line user is calling locally to the Bizfon’s extension or auto attendant.

SIP Settings

The SIP Settings provide information on the SIP receive UDP and TCP ports and allows to select DNS server configuration for SIP and SIP timers
scheme.

UDP Port indicates the SIP UDP (User Datagram Protocol)
receive port number. By default 5060 is selected and used.
The SIP UDP port cannot be in the selected RTP/RTCP port
range for FXS and IP lines (see
“Mapped port for SIP shouldn’t be in RTP port range” error
appears.

TCP Port indicates the SIP TCP (Transmission Control
Protocol) receive port number. By default 5060 is selected and
used.

Please Note: Bizfon will not use TCP protocol as a transport

for SIP messages if the TCP Port field is left empty.

Enable Session Timer enables advanced mechanisms for

connection activity checking. This option allows both user
agents and proxies to determine if the SIP session is still
active.

The DNS server for SIP radio button group allows to choose between regular DNS servers configured in the DNS Settings page and specific DNS
servers for the SIP traffic.

RTP Settings), otherwise the

Fig. II-83: SIP Settings page

• Use default is used to apply regular DNS servers for the SIP traffic.

• Specific is used to enable SIP specific DNS servers. For this selection, both primary and secondary SIP DNS servers should be defined in the

SIP DNS 1 and SIP DNS 2 text fields. At the least, a primary DNS server should be inserted.

The SIP Timers radio button group is used to define the timeouts of the SIP messages retransmission.

• RFC 3261 will apply standard SIP timers described in the corresponding specification.

• High availability will apply SIP timers to shorten the call establishment, registration confirmation and registration failure procedures. This

selection provides more firmness to the SIP connection but increases the network traffic on the Bizfon.

• Custom allows defining manually the Registration Timeout, Registration Failure Timeout, Transaction Duration and Session refresh

timeout SIP timers (in seconds).

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RTP Settings

The RTP Settings page allows the administrator to configure the codec’s packet size and silence suppression for each voice codec, to select the
G726 codec standard, to define RTP/RTCP port ranges, etc. All parameters listed on this page may be modified and submitted.

The Codec Properties table lists all codecs with the
corresponding packetization interval and information about
silence suppression.

Edit opens the Edit RTP Settings page where the codec
settings can be modified. To use Edit, only one codec may be
selected at a time, otherwise an error occurs: “One record
should be selected”.

The Packetization Interval is the time interval between two
RTP packets of the same stream. If the interval is increased, the
overhead is decreased but the voice quality may deteriorate as
a result. If the interval is decreased, the network load is
increased and the delay is reduced.

Silence Suppression disables RTP packet transmission in
case of no voice activity. This feature helps to avoid extra traffic
if the RTP stream contains no voice. It is activated after two
seconds of silence and restarted immediately if any audio
appears.

The G.726 Standard radio buttons are used to select between
packaging the G.726 codewords into octets. If you experience
problems with G.726 voice quality having one of these
packaging selected, try the other one.

• If Use ITU_T specification is selected, the ITU I.366.2

(“AAL2 type 2 service specific convergence sublayer for
narrow-band services”) type packaging of codewords is
used, where packing code words into octets is starting from
the most significant rather than the least significant digit in
the octet.

• If Use IETF RFC is selected, the IETF RFC (“RTP Profile

for Audio and Video Conferences with Minimal Control”)
type packaging of codewords is used, where packing code
words is starting from the least significant position in the
octet.

RTP/RTCP Port Range for FXS Lines and RTP/RTCP Port Range for IP Lines:

Fig. II-84: RTP Settings page

• Min - minimal port has to be higher than 1024 and lower than the maximal port range. Only even numbers are allowed.

• Max - maximal port has to be lower than 65536 and higher than the minimal port range. Only odd numbers are allowed.

As the specified maximum port has to be higher than the minimum port, the error message “Min port number should be less than max port number”
will occur if this condition is not met.
allowed” occurs. The difference between Max and Min RTP ports should be 50 ports or less (according to the system’s capabilities) otherwise the
corresponding warning appears. RTP/RTCP Port ranges cannot include the defined SIP UDP ports (see

Telephone Event Draft Support enables telephony events transmission according to the draft-ietf-avt-rfc2833bis-04. The checkbox needs to be
toggled if the SIP destination party phone or IVR has problems recognizing DTMFs generated by the Bizfon.

Enable RTCP Support enables Real Time Control Protocol support and allows for the RTCP packets transmission. RTCP protocol is used for
monitoring the RTP streams and changing RTP characteristics depending on Network conditions.

The port range may consist of digits only, otherwise the error “Incorrect Port Range: only Integer values

SIP Settings) otherwise an error appears.

The RTP Settings – Edit Entry page offers a drop down list and a
checkbox.

Packetization Interval contains possible values (in milliseconds)
to be configured for the selected codec.

The Enable Silence Suppression checkbox selection enables
voice activity detection for the selected codec.

To Edit Codec Parameters

1. Select the codec from the Codecs Table that is to be edited.

2. Press the Edit button on the RTP Settings page. The Edit Entry page will appear in the browser window.

3. Change values in Packetization Interval and/or enable/disable Silence Suppression.

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Fig. II-85: RTP Settings - Edit Entry

Bizfon Manual II: Administrator's Guide Administrator's Menus

4. To save the codec settings press Save, or to keep the initial data click Back.

NAT Traversal Settings

The NAT Traversal Settings page is divided into separate pages used to configure General NAT settings, SIP NAT parameters, RTP and STUN
parameters for NAT and a page where the NAT Exclusion table may be filled.

The General Settings page consists of a manipulation radio
buttons group to select the mode of the NAT Traversal usage for
the SIP traffic (any incoming and outgoing SIP messages from
and to the Bizfon will be routed through the NAT PC).

• Automatic – with this selection, system will analize the

Bizfon’s WAN IP address and if it is in the IP range
specified for local networks (according to RFC), the SIP
traffic will be parsed over NAT, otherwise, if Bizfon’s WAN
IP address is outside the specified IP range, no SIP traffic
will be routed through NAT server.

• Force – with this selection, all SIP traffic will be routed

through NAT server.

• Disable – with this selection, no SIP traffic will be routed

through NAT server.

The SIP Parameters page is used to configure NAT specific
settings for SIP. and offers two independent group of settings:

UDP Parameters:
Manipulation radio buttons allow to select the type of connection
over NAT:

Selecting Use STUN will switch to automatic discovery of
Mapped settings for the SIP UDP traffic over NAT. STUN
settings are configured on the STUN parameters page (see
below).

Selecting Use Manual NAT Traversal allows to define
manually the mapped settings for the SIP UDP traffic over
NAT:

Mapped Host requires the IP address of the mapped host for
SIP UDP traffic over NAT.

Mapped Port requires the port number on the mapped host
for the SIP UDP traffic over NAT.

TCP Parameters:

Mapped Host requires the IP address of the mapped host for

SIP TCP traffic over NAT.

Mapped Port requires the port number on the mapped host
for the SIP TCP traffic over NAT.

The RTP Parameters page is used to choose between the
STUN and Manual NAT traversal connection for the RTP traffic
and to define the RTP/RTCP ports for the connection over NAT.

Manipulation radio buttons allow to select the type of connection
over NAT:

Selecting Use STUN will switch to automatic discovery of
Mapped settings for the RTP UDP traffic over NAT. STUN
settings are configured on the STUN Parameters page (see
below).

Selecting Use Manual NAT Traversal allows to define manually
the RTP/RTCP port ranges for the RTP traffic over NAT:

Fig. II-86: General NAT traversal page

Fig. II-87: SIP Parameters page

• The Mapped Host text fields require the Mapped Host for

RTP traffic over NAT.

• Mapped RTP/RTCP Port Range for FXS Lines and

Mapped RTP/RTCP Port Range for IP Lines:

• Min - minimal port has to be higher than 1024 and lower

than the maximal port range. Only even numbers are

Fig. II-88: RTP Parameters page

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allowed.

• Max - maximal port has to be lower than 65536 and

higher than the minimal port range. Only odd numbers
are allowed.

Please Note: RTP/RTCP Mapped Port ranges should be

greater than or equal to the RTP/RTCP port ranges defined on
the

RTP Settings page.

The STUN Parameters page enables automatic NAT
configuration through the STUN server and is used to configure
the STUN (Simple Traversal of UDP over NAT) client on the
Bizfon. The page requires the following data to be inserted:

The STUN Server text field requires the STUN server’s
hostname or IP address. The STUN Port text field requires the
STUN server port number.

The Secondary STUN Server and Secondary STUN Port text
fields respectively require the parameters of the secondary
STUN server.

The Polling Interval drop down list contains the possible time
intervals between referrals to the STUN server.

The Keep-alive interval text field gives the possibility to select
the time interval (in seconds) for keeping NAT mapping alive.

The NAT IP checking interval text field indicates interval (in
seconds) between the NAT IP checking attempts (used to
distinguish the possible NAT IP address changes and to perform
registration on the new host). Value should be in a range from
10 to 3600.

The NAT Exclusion Table page includes a table where all possible IP ranges are listed that allows to exclude some network addresses from being
NATed. For example, if a Bizfon user needs to make SIP calls within the local network as well as outside of that network, all local IP addresses are
required to be excluded from NAT traversal settings by being listed in this table. Otherwise, a malfunction may occur in SIP operations.

The NAT Exclusion Table page offers the following
input options:

Each record in the table has its checkbox assigned to its
row. This checkbox is used to delete or to edit the
corresponding record. As only one record may be edited
at a time, an error message appears, if none or more
than one is selected.

Each column heading in the table is a link. By clicking on
the column heading, the table will be sorted by the
selected column. Upon sorting (ascending or
descending), arrows will be displayed close to the
column heading.

The Add Entry page includes the following text fields:

Add opens the Add Entry page where a new IP range
can be added.

Edit opens the Edit Entry page where the IP range can
be modified. The page includes the same components
as the Add Entry page.

The NAT Exclusion Table lists all possible IP ranges
that are not included into the NAT process, but may be
accessed directly. IP addresses that are not listed in the
NAT Exclusion Table are accessed over NAT.

Fig. II-89: STUN Parameters page

Fig. II-90: NAT Exclusion Table page

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IP address requires the IP address that is placed behind
NAT within the local network.

Subnet Mask requires the subnet mask corresponding
to the specified IP address.

To Configure the NAT Exclusion Table

1. Press the Add button on the NAT Exclusion Table page. The Add Entry page will appear in the browser window.

2. Specify an IP Address and its Subnet Mask in the corresponding text fields.

3. Press Save on the Add Entry page to add the selected IP range to the NAT Exclusion Table list.

To Delete an IP Range from the NAT Exclusion Table

1. Select the checkboxes of the corresponding IP range(s) that ought to be deleted from the NAT Exclusion Table. Press Select all if all IP

ranges ought to be deleted.

2. Press the Delete button on the NAT Exclusion Table page.

3. Confirm the deletion with Yes. The IP range will be deleted. To abort the deletion and keep the IP range in the list, press No.

Fig. II-91: NAT Exclusion Table - Add Entry page

Line Settings

The Line Settings are used to configure Bizfon FXS and IP Line (if available on the board) settings. The Line Settings page consists of two pages:
Onboard Line Settings page for onboard FXS lines configuration and IP Line Settings for IP Lines configuration.

Onboard Line Settings

The Onboard Line Settings page is used to configure Bizfon lines and to define the caller ID detection type, configure remote party disconnect
indication and select the ringer type on each of them. Additionally this page provides a possibility to enable Loopback diagnostics on the lines.

The page Onboard Line Settings shows the table Available

Lines where all active lines of Bizfon are listed with their Attached
Extension (if the line is attached to an extension, the

corresponding extension number is displayed in this column (else,
“none” is displayed if extension is not attached to the line), and
clicking on the extension number the
General Settings page will appear, where the line attached to the
extension can be reconfigured). Further, the table provides
information about the selected Ringer Type and Caller ID
detection method that is configured for the selected line. The caller
ID detection method is different for various types of phones and
can be found in the phone manual.

The Loopback Settings link takes you to the page where lines
can be configured for loopback diagnostics purposes.

When pressing on the line number under the Available Lines column
offers the following input options:

Extensions Management –

Fig. II-92: Line Settings Page

, the Onboard Line Settings page specific for the current line is opened and

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Caller ID drop down list contains various standards of Caller ID
transmission used to send the calling party's information to the
phone attached to the selected line:

• No Caller ID.

• FSK, send prior to the first ring.

• FSK, send between the first and second ring.

• FSK, send both prior to ring and between the first and second

ring.

• DTMF, send prior to the first ring.

• DTMF, send between the first and the second ring.

• Combined, send both DTMF prior to the first ring and FSK

between the first and the second rings.

The Bizfon sends the current time/date to the called phone
together with the caller’s information.

A group of Remote Party Disconnect Indication parameters are used especially to configure the private PBX attached to the Bizfon FXS port.

Fig. II-93: Line Codec and Caller ID Settings page

• The Enable Busy Tone Indication checkbox enables the busy tone transmission to the FXS port when the remote party being in call is

disconnected. The Busy Tone Duration drop down list is used to select the period (in seconds) when a busy tone will be transmitted to the
FXS port.

• The Enable Power Disconnect Indication checkbox enables the power cycling on the FXS line when the remote party being in call is

disconnected. Power Disconnect is applied after the busy tone transmission on the FXS line. The Disconnect Duration drop down list is
used to select the period (in milliseconds) when the FXS line power will be down.

The Ringer Type drop down list allows to select the frequency of the ringer supported by the phone attached to the line. Information can be found on
the phone enclosure or in the phone's manual. Problems with the ringer might occur if the ringer type selected here does not correspond to the one
supported by the phone.

Please Note: The supported ringer type can be found on the phone bottom, in the “Ren:x.xN” value where N is the ringer type supported by the

phone, (e.g., if N=A, the TypeA ringer type should be selected, if N=B, the TypeB&Z ringer type should be selected).

The Enable off-hook Caller ID checkbox enables Caller ID transmission to the phone in the off-hook state attached to the certain line. Service is
applicable to the phones supporting the Call Waiting Caller ID feature.

Information on the Caller ID system:

Caller ID is a service identifying the caller (when performing a call or sending a voice mail) and notifying the called party about the identity of the

caller. Caller ID service is available only for phones with a display to show that information. Two types of Caller ID notification are available on
Bizfon: FSK and DTMF.

FSK Standard

The FSK standard supports caller ID indication either with the phone handset on-hook or if the called party is already busy with another call or
operation (handset is off-hook). For internal calls, caller ID notification in FSK can show up to two lines of identifiable parameters on the called
phone’s display. The first line shows the caller’s extension number. The second line shows the caller’s nickname (if indicated in the configuration).
For external IP calls, caller ID notification in FSK can also show up to two lines of identifiable parameters on the called phone’s display. The first line
shows the caller’s user name. The second line shows the caller’s nickname (if indicated in configuration). If the nickname is not available and there is
a display name, provided by the caller party, the second line will display it, otherwise the URL in the format: username@host will be shown instead.
For calls from the PSTN network, the entire caller ID message will be shown, sent by the PSTN station.

DTMF Standard

The DTMF standard supports caller ID indication only if the phone handset is on-hook (phone is free and ready to accept calls). This standard also
has caller ID notification conditions but they are nonconfigurable as well. Caller ID notification in DTMF can show only one line of identifiable
parameter on the called phone’s display. For internal calls, it is the caller’s extension number. For external IP calls, it is the caller’s user name. For
calls from the PSTN network, caller ID will display the caller’s phone number only.

Please Note: DTMF supports only parameters consisting of digits. If any letter symbol has been used in the external caller user name, DTMF will

display no caller ID at all.

To Configure the Line Settings

1. Select the line number that ought to be configured from the Active Lines column on the Lines table on the Line Settings page

2. Press on the line number link from the Line Settings table. The Line Settings - Line# page will appear in the browser window.

3. Use the Caller ID drop down list to select the caller ID detection system mode corresponding to the phone type.

4. Enable Dialing Prefix With Caller ID checkbox if needed.

5. Configure Remote Party Disconnect Indication parameters by selecting the corresponding checkboxes.

6. Define a Ringer Type from the corresponding drop down list.

7. Enable Off-hook Caller ID if needed.

8. Press the Save button on the Line Settings - Line# page to save the caller ID system and other line specific configuration settings.

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IP Line Settings

The IP Line Settings page is used to configure IP lines for IP phones to be connected to the Bizfon. Bizfon provides the possibility to connect MGCP
and SIP phones to its LAN side, assign the corresponding IP line to some active extension, and use MGCP and SIP phone as a simple phone with
all telephony services of the Bizfon, for example, call hold, waiting, transfer, etc. 30 IP Lines are available on the Bizfon4000. More IP lines can be
enabled by entering the feature key in the

The IP Lines Settings page displays a table with the available IP lines on the Bizfon.

The IP Lines table lists all available IP lines with additional information
about each of them: number of the extension attached to it, information
about the phone type and the configuration details.

Each column heading in the tables is link. By clicking on the column
heading, the table will be sorted by the selected column. Upon sorting
(ascending or descending), arrows will be displayed close to the column
heading.

Pressing on the IP line link in the Available IP Lines column
IP Line page specific for the current IP line is opened and offers
a group of manipulation radio buttons that allows to enable the IP line
and to configure it to for use by the SIP or MGCP phones:

Inactive - selection disables the corresponding IP line.

MGCP Phone - selection configures the IP line for an MGCP phone to

be connected to the Bizfon’s LAN.

• The MGCP phone’s IP Address, Gateway Name (optional)

and the Endpoint Name will be required for this selection.
Endpoint Name is defined on the MGCP phone and should
match on Bizfon for the successful connection between the
MGCP device and the Bizfon.

SIP Phone - selection configures the IP line for a SIP phone to be
connected to the Bizfon’s LAN.

• Phone Model drop down list is used to select the IP phone

model to be used by the receptionist. The selection other than

Other enables the MAC address text fields used to insert the
MAC Address of the corresponding SIP phone.

• Line Appearance text field requires number of simultaneous

calls supported by the SIP phone.

• Username and Password are required for this selection,

which should match on both the Bizfon and the SIP Phone for
successful connection establishment.

For automatic SIP phone configuration, the SIP phone should be simply reset/rebooted and then, appropriate configuration will be automatically
downloaded from Bizfon to the SIP Phone. However, if you have decided to make the SIP phone configuration manually, it is recommended to select

Other from Phone Model drop down list and to make the configuration manually from the SIP phone's GUI.

Please Note: For automatic configuration, some SIP phones may require additional actions to follow the simple restart. For example, by default IP

Dialog SIP Tone II is in non-auto-provisioning mode, so it should be manually enabled on the phone. To find out how to perform factory reset or
reboot on any of the supported phones, what additional configuration is required for particular SIP phone, and the instructions on how to manipulate
with GUI, refer to the users manual of the corresponding SIP phone.

Features page.

, the Edit

Fig. II-94: IP Line Settings page

Fig. II-95: IP Line Edit page

Supported SIP Phones

Below is the list of SIP phones that can be automatically configured to work with Bizfon4000:

• Snom 190, Snom 200, Snom 220, Snom 320, and Snom 360

• Polycom Soundpoint IP 300SIP

• Cisco IP Phone 7960

• Swissvoice IP 10S

• IP Dialog SIP Tone II (ST201)

• Aastra 480i

• Sipura SPA-841 IP

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Loopback Settings

The FXS Lines Loopback Settings page is used to configure the lines for voice loopback diagnostics. When loopback is enabled on the line, any
incoming calls to the corresponding line will be automatically picked up on the first ring and any voice towards the line will be automatically sent back
to the caller, i.e., caller will hear themselves in the handset. Loopback Timeout gives a possibility to limit the voice loopback diagnostics duration,
i.e., caller will be disconnected from the Bizfon when the Loopback Timeout expires.

The FXS Lines Loopback Settings page shows the only table where all FXS lines of the Bizfon are listed. Here, loopback diagnostics may be
enabled/disabled and the Loopback Timeout can be adjusted for FXS lines.

The FXS Lines Loopback table lists all the FXS lines on the Bizfon
along with their loopback parameters (Loopback State and Loopback
Timeout).

The Edit functional link leads to the FXS Lines Loopback Settings -
Edit Entry page where Loopback Timeout (in seconds) may be
configured for one or more selected FXS line(s).

The Enable/Disable Loopback functional link is used to enable/disable
the Loopback service on the selected FXS line(s).

Fig. II-96: IP Line Settings –Loopback page

FXO Settings

The FXO Settings are used to configure the FXO support that allows Bizfon to connect to other PBXs or analog telephone lines. The FXO Settings
also give a possibility to limit incoming or outgoing calls for the selected FXO line if required. Depending on the Bizfon model, several FXO ports will
be available on the board, thus giving the possibility to connect several PSTN lines to the Bizfon and to use them simultaneously.

The administrator may assign a default recipient for each FXO line, where calls from the Central Office (PSTN) will be routed to. The assigned
recipients become the Bizfon “default users”. If the Bizfon Auto Attendant has been selected as “default user”, a caller from the PSTN needs to go
through the attendant menu to reach the desired extension.

The FXO Settings page lists the available local FXO lines,
shared FXO lines on the remote devices (if any) and their
settings. If the FXO service has been disabled, the Allowed
Call Type, Route Incoming Call to and PSTN number
columns are set to N/A.

Clicking on the FXO line number will open the FXO Settings -
FXO# page where the FXO line settings may be modified.

The Enable FXO checkbox selection activates FXO support for the selected FXO line.

The Allowed Call Type is used to choose the allowed call directions for the corresponding FXO line. The administrator may choose between:

Fig. II-97: FXO Settings page

• Enabling incoming calls (prohibiting outgoing calls) for the selected FXO line.

• Enabling outgoing calls (prohibiting incoming calls) for the selected FXO line.

• Enabling both incoming and outgoing calls for the selected FXO line.

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The Route incoming FXO Call to manipulation radio buttons
group allows to define the destination where incoming calls
addressed to the corresponding FXO line will be forward to.

• Extension – selection allows to choose the local PBX user

or auto attendant extension to forward calls to. If inactive
extension is chosen from this list, the voice mail system will
answer the call addressed to the corresponding FXO line. If
Auto Attendant extension is chosen, it will become the
“default user” of corresponding FXO line on the Bizfon.

• Routing – selection allows to forward the incoming calls to

the destination defined through
requires to enter a routing pattern to the corresponding
field. Based on the registered PSTN users, caller will be
able to reach the destination according to configuration in
Call Routing Table.

By choosing a destination, the Bizfon administrator virtually
assigns a default number that will start ringing whenever a call is
initiated to the Bizfon’s PSTN number.

PSTN Number text field allows entering the PSTN number current FXO line is attached to. Field value is optional, used as an identification
parameter for FXO lines and can be empty.

Alternative AC Termination Mode appears if the local country (Germany, Israel, France, etc.) selected for Bizfon has two kinds of COs that use
different types of AC termination. Contact your CO to learn about your AC termination mode. Selecting the checkbox may help if the voice quality
over FXO is poor or echo is noticed.

To modify the FXO Settings

1. Select the FXO line number from the FXO Settings table. The FXO Settings -FXO# will appear where the line settings may be modified.

2. Enable the FXO line to receive calls from PSTN. To reject calls from/to the PSTN deselect the Enable FXO checkbox.

3. If FXO has been enabled, select the Call Type from the Allowed Call Type drop down list and the extension from the Route FXO Call to drop

down list to route the FXO calls correspondingly.

4. Insert a PSTN number in the same named text field to identify the FXO line.

5. Enable Alternative AC Termination Mode if your CO so requires.

6. Press Save to submit the FXO line settings.

Call Routing. Selection

Fig. II-98: FXO Line Settings page

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Gain Control

The Gain Control settings are used to define the transmit and
receive gains. For FXS lines, Transmit Gain defines the phone
speaker volume and Receive Gain defines the phone
microphone volume. For FXO linesTransmit 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.

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), which 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-99: Gain Control page

Call Routing

The Call Routing service simplifies the calling procedure for Bizfon users, i.e., any kind of calls (internal, SIP, PSTN or IP-PSTN) can be placed in
the same way. No SIP registration is needed for extensions to make routing calls.

The Call Routing page offers the following components:

• The Route all incoming SIP calls to Call Routing

checkbox that is used to route ALL incoming SIP calls
(whether the pattern matches the extension’s SIP
registration username or not) to the Call Routing table. No
digits will be stripped in this case.

• The Call Routing Table link leads to the Call Routing

table where routing patterns may be defined manually.

• The Local AAA Table link leads to the page where local

AAA (Authentication, Authorization, and Accounting)

Fig. II-100: Call Routing page

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database can be managed.

Fig. II-101: Call Routing table

Defining patterns in the Call Routing Table avoids registering Bizfon at the routing management server and gives a possibility to establish a direct
connection to the destination or to use a SIP server for call routing.

The Call Routing Table lists manually defined routing patterns along with their parameters (pattern number, state, routing and inbound caller
settings, RTP Proxy and Date/Time period settings, metric and description), as well as automatically created and undeletable patters created from
the

System Configuration Wizard.

If the route has an Authentication or an Authentication&Accounting selected from the AAA Required checkbox group, it will have a link to the
Users List in the Call Routing table. Users List page contains a list of authorized users defined from the Local AAA Table, and gives a possibility
to enable/disable authentication of each user for the particular route.
Since Call Routing Table may have multiple entries that could match to same pattern, the table will be internally rearranged according to the rules
with these consequences:

• The pattern matching best to the Best Matching Algorithm will have the higher position in the rearranged list

• If multiple patterns equally match to the Best Matching Algorithm, the pattern with the lower metric will get the higher position in the rearranged

list

• If the multiple patterns with the same metric have been matched to the Best Matching Algorithm, the pattern in the higher position in the table

will get the higher position in the rearranged list.

The pattern in the highest position of the rearranged list will be considered as the preferred one. Second and subsequent matching patterns will be
used, if the destination refused the call due to the configured Fail Reason.

The Enable/Disable functional buttons are used to enable/disable the selected route(s). Disabled routes will take no effect while enabled routes will
be parsed when initiating routing calls. The State column in the Call Routing Table displays the current state of the routes (enabled/disabled).

Add starts the Call Routing Wizard where a new routing pattern may be defined. The Call Routing Wizard is divided into several pages: Page 1
displays the following components:

Pattern requires entering the routing pattern’s identification. To make a specified call, the appropriate routing pattern should be dialed. Wildcards are
allowed here (see chapter
used for exclusion (“!5a” inserted in Pattern field means all patterns except those equal to 5a). For example, 2{13-17, ww, a-c} means that the dialed
number may be 213, 214, 215, 216, or 217, 2ww, 2a, 2b and 2c to match the specified pattern; in the case of 2[3,7], the dialed number may be 23 or
27 to match the specified pattern.

Entering a SIP Addresses correctly). '[' , ']' , ',', '-', ‘{‘, ‘}’ are used to define a range or a quantity of numbers, ‘!’ symbol is

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Number of Discarded Symbols (NDS) requires the number of
symbols that should be discarded from the beginning of the
routing pattern. The field should be empty if no digits need to be
discarded. Only numeric values are allowed for this field,
otherwise an error message occurs: “Error: Number of
Discarded Symbols is incorrect - digits allowed only”.

Prefix requires entering the symbols (letters, digits and any
characters supported in the SIP username) that will be placed in
front of the routing pattern instead of the discarded digits.

Suffix requires entering the symbols (letters, digits and any
characters supported in the SIP username) that will be placed in
the end of the routing pattern. (For example, if the routing
Pattern is 12345, the Number of Discarded Symbols is two,
and the Prefix is 909 and Suffix is 0a, the final phone number
will be 9093450a.)

Call Type gives a possibility to select the call type (PBX, PBX­Voicemail, FXO, SIP, IP-PSTN). PBX call type is dedicated for
call routing to the local PBX extension, and PBX-Voicemail call
type is dedicated to route the calls directly to the voice mailbox
of the local PBX extension.

Metric allows entering a rating for the selected route in a range
from 0 to 20. If no value is inserted to this field, 10 will be taken
as the default. If two route entries match a user’s dial string, the
route with the lower metric will be chosen.

The Description text field requires an optional description of the
routing pattern.

The Filter on Caller / Call Type / Modify Caller ID checkbox
selection allows limiting the functionality of the current route to
be used by the defined caller(s) only. If this checkbox is
enabled, inbound caller information (Inbound Caller Pattern,
Inbound Call Type, Inbound Port ID, etc.) will be required later
in the Call Routing Wizard.

The Set Date / Time Period(s) checkbox selection allows to
define a validity period(s) for current routing pattern to take
place and to define pattern date/time rules. When this checkbox

Fig. II-102: Call Routing Wizard - page 1

is enabled, Call Routing Wizard - Page 5 will be displayed.

The second page of the Call Routing Wizard offers different
components depending on the Call Type selected on the
previous page.

Use Extension Settings is applicable to SIP and IP-PSTN call
types only and allows to select the extension (also Auto
Attendant) on behalf of the call that will be placed. The SIP
settings of the selected extension will be used as the caller
information. If no entry is selected in this list, the original caller
information will be kept. When Keep original DID checkbox is
selected, called destination will receive the original caller’s
information, rather than the information of extension selected
from the Use Extension Settings list.
Destination Host requires the IP address or the host name of
the destination (for a direct call) or the SIP server (for calls
through the SIP server).

Destination Port requires the port number of the destination or
of the SIP server.

User Name and Password require the identification settings for
the public SIP server or servers requiring authentication.

Enable Activity Timeout checkbox is used to limit time-to-live
period of routing pattern (makes sense if accept or failure
feedback arrives too late from the destination).

Checkbox selection enables the Activity Timeout text field
which is used to insert a routing pattern activity timeout (in the
range from 1 to 180 seconds). When timeout is configured, the
routing pattern will be active within the defined time frame and if
no response has been received from the destination during that

Fig. II-103: Call Routing Wizard - page 2

period, the pattern will be stopped and next routing rule might be

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optionally considered (depending on the Fail Reason
configuration on the corresponding pattern).

The Multiple Logons (ML) checkbox is available only for the IP-PSTN call type and allows/denies multiple logon to the public SIP server with the
same username at the same time.
Use RTP Proxy checkbox is available for SIP and IP-PSTN call types and is applicable only when route is used for calls through Bizfon between
peers both located outside the Bizfon. When this checkbox is selected, RTP streams between external users will be routed through Bizfon,
otherwise, when checkbox is not selected, RTP packets will be moving directly between peers.

A group of AAA Required checkboxes are used to choose one or more Authentication, Authorization, and Accounting (AAA) settings:

• Local Authentication – with this checkbox selected, callers will need to pass authentication through Local AAA table (see below) when

dialing the current pattern.

• RADIUS Authentication and Authorization – checkbox is present when RADIUS client is enabled. With this checkbox selected, when

dialing the current pattern, callers will need to pass the authentication through RADIUS server (see above).

• RADIUS Accounting - checkbox is present when RADIUS client is enabled. With this checkbox selected, no authentication will take place,

but a caller identifying CDR (call detail report) will be sent to the RADIUS server. Checkbox selection enables accounting on the RADIUS
for the certain call.

If the authentication is configured based on the caller’s address, callers will pass the authentication automatically; otherwise they will be
required to identify themselves by a username and a password.

The Fail Reason drop down list indicates available failure reasons and contains different failure reasons, depending on the call type selection on the
previous page. Following Fail Reasons may be available in this list:

• Cannot Establish Connection - failure reason is available for FXO calls only and indicates cases when connection cannot be

established.

• Wrong Number – available for PBX, SIP and IP-PSTN call types and indicates cases when the dialed number is wrong.

• Busy - available for PBX, SIP and IP-PSTN call types and indicates cases when the dialed destination is busy.

• Network Failure - available for SIP and IP-PSTN call types and indicates cases when system overload, network failure or timeout

expiration occurred.

• Other - available for SIP and IP-PSTN call types and indicates cases when authorization, negotiation, not supported or request rejected or

other unknown errors occur.

•

System Failure - available for SIP and IP-PSTN call types and indicates cases indicated in Network Failure and Other fail reasons.

• None – available for all call types and indicates no fail reason.

• Any - available for all call types and indicates any of above mentioned fail reason.

If the call cannot be established due to some of the selected Failure Reason, the call routing table will be parsed for the next matching pattern and, if
found, the call will be routed to the specified destination.

SIP Privacy manipulation radio buttons group is available for SIP call type only and allows to select the security of the SIP route by means of hiding
(or replacing, depending on the configuration of the SIP server) the key headers of the SIP messages used to establish the call.

• Default Privacy – with this selection, no Bizfon specific SIP privacy will be applied, all privacy will be relied on the configuration of the SIP

Server.

• Disable Privacy – with this selection, no SIP call security will be disabled, all headers of the SIP message will be transparently visible to

the destination.

• Enable Privacy - with this selection, SIP privacy will be specified for the corresponding route. Selection enables a group of checkboxes to

choose the key headers to be fully or partly hidden or replaced. Require Privacy checkbox selection is used to restrict the delivery of the
SIP message if either of the selected headers cannot be hidden (or replaced, depending on the configuration of the SIP server) before
being sent to the destination.

The Port ID drop down list is present for FXO call type and contains FXO line numbers. Any Local and Any@Any selections are available for the
FXO call type only and give a possibility to route calls via the first available local FXO line or any FXO lines (including shared on other Bizfon boards)
respectively.

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The Call Routing Wizard - Page 3 appears if the Fill Call Source
Information checkbox previously had been enabled on Page 1 of

the Call Routing Wizard, and it will require information about the
Inbound caller.
The Inbound Caller Pattern field requires the caller’s address
where the current route will be applied. Alphanumerics and any
characters supported in the SIP username are allowed for this
field. Wildcards are allowed here (see chapter

Addresses correctly

). '[' , ']' , ',', '-', ‘{‘, ‘}’ are used to define a

Entering a SIP

range or a quantity of numbers. For example, 2{13-17, ww, a-c}
means that the dialed number may be 213, 214, 215, 216, or 217,
2ww, 2a, 2b and 2c to match the specified pattern; in the case of
2[3,7], the dialed number may be 23 or 27 to match the specified
pattern.
The Inbound Number of Discarded Symbols and Inbound
Prefix text fields are hidden only when an FXO call type has been
selected from Page 1 of the Call Routing Wizard. The Number of
Discarded Symbols (NDS) text field requires the number of digits
that should be discarded from the beginning of the Inbound Caller
Pattern. The field should be empty if no digits need to be
discarded. Only numerics are allowed for this field, otherwise an
error message occurs: “Error: Number of Discarded Symbols is
incorrect - digits allowed only”.

Fig. II-104: Call Routing Wizard - page 3

The Inbound Prefix text field requires entering the symbols (alphanumerics and any characters supported in the SIP username) that will be placed
in front of the Inbound Caller Pattern instead of the discarded digits. (For example, if the routing pattern is 12345, the Number of Discarded
Symbols is two, and the prefix digits are 909, the final phone number will be 909345.) Wildcards are allowed here (see chapter

Addresses correctly

).

Entering a SIP

The Inbound Call Type drop down list gives a possibility to select the call type (PBX, SIP, FXO) used by the inbound caller to reach the Bizfon.
The Next button will open a Call Routing Wizard - Page 4 where different information about Inbound Caller will be required depending on the
selected Inbound Call Type. For the SIP Inbound Call Type, the Inbound Host text field will require one or more IP addresses or host names of SIP
server where the caller is registered, or the caller’s device in case of direct calls, separated by a space. If the FXO Inbound Call Type is selected, the
Inbound Port ID drop down list will require selecting the FXO line number.

The Call Routing Wizard - Page 5 appears if the Set Date / Time

Period(s) checkbox previously had been enabled on Page 1 of the
Local Vall Routing Wizard, and it will require information about

the pattern validity period(s).
Page provides selection between Typical and Custom date/time
rule definition.

Typical selection contains a group of radio buttons that are used
to select the frequency of the corresponding routing pattern to take
place:

• Daily

• Weekly – the preferred weekday(s) should be selected for

this option.

• Monthly – the calendar day should be selected for this

option.

• Annually – the calendar day and month should be selected

for this option.

In Available Time Period drop down lists, the time range of the
pattern validation should be defined. Any time selected in this field
will be considered corresponding to the Bizfon’s

Settings

.

Time/Date

Custom selection provides a possibility to manually define the
validity period(s). Use following format to insert pattern date/time
rule(s):

Fig. II-105: Call Routing Wizard - page 5

[Month,Month-Month,...][Day-Day,Day,...][hh:mm-hh:mm,...]; ...

Please Note: Established patterns based on the Emergency Codes and PSTN Access Codes Settings in the

System Configuration Wizard will be

marked in bold and will be placed at the first position in the Call Routing Table. Additionally they cannot be modified and deleted from the Call
Routing Table.

The Duplicate functional button is used to create a routing pattern with the settings of an exiting one. This is to avoid configuring a new routing entry
completely by duplicating an existing entry with different settings. To use the Duplicate button only one record may be selected, otherwise an error
will occur: “One row should be selected”. The Duplicate button opens the Call Routing Wizard where all fields except the Pattern field are already
filled in. Pattern for the new route will be required anyway.

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The Move Up/Move Down buttons are used to move call routing patterns one level up or down within the Call Routing table. The consecution of the
routing patterns is important when making routing calls as the Call Routing table is parsed from the top down and routing will take place according
to the first pattern that matches the dialed number. The Move To button is used to move the selected entry to some other position in the Call Routing
Table, which will increase or decrease the selected pattern’s priority. Pressing the button will open the page where the row number should be
specified, together with the position the selected entry is to be placed (before or after the defined row).

The Local AAA Table page allows to manage local authentication
and authorization database. Callers dialing the routes which have
an AAA (Authentication, Authorization, and Accounting) option
enabled, will pass the authorization on Local AAA Table by phone
number or username/password, depending on corresponding entry
configuration on this page.

If the detected phone number of the caller dialing a route which
has AAA option enabled, is registered in the Local AAA Table,
caller passes authorization automatically. If the caller ID service is
disabled or the caller’s phone number is not registered, the caller is
asked to enter registration user name and password.

The Add functional button opens the Call Routing – Local AAA
Table - Add Entry page where new local AAA record can be
created.

Fig. II-106: PSTN User Registration page

The Call Routing – Local AAA Table - Add Entry page offers a group of manipulation radio buttons to select the way of authorization and other
parameters:

• Authentication by Caller ID – selection is used to set the authentication based on the caller’s phone number (which is considered to be

automatically detected). The Phone Number text field requires caller’s phone number. Only numeric and wildcard characters (see chapter

Entering a SIP Addresses correctly) are allowed for this field. '[' , ']' , ',', '-', ‘{‘, ‘}’ are used to define a range or a quantity of numbers. For

example, 2{13-17, ww, a-c} means that the dialed number may be 213, 214, 215, 216, or 217, 2ww, 2a, 2b and 2c to match the specified phone
number; in the case of 2[3,7], the dialed number may be 23 or 27 to match the specified phone number.

• Authorization by Username and Password - selection is used to set the authentication based on the username and password inserted by the

user upon login. The Username text field requires the authentication user name. Only numeric values are allowed for this field, otherwise the
“Incorrect Username - digits allowed only” error message occurs. The Password text field requires the authentication password. Only numeric
values are allowed for this field, otherwise the “Incorrect Password - digits allowed only” error message occurs.

The Expiration Date and Time drop down lists are used to set the
date and time when the registration is to expire.
The Expires in checkbox is used to enable the Expiration Date
and Time feature.
The Description text field required an optional description about

Fig. II-107: PSTN User Registration - Add Entry page

the calling party.

To make a Call Routing pattern

1. Click on the Call Routing Table link on the Call Routing page.

2. Press the Add button on the Call Routing page.

3. Specify the Pattern in the corresponding field.

4. Select the Number of Discarded Symbols and Prefix if required.

5. Select the Call Type from the drop down list.

6. Define the Metric or leave the default.

7. Enter a Description if needed.

8. Enable the Filter on Caller / Call Type / Modify Caller ID checkbox, if the route functionality should be limited depending on inbound caller

information.

9. Enable Set Date/Time Period(s) checkbox, if route should be functional within certain time/date interval.

10. Press Next.

11. Select user or attendant extension from Use Extension Settings drop down on behalf of which the call will be placed.

12. Specify the Destination Host and Port Number, Username and Password if IP or IP-PSTN call type has been selected. For IP-PSTN call

type, enable Multiple Logons if necessary. Enable Use RTP Proxy checkbox, if needed.

13. Choose the Authentication and Accounting method from AAA Required drop down list.

14. Choose a Fail Reason from the corresponding drop down list.

15. Press the Next button.

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16. If Filter on Caller / Call Type / Modify Caller ID checkbox has been previously enabled and the call type is different from the FXO, fill Inbound

Caller Pattern in the corresponding text field, choose the needed value from Inbound Call Type drop down list, as well as Inbound Number
of Discarded Symbols and Inbound Prefix values.

17. Press the Next button.

18. If IP has been selected on the previous step in the Inbound Call Type drop down list, then Inbound Host should be inserted in the current

page. If FXO has been selected in the Inbound Call Type drop down list, then the FXO line number should be selected here.

19. If Set Date/Time Period(s) checkbox has been selected on the first page, pressing Next will open Date/Time Rules page where route validity

should be defined.

20. Press the Finish button to establish a local route with the inserted settings.

To create a local AAA entry

1. Click on the Local AAA Table link on the Call Routing page.

2. Press the Add button on the Local AAA Table page.

3. Choose the Authentication type.

4. Enter the Phone Number or the Username and Password depending on the selected Authentication type.

5. Use the Expiration Date and Time checkbox to enable the expiration timeout.

6. Select the Expiration Date and Time from the corresponding drop down lists.

7. Press Save to apply these settings.

Best Matching Algorithm

The Best Matching Algorithm is used by the Routing Agent (RA) to sort the list of the patterns that match a dialed number. Sorting is done by the
following principle: the more the pattern matches the dialed number, the higher its priority.

To decide which of the selected patterns matches the dialed number more in comparison with the others, the following list of criteria is used (List 1).
The criteria are ordered by their priorities: that is Criterion 2 is calculated only if more than one pattern takes the same value for Criterion 1, Criterion
3 is calculated only if more than one pattern takes the same value for Criterion 2 (obviously for Criterion 1 as well) etc. Each consecutive criterion

is calculated only if more than one pattern takes the same value for the preceding criteria.

List 1

Criterion 1 The presence of asterisks (“*”) in a pattern

The patterns without “*” have higher priority.

Criterion 2 The number of matching digits/symbols

The more matching digits a pattern has, the higher its priority.

Criterion 3 The number of square brackets (“[]”)

The more ranges a pattern has, the higher its priority.

Criterion 4 The number of question marks (“?”)

The more question marks a pattern has, the higher its priority.

Criterion 5 The number of braces (“{}”)

The more ranges a pattern has, the higher its priority.

Criterion 6 The number of asterisks (“*”)

The fewer asterisks a pattern has, the higher its priority.

Criterion 7 The value of the metric

The lower the metric of a pattern is, the higher its priority.

Criterion 8 The position in the routing table

The higher the position of a pattern in the routing table is, the higher its
priority.

The algorithm is discussed in the example below.

Example The user has dialed 1231 and Routing Agent has found the following list of matching patterns.

The list of matching

patterns found by RA

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*1*

123*

{11-15}3*

?2?1
123?

[1-3]*

[1-3]???

{100-150, asd, \*\?}1

12*31

1[1-3]3[0-8]

1231

*2*1

*

The step by step discussion of the Best Matching Algorithm is as follows.

Step 1: The list is split into two groups separating the patterns with “*” from the ones without (Criterion 1). The patterns with “*” form a group with

lower priority and are pushed back to the end of the list (Table 1).

Table 1

The list split into two

subgroups

?2?1
123?

[1-3]???

{100-150, asd, \*\?}1

1[1-3]3[0-8]

1231

*1*

123*

{11-15}3*

[1-3]*

12*31

*2*1

*

Step 2: The two groups of the patterns are sorted separately from each other by the number of matching digits in descending order (Criterion 2,

Table 2). The patterns that have the same number of matching digits are grouped into sub-lists (Table 3). If a sub-list consists of one
pattern, it stays in its position and does not participate in further discussions.

Table 2

The list of patterns Criterion 2

1231 4

123? 3

?2?1
1[1-3]3[0-8]

2
2

{100-150, asd, \*\?}1 1

[1-3]??? 0

12*31 4

123* 3

*2*1 2

*1*
{11-15}3*

[1-3]*
*

1
1

0
0

Table 3

The list of patterns Criterion 2

1231 4

123? 3

?2?1
1[1-3]3[0-8]

{100-150, asd, \*\?}1 1

[1-3]??? 0

12*31 4

123* 3

*2*1 2

*1*
{11-15}3*

[1-3]*
*

2
2

1
1

0
0

The principle by which the patterns have been sorted in Step 1 is applied in all further steps with a different criterion.

Step 3: Each sub-list is sorted separately from the others by the number of square brackets (“[ ]”) in the pattern in descending order (Criterion 3,

Table 4). The patterns that have the same number of ranges are grouped into sub-lists (Table 5). If a sub-list consists of one pattern, it stays
in its position and does not participate in further discussions.

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Table 4

The list of patterns Criterion 3

1231 -

123? -

1[1-3]3[0-8]
?2?1

2
0

{100-150, asd, \*\?}1 -

[1-3]??? -

12*31 -

123* -

*2*1 -

*1*
{11-15}3*

[1-3]*
*

0
0

1
0

Table 5

The list of patterns Criterion 3

1231 -

123? -

1[1-3]3[0-8] 2

?2?1 0

{100-150, asd, \*\?}1 -

[1-3]??? -

12*31 -

123* -

*2*1 -

*1*
{11-15}3*

[1-3]* 1

* 0

0
0

Step 4: Each sub list is sorted separately from the others by the number of question marks in the pattern in descending order (Criterion 4, Table 6).

The patterns that have the same number of question marks are grouped into sub-lists. If a sub-list consists of one pattern, it stays in its
position and does not participate in further discussions.

Table 6

The list of patterns Criterion 3

1231 -

123? -

1[1-3]3[0-8] -

?2?1 -

{100-150, asd, \*\?}1 -

[1-3]??? -

12*31 -

123* -

*2*1 -

*1*
{11-15}3*

0
0

[1-3]* -

* -

Step 5: Each sub-list is sorted separately from the others by the number braces (“{ }”) in the pattern in descending order (Criterion 5, Table 7). The

patterns that have the same number of ranges are grouped into sub-lists (Table 8). If a sub-list consists of one pattern it stays in its position
and does not participate in further discussions.

Table 7

The list of patterns Criterion 4

1231 -

123? -

1[1-3]3[0-8] -

?2?1 -

{100-150, asd, \*\?}1 -

[1-3]??? -

12*31 -

123* -

*2*1 -

{11-15}3*
*1*

1
0

[1-3]* -

* -

Table 8

The list of patterns Criterion 4

1231 -

123? -

1[1-3]3[0-8] -

?2?1 -

{100-150, asd, \*\?}1 -

[1-3]??? -

12*31 -

123* -

*2*1 -

{11-15}3* 1
*1* 0

[1-3]* -

* -

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Step 6: This step is applicable to the subgroup containing patterns with “*”, the group with lower priority. Each sub-list is sorted separately from the

others by the number of asterisks (“*”) in ascending order (Criterion 6). The patterns that have the same number of asterisks are grouped
into sub-lists. If a sub-list consists of one pattern it stays in its position and does not participate in further discussions.

Step 7: Each sub-list is sorted separately from the others by the value of metric in ascending order (Criterion 7). The patterns that have the same

value of metric are grouped into sub-lists. If a sub-list consists of one pattern it stays in its position and does not participate in further
discussions.

The values of metrics are taken from the routing table.

Step 8: The patterns in each sub-list are arranged by their positions in the routing table (Criterion 8).

The subgroup containing patterns with “*” is attached to the end of the subgroup without “*” forming a single list of sorted patterns. The obtained list
is the sorted list of the patterns by the Best Matching Algorithm (Table 9).

Table 9

The sorted list of

patterns

1231
123?

1[1-3]3[0-8]

?2?1

{100-150, asd, \*\?}1

[1-3]???

12*31

123*
*2*1

{11-15}3*

*1*

[1-3]*

*

VoIP Carrier Wizard

VoIP Carrier Wizard is used to define access codes for available VoIP Carrier account which will particularly allow to reach users over IP-PSTN
providers or to call to the peers registered on the certain SIP servers by dialing simple digit combinations.

For each configured VoIP carrier, wizard creates specific IP-PSTN routing rule in the
automatically generated in

Extensions Management will be registered on the defined VoIP Carrier’s SIP server and on behalf of which the calls from

Bizfon’s users towards the created VoIP Carrier will be placed.

VoIP Carrier Wizard – Page 1 provides a possibility to
describe VoIP carrier:

When predefined carrier is selected in VoIP Carrier drop
down list, SIP Server and Port will be already predefined in
the next page. Manual selection allows to set up the VoIP
Carrier settings manually.

Description field allows to insert an optional description of
the VoIP Carrier.

Call Routing table. Additionally, a virtual extension will be

Fig. II-108: VoIP Carrier Wizard page

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VoIP Carrier Wizard – Page 2 is used to define VoIP
Carrier Settings. Page contains following components:

1. VoIP Carrier Common Settings

Account Name text field requires a username for

authentication on the defined SIP server.
Password requires a password for authentication on the

defined SIP server.

Confirm Password requires a password confirmation. If
the input is not corresponding to the one in the Extension
Password field, the error will appear: “Incorrect Password
confirm”.

SIP Server text field requires an IP address or the
hostname of the SIP server destination party is registered
on.

SIP Server Port text field requires the port number of the
SIP server destination party is registered on.

2. VoIP Carrier Advanced Settings
Use RTP Proxy checkbox is applicable only when route is used for calls towards configured VoIP Carrier from peer located outside the Bizfon.

When this checkbox is selected, RTP streams between external users will be routed through Bizfon, otherwise, when checkbox is not selected, RTP
packets will be moving directly between peers.

UserID requires an identification parameter to reach the SIP server. It should have been provided by the SIP service provider and can be requested
for some SIP servers only, for others, the field should be left empty.

Send Keep-alive Messages to Proxy enables the SIP registration server accessibility verification mechanism. Timeout indicates the timeout
between two attempts of SIP registration server accessibility verification. If no reply is received from the primary SIP server within this timeout, the
secondary SIP server will be contacted. When the primary SIP server recovers, SIP packets will be sent to it once again.

A group of Host address and Port text fields respectively require the host address (IP address or the host name), the port number of the Outbound
Proxy, Secondary SIP Server and the Outbound Proxy for the Secondary SIP Server. These settings are provided by the SIP servers’ providers
and are used by Bizfon to reach the selected SIP servers.

VoIP Carrier Wizard – Page 3 contains VoIP Carrier access
code selection components:

Access Code text field requires a digit combination by dialing
which the corresponding VoIP Carrier will be reached.

Route Incoming Calls To drop down list allows to select an
extension (or Auto Attendant) on the Bizfon where incoming
calls from the configured VoIP Carrier should be routed to.

Failover to PSTN checkbox selection will route the call to
PSTN through local FXO line in case if VoIP Carrier is not
available. When this checkbox is selected, an additional entry
will be added to the
transmission to local PSTN when IP call towards the
configured VoIP Carrier cannot be established.

Please Note: Warning message will inform that the defined

Access Code already exists in the Call Routing table or

causes a conflict with entries already in the Call Routing table.
In this case, when proceeding the VoIP Carrier Wizard,
existing entry in the Call Routing table will be automatically
overwritten by the new settings.

Call Routing table maintaining digit

RADIUS Client Settings

The RADIUS (Remote Authentication Dial In User Service) specifies the RADIUS protocol used for authentication, authorization and accounting, to
differentiate, to secure and to account for the users. The RADIUS Server gives an extra possibility for caller from/through Bizfon to pass
authentication to be able to dial the specific number.

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When RADIUS client is enabled on the Bizfon, and according to configuration of AAA Required option (see Call Routing table), RADIUS server will
be used to authenticate user and/or to account the call. This can be accomplished by caller’s number automatic detection or a customizable login
prompt, where caller is expected to enter username and password.

Transactions between the client and the RADIUS server are authenticated through the use of a shared Secret Key, which is never sent over the
network. In addition, any user passwords are sent encrypted between the client and RADIUS server, to eliminate the possibility that someone
snooping on an insecure network could determine a user's password. If no response from the RADIUS Server is returned after Receive Timeout
expires, the request is resent a number of times, defined in the Retry Count list. The client also can forward requests to an alternate server or servers
if the primary server is down or unreachable. An alternate server can be used after a number of failed tries to the primary server.

Once the RADIUS server receives the request, it determines if the sending client is valid. A request from a client that the RADIUS server does not
have a shared secret must be silently discarded. If the client is valid, the RADIUS server consults a database of users to find the user whose name
matches the request. The user entry in the database contains a list of requirements (username, password, etc.) that must be met to give access to
the user. If all conditions are met, the user gets access to the Bizfon Network.

The RADIUS Client Settings page contains the Enable RADIUS Client checkbox that enables RADIUS client on the Bizfon.

Please Note: RADIUS Client cannot be disabled if there is at least one route with RADIUS Authentication and Authorization or RADIUS

Accounting values configured in the AAA Required drop down list at the

Call Routing table. To be able to disable the RADIUS Client on the Bizfon,

appropriate routes should be remove first.

The other RADIUS Client settings are divided into three groups:

1. Registration Settings
Primary Server requires the IP address of the primary Radius

Server.
Secondary Server requires the IP address of the secondary
Radius Server.
NAT Station IP text fields require the NAT PC WAN IP address. If
no NAT Station is specified here, Bizfon’s IP address will be sent
to the RADIUS server.
Secret Key is used to insert the secret key between the Radius
client and the server. Contact the Radius server administrator to
get the secret key for your Bizfon.
Confirm Secret Key field is used to verify the secret key. If the
entered Secret Key does not correspond to the one in the
Confirm Secret Key field, the error will appear: “The Secret Key
does not match. Please try again”.
Retry Count allows selecting the number of attempts before
canceling the registration.
Receive Timeout allows selecting the timeout (in seconds)
between two attempts to register.
Encoding Type allows selecting the encoding type (PAP or
CHAP) that should be unique on both the client and the server
sides for the establishment of a successful connection. Encoding
type also should be requested from the Radius Server
administrator.
The Authorization Port text field requires the port number on the
RADIUS server where Bizfon will send the authentication requests.
The Accounting Port text field requires the port number on the
RADIUS server where Bizfon will send the accounting messages.

Fig. II-109: Radius Client Settings page

2. Authentication Settings

Enable common login for all users in time of by Phone authentication checkbox enables custom settings for the callers passed an authorization

by phone on the Bizfon. Checkbox enables Username and Password text fields to insert the custom settings that will stand instead of source
caller’s settings when being delivered to the RADIUS server.

Authentication on Destination RADIUS Server parameters group is used to insert a Username and a Password (followed by the password

confirmation) used by PSTN callers to pass the authentication on to the RADIUS Server of the destination Bizfon. If these fields are left empty, the
original authentication settings that PSTN users enter for authentication will be used.

3. Accounting Settings

The Username field is dedicated for accounting service only and is used to insert an identification username accounting will be performed on behalf
of. When no username is specified in this field, source username will be used for accounting.

Send Accounting messages manipulation radio buttons group is used to select the whether both Start and Stop Accounting messages should be
sent or Stop Accounting message is delivered only.

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Voice Mail Common Settings

The Voice Mail Common Settings page is used to configure the Voice Mail recording codec and memory allocation for voice mails and user defined
system messages. Bizfon allows using USB flash memory for voice data files storage. USB Flash Memory is basically a portable hard drive. Using
flash memory, the USB Flash Disk is compact, lightweight, durable and easy to use. It can be plugged into one of the available USB ports on the
Bizfon.

The page offers the following components:

The Recording Codec drop down list contains the existing codecs for voice mail compression. Changing the Voice Mail recording codec will directly
effect the allocated memory size for users.

The Memory Allocation manipulation radio buttons allow
choosing the location where the user’s voice mails are to be
saved. Either in the Embedded Memory Storage (User’s Space
assigned to the Bizfon extensions) or the External USB Flash
memory.

• Selecting Embedded Memory Storage will save the

user’s voice mails in the device’s internal flash memory.
The available memory for every user is configured on the

Extensions Management page.

• Selecting External USB Flash memory allows keeping the

user's voice mails on the external USB device attached to
the Bizfon. This will save the assigned User's space.
Selecting the External USB Flash and saving the page
settings will start the USB Flash memory automatic
configuration.

The USB Flash memory needs to be formatted to be available to the Bizfon. A confirmation message will ask for the USB Flash memory formatting
verification.

Attention: Formatting will erase all data on your USB Flash Disk.

Once the USB Flash memory is formatted, it can be reused by Bizfon. Every time the USB Flash memory is plugged into the Bizfon, it will be
mounted automatically. For safety, removing the USB Flash memory will result in it being automatically unmounted first. Error messages will appear
when formatting, mounting, or unmounting procedures are unsuccessful, as well as when the USB Flash memory is write protected or undetected.

Replacing or removing the USB Flash memory from the Bizfon will result in the Bizfon to switch back automatically to the Embedded Memory

Storage, and all further voice data will be stored in the Bizfon’s local flash memory.

Attention: It is highly recommended to select the Embedded Memory Storage prior to unplugging the USB Flash memory otherwise data stored on

the USB Flash memory may be corrupted or lost. It is not recommended to remove one USB Flash Memory stick and plug another one within 10
seconds; otherwise the system may fail detecting the new USB Flash device.

When attaching the USB Flash memory to the Bizfon, its voice data (if any) will be copied automatically from the Bizfon’s local flash to the external
USB Flash memory. If the USB Flash memory is too small to fit the voice data on the Bizfon, the Memory Allocation mode will still be switched to the
External USB Flash, however, no voice data will be copied to the USB Flash memory.

Please Note: As Bizfon allows using both its local flash space and the External USB Flash memory, voice data can be stored in both locations

depending on the selected Memory Allocation mode. So if the user complains about the loss of a specific voice mail in her mailbox or an
uploaded/recorded system message in her configuration, please verify the memory allocation mode Bizfon has been acting in at the time the voice
data was recorded.

The Clear USB Flash button appears only when the USB Flash memory stick is attached to the Bizfon and is used to clear all existing data on the
Flash.

Below is a list of the USB Flash memory devices tested and found to be functional with the Bizfon:

Fig. II-110: Voice Mail Coming Settings page

1. SanDisk Cruzer mini 256MB

2. Kingston Data Traveler 128MB

3. Kingston 128MB

4. Twin MOS Mobile Disk 128MB

Bizfon4000 (SW Version 3.1.x) 69

6. PNY technologies Attaché Flash Drive 128MB

7. Lexar Media JumpDrive

8. Crucial Technology Gizmo 128

9. QuickDrive 128MB

11. JMTEK 128MB

12. Apacer 64MB

13. Pen Drive 64MB

14. Kingston 64MB

Bizfon Manual II: Administrator's Guide Administrator's Menus

5. PQI 128MB Intelligent Stick 10. LinkSys Instant USB Disk 128MB 15. JMTek USBDrive 32MB

Please Note: It is strongly recommended to use one USB flash memory. Two sticks cannot be used simultaneously.

Dial Plan Settings

The Dial Plan Settings page is used to adjust the
dialing timeouts for the routing calls over Bizfon.

This page consists of the only drop down list used to
configure the dialing timeout for the Routing calls.
Values selected in the lists indicate the interval between
the dialed number and it being applied to the network.

Fig. II-111: Dial Plan Settings page

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Internet Uplink Menu

Fig. II-112: Internet Uplink menu in Dynamo theme

Fig. II-113: Internet Uplink menu in Plain theme

PPP/ PPTP Settings

The PPP/PPTP Settings page is used to establish a connection over the DSL link or any other type of uplink, to the ISP (provider party). A
connection is needed to set up and to make or receive calls through PPP over Ethernet. The connection may be configured for manual setup or to be
always up. Once a connection has been established between the Bizfon and the provider, Bizfon users will be able to make and receive calls at any
time.

The PPP/PPTP Settings page offers the following components:

Advanced PPP Settings link refers to same named page where certain parts of the negotiation process during connection establishment can be

adjusted. Link is not available when accessing this page through

PPTP Server text fields are only enabled when Bizfon is running with the PPTP interface and require the IP address of the PPTP server.

Encryption drop down list is only enabled when Bizfon is running with the PPTP interface and is used to select the encryption for the traffic over the

PPTP interface.

Authentication Settings require the Username and the Password used for the authentication on the ISP server.

Internet Configuration Wizard.

Dial Behavior radio buttons:

• Dial Manually - if this radio button is activated, a

button will be displayed in the main management
window that serves to switch the Internet connection
on/off. When accessing the Internet, every station of
the connected LAN has to connect to Bizfon first.

• Always connected - Bizfon stays in the always

connected mode. This will allow remaining always
online in the network.

IP Address Assignment radio buttons are used to define
the way of IP address assignment for the PPP interface:

• Dynamic IP Address – the IP address to the PPP

interface will be assigned dynamically by the DHCP
server.

• Fixed IP Address – the fixed user defined IP address

will be assigned to the PPP interface.

Keep connection alive checkbox enables keeping the
connection alive by sending control packets dedicated for
the link state verification.

Fig. II-114: PPP/PPTP Settings page

Advanced PPP Settings

The Advanced PPP Settings are used to enable/disable certain parts of the negotiation process during connection establishment. These settings
are available only if Bizfon has a PPPoE WAN interface.

Attention: Disabling any of the services below may cause problems when establishing a connection up to complete connection failure. The default

settings should be changed only if the ISP (Internet Service Provider) requires it explicitly or if the peer system has problems with one of the services
listed below. More information about these services can be found at:

The Advanced PPP Settings page offers a group of checkboxes:

http://www.protocols.com/pbook/ppp.htm.

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Enable automatic PPP restart at checkbox is used to select the time when PPP connection will be automatically restarted. Checkbox selection
enables LCP echo failures text field that indicates the number of LCP echo failure packets received before the PPP connection will be considered
as dead and will be restarted.

Disable CCP (Compression Control Protocol) negotiation -

this option should only be selected if the peer system is not
working properly, e.g., not accepting the requests from the
PPPD (Point-to-Point Daemon) for CCP negotiation.

Disable magic number negotiation - with this option, PPPD
cannot detect a looped-back line. This option should only be
selected if the peer is not working properly.

Disable protocol field compression negotiation in both the
receive and the transmit direction - no protocol field

compression will take place.

Disable Van Jacobson style TCP/IP header compression in
both the transmit and the receive direction - no negotiation of

TCP/IP header compression will take place, the header will
always be sent uncompressed.

Disable the connection-ID compression option in Van
Jacobson style TCP/IP header compression - with this

option, PPPD will not compress the connection-ID byte from
Van Jacobson, nor ask the peer to do so.

Disable the IPXCP and IPX protocols - this option should only
be selected if the peer is not working properly and cannot
handle requests from PPPD for IPXCP negotiation.

Fig. II-115: Advanced PPP Settings page

VPN Configuration

A VPN (Virtual Private Network) is established to connect two local networks (intranets) over the Internet securely. VPN routers manage
authentication between servers and clients and handle data encryption for the connection. Only authorized users may access the network, and the
data exchange cannot be intercepted.

VPN connections are, in many ways like every Internet connection;, they are based on IP addresses, which means, the concerned VPN gateways
must authenticate the IP addresses of their respective partner’s VPN gateways. Each time a specific VPN is to be established, usually, the same IP
addresses are expected,. That fact won’t cause any problems if both VPN partners have fixed WAN IP addresses. But, there may be good reasons
to prefer dynamically allocated IP addresses. To enable devices that use a variable IP address to become part of a VPN, they are turned into so­called Road Warriors. Then they are able to reach their corporate network via authentication at the company's VPN gateway device, for example.
This VPN gateway device has to have a fixed IP address for Internet access, because every VPN needs at least one VPN gateway with a fixed IP
address.

The partner devices of a VPN must have different WAN IP addresses, and if they are connected to local area networks, these LAN’s must have
different IP addresses. As all Bizfon devices have the same default IP addresses on delivery, at least one of them must be reconfigured in order to
set a new IP addresses.

Bizfon supports several kinds of VPN connections such as IPSec, L2TP and PPTP.

The VPN Configuration page offers four links (IPSec
Configuration, PPTP Client Configuration, PPTP Server
Configuration and L2TP Configuration), which leads to the
corresponding feature settings pages.

Attention: It is strongly recommended not to run different kinds

of VPN tunnels between the same endpoints simultaneously.

Fig. II-116: VPN Configuration page

An IPSec connection includes authentication and encryption to protect data integrity and confidentiality. VPNs are “virtual” in the sense that
individuals can use the public Internet as a means of securely accessing an internal network. Once the IPSec connection is established, users have
access to the same network resources, addresses, and so forth as if they were connected locally. VPNs are “private” because the data is encrypted
between two VPN gateways. Encryption makes it very difficult for anyone to intercept data and capture sensitive information such as passwords. The
Bizfon can be set up to act as a VPN router when connected to the Internet with a fixed IP address or as an IPSec connection Road Warrior when
using dynamic IP addresses.

Establishing an IPSec connection normally requires the functionality of a VPN gateway on each side of the communication line. An intelligent Internet
access router, for example Bizfon, delivers this function but also PCs or workstations may be equipped with VPN gateway functionality. For home
offices it may be too expensive to get fixed IP addresses so they prefer dynamically allocated IP addresses.

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When Bizfon is connected to the Internet with a fixed IP address, it will be set up to act as a VPN gateway. Then Bizfon is prepared to establish an
IPSec connection with another VPN gateway device, but allows access to Road Warriors, too. A traveling salesperson's notebook for example could
be such a Road Warrior. Access to their company’s intranet via IPSec connection can be obtained regardless of location.

Besides being a VPN gateway, Bizfon can be set up to act as a Road Warrior. If a home office for example is connected to the Internet via Bizfon
with PPPoE (Point-to-Point Protocol) and dynamic IP addressing, setting up Bizfon as a Road Warrior will allow a IPSec connection to the corporate
network.

For the encryption and decryption of the data transmitted via the IPSec connection, a key is used. RSA used by Bizfon is an asymmetric key system.
It has to be available on both sides of the IPSec connection and will generate a different pair of keys on each side, a private and a public key. During
the connection establishment, some data is encrypted with the remote party’s public key and can be decrypted with their private key by themselves
and vice versa (the data encrypted there with Bizfon’s public key can be decrypted with Bizfon’s private key). Since the private key is never
transmitted in any way, it stays completely unknown for everybody, thus the system remains safe. Even if someone gets hold of the public key,
decryption cannot be possible without the private key. Bizfon generates such a pair of keys automatically when it is set up. The user cannot see the
private key, but must know the public one, as their IPSec connection partner will need it.

Please Note: Always a pair of keys will be generated, a public one and a private one, the former pair of keys will become invalid as well as all

existing IPSec connections that use RSA keying.

The IPSec Configuration link refers to the IPSec Connection
Settings page, which gives an overview of all existing IPSec
connections characterized by their Connection Name, the
Remote Gateway (the IP address or the hostname of the IPSec
connection partner), the State of the IPSec connection
(Stopped, Connecting, Activated, Waiting or Connected) and the
dedicated Keying Type (the encryption type). The content of
the table can be sorted in ascending or descending order by
clicking on the header of the respective column. There is a
checkbox for every IPSec connection to select it for further
editing.

Start activates the connection establishment of the selected
IPSec connection. The State of the IPSec connection will
change into “Connected” or “Activated” depending on the IPSec
connection type. If no record is selected, the “One Record
should be selected” error message occurs.

Attention: It is not recommended to start a static and a dynamic

connection configured to use the same secret key
simultaneously. A dynamic connection may capture the static
connection peer and vice versa, depending on which connection
established first.

Stop disconnects the selected IPSec connection. The state of
the IPSec connection will change into “Stopped”. If no record is
selected, the “One Record should be selected” error message
will occur. More than one record may be selected at a time to be
stopped.

Fig. II-117: IPSec Connection Settings page

Add leads to the Add IPSec Connection wizard where a new IPSec connection can be defined and specified. The wizard provides several pages.

Edit leads to a set of IPSec Connection Properties pages to modify the parameters of the selected IPSec connection. The page includes the same

components as the Add IPSec Connection page. To operate with Edit, only one record may be selected, otherwise an error will occur: “One row
must be selected”.

Restart all Connections restarts all active IPSec connections. The State of these IPSec connections will turn into Connected or Activated if the
restart procedure has been completed successfully.

RSA Key Management leads to the RSA Key Management page to see the current RSA key, to generate a new one and to send it to the peer via
e-mail.

The first IPSec Connection Wizard page Add IPSec Connection has the Connection Name text field that requires a new IPSec connection name,
which is mandatory, and should be filled out, otherwise an error will occur: “Error: Incorrect connection name”.

Please Note: The input in the Connection Name field should be only in Latin characters, otherwise an error occurs and no IPSec connection can be

created.

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The Peer type drop down list is used to choose the remote
machine type for the IPSec Connection to be established. If the
list does not include the required type of machine, choose

Other.

VPN Network Topology drop down list allows to select the

location of the peers participating to the VPN connection.
Following selections are present in the list:

• Bizfon<>Peer – direct connection between Bizfon and a

peer.

• Bizfon<>[Internet]<>Peer – connection between Bizfon and

peer over Internet.

• Bizfon<>NAT<>[Internet]<>Peer – connection between

Bizfon and peer over Internet through Bizfon provider’s
NAT.

• Bizfon<>[Internet]<>NAT<>Peer – connection between

Bizfon and peer over Internet through peer provider’s NAT.

The second page of the IPSec Connection Wizard, IPSec
Connection Properties serves to specify the members of
the IPSec Connection and to set the basic parameters for
encryption.

A group of radio buttons are used with Dynamic IP/Road
Warrior and Static IP/ Remote Gateway to select if the
remote Bizfon (or another VPN gateway device) is
connected to the Internet with a dynamic IP address and is
acting as a Road Warrior, or is connected to the Internet
with a fixed IP address and is acting as a VPN Gateway.

If Dynamic IP / RoadWarrior is selected, the Remote
Gateway IP Address text field automatically will get the
value “any”, to allow access independent from the sending
IP address.

Selecting Static IP / Remote Gateway requires entering
the IP address or the hostname of the remote Bizfon (or
another VPN gateway device) in the Remote Gateway text
field.

Please Note: Static IP/ Remote Gateway selection is not

possible if this Gateway is positioned behind NAT, since the
IP-address of the remote gateway is not reachable directly
in this case.

Bizfon <> Remote Gateway allows access from the local Bizfon to the remote VPN gateway (local subnet and remote subnet are not included). This
includes management access. Checkbox is disabled when “Bizfon<>NAT<>[Internet]<>Peer” or “Bizfon<>[Internet]<>NAT<>Peer” is selected from

VPN Network Topology drop down list on the first page of IPSec Connection Wizard.

Local Subnet <> Remote Gateway allows access from all stations connected to the local network to the remote VPN gateway device (local Bizfon

and remote subnet are not included). Checkbox is disabled when “Bizfon<>[Internet]<>NAT<>Peer” is selected from VPN Network Topology drop
down list on the first page of IPSec Connection Wizard.

Bizfon <> Remote Subnet allows access from the local Bizfon to all stations of the remote LAN (local subnet and remote VPN gateway devices are
not included). Checkbox is disabled when “Bizfon<>NAT<>[Internet]<>Peer” is selected from VPN Network Topology drop down list on the first
page of IPSec Connection Wizard.

Local Subnet <> Remote Subnet allows access from all stations of the local network to all stations of the remote LAN (VPN gateway devices are
not included). In this case the local and remote subnet IP addresses and subnet masks have to be entered in the corresponding text fields Local
Subnet IP and Remote Subnet IP.

More than one of the above checkboxes may be selected to specify the desired communication relations.

The Stop Connection if not successful checkbox allows to stop the IPSec connection attempts if the partner is still unreachable after the timeout
period. If the checkbox is unselected, the system will continue to try to reach the IPSec connection partner.

The right side of the page offers security settings for key exchange, data encryption and authentication:

The area Keying Type offers the choice between automatic and manual keying. To use manual keying, the Static IP / Remote Gateway needs to
be selected.

Fig. II-118: IPSec Connection Wizard - Add IPSec Connection

Fig. II-119: IPSec Connection Wizard -IPSec Connection Properties

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Auto Keying requires the ESP (Encapsulated Security payload) and IKE (Internet Key Exchange) settings (in addition with Diffie-Helman Group
settings) to be selected for the automatic keying exchange. Encryption and Authentication parameters should be defined for each of these
standards, as well as for the Manual Keying.

The Encryption drop down list offers the following standards for selection:

DES (Data Encryption Standard) is a block cipher algorithm with 64-bit blocks and a 56-bit key. This algorithm is considered to be insecure for
sensitive information.

3DES (Triple DES) uses three DES encryptions on a single data block with three different keys to achieve a higher security than is available from a
single DES pass.

AES (Advanced Encryption Standard) is a computer security standard, which became effective on May 26, 2002 by NIST to replace DES. The
cryptography scheme is a symmetric block cipher, which encrypts and decrypts 128-bit blocks of data. Lengths of 128, 192, and 256 bits are
standard key lengths used by AES.

The area Authentication offers the following parameters to be selected:

SHA (Secure Hash Algorithm) is a strong digest algorithm proposed by the US NIST (National Institute of Standards and Technology) agency as a
standard digest algorithm and is used in the Digital Signature standard, FIPS number 186 from NIST. SHA is an improved variant of MD4 producing
a 160-bit hash. SHA and MD5 are the message digest algorithms available in IPSEC.

SHA1 is an enhanced version of SHA. It works with checksums like MD5 does, but it makes a longer hash.

MD5 (Message Digest) is a hash algorithm that makes a checksum over the messages. The checksum is sent with the data and enables the receiver

to notice whether the data has been altered.

The Diffie-Hellman parameter is used to determine the length of the base prime numbers used during the key exchange process. The cryptographic
strength of any key derived depends, in part, on the strength of the Diffie-Hellman group, which is based upon the prime numbers.

Group 2048 (high) is stronger (more secure) than Group 2 (medium), which is stronger than Group 1 (low). Group 1 provides 768 bits of keying
strength, Group 2 provides 1024 bits, and Group 2048 provides 2048 bits. If mismatched groups are specified on each peer, negotiation fails.

Depending on whether the automatic keying type or the manual one has been selected, the button Next will lead you to the Automatic Keying or
Manual Keying page.

The third page of the IPSec Connection wizard,
Automatic Keying, is used to setup a type of
password (Shared Secret) or the RSA public key to
secure your IPSec Connection. The functionality of
Perfect Forward Secrecy (PFS) can be added to
both.

Shared Secret is a type of password consisting of
any characters that both of the IPSec Connection
partners must know. The authentication will be done
with this shared secret. All encryption functions below
will remain concealed.

RSA requires the public RSA key of your IPSec
Connection partner.

The Local ID requires an IP address, Bizfon FQDN
(Fully Qualified Domain Name) that is resolved to an
IP address, or any @-ed string that is used in the
same way.

Remote ID also requires an IP address, the IPSec
Connection partner’s FQDN (Fully Qualified Domain
Name) that is resolved to an IP address, or any @-ed
string that is used in the same way.

PFS (Perfect Forward Secrecy) is a procedure of
system key exchange, which uses a long-term key
and it generates a short-term keys as is required.
Thus an attacker who acquires the long-term key can
neither read previous messages that she may have
captured nor read future ones.

Use IPSec Compression enables IPSec data
compression. This option is displayed only if the
IPSec-VPN partner supports it.

Fig. II-120: IPSec Connection Wizard - Automatic Keying Settings page

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The Manual Keying page offers the following
components:

Depending on the selected encryption and
authentication services of the prior page (IPSec
Connection Properties) you will get some of the
following text fields:

• DES Encryption Key

• 3DES Encryption Key

• SHA1 Authentication Key

• MD5 Authentication Key

Manual keys must be entered in the hexadecimal
format, otherwise the “Incorrect Encryption Key” error
appears.

The SPIs (Security Parameter Index) are indices to
keep the IPSec Connection tunnels distinct. A security
association (SA) is defined by destination, protocol
and SPI. Without the SPI, connections to the same
gateway using the same protocol can not be
distinguished.

The public key is displayed in the RSA Public Key
text field so the user may inform their IPSec
connection partner about it, for example, via fax.

Furthermore, the user has a possibility to generate a
new pair of keys by specifying the key length with the
corresponding radio buttons Generate a new 1024bit
RSA Key and Generate a new 2048bit RSA Key
and the clicking the Generate Button.

A valid RSA key should fit to following requirements:

• RSA key doesn't start with "0s"

• RSA key doesn't end with "=="

• RSA key contains symbols other than

Alphanum, +, /, =

The Email this to the peer text field requires the
mailing address of the IPSec connection partner. The
Send button will insert Bizfon’s public RSA key into
an e-mail and send it to the IPSec connection partner.

PPTP (Point-to-Point Tunneling Protocol) is used to establish a virtual private network (VPN) over the Internet. Remote users can access their
corporate networks via any ISP that supports PPTP on its servers. PPTP encapsulates any type of network protocol (IP, IPX, etc.) and transports it
over IP. Thus if IP is the original protocol, IP packets ride as encrypted messages inside PPTP packets running over IP. PPTP is based on point-to­point protocol (PPP) and the Generic Routing Encapsulation (GRE) protocol. Encryption is performed by Microsoft's Point-to-Point Encryption
(MPPE), which is based on RC4.

L2TP (Layer 2 Tunneling Protocol) is a protocol from the IETF, which allows a PPP session to run over the Internet, an ATM, or frame relay network.
L2TP does not include encryption (as does PPTP), but defaults to using IPSec in order to provide virtual private network (VPN) connections from
remote users to the corporate LAN. Derived from Microsoft's Point-to-Point Tunneling Protocol (PPTP) and Cisco's Layer 2 Forwarding (L2F)
technology, L2TP encapsulates PPP frames into IP packets either at the remote user's PC or at an ISP that has an L2TP remote access
concentrator (LAC). The LAC transmits the L2TP packets over the network to the L2TP network server (LNS) at the corporate side. Large carriers
also may use L2TP to offer remote POPs to smaller ISPs. Users at the remote locations dial into the modem pool of an L2TP access concentrator,
which forwards the L2TP traffic over the Internet or private network to the L2TP servers at the ISP side, which then sends them on to the Internet.

For PPTP and L2TP Connections, two parties are required: a Client and a Server. The client is responsible for establishing the connection, hence
active. The server is waiting for clients; it is not able to initiate the connection itself, hence passive.

Attention: L2TP tunnels have no data encryption mechanism.

The Host Name and a Password specify each side. The client should know the server’s name and password (the Bizfon server has no password)
and the server should set the client’s host name and a password. The client and server settings have to match on both sides for successful
connection establishment.

Clients and Servers are identified by their hostnames, which means that only one client can be connected to the server in the same network. Servers
also define the range of IP addresses that are assigned to the Server and Client hosts participating in a connection.

Fig. II-122: IPSec Connection Wizard - IPSec Connection RSA Key Settings page

Fig. II-121: IPSec Connection Wizard - Manual Keying Settings page

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The PPTP Client Configuration link displays a page where all
existing PPTP client connections are listed, characterized by
their Connection Name, the State of the PPTP connection
(Pending, Disabled, Trying…, Authentication Failure, No
Connectivity - still trying, Unknown, Broken, or Connected) and
the Remote IP/Hostname (the IP address or the hostname of
the PPTP server). PPTP Connections’ states, except the
“Disabled” state, are established as a link that refers to the page
where logout information about PPTP connection status is
displayed. Logs can be useful to determine problems on PPTP
connections failure.

Start initiates the PPTP client(s) activity (reaching the server).
Several client connections may be selected at once.

The Stop button is used to stop the selected PPTP client(s)
activity. Several client connections may be selected at once
using this function.

Fig. II-123: PPTP Client Configuration page

Add leads to the PPTP Client Connection - Add Entry page
where a new PPTP client connection can be established:

Server Host Name requires the server’s hostname.

Please Note: The input in the Server Host Name field should

be only in Latin characters, otherwise an error occurs and no
PPTP connection can be created.

Client Host Password requires the local peer password.

The Server Host IP radio buttons allow selecting the PPTP
server.

IP requires the IP address of the PPTP server.
Hostname requires the FQDN (Full Qualified Domain Name) of

the PPTP server.

Please Note: All settings should be configured the same way on

both the PPTP client and server hosts, that is to say, for
successful PPTP tunnel establishment; all settings should match
each other.

Fig. II-124: PPTP Client Configuration - Add Entry page

The PPTP Server Configuration link displays a page to
configure the PPTP server connections. It is divided into two
pages with a table of the existing PPTP server connections on
one page and the PPTP server settings on the other.

The page PPTP Server Connections offers the following input
options:

The table PPTP Server Connections lists all the PPTP
connections, characterized by their Connection Name and the
State of the PPTP connection (Disabled, No Client Connected
or Connected: IP address). Each PPTP Server connection can
hold only one PPTP tunnel.

Start is used to activate the selected PPTP connection(s).
Several server connections may be selected at once.

Stop is used to deactivate the selected PPTP connection(s).
Disabling the connection will disconnect all connected clients
and close the PPTP tunnel. Several server connections can be
selected at once to operate with this function.

Fig. II-125: PPTP Server Configuration page

Add leads to the PPTP Server Connection - Add Entry page
where a new PPTP Server connection can be made.

Client Host Name and Client Password require the
corresponding client’s host name and password.

Please Note: The input in the Client Host Name field should be

only in Latin characters, otherwise an error occurs and no PPTP
connection can be created.

In certain cases, Client Host Name can be a conditional

Fig. II-126: PPTP Server Configuration - Add Entry page

username defined by the client and used for PPTP connection

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establishment.

Please Note: All settings should be configured the same way on

both the PPTP client and server, that is to say, for successful
PPTP tunnel establishment, all settings should match.

The PPTP Server Configuration page is used to configure the
PPTP server settings and offers the following components:

The PPTP Subnet text fields are used to enter the IP address
range for the PPTP server and clients within the PPTP tunnel.
The value specified for the subnet mask is fixed to 24 to restrict
the possible number of clients for the PPTP connection.

Please Note: The first address specified in the PPTP Subnet

will be assigned to the PPTP server; others will be assigned to
the clients. PPTP server subnet should be different from the
L2TP server subnet, otherwise a corresponding error message
appears.

Fig. II-127: PPTP Server Configuration page

The L2TP Configuration link displays a page to configure the
L2TP connections.

Attention: L2TP tunnels have no data encryption mechanism.

The L2TP Configuration page has two pages with the table of
existing L2TP connections on one page and the L2TP server
settings on the other page.

The L2TP Connections page has the following components:

The L2TP Connections table lists all the L2TP connections
characterized by their Connection Name, connection Type
(active or passive), the State of the L2TP connection (Waiting,
Connected, Trying, Disabled or Down) and the Remote IP
address (the IP address or the hostname of the L2TP partner if
connection is active). Each L2TP passive connection can hold
only one L2TP tunnel.

Start is used to enable the selected L2TP server or client
connection(s). Several records may be selected at with this
function.

Stop is used to disable the selected L2TP server or client
connection(s). Stopping the server will disconnect all connected
clients and close the L2TP tunnel. Several records may be
selected at once with this function.

Fig. II-128: L2TP Configuration page

Add leads to a page where the connection type (active or
passive) has to be selected. Afterwards the appropriate Add
L2TP Connection page will be displayed where a new L2TP
connection can be established.

The Passive L2TP Connection - Add Entry page is used to
specify passive L2TP server connections.

Client Host Name and Client Password text fields require the
corresponding client’s host name and password.

Please Note: The input in the Client Host Name field should be

only in Latin characters, otherwise an error occurs and no L2TP
connection can be created.

In certain cases, Client Host Name can be a conditional
username defined by the client and used for L2TP connection
establishment.

Please Note: All settings on this page should be configured in

the same way on both the L2TP client and server, that is to say,.
for successful L2TP tunnel establishment, all settings should
match.

Fig. II-129: L2TP Configuration - Add Passive Connection page

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The Active L2TP Connection - Add Entry page is used to
specify L2TP client connections and offers the following
components:

The Server Host Name text field requires the L2TP server’s
name.

Please Note: The input in the Server Host Name field should

be only in Latin characters, otherwise an error occurs and no
L2TP connection can be created.

The Client Host Password text field requires the local peer
password.

The Server IP text field requires the IP address of the L2TP
server.

Please Note: All settings have to be configured the same way

both on the L2TP client and the server hosts. For successful
L2TP tunnel establishment all the settings should match each
other.

Fig. II-130: L2TP Configuration - Add Active Connection page

The Configuration page is used to configure the L2TP server
settings and provides the following input options:

The L2TP Subnet text fields are used to enter the IP address
range for the L2TP server and clients within the L2TP tunnel.
The value specified for the subnet mask is fixed to 24 to restrict
the possible number of clients for the L2TP connection.

Please Note: The first address specified in the L2TP Subnet will

be assigned to the L2TP server; others will be assigned to the
clients. L2TP server subnet should be different from the PPTP
server subnet, otherwise a corresponding error message
appears.

Fig. II-131: L2TPServer Configuration page

To Specify an IPSec Connection

1. Press the Add button on the IPSec Connection Settings page. The IPSec Connection Wizard will appear in the browser window.

2. Select a VPN Peer Type and assign a name to the IPSec Connection. Press Next to go to the next page of the IPSec Connection wizard.

3. Enter the remote side IP parameters, check subnets/gateways for the connection, select the NAT traversal option (if needed), and the desired

keying type. Press Next to advance to the next page of the IPSec Connection wizard.

4. If the Automatic Keying type has been selected enter the automatic keying parameters and select the PFS and IPSec compression options (if

needed). If the Manual Keying type has been selected enter the encryption and authentication keys and SPI(s).

5. To specify an IPSec connection with these parameters press Finish. Use Cancel to abort the operation.

To Manage an RSA key for the IPSec Connection

1. Press the RSA Key Management button on the IPSec Connection Settings page. The IPSec Connection RSA Key will appear in the

browser window.

2. Select the RSA key length and press Generate to generate a new RSA public key. This may take several seconds.

3. Enter a destination e-mail address in the Email this key to peer text field, then press Send to send the new RSA public key.

To Specify a PPTP Client Connection

1. Press the Add button on the PPTP Client Configuration page.

2. Enter the server host name in the Server Host Name text field.

3. Specify the PPTP client password in the Client Password text field.

4. Using the radio buttons on the page, select the server address representation (IP address or hostname) and provide the corresponding

information in the Server Host IP fields.

5. Press Save to add a PPTP connection with these settings.

To Add a PPTP Server Connection

1. Press the Add button on the PPTP Server Configuration page.

2. Enter the client host name and the password in the Client Host Name and Client Password text fields.

3. Press Save to add a PPTP connection with these settings.

To Add an Active L2TP Connection

1. Press the Add button on the L2TP Configuration page.

2. Select the Active Connection link.

3. Enter the server host name in the Server Host Name text field.

4. Specify the L2TP client password in the Client Password text field.

5. Specify the server’s IP address in the Server IP text field.

6. Press Save to add an active L2TP connection with these settings.

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To Add a Passive L2TP Connection

1. Press the Add button on the L2TP Configuration page.

2. Select the Passive Connection link.

3. Enter the client host name and the password in the Client Host Name and Client Password text fields.

4. Press Save to add a passive L2TP connection with these settings.

To Delete/Stop/Start/Enable/Disable a VPN Connection

1. Select one or more checkboxes of the corresponding connections that ought to be deleted/stopped/started from the Connection tables. Press

Select all to delete/stop/start all connections.

2. Click on the Delete/Stop/ Start button from the table’s menu to perform the corresponding operation for the selected VPN connection(s).

3. If deleting confirm it with Yes. The VPN connection will be deleted. To abort the deletion and keep the VPN connection in the list, click No.

Dynamic DNS Settings

The Dynamic DNS (DynDNS) is a service that is used to map a dynamic IP address to a host name. Thus this service only makes sense if you are
connected to the Internet with a dynamic IP address (and PPP, DHCP client) and want to allow access from the Internet to a device behind the
firewall. For example, if you want to run your own WEB server.

To enable the DynDNS service on Bizfon you first have to choose a DynDNS provider and register at their website.

The Dynamic DNS Settings page provides the following
components:

The Enable Dynamic DNS checkbox selection enables the
dynamic DNS service.

The User text field requires the username specified during the
registration at the DynDNS provider.

The Password text field requires the password specified during
the registration at the DynDNS provider.

The Max time between updates text field requires entering the
period between two updates (in hours). The values entered in
these fields should be greater than 24, otherwise an error
occurs: “Update interval times smaller than 24 hours are too
small”. Normally, whenever you set up a connection to the
Internet, the DynDNS is updated at least once in the period
indicated in this field.

The Use predefined service radio button leads to the manual
configuration of the DynDNS service. The selection enables the
following optional settings:

The Service drop down list contains the provider list where the
administrator needs to select the one that has been subscribed
to.

The Host text field requires the name of the host on the Internet.

The TZO Connection Type text field is used for a special
parameter required by the DynDNS provider TZO.

The DHS Cloak-Title text field is used for a special parameter
required by the DynDNS provider DHS.

The Mail Exchange text field requires the address of the e-mail
server where the DynDNS service provider will relay your e­mails.

Attention: If this service is used, make sure, that there is port

forwarding configured for SMTP (port 25) to the internal e-mail
server.

The easyDNS Partner text field is used for a special parameter
required by the DynDNS provider easyDNS.

Selecting the Create Custom HTTP GET Request radio button switches to the custom settings of the DynDNS service. Normally, the DynDNS
provider uses HTTP get requests to map dynamic IP addresses to host names. If the user knows this HTTP get request exactly, the radio button
Create Custom HTTP GET Request together with the text field URL allows to enter it directly.

The selection enables the following optional settings:

The URL text field requires the complete request to be sent to the DynDNS server. Normally it has the format:

[http://]www.server.domain:port/scriptpath/scriptname?param1=value1¶m2=value2

Fig. II-132: Dynamic DNS Settings page

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The request modifies the nameserver database so that the hostname will be resolved to the new IP address.

The Basic Authentication checkbox enables the encoding of the username and password entered in the text fields above, and then uses the Basic
Authentication method to notify the provider about the user authentication settings.

Most of the DynDNS providers require an authentication for security. The user can do that either together with the HTTP get request in the text field
URL or by selecting the Basic Authentication checkbox.

Firewall and NAT

The Firewall Configuration page allows setting up a firewall, configuring the security level and enabling the NAT and IDS services of Bizfon.

A Firewall is a security service configured by the Bizfon administrator based on various criteria. The firewall allows or blocks traffic based on
policies, services and/or IP addresses. The firewall has several levels of security policies (low, medium or high). The administrator may add
additional service-based rules. Filtering rules will take effect only if the Firewall has been enabled and are independent from the selected firewall
security level.

NAT (Network Address Translation) is used to allow Bizfon LAN members to connect to the Internet, using Bizfon's WAN IP address. The
Bizfon/NAT also handles forwarding incoming packets from the WAN to the PCs or devices on Bizfon’s LAN.

The IDS (Intrusion Detection System) is a type of firewall, but together with deleting dangerous packets or packets containing intrusion attacks, IDS
generates a log file with information about these dropped packets and the senders responsible for those packets. The log can be viewed on the

Log

page and notifications about them can be sent to the user in various ways (e-mail, flashing LED and display notification).

The Firewall Configuration page offers the following
components:

The Enable IDS checkbox selection enables the Intrusion
Detection System.

The Enable NAT checkbox selection enables Network Address
Translation.

The Enable Firewall checkbox selection enables the firewall
security service. The firewall security level has to be selected,
otherwise the firewall cannot be enabled.

The Firewall Security radio buttons are:

IDS

• Low Security - Everything that is not explicitly forbidden is

allowed. This security level doesn't block anything by
default. It is recommended if the device is already located
behind another firewall or if every filter has been configured
correctly.

• Medium Security - Traffic originating from the LAN side

may pass and traffic from the WAN side will be blocked by
default. This is the recommended security level.

• High Security - Everything that is not explicitly allowed will

be blocked, including traffic from the LAN side.

Advanced Firewall Settings link refers to page where Bizfon’s

privacy can be configured.

The View Filter Rules link opens the

Filtering Rules page.

Fig. II-133: Firewall and NAT Settings page

Advanced Firewall Settings

Advanced Firewall Settings are used to deny Ping and
Portscanning operations addressed toward the device. With
these features enabled, Bizfon will answer with inscrutable
messages to the Ping and Portscanning operations.

Please Note: Operations are available only when Firewall is

enabled from the

The page offers the following components:

The Ping Stealth checkbox selection prohibits a Ping
operation toward Bizfon from its WAN.

The Fool Portscanner checkbox selection prohibits Bizfon
portscanning from its WAN. As a reply to a Portscanning
operation, "network unreachable" or "host unreachable"
feedback messages will be sent.

Firewall and NAT page.

Fig. II-134: Advanced Firewall Settings page

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Filtering Rules

The Filtering Rules page allows the configuration of filters for the incoming and outgoing traffic.

To prevent misconfiguration, only one rule per service is allowed. The user may use IP groups to include several IP addresses for this rule. As the
filtering rules specify the operation mode of the firewall, they only take effect if the firewall has been enabled (additionally NAT should be enabled to
use the Port Forwarding function in the Incoming Traffic / Port Forwarding filtering rules). The filtering rules are independent from the security
level, so they will work if enabled, no matter what security level has been selected.

Please Note: Applying firewall rules will just prevent the establishment of new connections that violate the rules. Applying rules does not kill existing

connections that violate the rule.

View All displays all configured filters specified by their State
(enabled or disabled), the selected Service, the set Action
(allowed or blocked), the IP addresses the filters apply to (if
Restricted) and the destination of port forwarding (Redirect to,
in case of Incoming Traffic/Port Forwarding). As it is read-only,
no modifications are allowed and no functional buttons are
available.

The Incoming Traffic/Port Forwarding filter is for incoming
traffic. The rules here allow or deny systems on the Internet to
reach the services of Bizfon’s LAN. NAT service should be
enabled on the Bizfon to provide the possibility of Port
Forwarding in the Incoming Traffic/Port Forwarding filtering
rules. The Port Forwarding function will be unavailable if NAT is
disabled on the Bizfon.

The Outgoing Traffic filter is for outgoing traffic. The rules here
allow or deny Bizfon’s LAN users to reach external services.

Management Access is used to enable management access to
the Bizfon from the Internet. A host on the Internet can be allowed
to reach the Bizfon.

SIP Access is to allow or deny the SIP access to or from the
particular SIP servers, SIP hosts or a group of them. The SIP
Access filtering rule may prevent or allow incoming or outgoing
SIP calls to or from specified SIP server(s) or host(s).

When Blocked IP List is used, traffic from specific hosts may be
blocked, no matter what services are opened in the other filters.
NO traffic will be allowed to the specified hosts. The Blocked IP
List service has a higher priority if the same host is also listed in
the Allowed IP List table.

Allowed IP List allows trusted hosts to reach your network and
vice versa. It is an exception to other rules and only all services
may be allowed for a single host.

Restricted IPSec - Generally hosts in a VPN are allowed to have
access to any service, i.e., no traffic will be blocked. They are
treated as if they were part of the Bizfon LAN. However, this
service can be manually denied here.

The Filtering Rules page provides several links. Each link opens its specific parameters on the same page. Only Change Policy (see chapter

Firewall and NAT), Manage user Defined Services (see chapter Service Pool) and Manage IP Pool Groups (see chapter IP Pool) are leading to

separate pages. The Filtering Rules page also includes the currently selected firewall security (Policy) level and its description.

The table displayed on the bottom of the page shows the filters selected above, specified by their State (enabled or disabled), the selected Service,
the set Action (allowed or blocked), the IP addresses the filters apply to (if Restricted) and the destination of port forwarding (Redirect to, in case of
Incoming Traffic/Port Forwarding). With the exception of View All, the table offers the following functional buttons:

Fig. II-135: Filtering Rules page

• Enable is used to enable the rule. If no records are selected the “No record(s) selected” error occurs.

• Disable is used to disable the rule. If no records are selected the “No record(s) selected” error occurs.

• Add opens a filter specific page where new rules may be defined by a Service, an Action, a Restriction to certain IP address(es) or IP groups,

and if adding a rule for Incoming Traffic/Port Forwarding, the destination IP address for Forwarding:

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For example, the page to add a rule for Incoming Traffic/Port
Forwarding offers the following input options:

Service includes a list of possible services to be configured. All

user defined services also will be displayed in this list.

Action includes possible actions to setup the rule.

Forward to IP requires the destination IP address where traffic

should be transferred to, if it comes from the restricted host. The
IP address defined in this field will be ignored for blocked action
of the Incoming Traffic/Port Forwarding rule.

Note: It is not allowed to forward incoming packets when NAT

service is disabled on the Bizfon.

Port Translation text field is available for “Allowed” action only
and optionally requires the port number that will stand instead of
original port number when incoming packet is being forwarded. If
this field is left empty, original port number will be used upon
forwarding the packet.

Restriction radio buttons:

• Selecting Any blocks or allows all host IP addresses. This

selection is not present for the Management Access,
Blocked and Allowed IP List rules.

• Selecting Single IP will require the IP address of the

allowed or blocked host.

• Selecting IP/Mask will require the subnet to be allowed or

blocked, specified by an IP address and the Maskbits.
Maskbit examples:

255.0.0.0= /8,

255.255.0.0 = /16,

255.255.255.0 = /24,

255.255.255.255= /32

• Group indicates the user defined groups that include IP

addresses that ought to be allowed or blocked.

Description field is used to insert an optional description of the
filtering rule.

Fig. II-136: Filtering Rules - Page to add a rule for Incoming Traffic

To Add a Filtering Rule

1. Select the Filter link (Incoming Traffic/Port Forwarding, Outgoing Traffic, Management Access, SIP Access, Blocked IP List, Allowed IP List or

Restricting IPSec) to add a rule for it. The corresponding Filter table will appear in the same window.

2. Click Add on the Filtering Rules page. A page where a new rule may be added will appear in the browser window. The page will be named

corresponding to the selected filter.

3. Select a service name from the Service list to configure a rule for it. If the list has a default value, leave it as is.

4. Select an action from the Action list that is used in the rule. If the list has a default value, leave it as is.

5. Enter the IP address in the Forward to IP field if an Incoming Traffic Rule is to be added.

6. Choose the restriction type by selecting Any, SingleIP or IP/Mask and enter the required information in the text fields or select a group.

7. Insert a Description, if needed.

8. To add a rule with these parameters press Save.

To Delete Filtering Rules

1. Select the Filter link to delete a rule from its table. The appropriate Filter table will appear in the same window.

2. Check one or more checkboxes of the corresponding rules that ought to be deleted from the rules table. Press Select all if all rules ought to be

deleted.

3. Press the Delete button on the Filtering Rules page.

4. Confirm the deletion with Yes, or cancel it with No.

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Service Pool

The Service Pool table is a list of all created services and their
parameters. It is used to add new services with the appropriate
settings (protocol type and port range). New services can be
used to add a restriction or permission by defining a new filtering
rule:

Add opens the Add New Service page where new services may
be added.

Edit opens the Edit Service page where the service parameters
(except for the service name) can be modified. This page
includes the same components as the Add New Service page.
To operate with Edit only one record may be selected, otherwise
an error will occur: “One row must be selected”.

Fig. II-137: Service Pool page

The Add page is used to add new services and includes the
following text fields and buttons:

Service Name requires a name for the service that ought to be
added.

Protocol includes a list of possible protocols to be selected.

Port Range requires a port range for the defined service.

Fig. II-138: Service Pool - Page to add a new Service

To Add a new Service

1. Select the Manage User Defined Services link on the Filtering Rules page.

2. Click on the Add button on the Service Pool Configuration page. A page where a new service may be added will appear in the browser

window.

3. Define a service name in the Service Name text field.

4. Select the protocol type for the service from the Protocol drop down list.

5. Enter the port range in the Port Range text fields or leave one of them empty to define a particular port for the service.

6. To add a service with these parameters click on Save.

To Delete a Service

1. Select the Manage User Defined Services link. The Service Pool Configuration page appears with the table of services (if any).

2. Check one or more checkboxes of the corresponding services that ought to be deleted from the Service Pool table. Press Select all if all

services ought to be deleted.

3. Click on the Delete button on the Service Pool Configuration page.

4. Confirm the deletion with Yes, or cancel it with No.

IP Pool

The Manage IP Pool Groups link opens the IP Pool Configuration page.

The IP Pool table is the list of all added groups and the members
assigned to these groups. If a group is empty, EMPTY will be
indicated in the Members column. If hidden, group members will
still remain active but HIDDEN will be displayed in the Members
column.

The IP Pool Configuration is used to add groups of IP addresses
that have the same restriction criteria. Whenever adding a new
filtering rule, groups may be used instead of several IP addresses.

IP Pool Configuration offers the following components:

View makes hidden groups visible.

Fig. II-139: IP Pool Configuration page

Hide makes group members hidden and adds the HIDDEN

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comment in the member column.

Add opens the Add Group page where a new group may be
added. This page consists of the Group Name text field (requiring
the group name) and the Group Description text field (requiring
the optional group description), as well as standard Save and

Back buttons to apply or abort changes.

Edit opens the Edit Group page where the service parameters
can be modified. It provides the same components as the Add
Group page. To operate with Edit, only one record may be

selected, otherwise an error will occur: “One row must be
selected”.

Please Note: Changing a group name will also change the

references to this group, including groups where this group is a
member of, and all affected filter rules (enabled and disabled ones,
in all chains).

Clicking on the Group name will display an IP Pool Group
Configuration page with the Members list for the current group.

Fig. II-140: IP Pool configuration – Add Group page

The IP Pool Group Configuration page displays a list of all the
added member IP addresses for the selected group. It offers the
following components:

Current Group provides read-only information about the current
group name the members are listed for.

Add opens the Add Member page where a new member may be
added.

Edit opens the Edit Members page where the service parameters
can be modified. This page includes the same components as the
Add Member page. To operate with Edit, only one record may be
selected, otherwise an error will occur: “One row must be
selected”.

Fig. II-141: IP Pool Group Configuration page

The Add Members page provides the following radio buttons:

IPaddress requires the member IP address that is to be added to
the group.

IP Subnet requires the subnet specified by the IP address and the
Maskbits. See above for more information about Maskbits.

The User-defined Group includes previously added groups that
also may be added as a member to another group.

Member description text fields can be used to enter an optional
description of the member.

Fig. II-142: IP Pool Group Configuration – Add Member

To Add a new Group with Members

1. Select the Manage IP Pool Groups link on the Filtering Rules page.

2. Click on the Add button on the IP Pool Configuration page. A page where a new group may be added will appear in the browser window.

3. Define a group name in the Group Name text field and fill in the Group Description, if needed.

4. To add a group with the given parameters press Save.

5. Open the IP Pool Group Configuration page by clicking on the group name.

6. Select the Add button on the IP Pool Group Configuration page. A page opens where new members may be added to the group.

7. Enter an IP address for the member in the IP Address text fields, select a IP subnet or IP group from the User defined Group drop down list

to assign it to the currently selected group.

8. Enter a Member Description in the corresponding text field, if needed.

9. To add a member with these parameters to the selected group press Save.

To Delete a Member

1. Select the Manage IP Pool Groups link. The IP Pool Configuration page appears with the table of groups (if any).

2. Click on the desired members that ought to be deleted. The IP Pool Group Configuration list appears.

3. Check one or more checkboxes of the corresponding members that ought to be deleted from the Members table. Press Select all if all

members ought to be deleted.

4. Press the Delete button on the IP Pool Group Configuration page.

5. Confirm the deletion with Yes or quit with No.

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Bizfon Manual II: Administrator's Guide Administrator's Menus

To Delete a Group

1. Select the Manage IP Pool Groups link. The IP Pool Configuration page appears with the table of groups (if any).

2. Check the one or more checkboxes of the corresponding groups that ought to be deleted from the groups table. Press Select all if all groups

ought to be deleted.

3. Press the Delete button on the IP Pool Configuration page.

4. Confirm the deletion with Yes or quit with No.

IDS Log

The IDS logging page contains information about dropped packets
and the senders responsible for those packets. IDS discards
dangerous packets or packets including intrusion attacks and
generates a table with the IDS log report. The administrator can be
notified about newly logged entries in various ways (mail, display
notification and Flashing LEDs) depending on the settings on the
Event Settings page. To make an IDS log reporting table, IDS
needs to be enabled on the

The IDS Logs table is a list of new or read IDS entries and
descriptions referring to them. The table provides a status row that
has the value New if the entry is still unread or that is empty if the
entry has already been read.

Mark All as Read marks all IDS logged entries as read and
removes the New status from the Status row of the IDS entries
table.

Delete Log is used to delete all entries from the IDS table.

A detailed log of the selected entry can be seen by clicking on the
Description link of the corresponding entry in the IDS Entries
table.

Firewall and NAT page.

Fig. II-143: IDS Log page

The IDS Logs detailed page has a following preview:

The Issue Detailed Log table is a detailed list of new and read
IDS entries. The table contains a Status row that has the value
New if the entry is still unread or that is empty if the entry has
already been read.

Fig. II-144: IDS issue detailed preview

Bizfon4000 (SW Version 3.1.x) 86

Bizfon Manual II: Administrator's Guide Administrator's Menus

LAN Services Menu

Fig. II-146: LAN Services menu in Plain theme

Fig. II-145: LAN Services menu in Dynamo theme

DNS Settings

The DNS Settings page gives the possibility to setup a name
server for the Bizfon. It offers the following components:

NameserverAssignment radio buttons:

• Dynamically by provider selection automatically

configures the assignment of the name server address from
the provider party.

• Fixed Nameserver address is a manually selected name

server. The Nameserver text field requires the IP address
of an external name server. The Alternative Nameserver
text field requires the IP address of the secondary name
server. The Alternative Nameserver is used if the main
name server cannot be accessed.

Fig. II-147: DNS Settings page

DHCP Settings for the LAN Interface

The DHCP Settings page gives the possibility to enable a DHCP server and control the Bizfon user’s LAN settings. Thus Bizfon LAN users will be
provided automatically with the following settings using the configured parameters:

• IP addresses

• NTP (corresponds to the Bizfon’s IP address)

• WINS server

• Nameserver (corresponds to the Bizfon’s IP address)

• Domain name

The DHCP Settings page offers the following input
options:

Enable DHCP Server activates the DHCP server on
Bizfon.

IP Address Range defines a range of IP addresses that
will be assigned to the Bizfon LAN users. The IP range
must be at least six, otherwise the “Address Range too
small” error will prevent saving. The “Address Range too
large” error occurs if the IP range is greater than 254.

WINS Server defines a WINS server IP address for the
Bizfon LAN users.

View DHCP Leases leads to the page where the DHCP
leased LAN IP addresses are listed.

Fig. II-148: DHCP Settings page for LAN interface

Bizfon4000 (SW Version 3.1.x) 87

Bizfon Manual II: Administrator's Guide Administrator's Menus

The DHCP Leased IP Addresses page includes a list of
the leased host addresses that are part of the Bizfon’s
LAN. For these hosts, Bizfon acts as a server supplying
them with a unique IP address. It displays a read-only
table describing all the leased IP hosts and their
parameters. The table contains the following columns:

IP address - host IP address, assigned by Bizfon.

MAC address - host MAC address, provided by the host

itself.

Lease Start - date and time when the leased IP address
has been activated.

Lease End - date and time when the leased IP address
has been or will be deactivated.

Hostname - hostname, provided by the host itself.

Fig. II-149: DHCP Leases page for LAN interface

Bizfon4000 (SW Version 3.1.x) 88

Bizfon Manual II: Administrator's Guide Administrator's Menus

Administrator’s Additional Features

Incoming Call Blocking and Outgoing Call Blocking

The Incoming Call Blocking and Outgoing Call Blocking pages offer extended features for the administrator to activate incoming/outgoing call
blocking services for certain callers. This configuration cannot be changed by the users.

For more information on the Call Blocking Settings pages, see Incoming Call Blocking and Outgoing Call Blocking chapters of the Extensions
Users Guide - Manual III.

The Call Blocking pages accessed from Caller ID Based
Services table by clicking on the corresponding address, give
administrator a possibility to enable blocking services which
could not be disabled by the users.

Besides the components seen for the user, an additional

Protect this entry checkbox is available in the Call Blocking ­Add Entry pages for administrator access only. With this

checkbox selected, user will be unable to deactivate the
blocking services configured by administrator.

Fig. II-150: Blocking Page for the Administrator

Logout

This option is used to close the session between the user PC and Bizfon and to leave the Bizfon Web Management or to enter the management with
another login. By selecting the Logout button, the startup page will be displayed and the user needs to login again.

Bizfon4000 (SW Version 3.1.x) 89

Bizfon Manual II: Administrator's Guide Welcome Page

Appendix: Extension User's Welcome Page

This welcome page may be helpful, if administrators want to inform their extension users about individual data, they need to use the extensions.
Such as phone numbers, phone lines, IP addresses and SIP numbers. To get a word form that may be edited and sent by mail, double-click on the
paperclip sidewise.

Welcome

You are using a Bizfon Voice Router made by Bizfon Inc. This product incorporates SIPVoice™ Digital
Signal Processing technology to send crystal clear voice around the globe without associated fees for long
distance. But, you will soon learn, it does much more. Your Bizfon Voice Router, The Global Phone
Network in a Box, operates in much the same way as systems with which you are already familiar: a
telephone, a PBX, voice mail, a phone book, et cetera. Beyond that the Bizfon Voice Router provides
capabilities you never believed were accessible in a customer premise telephony product. Soon you will
experience the freedom and power of the Bizfon Voice Router, The Global Phone Network in a Box.

To get started the following information is helpful.

PHONES

Your extension number is <extension number> and your password is <password> (optional).

Remember to type 00 when you pick up your phone receiver to find THE WELCOME SPOT. *0 will take you
directly to voice mail for your extension. *4 will confirm your extension number.

LOCAL PHONE LINES

Bizfon4000 offer 4 external phone lines. They are:

<1. local phone line> <2. local phone line> <3. local phone line> <4. local phone line>.

IP

To reach your Bizfon Voice Router from a network connection inside your office, home or place of utilization,
connect a Web browser to <IP address> (172.30.0.1 is the default IP address).

The email address of your Bizfon Voice Router System Administrator is <email address>.

His phone numbers are <phone numbers>.

SIP

Your SIP number (an Internet phone number) is <SIP number>@sip.bizfon.com.

This is a number you can give people to reach you.

The SIP number to reach the Auto Attendant of your local Bizfon is <SIP number>@sip.bizfon.com.

The email address of your SIP System Administrator is <email address>.

His phone numbers are <phone numbers>.

Bizfon4000 (SW Version 3.1.x) 90

Bizfon Manual II: Administrator's Guide System Default Values

Appendix: System Default Values

Administrator Settings

Parameter System Default Value

Admin Settings

Bizfon Hostname bizfon

LAN IP Address

DHCP Server

Regional Settings and Preferences

Emergency and PSTN codes

WAN Interface Protocol Ethernet

WAN Interface Bandwidth

WAN IP Automatically through DHCP

Mac Address

DNS Server Dynamically

IP Routing Configuration

Event Settings

Time/Date Settings

Mail Settings Disabled

Login name -admin
Password - 19

172.30.0.1
Subnet Mask - 255.255.0.0

Enabled,
IP Range - 172.30.0.100-172.30.0.254,
WINS - 0.0.0.0.

Locale – US,
TimerZone - US/Central,
Theme – Dynamo.

Emergency code -911,
PSTN code – 9.

Upstream – 10000,
Downstream – 10000,
Min Data Rate – 0.

Assigned by device,
MTU - 1500 Bytes.

No Routes

”Display notification” for all except Login event, which has “Do nothing” action
assigned.

NTP Server and Client – enabled,
Predefined NTP Server - ntp1.bizfon.com,
Polling interval – 6.

SMS Settings Disabled

Features IP Phone support - disabled

Language Pack

User Rights Management

Extensions Management 00, 11-14, 31-46 extensions exist

Extension Settings – General

Extension Settings – SIP

Bizfon4000 (SW Version 3.1.x) 91

Default - English
Custom Language Pack - none

Users - admin (enabled), localadmin (disabled).
Roles - Extension (all accessible pages for extension), Local Administrators (all
accessible pages for localadmin).

Display name – none,
Password – empty,
11-14 extensions attached to the FXS lines 1-4,
31-46 extensions attached to the IP lines 1-16,
Call Relay – disabled,
Call Park – disabled,
External Call Policy – disabled,
Percentage of System Memory – 4%.

Registration username and password - automatically generated,
SIP server - sip.bizfon.com,
SIP Server port – 5060,
SIP Server Registration – enabled.

Bizfon Manual II: Administrator's Guide System Default Values

Parameter System Default Value

User ID – undefined,
Send Keep-alive Messages to Proxy – disabled,

Extension Settings – SIP Advanced

RTP Priority Level – medium,
Outbound Proxy, Secondary SIP Server and Outbound Proxy for Secondary SIP

Server – undefined.

Extension Settings – Remote Remote Extension – disabled

Extension Settings – Call Queue Call Queue – disabled

Extension Settings – Voice Mailbox

Internal Voice Mail for all extensions except 14.
Extension 14 – Disabled Voice Mail.

Codecs - G711u (preferred), G711a, G729a, G726/32, G726/16, G726/24, G726/40
Out of Band DTMF Transport – enabled,

Extension Settings – Codecs

T.38 FAX – enabled,
Pass Through FAX – enabled,
Pass Through Modem – disabled,
Force Self Codecs Preference for Inbound Calls - disabled.

Display name – Attendant,

Attendant 00 Settings – General

FAX forwarding – enabled, Extension to forward – 14,
Percentage of System Memory – 3%.

Scenario – default,

Attendant 00 Settings – Attendant
Scenario

Send AA digits to Routing Table – disabled,
Welcome and Menu messages – default,
Authorized Phones Database – undefined.

Attendant 00 Settings – SIP and SIP
Advanced

Same as for an extension.

Codecs - G711u (preferred), G711a, G726/16, G726/24, G726/32, G726/40, G729a,
G723, iLBC

Out of Band DTMF Transport – enabled,

Attendant 00 Settings - Codecs

T.38 FAX – enabled,
Pass Through FAX – enabled,
Pass Through Modem – disabled,

Force Self Codecs Preference for Inbound Calls - disabled.

Universal Extension Recordings Default

Receptionist Management No entries

Extension Directory

Call Statistics

No entries

Enabled
100 entries for all type of calls

UDP and TCP Port – 5060,

SIP Settings

Session Timer – disabled,
DNS Server for SIP – default,
SIP timers – RFC 3261.

Properties for all Codecs except G723 and iLBC :

Packetization -20ms
Silence Suppression -yes

G723 and iLBC properties:

Packetization - 30ms

RTP Settings

Silence Suppression – yes
RTP/RTCP port range for FXS lines - 6000-6049
RTP/RTCP port range for IP lines - 6050-6099
G276 Standard - ITU-T specification
Telephone Event Draft Support - enabled
RTCP Support - disabled

NAT Traversal for SIP – force
SIP and RTP Parameters - Use STUN

NAT Traversal Settings

SIP TCP Port – 5060
STUN Parameters:

Primary STUN Server – stun.bizfon.com

Primary STUN Port – 3478

Bizfon4000 (SW Version 3.1.x) 92

Bizfon Manual II: Administrator's Guide System Default Values

Parameter System Default Value

Secondary STUN Server – undefined

Secondary STUN Port - undefined

Polling Interval: 1 hour

Keep-alive interval: 120 seconds

NAT IP checking interval: 300 seconds

No entries in NAT Exclusion table

Onboard Lines Configuration:

CallerID- Standard 2 FSK for all lines

Ringer type: Type A for all lines

Busy Tone and Power Disconnect indications: disabled for all lines

Line Settings

FXO Settings

Gain Control Settings

Call Routing

RADIUS Settings

Voice Mail Common Settings

Dial Timeouts 4 seconds

Off-hook caller ID - disabled for all lines
IP Lines Configuration:

1-16 IP Lines attached to 31-46 extensions. IP Lines 1-4 enabled, others

disabled.

FXS Lines Loopback Settings – Loopback is disabled for all FXS lines, Loopback

timeout is 30.

4 FXO lines – all enabled, incoming and outgoing calls allowed and routed to 00
Attendant on all lines, Use FXO lines of the other device – disabled.

FXS lines:
Transmit Gain: - 6
Receive Gain: 0
FXO lines:
Transmit Gain: 0
Receive Gain: 6

Route all incoming SIP calls to Call Routing - disabled
Local Routing table - 6 entries. Entries are defined for IP, PBX and PSTN calls
establishment.
Local AAA Table - no entries.

RADIUS client - disabled

Voice Mail Recording - G729a
Memory Allocation - Embedded Memory Storage

IPSec, PPTP and L2TP No connections

Dynamic DNS Disabled

Firewall

IDS Enabled

NAT Enabled

Filtering Rules

Enabled, Medium level,
Ping Stealth - enabled
Fool Portscanner - disabled

Outgoing Traffic - MS File Sharing
SIP Access (Allowed for all),
No user defined services and IP pool groups

(Blocked for all),

Bizfon4000 (SW Version 3.1.x) 93

Administrator's Manual Appendix: System Default Values

Extension Settings

Parameter System Default Value

Maximal mail message duration - 5 min,
Send end of greeting message – disabled,
Ask password before granting local access to mail box – disabled,
Ask password before granting remote access to mail box – enabled,
Send welcome message – disabled,
Play Voice Mail help – enabled,
Automatically play messages - enabled,
Send mails count information message – disabled,

Voice Mail Settings

Group List No entries

Send date/time information message – enabled,
Send beep at the end of message – enabled,
Send new voice messages via e-mail – disabled,
Send new voice message notifications via SMS – disabled,
Send new voice message notifications via phone call – disabled,
Voice Mail Indication - Tone indication,
Zero Out – enabled, to 00 default Attendant,
FAX Redirection – disabled,
Out of Office – disabled,
Greeting message – default.

Speed Calling No entries

Display Name – undefined,

Account Settings

Caller ID Based Services

Basic Services - General

Basic Services - Hold Music

Basic Services - Do Not Disturb

Basic Services - Hotline Disabled

User Password Protection – disabled both for incoming and outgoing calls,
User’s Name for Extensions Directory – default,
Custom Voice Messages – default.

No entries in the table.
For Any Callers – all services disabled,
Blocking Voice Messages - default

No answer timeout – 20 sec,
Call Waiting Service – enabled,
Autoredial Interval - 10 sec,
Autoredial Period - 15 min.

Send Hold Music to remote party – disabled,
Hold Music - Own Music.
Music file – default

Disabled.
Timeout - 30 min,
Send message to Caller Party – enabled.

Bizfon4000 (SW Version 3.1.x) 94

Bizfon Manual II: Administrator's Guide About this Administrator's Guide

Appendix: Software License Agreement

BIZFON Inc.

Software License Agreement

CAREFULLY READ ALL THE TERMS AND CONDITIONS CONTAINED IN THIS AGREEMENT.
USE OF THE BIZFON HARDWARE AND OPERATIONAL SOFTWARE PROGRAM INDICATES
YOUR ACCEPTANCE OF THESE TERMS AND CONDITIONS. IF YOU DO NOT AGREE TO
THESE TERMS AND CONDITIONS, YOU MAY NOT USE THE HARDWARE OR SOFTWARE.

1. License.

documentation for the software and such revisions for the software and documentation as the Licensor may make available to you from time to
time (collectively, the "Licensed Materials"). You may use the Licensed Materials only in connection with your operation of your Bizfon. You may
not use, copy, modify or transfer the Licensed Materials, in whole or in part, except as expressly provided for by this Agreement.

2. Ownership.

of this Agreement. The Licensor, however, retains sole and exclusive title to, and ownership of, the Licensed Materials, regardless of the form
or media in or on which the original Licensed Materials and other copies may exist. You acknowledge that the Licensed Materials are not your
property and understand that any and all use and/or the transfer of the Licensed Materials is subject to the terms of this Agreement.

3. Term.

Agreement or you transfer possession of the Licensed Materials to a third party in violation of this Agreement. You agree that upon such
termination, you will return the Licensed Materials to the Licensor, at its request.

4. No Unauthorized Copying or Modification.

secrets of the Licensor. Unauthorized copying, modification or reproduction of the Licensed Materials is expressly forbidden. Further, you may
not reverse engineer, decompile, disassemble or electronically transfer the Licensed Materials, or translate the Licensed Materials into another
language under penalty of law.

5. Transfer.

Gateway product. If you sell your license rights in the Licensed Materials you must at the same time transfer the documentation to the acquirer.
Also, you cannot sell your license rights in the Licensed Materials to another party unless that party also agrees to the terms and conditions of
this Agreement. Except as expressly permitted by this section, you may not transfer the Licensed Materials to a third party.

6. Protection And Security.

the Licensed Materials or any part thereof to any person other than the Licensor or its employees, without the prior written consent of the
Licensor. You agree to use your best efforts and take all reasonable steps to safeguard the Licensed Materials to ensure that no unauthorized
person shall have access thereto and that no unauthorized copy, publication, disclosure or distribution thereof, in whole or in part, in any form,
shall be made.

7. Limited Warranty.

Materials are recorded will be free from defects in materials and workmanship under normal use for a period of one (1) year from the date of
purchase (the "Warranty Period"). If you determine within the Warranty Period that the media on which the Licensed Materials are recorded are
defective, the Licensor will replace the media without charge, as long as the original media are returned to the Licensor, with satisfactory proof
of purchase and date of purchase, within the Warranty Period. This warranty is limited to you as the licensee and is not transferable. The
foregoing warranty does not extend to any Licensed Materials that have been damaged as a result of accident, misuse or abuse.

EXCEPT FOR THE LIMITED WARRANTY DESCRIBED ABOVE, THE LICENSED MATERIALS ARE PROVIDED ON AN "AS IS" BASIS.
EXCEPT AS DESCRIBED ABOVE, THE LICENSOR MAKES NO REPRESENTATIONS OR WARRANTIES THAT THE LICENSED
MATERIALS ARE, OR WILL BE, FREE FROM ERRORS, DEFECTS, OMISSIONS, INACCURACIES, FAILURES, DELAYS OR
INTERRUPTIONS INCLUDING, WITHOUT LIMITATION, TO ANY IMPLIED WARRANTIES OF MERCHANTABILITY, FITNESS FOR A
PARTICULAR PURPOSE, LACK OF VIRUSES AND ACCURACY OR COMPLETENESS OF RESPONSES, CORRESPONDENCE TO
DESCRIPTION OR NON-INFRINGEMENT. THE ENTIRE RISK ARISING OUT OF THE USE OR PERFORMANCE OF THE LICENSED
MATERIALS REMAINS WITH YOU.

8. LIMITATION OF LIABILITY AND REMEDIES.

INVOLVED IN THE CREATION, PRODUCTION OR DELIVERY OF THE LICENSED MATERIALS BE LIABLE FOR ANY CONSEQUENTIAL,
INCIDENTAL, DIRECT, INDIRECT, SPECIAL, PUNITIVE OR OTHER DAMAGES, INCLUDING, WITHOUT LIMITATION, LOSS OF DATA,
LOSS OF BUSINESS PROFITS, BUSINESS INTERRUPTION, LOSS OF BUSINESS INFORMATION, OR OTHER PECUNIARY LOSS,
ARISING OUT OF THE USE OF OR INABILITY TO USE THE LICENSED MATERIALS, EVEN IF THE LICENSOR OR SUCH OTHER PARTY
HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. YOU AGREE THAT YOUR EXCLUSIVE REMEDIES, AND THE
LICENSOR'S OR SUCH OTHER PARTY'S ENTIRE LIABILITY WITH RESPECT TO THE LICENSED MATERIALS, SHALL BE AS SET
FORTH HEREIN, AND IN NO EVENT SHALL THE LICENSOR'S OR SUCH OTHER PARTY'S LIABILITY FOR ANY DAMAGES OR LOSS TO
YOU EXCEED THE LICENSE FEE PAID FOR THE LICENSE MATERIALS.

The foregoing limitation, exclusion and disclaimers apply to the maximum extent permitted by applicable law.

9. Compliance With Laws.

domestic or foreign law. You are responsible for compliance with all domestic and foreign laws governing Voice over Internet Protocol (VoIP)

This license is effective until terminated. This license will terminate if you fail to comply with any terms or conditions of this

Bizfon, Inc. (the "Licensor"), hereby grants to you a non-exclusive right to use the Bizfon Operational Software program, the

By paying the purchase price for the Licensed Materials, you are entitled to use the Licensed Materials according to the terms

You may sell your license rights in the Licensed Materials to another party that also acquires your Bizfon4000 or any Bizfon SIP

Except as permitted under Section 5 of this Agreement, you agree not to deliver or otherwise make available

The only warranty the Licensor makes to you in connection with this license is that the media on which the Licensed

You may not use the Licensed Materials for any illegal purpose or in any manner that violates applicable

THIS IS A CONTRACT.

The Licensed Materials are copyrighted and contain proprietary information and trade

IN NO EVENT SHALL THE LICENSOR OR ANY OTHER PARTY WHO HAS BEEN

Bizfon Manual II: Administrator's Guide About this Administrator's Guide

calls.

10. U.S. Government Restricted Rights.

the Government is subject to restrictions as set forth in subparagraphs (c)(1) and (2) of the Commercial Computer Software—Restricted Rights
clause at 48 C.F.R. section 52.227-19, or subparagraph (c)(1)(ii) of the Rights in Technical Data and Computer Software clause at DFARS

252.227.7013, as applicable.

11. Entire Agreement.

complete and exclusive agreement between you and the Licensor and supersede any proposal or prior agreement or license, oral or written,
and any other communications related to the subject matter hereof. If one or more of the provisions of this Agreement is found to be illegal or
unenforceable, this Agreement shall not be rendered inoperative but the remaining provisions shall continue in full force and effect.

12. No Waiver.

in no way be considered to be a waiver of such provisions or rights, or to in any way affect the validity of this Agreement. If one or more of the
provisions contained in this Agreement are found to be invalid or unenforceable in any respect, the validity and enforceability of the remaining
provisions shall not be affected.

13. Governing Law.

choice of law provisions that would cause the application of the law of another jurisdiction.

14. Attorneys' Fees.

any such litigation or other dispute shall be entitled to, in addition to any other damages assessed, its reasonable attorneys’ fees, and all other
costs and expenses incurred in connection with settling or resolving such dispute.

If you have any questions about this Agreement, please write to Bizfon at 50 Stiles Road, Salem, NH 03079 or call Bizfon at [phone] or
[phone].

It is understood that this Agreement, along with the Bizfon Installation Guide and User’s Manual, constitute the

Failure by either you or the Licensor to enforce any of the provisions of this Agreement or any rights with respect hereto shall

This Agreement shall be governed by and construed in accordance with the laws of the state of Texas, without regard to

In the event of any litigation or other dispute arising as a result of or by reason of this Agreement, the prevailing party in

The Licensed Materials are provided with RESTRICTED RIGHTS. Use, duplication or disclosure by