Tau 2m-Ip User-Manual 1.13.1 en

TAU-2M.
IP
Operation Manual
version 1.4 (05.2016)
VoIP Gateway
IP-address: http://192.168.1.1
username: admin
password: password
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Document version
Version 1.4
Version 1.3
Issue data
23.05.2016
03.02.2016
Version 1.2
17.03.2015
Version 1.1
10.11.2014
Version 1.0
Firmware version
Revisions
Updated according to firmware version 1.13.1
Updated according to 1.13.0
Added:
- 3.6.2.7 filter submenu
Changed:
- 3.5.1 Internet
- 3.6.2.1 Internet submenu
- 3.6.2.2 address settings submenu
- 3.6.2.3 DHCP-server submenu
- 3.6.2.8 Routing submenu
- 3.6.3.2 Line configuration submenu
- 3.6.3.3 Profiles submenu
- 3.6.5.1 Time submenu
- 3.6.5.8 Autoconfiguration submenu
- 3.7.2 VoIP submenu
- 3.7.5 ARP submenu
- 4.1 Call transfer
Updated according to firmware version1.12.0
Added:
- 3.6.3.9 Call history menu
- 3.6.5.9VLAN managment submenu
- 3.7.9 Call history submenu
- APPENDIX B. Running user-defined script upon system startup
Changed:
- 3.6.2.5 NAT and port forwarding submenu
- 3.6.3.1 Network configuration submenu
- 3.6.3.2 Line configuration submenu
- 3.6.3.6 VAS management prefix submenu
- 3.6.5.2 Access submenu
- 3.6.5.5 Configuration management submenu
- 3.6.5.8 Autoconfiguration submenu
- 3.6.5.10 Additional settings submenu
- 3.7.1 Internet submenu
- 3.7.2 VoIP submenu
- 6 Device automatic update algorithm based on DHCP
Updated according to firmware version 1.10.1
Added:
- 3.6.3.6 Call signal submenu
Changed:
- 3.6.3.1 Network configuration submenu
- 3.6.6.1 Time submenu
27.02.2014
First issue
Firmware version: 1.13.1
Web-interface version: 1.13.80
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SYMBOLS
Symbol
Bold font face

Calibri Italic
Description
Notes, warnings, chapter headings, titles and table titles are written in bold.
Important information is written in Calibri Italic.
Notes and warnings

Notes contain important information, tips or recommendations on device operation
and setup.
Warnings are used to inform the user about harmful situations for the device and
the user alike, which could cause malfunction or data loss.
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CONTENTS
1 INTRODUCTION ................................................................................................................................................ 6
2 PRODUCT DESCRIPTION .................................................................................................................................... 7
2.1 Purpose ..................................................................................................................................................... 7
2.2 Device specification ................................................................................................................................... 7
2.3 Device design and Operating Principle ....................................................................................................... 9
2.4 Main specifications .................................................................................................................................. 11
2.5 Design ..................................................................................................................................................... 12
2.5.1 Top panel of the device ................................................................................................................... 12
2.5.2 Rear panel of the device.................................................................................................................. 12
2.6 Light indication ........................................................................................................................................ 13
2.7 Reset to factory defaults.......................................................................................................................... 14
2.8 Delivery Package ..................................................................................................................................... 14
3 DEVICE MANAGEMENT VIA WEB CONFIGURATOR .......................................................................................... 15
3.1 Getting started ........................................................................................................................................ 15
3.2 Changing Users ........................................................................................................................................ 15
3.3 WEB-interface operation mode ............................................................................................................... 16
3.4 Applying and Discarding Changes Made to Configuration ........................................................................ 17
3.4.1 Applying configuration .................................................................................................................... 17
3.4.2 Discarding changes ......................................................................................................................... 17
3.5 Quick Configuration Menu ....................................................................................................................... 18
3.5.1 Internet .......................................................................................................................................... 18
3.5.2 VoIP ................................................................................................................................................ 21
3.5.3 IPTV ................................................................................................................................................ 21
3.5.4 System ............................................................................................................................................ 22
3.6 Advanced settings ................................................................................................................................... 23
3.6.1 WEB interface basic element........................................................................................................... 23
3.6.2 Network menu .............................................................................................................................. 24
3.6.2.1 Internet submenu ...................................................................................................................... 24
3.6.2.2 'MAC managment' submenu ........................................................................................................ 32
3.6.2.3 DHCP server submenu ................................................................................................................ 33
3.6.2.4 Local DNS submenu.................................................................................................................... 34
3.6.2.5 NAT and port forwarding submenu ............................................................................................ 35
3.6.2.6 Firewall submenu ....................................................................................................................... 36
3.6.2.7 filter submenu ................................................................................................................... 38
3.6.2.8 Static routes submenu ............................................................................................................... 38
3.6.2.9 Dynamic DNS submenu .............................................................................................................. 39
3.6.2.10 SNMP submenu........................................................................................................................ 40
3.6.3 VoIP Menu .................................................................................................................................... 41
3.6.3.1 Network settings submenu ........................................................................................................ 41
3.6.3.2 Line settings submenu................................................................................................................ 43
3.6.3.3 Profiles submenu ....................................................................................................................... 47
3.6.3.4 Hunt group submenu ................................................................................................................ 60
3.6.3.5 QoS submenu............................................................................................................................. 62
3.6.3.6 Supplementary service prefixes submenu.................................................................................. 62
3.6.3.7 Pickup group submenu ............................................................................................................... 64
3.6.3.8 Cadence submenu ...................................................................................................................... 64
3.6.3.9 Call history menu ....................................................................................................................... 66
3.6.4 IPTV menu ...................................................................................................................................... 67
3.6.4.1 IPTV submenu ............................................................................................................................ 67
3.6.5 System Menu ................................................................................................................................ 69
3.6.5.1 Time submenu ........................................................................................................................... 69
3.6.5.2 Access submenu......................................................................................................................... 70
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3.6.5.3 Log submenu .............................................................................................................................. 73
3.6.5.4 'Passwords' submenu ................................................................................................................... 75
3.6.5.5 'Configuration management' submenu ......................................................................................... 76
3.6.5.6 'Firmware update' submenu ......................................................................................................... 76
3.6.5.7 Reboot submenu ....................................................................................................................... 77
3.6.5.8 Autoprovisioning submenu ......................................................................................................... 78
3.6.5.9 'Management interface' submenu ................................................................................................ 81
3.6.5.10 Advanced submenu .................................................................................................................. 82
3.7 System monitoring................................................................................................................................... 83
3.7.1 Internet submenu ......................................................................................................................... 83
3.7.2 'VoIP' submenu ............................................................................................................................... 84
3.7.3 'Ethernet ports' submenu ................................................................................................................ 87
3.7.4 DHCP submenu ............................................................................................................................. 88
3.7.5 ARP submenu ................................................................................................................................ 88
3.7.6 Device submenu ............................................................................................................................ 89
3.7.7 Conntrack submenu ...................................................................................................................... 89
3.7.8 Static routes submenu................................................................................................................... 90
3.7.9 'Call history' submenu ..................................................................................................................... 91
3.8 Configuration example............................................................................................................................. 93
4 VALUE ADDED SERVICE USAGE........................................................................................................................ 97
4.1 Call transfer ............................................................................................................................................. 97
4.2 Call Waiting ........................................................................................................................................... 100
4.3 Three-way conference call ..................................................................................................................... 100
4.3.1 Local conference ........................................................................................................................... 100
4.3.2 Remote conference ....................................................................................................................... 101
5 CONNECTION ESTABLISHMENT ALGORITHM ................................................................................................. 103
5.1 Algorithm of a Successful Call via SIP Protocol ........................................................................................ 103
5.2 Call Algorithm involving SIP Proxy server................................................................................................ 104
5.3 Call Algorithm Involving Forwarding Server ............................................................................................ 105
6 DEVICE AUTOMATIC UPDATE ALGORITHM BASED ON DHCP ......................................................................... 106
7 SYSTEM RECOVERY AFTER FIRMWARE UPDATE FAILURE ............................................................................... 109
APPENDIX . CALCULATION OF PHONE LINE LENGTH ....................................................................................... 110
APPENDIX B. RUNNING USER-DEFINED SCRIPT UPON SYSTEM STARTUP .......................................................... 111
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1 INTRODUCTION
Today, VoIP is one of the most rapidly evolving communication services. TAU-2M.IP series gateways
(hereinafter the "device") are designed to provide VoIP services to the network clients.
TAU-2M.IP VoIP gateway with an integrated router allows to connect analog phones to packed-based data
networks accessible via Ethernet.
The device is intended for operation in home or small office (SMB) environment.
This operation manual describes intended use, key specifications, configuration, monitoring, and firmware
update for TAU-2M.IP VoIP gateways.
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2 PRODUCT DESCRIPTION
2.1 Purpose
TAU-2M.IP is a high-performance VoIP gateway with the full set of features which allow users to leverage
VoIP functionality.
TAU-2M.IP gateway allows connecting an analog phone or a fax modem to IP networks. With built-in
router, the device enables connection of local network equipment to broadband access network. Device may be
connected up to 2 PCs that will be able to access the Internet through integrated NAT/DHCP server features.
USB port allows for the external storage device, 3G/4G USB modem or Wi-Fi-adapter connection1.
2.2 Device specification

Interfaces:
FXS:
LAN:
WAN:
USB:
2 x RJ-11 ports;
1 x Ethernet RJ-45 10/100BASE-T port;
1 x Ethernet RJ-45 10/100BASE-T port;
1 x USB2.0 port.
Gateway is powered by external 12V DC adapter for 220V electrical networks.

Functions:
Network functions:
Operation in bridgeor router mode;

PPPoE support (PAP, SPAP and CHAP authorization, PPPoE compression);
PPTP support;
L2TP support;
Static IP address and DHCP support (DHCP-client on WAN side, DHCP-server on LAN side);
DNS support;
NAT support;
firewall;
NTP support;
QoS support (QoS via DSCP and 802.1P);
IPTV function support;

VoIP protocols: SIP;
Echo cancellation (G.168 recommendations);
Voice activity detection (VAD);
Comfort noise generator;
DTMF signals detection and generation;
DTMF transmission(INBAND, rfc2833, SIP INFO);
Fax transmission:
Wi-Fi adapters are not supported in the current firmware version.
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G.711A/G.711U;
T.38;
Operartion w/ and w/o SIP-server;

Value Added Service:
Call Hold;
Call Transfer;
Call Waiting;
Call Forward at Busy;
Call Forward at No answer;
Call Forward Unconditional;
DND (Do not desterb);
Caller ID: FSK, DTMF;
Hotline;
Group call;
Call Pickup;
Three-way conference call;
Flexible numbering scheme;
Firmware update via web-interface;

DHCP-based auto provisioning support;
TR-069;
Remote monitoring, configuration and setup: Web-interface, Telnet.
Fig. 1 shows TAU-2M.IP connection diagram.
Fig. 1 TAU-2M.IP operation diagram
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2.3 Device design and Operating Principle

TAU-2M.IP terminal includes the following subsystem:
Controller featuring:
Realtek RTL8972C highly-integrated System-on-a-Chip (SoC), including a CPU, 100 Mbits

switch with a built-in PHY, hardware L2/L3/L4 acceleration, USB 2.0 ports, PCI-E controller
and 8 PCM channels for VoIP application;
Flash memory 8M;
SDRAM 128;
2 x SLIC subscriber units;

1 x LAN port: RJ-45 10/100BASE-T;
WAN Ethernet module: RJ-45 10/100BASE-T;
USB Host port.
Design diagram for device is depicted in Fig. 2.
Fig. 2 TAU-2M.IP Design diagram

The device runs under Linux operating system. Basic control functions are performed by Realtek processor
which enables IP packet routing, VoIP operation, group traffic proxing and etc.
The device may functionally be divided into four block:
Support block of the device network functions;

VoIP block;
Multicast-traffic processing block;
Control block (Linux operating system).
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Support block of the device network functions enables IP packet transmission and switching according to
the device routing table; depending on the network interface configuration, can process both tagged and
untagged frames. Supports DHCP, PPPoE, PPTP, L2TP.
VoIP block enables SIP protocol operation for transmission of voice signals through the network that
features packet switching. Subscriber's voice signal is transferred to SLIC subscriber unit module to be digitized.
Digitized signal is transferred to VoIP block to be encoded using one of the selected standards and is transferred
further in the form of digital packets to the controller via the intrasystem backbone. In addition to voice signals,
digital packets contain control and interaction signals.
Multicast traffic processing block enables processing of IGMP messages and multicast traffic for IPTV
functions support.
Control block based on Linux operating system monitors operation of all the other blocks and the device
subsystems and manages their interaction.
Fig. 3 shows TAU-2M.IP functional diagram.
Fig. 3 TAU-2M.IP series functional diagram
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2.4 Main specifications

Table 2 lists main specification of the device.
Table 1 Main specifications
VoIP Protocols
Supported protocols
SIP
Audio codecs
Codecs
G.729, annex A, annex B

G.711, G.711u,
G.723.1
Modem transmission: G.711, G.711u
Fax transmission: G.711, G.711u, T.38
Ethernet WAN interface specifications

Number of ports
Electric port
Data rate, Mbps
Supported standards
1
RJ-45
10/100, autodetection
BASE-T
Ethernet LAN-interface specification

Number of interface
Electric port
Data rate, Mbps
Supported standards
1
RJ-45
10/100, autodetection
BASE-T
Analogue user ports example

Number of ports
Loop resistance (phone resistance excluded)
Dialling reception
Subscriber terminal protection
Caller ID broadcasting
2
Up to 80
pulse/frequency (DTMF)
current and voltage
FSK BELL202/FSK V.23/DTMF
Control
Remote control
Access restriction
Web-interface, Telnet, SSH, SNMP, TR-069

password
General parameters
power supply
power consumption
Operating temperature range
relative humidity at 25
Dimensions
Weight
power adapter 12 V DC, 1.5

7.2 W max (max. current consumption is 0.6 )
from +5 to +40
Up to 80%
122x96x33 mm
0,15 kg.
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2.5 Design
TAU-2M.IP terminal is enclosed into 1229633 mm plastic case.
2.5.1 Top panel of the device

Fig. 4 shows TAU-2M.IP top panel layout.
Fig. 4 TAU-2M.IP top panel layout

TAU-2M.IP top panel is equipped with LED indicators, see Table 2.
Table 2 Description of LEDs and controls located on the front panel
Front panel element

1
Power
USB
Description
Device power and activity status indicator
External USB device activity indicator (USB flash, external HDD, 3G/4G USBmodem)
Analog phone indicator
LAN
LAN-interface indicator
WAN
WAN-interface indicator
2.5.2 Rear panel of the device

Fig. 5 shows TAU-2M.IP rear panel layout.
.
Fig. 5 TAU-2M.IP rear panel layout
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TAU-2M.IP rear panel is equipped with the following connectors and controls, see Table 3.
Table 3 Description of LEDs and controls located on TAU-2M.IP rear panel
Rear panel element
Description
12V
Connector for power adapter
USB
USB port for external USB device connection (USB flash, HDD, 3G/4G USBmodem)
Phone
2 x RJ-11 port for analogue unit connection
LAN
10/100BASE-T Ethernet (RJ-45) port for local device network connection
10
WAN
10/100BASE-T (RJ-45) port for external network connection
2.6 Light indication

WAN, LAN, Phone, Power indicators located on TAU-2M.IP top panel show the device current status.
Table 5 lists possible states of the LEDs.
Table 4 Light indication of TAU-2M.IP series states
LED
WAN
LAN
LED state
Solid (green 10 Mbps, orange
100Mbps)
Flashes
solid (green 10 Mbps, orange
100Mbps
Flashes
green, solid
off
Green , flashes with 20 Hz frequency
for 1 second, then 4 seconds pause
Green flashes slowly in period
USB
Power
Green, double short flashes in 3

seconds intervals
Green, solid
off
green solid
orange solid
Red solid
Flashes red and green intermittently in
periods
Device state
Connection between station terminal and
subscriber device is established
Transferring data packets via WAN interface
Connection to the network device is established
Transferring data packets via LAN-interface
Phone is off-hook (line is active)
Phone is on-hook, normal operation
incoming call is on the phone port
Subscriber port is not registered at SIP proxy
server
Line test is in progress
USB-device is connected
USB-device is not connected
Device power is on, normal operatipon
Internet is not accesible
Device start up
Device is being reset to factory defaults
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2.7 Reset to factory defaults

In order to reset the device to factory settings, press the 'F' button located on the device side panel when
the device is powered up and hold it until the 'Power' indicator begins to flash red and green intermittently in
periods. Device will be rebooted automatically. Factory settings: DHCP client is launched on WAN interface, LAN
interface address192.168.1.1, subnet mask255.255.255.0;username/password for web interface access:
admin/password.
2.8 Delivery Package

TAU-2M.IP series standard delivery package includes:
Multi-purpose terminal;
220/12 V 1.5 power adapter;
Installation and configuration guide.
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3 DEVICE MANAGEMENT VIA WEB CONFIGURATOR

3.1 Getting started
In order to start the operation, you should connect to the device via LAN interface using a web browser:
1. Open a web browser (hypertext document viewer) such as Firefox, Opera or Chrome.
2. Enter the device IP address in the browser address bar
Factory default IP address: 192.168.1.1, subnet mask: 255.255.255.0
When the device is successfully detected, username and password request page will be shown in the
browser window.
3. Enter
your
username
into
'Login'
field
and
password
into
'Password'
field.
Factory settings: login: admin, password: password.
4. Click 'Log in button. Quick configuration menu will be shown in the browser window, see Fig. 6.
3.2 Changing Users

There are three user types for the device: admin, user, and viewer. Admin (administrator, default
password: password) has the full access to the device: read/write any settings, full device status monitoring.
User (non-privileged user , default password: user) may configure PPPoE in order to connect to the Internet and
configure Wi-Fi, may not access the device status monitoring. Viewer (spectator, default password: viewer) may
only view full device configuration without editing privileges; may access full device status monitoring.
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When you click Logout button, the current user session will be terminated; login window will be
displayed:
To change the access, you should specify the corresponding username and password and click Log in
button.
3.3 WEB-interface operation mode

TAU-2M.IP WEB-interface of the device can operate in three modes:
Monitoring system monitoring modeallows you to view various device operation

information: Internet connection activity, phone port status, amount of received/sent
data via network interfaces, etc.;
Tiles quick system configuration modeeach tile contains settings grouped by their
function: Internet, VoIP, IPTV, and etc. A tile only displays basic parameters that allow the
quickest possible configuration of a specific device function.
Settings advanced system configuration mode (full configuration mode)enables full
device configuration.
To switch between WEB interface modes, use the panel located on the left hand side in WEB interface.
The panel will open, when you move mouse cursor on it:
To proceed from 'Tiles' mode into 'Settings', you may also click 'Details' link in the tile.
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3.4 Applying and Discarding Changes Made to Configuration

3.4.1 Applying configuration
Click 'Apply' button to save the configuration into the device flash memory and apply new settings.
All settings will take effect without device restart.
'Apply' button in the quick configuration menu and the advanced settings menu will appear as follows:
;
.
WEB-interface visual indication of the current status of the setting application process, see Table 5.
Table 5 Visual indication of the current status of the setting application process
Appearence
Status description
When you click the 'Accept' button, settings will be applied and stored into the device
memory. This is indicated by the
icon in the tab name and on the 'Apply' button
Successful settings saving and application are indicated by
icon in the tab name.
If the parameter value being specified contains an error, you will see a message with the
reason description and the
icon will appear in the tab name, when you click 'Apply'
button.
3.4.2 Discarding changes

You may discard changes only until 'Apply' button is clicked. In this case, edited parameters on the
page will be updated with the values currently stored in the device memory. After you click
'Apply', you will not be able to restore previous settings.
Discard changes button in the quick configuration menu and the advanced settings menu will appear as
follows:
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3.5 Quick Configuration Menu

In the quick configuration menu, you will find basic device settings, see Fig. 6.
Fig. 6 Quick configuration menu

Settings are divided into the following categories:
Internet quick Internet access configuration;

VoIP quick VoIP configuration;
IPTV configure device to support IPTV features;
System configure access to web interface via WAN port.
3.5.1 Internet
In order to access the Internet, you should specify basic settings in the 'Internet' section. To specify
additional parameters, go to advanced settings mode by clicking the 'Details' link.
Work mode the device operation mode:
Router router mode is established between LAN and WAN (LAN is isolated from WAN );
Bridge bridge mode is established between WAN and LAN-interfaces: data is
transmitted transparently from LAN to WAN and back .
Protocol select the protocol that will be used for device WAN interface connection to provider
network:
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Static operation mode where IP address and all the necessary parameters for WAN
interface are assigned statically. When 'Static' type is selected, the following parameters will
be available for editing:
External IP addressspecify device WAN interface IP address in the provider

network;
Subnet maskexternal subnet mask;
Default gateway address where the packet will be sent to, when route for it is
not found in the routing table;
Primary DNS, Secondary DNSdomain name server addresses (allows to identify
the IP address of the device by its domain name). You may leave these fields
empty, if they are not required.
DHCP operation mode where IP address, subnet mask, DNS address, default gateway and
other necessary settings for network operation are automatically obtained from DHCP server.
Supported options:
1 network mask;
3 default network gateway address;
6 DNS address;
12 device network name;
28 network broadcast address;
33 - static routes;
42 NTP server address;
43 specific vendor information;
60 alternative Vendor ID;
66 TFTP server address;
67 firmware file name (to download via TFTP from the server specified in
Option 66)
121 classless static routes.
In Option 60 DHCP request, the device will send vendor information in the following format:
[VENDOR: device dvendor][DEVICE: device type][HW: Hardware version] [SN: serial
number][WAN: WAN interface - address][LAN: LAN interface MAC address][VERSION:
firmware version]
Example:
[VENDOR:Eltex][DEVICE:TAU-2M.IP][HW:1.0][SN:VI23000118][WAN:A8:F9:4B:03:2A:D0]
[LAN:02:20:80:a8:f9:4b][VERSION:#1.12.0]
PPPoE operation mode when PPP session is established on WAN interface. When 'PPPoE' is
selected, the following parameters will be available for editing:
Username username for authorization on PPP server;
Password password for authorization on PPP server;
Service-Name Service-Name tag value in PADI message for PPPoE connection
initialization (this parameter is optional, and configured only on the provider's
request);
Secondary access type of access to local area network resources.
You may select 2 options:
DHCP dynamic access when IP address and other required parameters are
obtained via DHCP;
Static in this case, you should specify PPTP server access settings manually: IPaddress, Subnet mask, DNS, Gateway.
PPTP operation mode when the Internet access is established via a tunnel, using PPTP.
When 'PPTP' is selected, the following parameters will be available for editing:
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PPTP-Server PPTP server address (domain name or IP address in IPv4 format);

User name username for authorization on PPTP server;
Password password for authorization on PPTP server;
Secondary access type of access to local area network resources and PPTP
server.
DHCP dynamic access when IP address and other required parameters are
obtained via DHCP;
Static in this case, you should specify PPTP server access settings manually:
IP-address when the static access is used, PPTP server will be accessed from
this address;
Subnet musk subnet mask for static access;
DNS when the static access is used, local area network DNS;
Gatewaywhen the static access is used, gateway for PPTP server access (if
necessary).
L2TP operation mode when the Internet access is established via a tunnel, using L2TP.
When 'L2TP' is selected, the following parameters will be available for editing:
L2TP server L2TP server address (domain name or IP address in IPv4 format);
Usernameusername for authorization on L2TP server;
Passwordpassword for authorization on L2TP server;
Secondary accesstype of access to local area network resources and L2TP
server.
DHCPdynamic access when IP address and other required parameters are
obtained via DHCP.
Staticin this case, you should specify L2TP server access settings manually:
IP addresswhen the static access is used, PPTP server will be accessed from
this address.
Netmasksubnet mask for static access.
DNSwhen the static access is used, local area network DNS.
Gatewaywhen the static access is used, gateway for L2TP server access (if
necessary).
PPTP and L2TP allow establishing secure communication link over the Internet
between the remote user's computer and organization's private network. PPTP and
L2TP are based on Point-to-Point Protocol (PPP) and act as its extension. First, the OSI
model higher level data is encapsulated into PPP, and then into PPTP or L2TP for
tunnel transmission via public data networks. PPTP and L2TP functionality differs.
L2TP may be used not only in IP networks, service messages for tunnel creation and
data transfer use the same format and protocols. PPTP may be used only in IP
networks, it requires a dedicated TCP connection for tunnel creation and usage. L2TP
over IPSec1 allows for the higher security level compared to PPTP and provides the
higher level of protection for business-critical data.
Due to its characteristics, L2TP is an attractive protocol for building virtual networks.
To apply a new configuration and store settings into the flash memory, click
changes, click
button. To discard
button.
To connect the device to the provider network, you should request the network settings from the
provider. If you use the static settings, select 'Static' value in the 'Protocol' field and fill the 'External IP address',
'Subnet mask', 'Default gateway', 'Primary DNS', and 'Secondary DNS' fields with the corresponding values
obtained from the provider. If devices in the provider network obtain network settings via DHCP, PPPoE, PPTP,
IPSec is not supported in the current firmware version.
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or L2TPselect the corresponding protocol in the 'Protocol' field and refer to provider's instructions to achieve
complete and correct device configuration.
3.5.2 VoIP
For VoIP operation, you should specify settings in the 'VoIP' section. To specify additional parameters, go
to advanced settings mode by clicking the 'Details' link.
In the Line 1 and Line 2 tabs you may configure the device phone ports respectivly:
Enablewhen selested, the current line is active;

Numbersubscriber number, assigned for this phone line;
User nameuser name for authentication on SIP server;
Passwordpassword for authentication on SIP server;
In SIP tab, you may configure basic settings for SIP proxy server:
SIP proxy servernetwork address of a SIP serverdevice that manages access to provider's
phone network for all subscribers. You may specify IP address as well as the domain name (specify
an alternative SIP server UDP port after the colon, default value is 5060);
Registrationwhen selected, subscriber port registration will be enabled on the registration
server;
Registration servernetwork address of a device that is used for registration of all phone network
subscribers in order to provide them with the communication services (specify an alternative
registration server port after the colon, default value is 5060). You may specify IP address as well
as the domain name. Usually, registration server is physically co-located with SIP proxy server
(they have the same address);
SIP domaindomain where the device is located (fill in, if required).
changes, click
. button. To discard
button.
3.5.3 IPTV
For IPTV operation, you should specify settings in 'IPTV' section. To specify additional parameters, go to
advanced settings mode by clicking the 'Details' link.
Enable IPTVwhen selected, enables IPTV packet transmission from TAU-2M.IP WAN interface
(from the provider network) to the devices connected to LAN interface.
Enable HTTP proxywhen selected, use HTTP proxy. HTTP proxy transforms UDP stream into
HTTP stream in order to improve stream image quality, when the quality of the communication
link in local area network is low.
HTTP portHTTP proxy port number that will be used for video streaming. Use this port to
connect to IPTV streams being broadcast by the device.
For example, if the device address on LAN interface is 192.168.0.1, proxy server port is 2354, and the desired
channel 227.50.50.100 is being broadcast to UDP port 1234, you should specify the following stream address for
VLC application: http://@192.168.0.1:2345/udp/227.50.50.100:1234.
changes,
button. To discard
button.
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3.5.4 System
In the 'System' section, you may configure access to the device web configurator. To specify additional
parameters, go to advanced settings mode by clicking the 'Details' link.
Access to Web via WAN:
HTTPwhen selected, WAN port connection to the device web configurator is enabled via HTTP
(insecure connection);
HTTPSwhen selected, WAN port connection to the device web configurator is enabled via
HTTPS (secure connection).
By default, access to the device Web interface is enabled only for LAN interface.
changes, click
button. To discard
button.
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3.6 Advanced settings

To proceed to the advanced settings mode, click 'Details' link in any tile or select 'Settings' item on the left
panel.
3.6.1 WEB interface basic element

Fig. 7 shows WEB configurator basic navigation elements in the advanced settings mode.
Fig. 7 Web configurator navigation elements

User interface window is divided into 7 areas:
1. Logged in user name and session termination button in the WEB interface ('Sign Out') for the
current user.
2. Changing the user interface language. One of two languages is available: Russian (ru) or English
(eng).
3. Menu tabs include submenu tabs grouped by a category: Network, VoIP, IPTV, System.
4. Submenu tabs allow you to control settings field.
5. WEB configurator mode changing panel (to get more detailes, see Chapter 3.3 WEB-interface
operation mode).
6. Device settings field based on the user selection; allows you to view device settings and enter
configuration data.
7. Configuration management buttons; for detailed description see Chapter 3.4 Applying and
Discarding Changes Made to Configuration.
Applyapply and save the current configuration into flash memory of the the device.
Discarddiscard changes (effective only until 'Apply' button is clicked).
8. Informational field showing firmware version and WEB interface version.
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3.6.2 Network menu
In the 'Network' menu, you may configure device network settings.
3.6.2.1 Internet submenu

In the 'Internet' submenu, you may configure an external network (via PPPoE, DHCP, PPTP, L2TP,
statically, in the router or bridge mode) and LAN.
Common Settings
Hostnamedevice network name.
WAN
Internet connectionexternal network connection method for the device:
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Wired connectionconnection to the Internet is established using Ethernet cable via WAN
port only;
3G/4G USB modemconnection to the Internet is established using 3G/4G USB modem (via
cellular data network), connected to the USB port of the device.
Speed and duplexspecify data rate and duplex mode for WAN Ethernet port of the gateway:
Autoautomatic speed and duplex negotiations;

100 Half100Mbps data transfer rate with half-duplex mode is supported;
100 Full100Mbps data transfer rate with duplex mode is supported;
10 Half10Mbps data transfer rate with half-duplex mode is supported;
10 Full10Mbps data transfer rate with duplex mode is supported.
Connection settings
When you choose 'Wired connection' method, the following connection settings will become available:
Work mode the device operation mode:
Routerswitch mode is established between LAN and WAN interfaces (LAN is isolated
from WAN);
Bridgebridge mode is established between LAN and WAN interfaces: data is transferred
transparently from LAN to WAN and backin fact, the device operates in the switch
mode.
When you choose 'Router' operation mode, the following connection settings will become available:
Protocolselect the protocol that will be used for device WAN interface connection to provider
network:
Staticoperation mode where IP address and all the necessary parameters for WAN
interface are assigned statically. When 'Static' type is selected, the following parameters
will be available for editing:
Device external IP addressspecify device WAN interface IP address in the

provider network;
Default gatewayaddress that the packet will be sent to, when route for it is not
found in the routing table;
Primary DNS, Secondary DNSdomain name server addresses (allows to identify
DHCPoperation mode where IP address, subnet mask, DNS address, default gateway
and other necessary settings for network operation are automatically obtained from DHCP
server.
Supported options:
1network mask
3default network gateway address
6DNS address
12device network name
28network broadcast address
33static routes
42NTP server address
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43specific vendor information
60alternative Vendor ID
66TFTP server address
67firmware file name (to download via TFTP from the server specified in
Option 66)
121classless static routes
For DHCP, you may specify the required value for Option 60.
Alternative Vendor ID (Option 60)when selected, the device transmits Vendor
ID (Option 60) field value in Option 60 DHCP messages (Vendor class ID). If the
field is empty, Option 60 will not be transmitted in DHCP messages.
If the Alternative Vendor ID (Option 60) checkbox is not selected, the default
value will be transmitted in Option 60 in the following format:
[VENDOR:device vendor][DEVICE:device type][HW:hardware version] [SN:serial
number][WAN:WAN interface address][LAN:LAN interface
address][VERSION:firmware version]
Example:
[VENDOR:Eltex][DEVICE:TAU-2M.IP][HW:1.0][SN:VI23000118]
[WAN:A8:F9:4B:03:2A:D0][LAN:02:20:80:a8:f9:4b][VERSION:#1.12.0]
Primary DNS, Secondary DNSDNS IP addressesif DNS addresses are not obtained
automatically via DHCP, you should define them manually. Manually defined addresses
will have a priority over DNS addresses obtained via DHCP.
PPPoE operation mode when PPP session is established on WAN interface. When
'PPPoE' is selected, the following parameters will be available for editing:
Usernameusername for authorization on PPP server;
Passwordpassword for authorization;
MTUmaximum block size for data transmitted via the network (1492 is
recommended);
Service-NameService-Name tag value in PADI message (this field is optional);
Secondary accesstype of access to local area network resources;
obtained via DHCP;
Staticin this case, you should specify access settings manually:
IP address, Subnet mask, DNS, Gateway.
Use secondary access for VoIPthis option is available, if there are no dedicated
interfaces for VoIP service ('Use Internet settings' checkbox is selected). When
the checkbox is not selected (default value), VoIP service uses PPP interface for
its operation; when selected, secondary access interface (IPoE);
Hardware traffic accelerationincrease device bandwidth for PPP traffic (if PPP is
selected) or IPoE traffic (if Ethernet is selected) transmission depending on the
selected value;
PPTPoperation mode when the Internet access is established via a special channela
tunnelusing PPTP. When 'PPTP' is selected, the following parameters will be available for
editing:
PPTP serverPPTP server IP address;
Usernameusername for authorization on PPTP server;
Passwordpassword for authorization on PPTP server;
recommended);
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Secondary accesstype of access to local area network resources and PPTP

server.
obtained via DHCP.
Staticin this case, you should specify PPTP server access settings manually:
IP addresswhen the static access is used, PPTP server will be accessed from this
address;
Subnet masksubnet mask for static access;
DNSwhen the static access is used, local area network DNS;
Gatewaywhen the static access is used, gateway for PPTP server access (if
necessary);
its operation; when selected, secondary access interface (IPoE).
Hardware traffic acceleration works only for secondary access interface (IPoE).
L2TP operation mode when the Internet access is established via a special channela
tunnelusing L2TP. When 'L2TP' is selected, the following parameters will be available for
editing:
L2TP serverL2TP server IP address;
recommended);
L2TP serverL2TP server IP address;
recommended);
Secondary accesstype of access to local area network resources and L2TP
server.
obtained via DHCP.
Staticin this case, you should specify L2TP server access settings manually:
IP addresswhen the static access is used, PPTP server will be accessed from this
address;
Subnet masksubnet mask for static access;
DNSwhen the static access is used, local area network DNS;
Gatewaywhen the static access is used, gateway for L2TP server access (if
necessary);
its operation; when selected, secondary access interface (IPoE).
Hardware traffic acceleration works only for secondary access interface (IPoE).
PPTP and L2TP allow you to establish secure communication link over the Internet
between the remote user's computer and organization's private network. PPTP and L2TP are
based on Point-to-Point Protocol (PPP) and act as its extension. First, the OSI model higher
level data is encapsulated into PPP, and then into PPRP or L2TP for tunnel transmission via
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public data networks. PPTP and L2TP functionality differs. L2TP may be used not only in IP
networks, service messages for tunnel creation and data transfer use the same format and
protocols. PPTP may be used only in IP networks, it requires a dedicated TCP connection for
tunnel creation and usage. L2TP over IPSec1 allows for the higher security level compared to
PPTP and guarantees the higher level of protection for business-critical data.
Due to its characteristics, L2TP is an attractive protocol for building virtual networks.
Use VLAN in external networkwhen selected, use VLAN identifier specified in 'VLAN ID' field for
the Internet access.
VLAN IDVLAN identifier used for the service.

802.1P802.1P marker (another name: CoS Class of Service), assigned to the outgoing
frames from this interface. It may take values from 0 (the lowest priority) to 7 (the highest
priority).
VLANvirtual local area network. VLAN consist of a group of hosts combined into a single
network regardless of their location. Devices grouped into a single VLAN will have the same
VLAN-ID.
Disable source address transmissionwhen selected, disable source address substitution for
packets sent from LAN (disables 'masquerading').
When you choose 'Bridge' operation mode, the following connection settings will become available:
Protocolselect the protocol that will be used for device WAN interface connection to provider
network:
Staticoperation mode where IP address and all the necessary parameters for WAN
interface are assigned statically. When 'Static' type is selected, the following parameters
IP addressspecify device WAN interface IP address in the provider network.
Default gatewayaddress that the packet will be sent to, when route for it is not
found in the routing table;
Primary DNS,Secondary DNSdomain name server addresses (allows to identify
DHCPoperation mode where IP address, subnet mask, DNS address, default gateway
and other necessary settings for network operation are automatically obtained from
Alternative Vendor ID (Option 60) when selected, the device transmits Vendor ID
(Option 60) field value in Option 60 DHCP messages (Vendor class ID). If the field
is empty, Option 60 will not be transmitted in DHCP messages.If the Alternative
Vendor ID (Option 60) checkbox is not selected, the default value will be
transmitted in Option 60 in the following format:
address][VERSION:firmware version]
Example:
IPSec is not supported in the current firmware version.
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1st DNS, 2nd DNSDNS IP addressesif DNS addresses are not obtained
automatically via DHCP, you should define them manually. Manually defined
addresses will have a priority over DNS addresses obtained via DHCP.
When '3G/4G USB modem' connection method is selected, the following fields will become available for
configuration:
Mobile provider3G/4G service provider name. You may select one of the six mobile service
providers operating in Russian Federation (their settings are stored in the device memory):
Megafon, Beeline, MTS, Skylink, Tele2, Yota. Click 'Default' button to fill in the connection settings
with the selected service provider parameters. If the service provider settings in your region differ
from the proposed ones, edit them accordingly;
If your provider is missing from the list, select 'Other' and enter your service provider settings into
fields;
Protocolthis field is available only when 'Other' is selected in the mobile service providers list.
For most cases, mobile service providers establish network access using PPPoE, however some
modems may require DHCP for proper operation;
Usernameusername used for authentication in the wireless network;
Passwordpassword used for authentication in the wireless network;
Dial-up numberdial-up number for wireless network connection (e.g. *99***1#).
Additional
parametersparameters
for
wireless
network
connection
(e.g.
AT+CGDCONT=1,IP,internetfor Megafon); do not use quotation marks in this string;
MTUmaximum block size for data transmitted via the network (1492 is recommended).
'Default' button allows you to fill in the service provider settings with preconfigured values from the
device memory, to free the user from searching for them in the Internet.
Local area network:
Device IP addressdevice IP address in LAN;

Subnet masksubnet mask in LAN.
If the local subnet address is changed, the local DHCP server address pool (NetworkDHCP
Server) will be changed automatically.
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IPSec settings:
In this section, you may configure IPSec encryption (IP Security).
IPSec is a set of protocols used for protection of data transmitted via Internet Protocol that enables
authentication, integrity check and/or encryption of IP packets. IPsec also includes secure Internet Key Exchange
protocols.
In the current firmware version, you may only access the device management interfaces (Web, Telnet,
SSH) using IPSec.
Enableenable IPSec protocol utilization for data encryption.

Interfacethis setting takes effect only when PPPoE, PPTP or L2TP are selected for the Internet,
and defines the interface that will be accessed with IPSec: Ethernet (secondary access interface)
or PPP (primary access interface). When DHCP or Static protocol is selected, there is only a single
interface (Ethernet) active for the service that may be accessed with IPSec only.
Local IP addressdevice address for IPSec operation.
Local subnet address together with Local subnet mask define a local subnet for creation of
network-to-network or network-to-point topologies.
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Remote subnet address together with Remote subnet mask define a remote subnet address used
for IPSec-encrypted communication. If the mask value is 255.255.255.255, communication is
performed with a single host. Mask that differs from 255.255.255.255 allows you to define a
whole subnet. Thus, device features allow you to establish 4 network topologies that utilize IPSec
traffic encryption: Point-to-Point, Network-to-Point, Point-to-Network, Network-to-Network.
Remote gatewaygateway used for remote network access.
NAT-Traversal IPsec NAT-T mode selection. NAT-T (NAT Traversal) encapsulates IPSec traffic and
simultaneously creates UDP packets to be sent correctly by a NAT device. For this purpose, NAT-T
adds an additional UDP header before IPSec packet so it would be processed as an ordinary UDP
packet and the recipient host would not perform any integrity checks. When the packet arrives to
the destination, UDP header is removed and the packet goes further as an encapsulated
IPSecpacket. With NAT-T technique, you may establish communication between IPSec clients in
secured networks and public IPSec hosts via firewalls. NAT-T operation modes:
OnNAT-T mode is activated only when NAT is detected on the way to the destination
host;
Forceuse NAT-T in any case;
Offdisable NAT-T on connection establishment.
The following NAT-T settings are available:
NAT-T UDP portUDP port for packets for IPSec message encapsulation. Default value is
4500;
NAT-T keepalive packet transmission interval, secondsperiodic message transmission
interval for UDP connection keepalive on the device performing NAT functions.
Aggressive modephase 1 operation mode when all the necessary information is exchanged
using three unencrypted packets. In the main mode, the exchange process involves six
unencrypted packets.
My identifier typedevice identifier type: address, fqdn, keyed, user_fqdn, asn1dn.
My iIdentifierdevice identifier used for identification during phase 1 (fill in, if required).
Identifier format depends on the type.
Phase 1 During the first step (phase), two hosts negotiate on the identification method, encryption
algorithm, hash algorithm and Diffie Hellman group. Also, they identify each other. For phase 1, there are the
following settings:
Pre-shared keya secret key used by authentication algorithm in phase 1. A string from 8 to 63
characters long;
IKE authentication algorithmselectan authentication algorithm from the list: MD5, SHA1;
IKE encryption algorithmselect an encryption algorithm from the list: DES, 3DES, Blowfish;
Diffie Hellman groupselect Diffie-Hellman group;
IKE SA lifetime, secondstime that should pass for hosts' mutual re-identification and policy
comparison (other name 'IKE SA lifetime'). Default value is 24 hours (86400 seconds).
Phase 2 During the second step, key data is generated, hosts negotiate on the utilized policy. This mode
also called as 'quick mode'differs from the phase 1 in that it may be established after the first step only, when
all the phase 2 packets are encrypted.
IKE authentication algorithmselectan authentication algorithm from the list: HMAC - MD5,
HMAC-SHA1, DES, 3DES.
IKE encryption algorithmselect an encryption algorithm from the list: DES, 3DES, Blowfish.
Diffie Hellman groupselect Diffie-Hellman group.
IPSec SA lifetime, secondstime that should pass for data encryption key changeover (other name
'IPSec SA lifetime'). Default value is 60 minutes (3600 seconds).
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To apply a new configuration and store settings into the flash memory, click 'Apply' button. To discard
changes, click 'Cancel' button.
This IPSec implementation works only when the sender address translation is disabled, as the
client IP address is involved in the key formation.
3.6.2.2 'MAC managment' submenu

In the 'MAC managment' submenu, you may change the device WAN interface MAC address.
Redefine when selected, MAC address from the MAC field is used on the Internet interface.
When you click a drop-down menu button in the 'MAC' field, you may specify MAC address of the
computer connected to WEB configurator. This may be helpful, when your ISP network employs MAC address
tethering. In this case, if you are planning to use TAU-2M.IP as a router, MAC address of your computer
(previously connected to the Internet) should be assigned to the WAN interface of the device.
To redefine MAC for 'VoIP' or 'ManagementVLAN ' interface, see sections 'Set MAC address for interface
'VoIP'' or 'Set MAC address for interface 'ManagementVLAN ''.
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3.6.2.3 DHCP server submenu
In the DHCP server submenu, you may configure a local DHCP server and define static address bindings.
With DHCP (Dynamic Host Configuration Protocol), TAU-2M.IP may automatically assign IP addresses and
parameters required for the Internet access to computers connected to the LAN interface and wireless Wi-Fi
access point. DHCP eliminates limitations associated with the manual TCP/IP protocol configuration. DHCP
server is available for configuration only when the Internet service is configured in the switch mode.
Settings of DHCP server
Enabledwhen selected, enable local DHCP server.

Start IP addressstarting address in the IP address pool.
Pool sizenumber of addresses in the pool.
Lease time, minset the maximum time for IP address lease issued by DHCP server to the
connected device, in minutes.
When you change a starting address to a value from another subnet in relation to the LAN
interface subnet, the pool will be automatically adjusted in accordance to the current local
subnet address value.
Static leases
To add a new static binding, click 'Add' button and fill in the following fields:
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Current static binding name;

MAC addressspecify a static address. Format: XX:XX:XX:XX:XX:XX, you may select
connected device addresses from the pop-up menu;
IP addressdefine a static IP address for the specific MAC address.
Static binding configuration may become useful, if you have to assign a specific IP address to the specific
PC connected to the device LAN interface.
Click 'Apply' button to enter the IP address into the static IP address list for DHCP server. To discard
To remove the address from the list, select the checkbox next to the respective record and click 'Delete'.
3.6.2.4 Local DNS submenu

In 'Local DNS' submenu, you may configure a local DNS server by adding 'IP addressdomain name' pairs
into the database.
Local DNSallows the gateway to obtain IP address of the communicating device by its domain name.
You may use local DNS in cases when DNS server is missing from the network segment that the gateway belongs
to, and you need to establish routing using network names, or when you have to use SIP server network name
as its address. Although, you have to know matches between hostnames (domains) and their IP addresses.
Configuration of domain names

To add the address into the list, click 'Add' button in the 'New domain name' window and fill in the
following fields:
Domain namehost name;

IP addresshost IP address.
Click 'Apply' to create 'IP addressdomain name' pair. To discard changes, click 'Cancel' button. To
remove the record from the list, select the checkbox next to the respective record and click 'Delete'.
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3.6.2.5 NAT and port forwarding submenu
In the 'NAT and port forwarding' submenu, you may configure port forwarding from WAN interface to LAN
interface. This submenu is available only when the Internet service is configured in the switch mode.
NAT (Network Address Translation) allows you to translate IP packet address and network port. Port
forwarding is required when TCP/UDP connection to a local computer (connected to LAN interface) is
established from the WAN In this settings menu, you may define the rules allowing packets to pass from the
external network to the specified address in the local network and thus enabling connection. In general, port
forwarding is necessary for torrent and P2P service operation. For this purpose, you should identify TCP/UDP
ports used by a torrent or p2p client in their settings and assign the respective forwarding rules for your
computer IP address.
Configuration of NAT rules

To add a new NAT rule, click 'Add' button and fill in the following fields in the 'Add a new rule' window:
Namename of the rule (this field is mandatory);

LAN IP addressIP address of the host in LAN used for packet translation falling under this rule;
Destination ports of LAN recipient TCP/UDP port values that will be used for packet translation
into LAN (a single port or port range delimited by '-' is permitted);
Protocolselection of the packet protocol falling under this rule: TCP, UDP, TCP/UDP;
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WAN IP addresssource IP address that sends packets into external networks falling under this
rule;
Destination ports of WANrecipient TCP/UDP port values in the external network that cause the
packet to fall under this rule (a single port or port range delimited by '-' is permitted).
Port forwarding rule will work as follows: For the packet that comes to device WAN interface address via
'Protocol' to the port from 'WAN ports' range and has a 'WAN IP address' source address (if this parameter is
empty, source address will not be analyzed), its destination address and port are substituted with values from
'LAN IP address' and 'LAN ports' fields.
Click 'Apply' button to add a new rule. To discard changes, click 'Cancel' button.
To remove the rule from the list, select the checkbox next to the respective record and click 'Delete'.
3.6.2.6 Firewall submenu

In the 'Firewall' submenu, you may set the rules for the incoming, outgoing, and transit traffic
transmission. You may restrict transmission of various traffic types (incoming, outgoing, transit) depending on
the protocol, source and destination IP addresses, source and destination TCP/UDP ports (for TCP or UDP
messages), ICMP message type (for ICMP messages).
Confoguration of firewall rules

To add a new rule, click 'Add' button and fill in the following fields in the 'Add a new rule' window:
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Namerule name;
Traffic typeselect the traffic type that will fall under this rule:
Incomingincoming device traffic (recipient is one of the device network interfaces).

When this traffic type is chosen, the following fields will become available:
Source addressdefine starting source IP address. Use '/' symbol to define a
mask in 'xxx.xxx.xxx.xxx' or 'xx' format, e.g. 192.168.18.0/24 or
192.168.18.0/255.255.255.0, when you need to highlight an address range (/24
mask record corresponds to /255.255.255.0).
Outgoingoutgoing device traffic (traffic generated locally by the device from one of the
network interfaces). When this traffic type is chosen, the following fields will become
available for editing:
Destination addressdefine destination IP address. Use '/' symbol to define a
192.168.18.0/255.255.255.0, when you need to highlight an address range.
Transittransit traffic (traffic being transferred between two network interfaces when
the source and destination are external devices). When this traffic type is chosen, the
following fields will become available:
Source IP addressdefine source IP address. Use '/' symbol to define a mask in
'xxx.xxx.xxx.xxx'
or
'xx'
format,
e.g.
192.168.16.0/24
or
192.168.16.0/255.255.255.0, when you need to highlight an address range.
Destination addressdefine destination IP address. Use '/' symbol to define a
192.168.18.0/255.255.255.0, when you need to highlight an address range;
Protocolpacket protocol that will fall under this rule: TCP, UDP, TCP/UDP, ICMP, any.
Actionaction to be performed on packets (reject/skip).
When TCP, UDP, TCP/UDP are selected, the following settings will become available for editing:
Source portslist of source ports (a single port or port range delimited by '-' is permitted);
Destination portslist of destination ports (a single port or port range delimited by '-' is
permitted).
When ICMP protocol is selected, the following settings will become available for editing:
Message typeyou may create a rule for the specific ICMP message type only or for all ICMP
message types.
Click 'Apply' button to add a new rule. To discard changes, click 'Cancel' button. To remove the record
from the list, select the checkbox next to the respective record and click 'Delete'.
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3.6.2.7 filter submenu
In the 'MAC filter' submenu, you may configure access filtering by client's MAC address .
Filter modedefine one of the three filter operation modes depending on the client's MAC
address:
DisableMAC address filtering is disabled, all clients are allowed to connect to the access
point;
Black listin this filter operation mode, clients with MAC addresses from the 'MAC
address list' are denied to connect to the access point. Subscribers with unlisted MAC
addresses are allowed to connect to the AP;
White listin this filter operation mode, clients with MAC addresses from the 'MAC
address list' are allowed to connect to the access point. Subscribers with unlisted MAC
addresses are denied to connect to the AP.
Address List
You may enter up to 30 client MAC addresses which may access the device in accordance to the specified
filtering mode.
To add a new client to the list, click 'Add' button and enter its MAC address.
3.6.2.8 Static routes submenu

In the 'Static routes' submenu, you may configure device static routes.
To add a new route, click 'Add' button and fill in the following fields:
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Nameroute name, used for for human perception convenience. You may leave this field empty.
Destination IPIP address of destination host or subnet that the route should be established to.
Netmask subnet mask. Subnet mask for host should be 255.255.255.255, for subnet
depending on its size.
Gatewaygateway IP address that allows for the access to the 'Destination IP'.
3.6.2.9 Dynamic DNS submenu

In the 'Dynamic DNS' submenu, you may configure the respective service.
Dynamic DNS (D-DNS) allows DNS server information to be updated in real time and (optionally) in
automatic mode. It's used for assigning a fixed domain name to a device (to a computer or router) with a
dynamic IP address.
Dynamic DNS is frequently used in local networks, where clients are obtaining IP addresses through DHCP
and then registering their names on local DNS-server.
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Enable D-DNSwhen seleced, D-DNS service is enabled; the following settings will become
available for editing:
ServerD-DNS provider name, select a provider from the list of available providers.
User nameuser name used to access D-DNS service account.
Passwordpassword used to access D-DNS service account.
Domain name (0..9)you may register up to 10 device domain names (in general, only one name
is required). Device IP address information is updated on the provider server periodically in 60
seconds.
3.6.2.10 SNMP submenu

TAU-2M.IP software allows you to monitor status of the device and configure it via SNMP protocol. In
SNMP submenu, you can configure settings of SNMP agent. Device supports SNMPv1, SNMPv2c protocol
versions.
Enable SNMPwhen checked, SNMP will be enabled for utilization.

roCommunitypassword for parameter reading (common: 'public').
rwCommunitypassword for parameter writing (common: 'private').
TrapSinkIP address or domain name of SNMPv1-trap message recipient in HOST [COMMUNITY
[PORT]] format.
Trap2Sink IP address or domain name of SNMPv2-trap message recipient in HOST [COMMUNITY
[PORT]] format.
InformSinkIP address or domain name of Inform message recipient in HOST [COMMUNITY
[PORT]] format.
Sys namedevice name.
Sys contactdevice vendor contact information.
Sys Locationdevice location information.
Trap communitypassword enclosed in traps (default value: trap).
In the current firmware version, you may configure specific device parameters via SNMP: SIP basic
settings, SIP profile settings, FXS port settings, call group settings, VAS management codes dialled from the
phone unit, SNMP settings, system log settings.
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Given below is the list of objects that may be read an configured via SNMP:
Enterprise.1.3.1SIP profile basic settings;

Enterprise.1.3.2.1SIP profile settings;
Enterprise.1.1.2.1FXS port settings;
Enterprise.1.5VAS activation codes for the phone unit;
Enterprise.2.1SNMP settings;
Enterprise.3.1system log settings.
where Enterprise1.3.6.1.4.1.35265.1.56 is the device identifier.

To store settings into the flash memory, click 'Apply' button. To discard changes, click 'Cancel' button.
3.6.3 VoIP Menu

In the 'VoIP' menu, you may configure VoIP (Voice over IP): SIP protocol configuration, FXS interface
configuration, installation of codecs, numbering schedule, fax and modem data transfer methods
3.6.3.1 Network settings submenu

In the 'Network settings' submenu, you may specify custom network settings for VoIP service.
Use Internet settingswhen seleced, use network settings specified in the 'Network' -> 'Internet'
menu, otherwise use settings specified in this menu.
Use VLAN1when seleced, VoIP service will use a dedicated interface in a separate VLAN for its
operation, with VLAN number specified in 'VLAN ID' field.
In the firmware version 1.9.1, you may specify custom settings for VoIP service only in the dedicated VLAN.
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802.1P802.1P marker (another name: CoS Class of Service), assigned to the outgoing
frames from this interface. It may take values from 0 (the lowest priority) to 7 (the highest
priority).
Protocolselect address assigning protocol for the VoIP service interface:
Staticoperation mode where IP address and all the necessary settings for WAN
interface are assigned manually. When 'Static' type is selected, the following parameters
IP addressspecify the IP address for VoIP service interface;

Subnet masksubnet mask for VoIP service interface;
Default gatewayIP address for VoIP service interface default gateway;
Primary DNS, Secondary DNSDNS server IP addresses required for VoIP service
operations.
DHCPoperation mode where IP address, subnet mask, DNS address and other necessary
settings for service operation (e.g. SIP and registration server static routes) are
automatically obtained from DHCP server. If you are unable to obtain DNS server
addresses from the provider, you may specify them manually using 'Primary DNS' and
'Secondary DNS' fields. Manually defined addresses will have a priority over DNS
addresses obtained via DHCP.
Alternative Vendor ID (Option 60)when checked, the device transmits Vendor
ID (Option 60) field value in Option 60 DHCP messages (Vendor class ID). If the
field is empty, Option 60 will not be transmitted in DHCP messages.
address][VERSION:software version]
Example:
IPSec settings:
In this section, you may configure IPSec encryption (IP Security).
IPSec is a set of protocols used for protection of data transmitted via Internet Protocol that enables
authentication, integrity check and/or encryption of IP packets. IPsec also includes secure Internet Key Exchange
protocols.
In the current firmware version, you may only access the device management interfaces (Web, Telnet,
SSH) using IPSec.
For detailed IPSec settings, see Section 3.6.2.1 Internet submenu.
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3.6.3.2 Line settings submenu
In the 'Line settings' submenu, you may configure Phone1 and Phone2 ports.
To edit settings, press and hold the left mouse buttom on the link with the number of adjastable line and
fill the following fields in the appeared Edit line window:
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Account configuration
Enablewhen seleced, the port is active.

Profileselect SIP profile from the list of available profiles. To configure profiles, use 'VoIP' ->
'Profiles' menu.
Phonesubscriber number, assigned to this phone port.
User nameusername associated with the port (shown in 'Display-Name' field of the 'From'
header in the outgoing SIP messages).
Use alternative numberwhen seleced, an alternative number will be inserted into the 'From'
header of SIP messages sent from this port (particularly, in order to hide the real number from the
Caller ID system of the callee).
Use alternative number - alternative number assigned to a phone port will be changed to specified
number and inserted into Contact hearder of SIP message.
SIP portUDP port for incoming SIP message reception for this account, and for outgoing SIP
message transmission from this account. It may take values from 1 to 65535 (default value: 5060).
Calling party categorycalling party categoryenables transmission of outgoing messages in the
'From' header; the latter is transmitted in Tel-URI format (see RFC3966).
Authentication login and passwordusername and password used for subscriber authentication
on SIP server (and on registration server).
Supplementary services
Flash mode flash function operation mode (short clearback):
Transmit flash transmit flash into the channel (using one of the methods described in
'Profiles' tab, 'Flash transmission' parameter);
Attended calltransfer flash will be processed locally by the device (call transfer will be
performed when the connection with the third party is established). For the 'Attended
calltransfer' detailed operation algorithm, see Section 4.1 Call transfer;
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Call transfer mode this setting is available for the 'Attended calltransfer' and Local calltransfer
modes only and governs call transfer service activation mode:
Call forward unconditional numbernumber that all incoming calls will be forwarded to
when Call forward unconditional service is enabled (in SIP profile being used, a prefix for
this direction should be defined in the numbering schedule).
Call forward on busywhen seleced, CFB (Call Forward at Busy) service is enabledforward the
call to the specified number, when the subscriber is busy. When seleced, fill in the following fields:
Hotline/warmline numberphone number that will be used for connection establishment

upon 'Delay timeout' expiration after the phone handset is picked up (in SIP profile being
used, a prefix for this direction should be defined in the numbering schedule);
Delay timeout, secondstime interval that will be used for connection establishment with
the opposite subscriber, in seconds.
Call forward unconditionalwhen seleced, CFU (Call Forward Unconditional) service is enabled
all incoming calls will be forwarded to the specified call forward unconditional number.When
seleced, fill in the following fields:
Mixedcall transfer is activated on clearback and pressing R 4.

Flash+4call transfer is activated on pressing R 4.
Onhookcall transfer is activated on clearback.
Direct numberwhen the phone goes offhook, dial the defined number immediately.
Call waitingwhen seleced, 'Call waiting' service will be enabled (this service is available in
'flashcall transfer' function operation mode).
Deliver calling party phone number during call waitingwhen selected, the subscriber number is
delivered for the call waiting service.
Stop dial at #when seleced, use # button on the phone unit to end the dialling, otherwise #
will be recognized as a part of the number.
Hotline/warmlinewhen seleced, 'Hotline/warmline' service is enabled. This service allows to
establish an outgoing connection automatically without dialling the number after the phone
handset is picked up with the defined delay (in seconds). When checked, fill in the following fields:
Unattended calltransfer flash will be processed locally by the device (call transfer will be
performed when the subscriber finishes dialling a third party number). For the
'Unattended calltransfer' detailed operation algorithm, see Section 4.1 Call transfer;
Local calltransfer call transfer inside of the device (without REFER message
transmission). For the Local calltransfer detailed operation algorithm see Section 4.1 Call
transfer.
Call forward on busy numbernumber that incoming calls will be forwarded to when the
subscriber is busy and Call forward on busy service is enabled (in SIP profile being used, a
prefix for this direction should be defined in the numbering schedule).
Call forward on no answerwhen seleced, CFNA (Call Forward at No Answer) service is enabled
forward the call, when there is no answer from the subscriber. When seleced, fill in the following
fields:
Call forward on no answer numbernumber that incoming calls will be forwarded to

when there is no answer from the subscriber and Call forward on no answer service is
enabled (in SIP profile being used, a prefix for this direction should be defined in the
numbering schedule);
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No answer timeout, secondstime interval that will be used for call forwarding when
there is no answer from the subscriber, in seconds.
Do not disturbwhen selected, temporary restriction is placed for incoming calls (DND service).
When multiple services are enabled simultaneously, the priority will be as follows (in the
descending order):
CFU;
DND;
CFB, CFNA.
Permit to pickup incoming calls when this option is enabled, incoming calls pickup is enabled for
the port (call pickup is allowed only within a single pickup group when ports use the same SIP
profile);
CLIR caller ID service restriction:
DisabledCLIR service is disabled;

SIP:FromAnonymous sip:[email protected] will be sent in the From header of
SIP messages;
SIP:From and SIP:ContactAnonymous sip:[email protected] will be sent in the
From and Contact headers of SIP messages.
Line parameters
Caller ID generation select the Caller ID mode. For Caller ID operation, subscriber's phone unit
must support the selected method:
OffCaller ID is disabled;
FSK Bell 202, FSK V.23FSK Caller ID method (using Bell202 standard, or ITU-T V.23). The
number is served between the first and second ringing tones by a stream of data with a
frequency modulation.
Hangup timeout, secondsdialling timeout for the first digit of a number. When there is no
dialling during the specified time, 'busy' tone will be sent to the subscriber, and the dialling will
end;
Busy timeout, seconds'busy' tone timeout for the subscriber. If the subscriber doesn't put the
phone onhook until the timeout expires, an error tone will be sent into the line;
Ringback timeout, secondslaunches when an incoming call is received and defines the maximum
call response time. When the defined timeout expires, 'busy' tone will be sent to the remote
subscriber;
Minimal on-hook time, msminimal clearback detection time, in milliseconds. At that, this
parameter represents the maximum flash detection time;
Minimal flash time, msminimal flash detection time, (801000) ms;
Gain receive, 0.1dBreceived signal gain (transmitted into the phone handset), measurement
unit0.1dB;
Gain transmit, 0.1dBtransmitted signal gain (received by the phone handset microphone),
measurement unit0.1dB;
Min pulse , msconfiguration is required for pulse dialling mode, (10-150) ms.
Interdigit configuration is required for pulse dialling mode, (150-20000) ms.
Polarity reversalwhen this option is enabled, line voltage polarity reversal occurs right after the
callee responses to the outgoing call. After the clearback, the voltage polarity returns to its
original state. This option is essential for payphone operation (polarity reversal indicates the start
of the paid time interval).
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3.6.3.3 Profiles submenu

In the 'Profiles' submenu, you may configure device SIP profiles. For each SIP profile, you may assign
custom SIP and registration server addresses, voice and fax/modem codecs, custom numbering schedule and
other parameters. Different SIP profiles are needed when different subscriber ports operate via different
communication directions (different SIP servers). At that, for each subscriber port, you may assign only one SIP
profile ('VoIP' -> 'Line settings).
To edit profile settings, left-click the link of the profile to be configured. In the 'Edit profile:' window, fill in
the following fields:
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SIP parameters
Profile includeslist of subscriber ports that the profile is assigned to; this field cannot be
changed.
Profile namecustom name of the configured profile.
Proxy pmode select SIP server operation mode from the drop-down list:
Off;
Parking SIP-proxy redundancy mode without main SIP-proxy management;
HomingSIP-proxy redundancy mode with main SIP-proxy management.
Gateway may operate with a single main SIP-proxy and up to four redundant SIP-proxies. For
exclusive operations with the main SIP-proxy, 'Parking' and 'Homing' modes are identical. In this
case, if the main SIP-proxy fails, it will take time to restore its operational status.
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For operations with redundant SIP-proxies, 'Parking' and 'Homing' modes will work as
follows. The gateway sends INVITE message to the main SIP-proxy address when performing
outgoing call, and REGISTER message when performing registration attempt. If on expiration
of 'Invite total timeout' there is no response from the main SIP-proxy or response 408 or 503
is received, the gateway sends INVITE (or REGISTER) message to the first redundant SIP-proxy
address. If it is not available, the request is forwarded to the next redundant SIP-proxy and so
forth. When available redundant SIP-proxy is found, registration will be renewed on that SIPproxy.
Next, the following actions will be available depending on the selected redundancy mode:
1 In the 'parking' mode, the main SIP-proxy management is absent, and the gateway will
continue operation with the redundant SIP-proxy even when the main proxy operation
is restored. If the connection to the current SIP-proxy is lost, querying of the subsequent
SIP-proxies will be continued using the algorithm described above. If the last redundant
SIP-proxy is not available, the querying will continue in a cycle, beginning from the main
SIP-proxy.
2 In the 'homing' mode, three types of the main SIP-proxy management are available:
periodic transmission of OPTIONS messages to its address, periodic transmission of
REGISTER messages to its address, or transmission of INVITE request when performing
outgoing call. First of all, INVITE request is sent to the main SIP-proxy, and if it is
unavailable, then to the next redundant one, etc. Regardless of the management type,
when the main SIP-proxy operation is restored, gateway will use it to renew its
registration. The gateway will begin operation with the main SIP-proxy.
Proxy servernetwork address of a SIP serverdevice that manages access to provider's phone
network for all subscribers. You may specify IP address as well as the domain name (specify SIP
server UDP port after the colon, default value is 5060).
Registrationwhen seleced, register ports that utilize this profile on registration server.
Registration servernetwork address of a device that is used for registration of all phone network
subscribers in order to provide them with the communication services (specify registration server
UDP port after the colon, default value is 5060). You may specify IP address as well as the domain
name. As a rule, registration server is physically co-located with SIP proxy server (they have the
same address).
Home server check methodselect availability control method for the primary SIP server in
'Homing' mode:
Invite control via transmission of INVITE request to its address when performing an
outgoing call;
Register control via periodic transmission of REGISTER messages to its address;
Options control via periodic transmission of OPTIONS messages to its address.
Home server keepalive timeout, seconds periodic message transmission interval in seconds;
used for primary SIP server availability check.
Reserved proxy
To add a redundant SIP proxy, click 'Add' button and enter the following settings:
Proxy servernetwork address of redundant SIP server. You may specify IP address as well as the
domain name (specify SIP server UDP port after the colon, default value is 5060);
Registration servernetwork address of redundant registration server (specify UDP port after the
colon, default value is 5060). You may specify IP address as well as the domain name. If the
'Registration server' checkbox is selected, the redundant server registration is enabled.
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To remove the redundant SIP proxy, select the checkbox next to the specified address and click 'Delete'
button.
SIP domaindomain where the device is located (fill in, if required).

Use Domain to registerwhen selected, apply SIP Domain for registration (SIP domain will be
inserted into the 'Request-Line' of 'Register' requests).
Outbound mode'Outbound' mode:
Offcalls will be routed according to the numbering schedule;
Outboundnumbering schedule is required for outgoing communications; however, all
calls will be routed via SIP server; if there is no registration, PBX response will be sent to
the subscriber in order to enable subscriber service management (VAS management);
Outbound with busynumbering schedule is required for outgoing communications;
however, all calls will be routed via SIP server; if there is no registration, VoIP will be
unavailable: error tone will be transmitted to the phone headset.
Expirestime for subscriber port registration on SIP server. At the average, port registration
renewal will be performed after 2/3 of the specified period;
Registration Retry Intervalwhen the registration is unsuccessful, time period between SIP server
registration attempts;
Public IP addressthis parameter is used as an external address of the device when it operates
behind the NAT (gateway). As a public address, you may specify an external address (WAN) of a
gateway (NAT) that TAU-2M.IP operates through. At that, on the gateway (NAT), you should
forward the corresponding SIP and RTP ports used by the device;
Use SIP Display Name in registerwhen seleced, use username in 'SIP Display Info' field of the
'Register' message;
Ringback on 183 Progresswhen seleced, 'ringback' tone will be sent upon receiving '183
Progress' message (w/o enclosed SDP);
User callprovisional response sent by the device to the caller equipment during the incoming
call.
180 Ringingcaller equipment will receive response 180; upon receiving this message,
caller equipment should send a local ringback tone into the line.
183 Progress with SDPcaller equipment will receive response 183+SDPused for voice
frequency path forwarding before the answer of the callee. In this case, TAU-2M.IP will
send a ringback tone remotely to the caller.
100reluse reliable provisional responses (RFC3262):
supportedreliable provisional responses are supported;
requiredreliable provisional responses are mandatory;
offreliable provisional responses are disabled.
SIP protocol defines two types of responses for connection initiating request (INVITE)
provisional and final. 2, 3, 4, 5 6-class responses are final and their transfer is reliable,
with ACK message confirmation. 1-class responses, except for '100 Trying' response, are
provisional, without confirmation (rfc3261). These responses contain information on the current
INVITE request processing step, therefore loss of these responses is unacceptable. Utilization of
reliable provisional responses is also stated in SIP (rfc3262) protocol and defined by '100rel' tag
presence in the initiating request. In this case, provisional responses are confirmed with PRACK
message.
Setting operation for outgoing communications:
Supportedsend the following tag in 'INVITE' requestsupported:100rel. In this case,

communicating gateway may transfer provisional responses reliably or unreliablyas it
deems fit;
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Requiredsend the following tags in 'INVITE' requestsupported: 100rel and

required:100rel. In this case, communicating gateway should perform reliable transfer of
provisional replies. If communicating gateway does not support reliable provisional
responses, it should reject the request with message 420 and provide the following tag
unsupported: 100rel. In this case, the second INVITE request will be sent without the
following tagrequired: 100rel;
Disableddo not send any of the following tags in INVITE requestsupported: 100rel and
required: 100rel. In this case, communicating gateway will perform unreliable transfer of
provisional replies.
Setting operation for incoming communications:

Supported, Requiredwhen the following tag is received in 'INVITE' requestsupported:
100rel, or required: 100relperform reliable transfer of provisional replies. If there is no
supported: 100rel tag in INVITE request, the gateway will perform unreliable transfer of
provisional replies;
Disabledwhen the following tag is received in 'INVITE' requestrequired: 100rel, reject
the request with message 420 and provide the following tagunsupported: 100rel.
Otherwise, perform unreliable transfer of provisional replies.
Timer enablewhen seleced, the 'timer' (RFC 4028) extension support is enabled. When
connection is established, and both sides support 'timer' extension, one of them periodically
sends re-INVITE requests for connection monitoring purposes (if both sides support UPDATE
method, wherefore it should be specified in the 'Allow' header, the session update is performed
by periodic transmission of UPDATE messages).
Min SE, secminimal time interval for connection health checks (90 to 1800s, 120s by default).
Session expires, secondsperiod of time in seconds that should pass before the forced session
termination if the session is not renewed in time (90 to 80000s, recommended value1800s, 0
unlimited session).
Keepalive NAT sessions mode select SIP server polling method:
OffSIP server will not be polled;
OptionsSIP server polling with OPTIONS messages;
NotifySIP server polling with NOTIFY messages;
CLRFSIP server polling with an empty UDP packet.
Keepalive timeout, secSIP server polling time period, in seconds;
Use Alert-Info headerprocess INVITE request 'Alert-Info' header to send a non-standard ringing
to the subscriber port.
Check RURI user part onlywhen seleced, only subscriber number (user) will be analyzed, and if
the number matches, the call will be assigned to the subscriber port. When unseleced, all URI
elements (user, host and portsubscriber number, IP address and UDP/TCP port) will be analyzed
upon receiving an incoming call. If all URI elements match, the call will be assigned to the
subscriber port.
Send IP address in Call-ID headerwhen seleced, during outgoing communications, device custom
IP address will be used in 'Call-ID' header in 'localid@host' format.
Three-way conference
Modethree-way conference call operation mode. Two modes are possible:
Localconference assembly is performed locally by the device after pressing 'flash+3';

Remote (RFC4579)conference assembly is performed at the remote server; after
pressing 'flash+3', 'Invite' message will be sent to the server using number specified in the
'Conference server' field. In this case, conference operation complies with the algorithm
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described in RFC4579. For detailed algorithm description, see Paragraph 4.3.2 Remote
conference.
Conference serverin general, address of the server that establishes conference using algorithm
described in RFC4579. Address is specified in the following format SIP-URI: user@address:port.
You may specify the 'user' URI part onlyin this case, 'Invite' message will be sent to the SIP proxy
address.
IMS settings
Enable VAS management using IMSyou may manage some service types using IMS (IP
Multimedia Subsystem) server. In this case, 'Three-way conference call' (complies with RFC4579
algorithm), 'Call hold', 'Call waiting', 'Hotline' services (regardless of whether they were enabled in
the configuration or not) will be enabled remotely by IMS server that sends 'Notify' messages
containing enable/disable commands in XCAP format (in fact, XML, RFC4825):
OffIMS will not be used;
Implicityou may manage some service types using IMS (IP Multimedia Subsystem)
server. In this case, 'Three-way conference call' (complies with RFC4579 algorithm), 'Call
hold', 'Call waiting', 'Hotline' services (regardless of whether they were enabled in the
configuration or not) will be enabled remotely by IMS server that sends 'Notify' messages
containing enable/disable commands in XCAP format (in fact, XML, RFC4825). In this case,
when the gateway finishes subscriber registration, SUBSCRIBE requests will not be sent,
only NOTIFY requests received from IMS and used for service management will be
processed;
Explicityou may manage some service types using IMS (IP Multimedia Subsystem)
server. In this case, 'Three-way conference call' (complies with RFC4579 algorithm), 'Call
hold', 'Call waiting', 'Hotline' services (regardless of whether they were enabled in the
configuration or not) will be enabled remotely by IMS server that sends 'Notify' messages
containing enable/disable commands in XCAP format (in fact, XML, RFC4825). In this case,
when the gateway finishes subscriber registration, it sends SUBSCRIBE requests, and if the
subscription is successfully completed, it will process NOTIFY requests received from IMS
and used for service management.
XCAP name for call hold name of the XML element located in the 'Notify' message body that is
used for transmission of 'Call hold' enabling/disabling commands. For example, if the service
name value is 'call-hold', enabling command will be as follows:
<call-hold active="true"/>
Disabling command:
<call-hold active="false"/>
XCAP name for call waiting name of the XML element located in the 'Notify' message body that
is used for transmission of 'Call waiting' enabling/disabling commands. For example, if the service
name value is 'call-waiting', enabling command will be as follows:
<call-waiting active="true"/>
Disabling command:
<call-waiting active="false"/>
XCAP name for three-party conference name of the XML element located in the 'Notify'
message body that is used for transmission of 'Three-way conference call' enabling/disabling
commands. For example, if the service name value is 'three-party-conference', enabling command
will be as follows:
<three-party-conference active="true"/>
Disabling command:
<three-party-conference active="false"/>
XCAP name for hotline name of the XML element located in the 'Notify' message body that is
used for transmission of 'Hotline' enabling command. In the enabling command, you should pass
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the hotline number and the call timeout. For example, if the service name value is 'hot-lineservice' and you should call the number 30001 in 6 seconds after the phone handset is picked up,
the enabling command will be as follows:
<hot-line-service>
<addr>30001</addr>
<timeout>6</timeout>
</hot-line-service>
If the enabling command is not received, the 'Hotline' service will be disabled.
XCAP name for call transfer name of the XML element located in the 'Notify' message body that
is used for transmission of 'Three-way conference call' enabling/disabling commands. For
example, if the service name value is 'call-transfer', enabling command will be as follows:
<call-transfer active="true"/>
Disabling command:
<call-transfer active="false"/>
By default, if the enabling command is not received, all the services mentioned above are disabled.
Dialplan
To define the numbering schedule, use regular expressions in the 'Numbering schedule configuration'
field.
The structure and format of regular expressions that enable different dialling features are listed below.
Structure of regular expressions:
Sxx, Lxx ( ),
where
arbitrary values of S and L timers;
() dialling plan margings.
Basis is the designations used for the dialled digit sequence recording. Digit sequence is recorded
using several designationsdigits dialled from the phone keypad: 0, 1, 2, 3, , 9, #, and *. If you
use # in the dialplan, you may block the dialling completion using this key!
Digit sequence enclosed in square brackets corresponds to any character enclosed in these
brackets.
Use a hyphen to define a range of characters. Mostly used inside square brackets.
Example 1: (15)any digit from 1 to 5.

Example 2: ([1-39])example listed above in the different entry format.
'X' character corresponds to any digit from 0 to 9.
Example: ([1239])corresponds to any digit1, 2, 3, or 9.
Example: (1XX)any 3-digit number that begins with 1.
'.' repeat previous character from 0 ad infinitum.

+repeat previous character from 1 ad infinitum.
{a,b}repeat previous character from 'a' to 'b' times.
{a,}repeat previous character more than 'a' times.
{,b}repeat previous character less than 'b' times.
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Example: (810X.) international number with any quantity of digits.
Settings affecting dialplan configuration:
Interdigit Long Timer ('L' character in numbering plan record)entry timeout for the next digit, if
there are no templates that correspond to the dialled combination;
Interdigit Short Timer('S')entry timeout for the next digit, if the dialled combination fully
matches at least one template and if there is at least one template that requires an extension
dialling for the full match.
Additional features:
1.
Dialled sequence replacement
Syntax: <arg1:arg2>
This feature allows you to replace the dialled sequence with any dialled character sequence. At that,
the second argument should be defined with the specific value, both arguments may be empty.
2.
Example: (<83812:> XXXXXX)this record will correspond to dialled digits 83812, but this
sequence will be skipped and will not be sent to the SIP server.
Tone insertion into dialling
For long-distance access (for city access in case of office PBX), it is common to hear a PBX response,
that may be implemented by inserting comma in a sequence of digits.
3.
Example: (8, 770)when number 8770 is dialled, the continuous tone will be played after
the digit '8'.
Dialling restriction.
When you specify an exclamation mark '!' at the end of the number template, dialling of numbers
corresponding to the template will be blocked.
4.
Example: (8 10Xxxxxxxx ! | 8 xxx xxxxxxx )expression allows long-distance dialling only

and denies outgoing international calls.
Replacement of dialling timer values
Timer values may be specified for the entire dialplan, as well as for the specific template only. 'S'
character deals with the 'Interdigit Short Timer', and 'L' with the 'Interdigit Long Timer' setting. Timer
values may be specified for all templates in the dialplan, when values are listed before the opening
parenthesis.
Example: S4 (8XXX.) or S4,L8 (XXX)
If these values are listed in one sequence only, they are effective only for this sequence. At that, you
are not required to delimit the key and timeout value with the colon, value may be specified anywhere
within the template.
5.
Example: (S48XXX. | XXX) or ([1-5] XX S0)record will trigger an instant call transfer,
when the 3-digit number beginning with 1,2, , 5 is dialled.
Direct address dialling (IP Dialing)
'@' placed after the number defines that the dialled call will be sent to the subsequent server address.
We recommend using 'IP Dialing', as well as call reception and transmission without registration (Call
Without Reg, Answer Without Reg). This may help when the server fails.
Also, IP Dialling address format may be used for numbers intended for the call forwarding.
Example 1: ( 8 xxx xxxxxxx )11-digit number beginning with 8.

Example 2: ( 8 xxx xxxxxxx | <:8495>xxxxxxx )11-digit number beginning with 8; if 7-digit
number is dialled, add 8495 to the number being sent.
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Example 3: (0[123] | 8 [2-9]xx [2-9]xxxxxx)dialling of emergency call numbers and

unusual sets of long-distance numbers.
Example 4: (S0<:82125551234>)quickly dial the specified number, similar to 'Hotline'
mode on other gateways.
Example 5: (S5 <:1000> | xxxx)this dialplan allows to dial any number that contains
digits, and if there was no entry in 5 seconds, dial number '1000' (for example, it belongs
to a secretary).
Example 6: (*5x*xxxx*x#|*2x*xxxxxxxxxxx#|#xx#|[2-7]xxxxx|8,
[2-9]xxxxxxxxx|8,
10x.|1xx<:@10.110.60.51:5060>).
Example 7: (1xx|0[1-9]|00[1-8]|*5x*xxxx*x#|*2x*xxxxxxxxxxx#|#xx#|[2-7]xxxxx|8, [29]xxxxxxxxx|8, 10x.).
Sometimes you have to make calls locally inside of the device. In this case, if IP address of a device is
not known or periodically changed, you may use the reserved "{local}" word as the server address. It
means sending an appropriate sequence of digits to the own device address.
Example: (123@{local}) 123 dialing number will be processed inside of the device locally.
Voice codecs configuration

The signal processor of the device encodes analogue voice traffic and fax/modem data into digital signal
and performs its reverse decoding. Gateway supports the following voice codec: G.711A, G.711U, G.729, G723.1.
G.711 is a PCM codec that does not employ a compression of voice data. This codec must be supported by
all VoIP equipment manufacturers. G.711A and G.711U codecs differ from each other in encoding law (-law is a
linear encoding and U-law is non-linear). The U-law encoding is used in North America, and the A-law
encodingin Europe.
G.723.1 is a voice data compression codec, allows for two operation modes: 6.3kbps and 5.3kbps. G.723.1
codec has a voice activity detector and performs comfort noise generation at the remote end during period of
silence.
G.729 is also a voice data compression codec with the rate of 8kbps. As with G.723.1, G.729 codec
supports voice activity detector and performs comfort noise generation.
Codec 1..4allows you to select codecs and the order of their usage. The highest priority codec
should be specified in the 'Codec 1' field. For operation, you should specify at least one codec:
Disabledcodec will not be used;

G.711ause G.711A codec;
G.711uuse G.711U codec;
G.723use G.723.1 codec;
G.729use G.729 codec.
Packet timeamount of voice data in milliseconds (ms) transmitted in a single RTP packet for the
corresponding codec (G.711, G.729, G.723).
Dispersion time, msparameter that cancels an echo caused by the voice signal dispersion.
Parameter values may be specified in the interval from 2 ms to 128 ms.
NOTE! You may specify alternative voice codecs for the selected direction. For each direction, you may
specify the preferred codec for voice communication in the numbering plan. Configuration is performed in the
numbering plan. For each direction, an additional codec configuration may be specified in parentheses after the
'codecs:' word.
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If you have to use multiple codecs, you may separate them with a comma ','. Multiple parameters may be
specified for a single direction. In this case, separate them with a semicolon ';'
(param1:subparam1,subparam2;param2:subparam1,subparam2). subparamX permitted values: 'g711a', 'g711u',
'g729', 'g723'.
param1 and param2 permitted values'codecs' and 'rfc2833_PT' respectively.
Example: ([23]xxx(codecs:g729; rfc2833_PT:96)|8x.(codecs:g711a;g711u)).
Jitter buffer
Jitter is a deviation of time periods dedicated to packet delivery. Packet delivery delay and jitter are
measured in milliseconds. Jitter value is higher for real time data transfers (e.g. voice or video data).
In RTP, also known as 'media stream protocol', there is a field for precision transmission time tag related
to the whole RTP stream. Receiving device uses these time tags to learn when to expect the packet and whether
the packet order has been observed. On the basis of this information, the receiving side will learn how to
configure its settings in order to evade potential network problems such as delays and jitter. If the expected
time for packet delivery from the source to the destination for the whole call period corresponds to the defined
value, e.g. 50ms, it is fair to say that there is no jitter in such a network. But packets are delayed in the network
frequently, and the delivery time period may fluctuate significantly (in the context of time-critical traffic). If the
audio or video recipient application will play packets in the order of their reception time, voice (or video) quality
will deteriorate significantly. For example, if the voice data is being transferred, there will be interruptions and
interference in the voice.
The device features the following jitter buffer settings:
Min delay, msminimum expected IP package network propagation delay;

Max delay, msmaximum expected IP package network propagation delay;
Deletion Threshold (DT), msmaximum amount of time for voice package removal from the
buffer. The parameter value should be greater or equal to maximum delay;
Jitter factorparameter used for jitter buffer size optimization. Recommended value is 0.
Fax and modem transfer

Fax may be transmitted using 711 voice codec or T.38 specialized codec for sending facsimile messages.
T.38 is a standard for sending facsimile messages in real time over IP networks. Signals and data sent by
the fax unit are copied to T.38 protocol packets. Generated packets may feature redundancy data from previous
packets that allows you to perform reliable fax transmissions through unstable channels.
Modem transferselect a codec to be used for data transmission when the gateway detects
modem signals:
Offmodem signals will not be detected;

G.711a VBDuse G.711A codec in VBD mode;
G.711u VBDuse G.711U codec in VBD mode.
In VBD (Voice band data) mode, the gateway disables voice activity detection (VAD), comfort noise
generator (CNG), and echo cancellers; this is necessary for establishing a modem connection.
Selected codec should also be enabled in voice codec list.
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Fax codec 1..3you may select a codec and the .and an order of their usage. The highest priority
codec should be specified in the 'Fax codec 1' field. For operation, you should specify at least one
codec:
Offcodec will not be used;

G.711ause G.711A codec;
G.711uuse G.711U codec;
T.38use T.38 protocol.
All fax codecs should be different! Also, when G.711a or G.711u codec is selected,
the respective codec should be enabled in the device voice codec list.
Take the transition to T.38when seleced, incoming re-inviteto T.38 from the opposite gateway
will be enabled.
T.38 redundancy countadd redundancy to T.38 packets; the value corresponds to the number of
previous packets which is duplicated in each new T.38 packet. Such redundancy method is
intended for packet loss during transmission.
Additional parameters
DTMF TransferDTMF tone transmission method:
Inbandinband transmission;
RFC2833according to RFC2833 recommendation, as a dedicated payload in RTP voice
packets;
SIP infotransfer messages via SIP in INFO requests.
Flash transferFlash transmission method:
SIP info (Hookflash)send messages to the opposite side via SIP in INFO requests. flash
event is sent in Application/ Hook Flash extension as 'signal=hf';
SIP info (DTMF Relay)send messages to the opposite side via SIP in INFO requests. flash
event is sent in Application/dtmf-relay extension as 'signal=hf';
SIP info (Broadsoft)send messages to the opposite side via SIP in INFO requests. flash
event is sent in Application/Broadsoft extension as 'event flashhook';
SIP info (SSCC)send messages to the opposite side via SIP in INFO requests. flash event is
sent in Application/sscc extension as 'event flashhook'.
In the current firmware version, flash transmission is possible via SIP only.
RFC2833 payload type payload type for packet transmission via RFC2833 (permitted values:
from 96 to 127).
Use the same PT both for transmission and receptionoption is used in outgoing calls for payload
type negotiation of events sent via RFC2833 (DTMF and Flash). When seleced, event transmission
and reception via RFC2833 is performed using the payload from 200Ok message sent by the
opposite side. When uncseleced, event transmission is performed via RFC2833 using the payload
from 200Ok being received, and receptionusing the payload type from its own configuration
(specified in the outgoing Invite).
Silence detectorwhen seleced, enable voice activity detector.
Echocancellationwhen seleced, use echo cancellation.
RTCPwhen seleced, use RTCP for voice link monitoring.
Sending intervalRTCP packet transmission period, in seconds.
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Receiving periodRTCP message reception period measured in transmission period units;

if there is not a single RTCP packet received until the reception period expires,TAU-2M.IP
will terminate the connection.
SIP profile basic settings
Use STUNwhen seleced, use STUN (Session Traversal Utilities for NAT) protocol in order to
define device public address (external NAT address). We recommend using this protocol for
device operation through NAT.
STUN server addressSTUN server IP address or domain name; specify an alternative server port
after the colon (default value is 3478).
STUN server polling period, secondstime period that defines transmission of a request to STUN
server. The less the polling period, the faster the response to the public address changes.
Transportselect the protocol for SIP message transmission.
3.6.3.4 Hunt group submenu

Use the menu to manage call groups.
Hunt groups provides functions of call processing center. The device supports 3 call group operation
modes:
Group mode, when call is transmitted to all the free group ports simultaneously. When one of
the group paticepants answers, the call to the other ports is finished;
Serial mode when the call is transmitted to the first free port from the group list and then, after
defined time range (Timeout of the following port call), the next free port is added to the basic
port and etc. If one from the group participants answers to the call, the call to the other ports is
broken up;
Cyclic mode when the free participant is successively selected from the group members upon
timeout (Timeout of the following port call) and the call is switched to its number.
Group name call group name;

Status call group status: enable or disable;
Profile SIP-profile that is used by call group;
Phone call group phone number;
List of the ports line (port) list of group members.
To apply settings of call group, click on the corresponding link of Group name column:
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Enable when seleced, use a group;

Profile SIP-profile assinged to the call group. Settings are performed in Section VoIP-> Profiles;
Group name group ID name;
Phone phone number of call group;
User name username for authentication on SIP server;
password password for authentication on SIP server;
SIP port alternative SIP-port of group (by default 5060);
Group type call group type:
Group call signal is transmitted to all the ports of group simultaneously;

Serial port number for call transmission has been increased by one after timeout
expiration of the following port;
Cyclic interval call signal is transmitted to the group port each after each. (interval is
equal to the call timeout of the following port). When the last port of call group is
achieved, call-down continues againg from the first port;
Next port calling timaout, s option is used by serial and cyclic group types and option assignes
time interval for the call to following port(s);
Call queue size setting allows you to limit maximum number of unanswered calls in call group
queue. If the group has free port incoming call is not put on the queue;
Call reply timeout, s if group call is not answered, it will be dropped after timeout expiration;
Group call pickup enable when seleced, group call interception is permitted. Call interception is
available only if call group subscribers belong to the same group. (See section 3.6.3.7 Pickup
group submenu).
List of the ports when seleced, line (port) will be a member of call group.
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3.6.3.5 QoS submenu
In the 'QoS' submenu, you may configure Quality of Service functions.
RTP port range configuration
Min RTP port the lower limit of the RTP port range used for voice traffic transmission.
Max RTP portthe upper limit of the RTP port range used for voice traffic transmission.
DSCP configuration
Signalling DSCP (SIP)DSCP field value of IP packet header for signalling traffic (should be
specified decimally, may take values from 0 to 63).
RTP DSCPDSCP field value of IP packet header for voice traffic (should be specified decimally,
may take values from 0 to 63).
To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To
discard changes, click 'Cancel' button.
3.6.3.6 Supplementary service prefixes submenu

In the 'Supplementary service prefixes' submenu, you may configure codes dialled from the phone unit
in order to enable or disable VAS.
Subscribers may manage state of services from their phone units. The following functions are available:
Service activation* service_code #;
Service activity check*# service_code #;
Service cancellation# service_code #.
In order to activate 'Call forward unconditional', 'Call forward on busy', 'Call forward on no answer', or
'Hotline/warmline' service, you should specify a phone number:
* service_code * phone_number #
When the activation code is entered or the service is cancelled, subscriber may hear a 'confirmation' tone
(3 short tones) which means that the service has been activated or cancelled successfully.
After service confirmation code entry, the subscriber may hear either 'PBX response' tone (continuous) or
a 'busy' tone (intermittent). 'PBX response' tone means that the service has been enabled and activated, 'busy'
tonethat this service is disabled.
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Subscriber service management
Supplementary services:
Call forward unconditionalwhen active, all incoming calls will be forwarded to the
specified number;
Call forward on busywhen active, all incoming calls will be forwarded to the specified
number, if the subscriber is busy;
Call forward on no answerwhen active, all incoming calls will be forwarded to the
specified number, if there is no answer from the subscriber;
Permit to pickup incoming callswhen activated by the subscriber, incoming calls may be
picked up by other subscribers from the same pickup group;
Hotline/warmlinewhen active, the defined phone number will be dialled upon
expiration of the specific time period after the phone handset will have been picked up;
Call waitingwhen active, the subscriber will receive notifications on incoming calls while
being in a call state. Subscriber may accept, reject or ignore waiting call;
Do not disturbthis service allows the subscriber to put temporary restriction on all
incoming calls.
Activation codeservice activation code.
Deactivation codeservice deactivation code.
Check codeservice activity check code.
Deactivation code and service status check code are generated automatically based on the activation
code.
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3.6.3.7 Pickup group submenu
Use this menu to configure pickup groups.
Pickup group subscriber group that can pickup any call directed to another subscriber inside of the
group. In other words, each subscriber inside of the group may pick up a call transmitted to any other client of
the group by dialing intercept code. Use the Section 3.6.3.3. Profiles submenu for settings of the intercept
code.
To add/delete line into/from group select/deselect the checkbox next to the respective group.
Service usage:
The call is transmitted to the subscriber phone unit covered by pickup group. If the subscriber cant
answer to a call - another subscriber, that belongs to the pickup group and use the same SIP profile, can pick up
incoming call. To do it, user should dial intercept code and after that connection with the calling user will be
established.
Pay attention that the call interception is possible only when called client and picking up client use the
same SIP profile.
Pickup group can be used together with call group if all the call group ports are contained in pickup group.
In this case, port is contained in pickup group can intercept incoming call of group number.
If subscriber is dialing code at the moment when group or phone port does not get calls, Busy tone will
be transmitted to subscriber.
3.6.3.8 Cadence submenu

In the 'Cadence' submenu, you may configure an alternative call control signal (cadence) depending on
the caller number or 'Alert-Info' header value in the incoming 'Invite'. Cadence value for each call signal is
represented by the sequence of alternating pulses and pauses delimited by ',' or ';'. Pulse/pause duration should
be defined in milliseconds and divisible by 100. Minimum pulse/pause duration is 200 ms, maximum is 8,000 ms.
To map a specific cadence to 'Alert-Info' header value in the incoming 'Invite', you should select the
'Process Alert-Info header' checkbox in the respective SIP profile, and define the signal name in 'Signal name'
field in cadence settings (e.g. Example-cadence). When 'Alert-Info' header value in the incoming 'Invite' is
<http://127.0.0.1/Example-cadence>, cadence will be output into the line.
If the cadence is not found by the 'Alert-Info' header, attempt to find the cadence by the caller number
will be taken. If the latter is absent, the standard call signal with the cadence '1000,4000' will be output.
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To edit the specific signal, click the respective link in the 'Signal name' column.
To add a signal, click 'Add' button and enter the following settings:
Cadence namename of the signal.

Cadenceduration of the call voltage application to the phone unit, both values should be
divisible by 100 ms, minimum value is 200 ms, maximum is 8,000 ms.
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3.6.3.9 Call history menu
In the call history submenu, you may configure call history logging.
Call history sizemaximum number of log records, may take values from 0 to 10,000 strings.
Enter '0' value to disable call history logging. When the defined log limit is reached, each
consequent record will delete the oldest record in the beginning of the log.
Download call history fileto save 'voip_history' file on a local PC, click 'Download' button.
Clear call historyto clear call history, click 'Clear' button.
To view the call history, follow the 'View call history' link. For parameter monitoring description, see
Section 3.7.9 'Call history' submenu.
To apply a new configuration and store settings into the non-volatile memory, click 'Apply' button. To
discard changes, click 'Cancel' button.
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3.6.4 IPTV menu
3.6.4.1 IPTV submenu
In the 'IPTV' submenu, you may configure the IPTV service.
Enable IPTVwhen seleced, enable IPTV signal transmission from TAU-2M.IP WAN interface
(from the provider network) to the devices connected to LAN interface.
IGMP versionIGMP version used for IGMP message transmission from the WAN interface (IPTV
channel subscription activation or deactivation messages). Versions 2 and 3 are supported.
Renew subscription
Enablewhen chosen, periodic transmission of messages containing a list of active IPTV channels
is performed from WAN interface to the upstream server broadcasting IPTV signals. Disabling the
periodic subscription update function is required if the upstream server disables IPTV channel
broadcasting after the definite time period.
Renew subscription interval, secondstransmission interval for messages containing a list of
active IPTV channels, in seconds. Define the update rate value less than the uplink server signal
broadcasting timeout.
Fast leave mode
Enablewhen seleced, enable quick group leave mode. This function minimizes the multicast
stream switching delay (stream disconnects right after receiving 'Leave Group' message from the
client without an additional acknowledgement request). Avoid using this mode, when there are
multiple IPTV recipients connected to a single LAN port.
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VLAN IPTV
Use VLANwhen seleced, use dedicated VLAN for IPTV service (VLAN number may be the same
as for the Internet service or STB), otherwise IPTV will use the Internet service interface. This
setting allows you to configure the interface for IPTV signal reception from the external network.
VLAN IDVLAN identifier for IPTV signal reception.
802.1P802.1P marker (another name: CoS Class of Service), assigned to the outgoing frames
from this interface. It may take values from 0 (the lowest priority) to 7 (the highest priority).
Utilized by QoS algorithms.
HTTP proxy settings
Enablewhen seleced, enable HTTP proxy feature. HTTP proxy transforms UDP stream into HTTP
stream utilizing TCP (reliable packet delivery protocol) in order to improve stream image quality,
when the quality of the communication link in local area network is low.
HTTP portHTTP proxy port number that will be used for video streaming. Use this port to
connect to IPTV streams being broadcast by TAU-2M.IP.
For example, if TAU-2M.IP address on LAN interface is 192.168.0.1, proxy server port is 2354, and
the desired channel 227.50.50.100 is being broadcast to UDP port 1234, you should specify the
following stream address for VLC application:
http://@192.168.0.1:2345/udp/227.50.50.100:1234.
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3.6.5 System Menu
In the 'System' menu, you may configure settings for system, time and access to the device via various
protocols, change the device password and update the device firmware.
3.6.5.1 Time submenu

In the 'Time' submenu, you may configure time synchronization protocol (NTP).
Time Settings
Time zoneallows you to set a timezone from the list according to the nearest city in your region.
Daylight saving time enablewhen seleced, automatic daylight saving change will be performed
automatically within the defined time period.
DST startdaylight saving change starting day.
DST enddaylight saving change ending day.
DST offset (minutes)time shift period in minutes.
Enable NTPselect this checkbox to enable device system time synchronization with the
particular NTP server.
NTP servertime synchronization server IP address/domain name. You may enter server address
manually or select it from the list.
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3.6.5.2 Access submenu
In the 'Access' submenu, you may configure the device access via WEB interface, Telnet and SSH, and FTP
access to the USB storage device.
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Access ports
In this section, you may configure TCP ports for the device access via HTTP, HTTPS, Telnet, SSH.
HTTP portnumber of the port that allows for the device WEB interface access via HTTP, default
value is 80.
HTTPS portnumber of the port that allows for the device WEB interface access via HTTPS
(secure connection), default value is 443.
Telnet portnumber of the port that allows for the device WEB interface access via Telnet,
default value is 23.
SSH portnumber of the port that allows for the device WEB interface access via SSH, default
value is 22.
You may use Telnet and SSH protocols in order to access the command line (Linux console).
Username/password for console connection: admin/password.
Access to service Internet
To get device access from the Internet service interfaces, set the following permissions:
Web, external network:
HTTPwhen seleced, WAN port connection to the device web configurator is enabled via HTTP
(insecure connection);
HTTPSwhen seleced, WAN port connection to the device web configurator is enabled via HTTPS
(secure connection).
Web, internal network:
HTTPwhen seleced, LAN port (or Wi-Fi wireless access point) connection to the device web
configurator is enabled via HTTP (insecure connection);
HTTPSwhen seleced, LAN port (or Wi-Fi wireless access point) connection to the device web
configurator is enabled via HTTPS (secure connection).
Telnet:
Telnet is a protocol that allows you to establish mechanisms of control over the network. Allows you to
remotely connect to the gateway from a computer for configuration and management purposes.
To enable the device access via Telnet protocol from the external (via WAN port) or internal (via LAN port
or Wi-Fi wireless access point) network, select the appropriate checkboxes.
SSH:
SSH is a secure device remote control protocol. However, as opposed to Telnet, SSH encrypts all traffic,
including passwords being transferred.
To enable the device access via SSH protocol from the external (via WAN port) or internal (via LAN port or
Wi-Fi wireless access point) network, select the appropriate checkboxes.
Access to service VoIP:
In this section, you may configure access to VoIP service interface (to configure VoIP service interface, use
IPVoIPNetwork configuration) through the WEB (HTTP or HTTPS), and also via Telnet and SSH protocols. To
enable access to any protocols listed above, select the appropriate checkboxes.
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Access to service Management VLAN:
Use this section to configure access to the management VLAN interface that allows for device
management via HTTP, HTTPS, Telnet or SSH protocols. To configure the interface, use SystemManagement
VLAN page.To enable access to any protocols listed above, select the appropriate checkboxes.
For Telnet and SSH protocol authorization, you may use default username admin and password
password. After authorization, Linux console will become available that supports basic commands
of the 'shell' command interpreter.
Access to USB:
In this section, you may configure FTP access to the USB storage device.
To enable the access to the connected USB device via FTP protocol from the external (via WAN port) or
internal (via LAN port or Wi-Fi wireless access point) network, select the appropriate checkboxes.
To allow the anonymous user access to the USB device, select the 'Allow anonymous user access'
checkbox.
To allow the anonymous user to write data to USB device, select the 'Allow anonymous user writes'
checkbox.
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3.6.5.3 Log submenu
In the 'Log' submenu, you may configure output for various debug messages intended for device
troubleshooting. Debug information is provided by the following device firmware modules:
VoIP managerdeals with VoIP functions operations;

System managerdeals with the device configuration according to the configuration file;
Configuration managerdeals with the configuration file operations (config file reads and
writes from various sources) and the device monitoring information collection.
VoIP log
Log outputlog message output direction:

Disabledlog is disabled;
Syslogmessages are output to remote server or local file via syslog protocol (for
protocol configuration, see below);
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Consolemessages are output to the device console (requires connection via COM port
adapter);
Telnetmessages are output to the telnet session; create telnet protocol connection first.
Given below is the configuration of VoIP log message types:
Errorsselect this checkbox, if you want to collect 'Error' type messages;

Warningsselect this checkbox, if you want to collect 'Warning' type messages;
Debugselect this checkbox, if you want to collect debug messages;
Infoselect this checkbox, if you want to collect information messages;
SIP trace level defines output level of VoIP SIP manager stack messages.
Networkd log

adapter);
Given below is the configuration of system manager log message types:

Debugselect this checkbox, if you want to display debug messages;
Infoselect this checkbox, if you want to display information messages.
Configdr log

adapter);
Given below is the configuration of configd log message types:

Debugselect this checkbox, if you want to display debug messages;
Infoselect this checkbox, if you want to display information messages.
Syslog settings
If there is at least a single log (VoIP manager, system manager or configuration manager) is configured for
Syslog output, you should enable Syslog agent that will intercept debug messages from the respective manager
and send them to remote server or save them to a local file in Syslog format.
Enablewhen seleced, user Syslog agent is launched.

ModeSyslog agent operation mode:
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Serverlog information will be sent to the remote Syslog server (this is the 'remote log'
mode);
Local filelog information will be saved to the local file;
Server and filelog information will be sent to the remote Syslog server and saved to the
local file.
Next, the following settings will be available depending on the selected Syslog agent mode:
Syslog server addressSyslog server IP address or domain name (required for 'Server' mode);
Syslog server portport for Syslog server incoming messages (default value is 514; required for
'Server' mode);
File namename of the file to store log in Syslog format (required for 'File' mode);
File size, KBmaximum log file size (required for 'File' mode).
3.6.5.4 'Passwords' submenu

In the 'Passwords' submenu, you may define passwords for administrator, non-privileged user, and viewer
access.
Defined passwords allow for the device access via WEB interface and also via Telnet and SSH protocols.
When signing into WEB interface, administrator (default password: password) has the full access to the
device: read/write any settings, full device status monitoring. Non-privileged user (default password: user) may
configure network settings (except for the Internet connection settings), may access the device status
monitoring. Viewer (default password: viewer) may view the configuration and the device monitoring data only,
without any editing permissions.
Administrator login: admin

Non-privileged user login: user
Viewer login: viewer
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Administrator passwordenter administrator password in the respective fields and confirm it.
User passwordenter non-privileged user password in the respective fields and confirm it.
Viewer passwordenter viewer password in the respective fields and confirm it.
3.6.5.5 'Configuration management' submenu

In the 'Configuration management' submenu, you may save and update the current configuration.
Backup configuration
To save the current device configuration to a local PC, click 'Download' button.
Updating configuration
Restore Configurationselect configuration file stored on a local PC. To update the device
configuration, click 'Select file' button, specify a file (in .tar.gz format) and click 'Upload' button.
Uploaded configuration will be applied automatically and does not require device reboot.
Reset to default configurationto reset the device to default factory settings, click 'Reset' button.
3.6.5.6 'Firmware update' submenu

In 'Firmware update' submenu, you may update the firmware of the device.
Firmware versioninstalled firmware version.

Check for updatesclick this button to check the availability of the latest firmware version. With
this function, you may quickly check the latest firmware version and update the firmware, if
necessary.
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Firmware update check function requires Internet access.
You may update the device firmware manually by downloading the firmware file from the web site
http://eltex.nsk.ru/downloads and saving it on the computer. To do this, click the 'Select file' button in the
'Firmware update file' field, and specify path to firmware .tar.gz format file.
To launch the update process, click 'Upload file' button. The process may take several minutes (its current
status will be shown on the page). The device will be automatically rebooted when the update is completed.
Do not switch off or reboot the device during the software update.
3.6.5.7 Reboot submenu

In the 'Reboot' submenu, you may reboot the device.
Click the 'Reboot' button to reboot the device. Device reboot process takes approximately 1 minute to
complete.
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3.6.5.8 Autoprovisioning submenu
In the 'Autoprovisioning' submenu, you may configure DHCP-based autoprovisioning algorithm and TR069 subscriber device automatic configuration protocol.
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DHCP-based autoconfiguration:
Provisioningselect the device update mode; there are several options available:
Disabledautomatic updates of the device configuration and firmware are disabled;

Configuration and softwareperiodic updates of the device configuration and firmware
are enabled;
Configuration onlyonly periodic updates of the device configuration are enabled;
Firmware onlyonly periodic updates of the device firmware are enabled.
Priority fromthis parameter manages names and locations of configuration and firmware files:
Static settingspaths to configuration and firmware files are defined by the

'Configuration file' and 'Firmware file' settings accordingly; for detailed algorithm
operation, see Section 6 DEVICE AUTOMATIC UPDATE ALGORITHM BASED ON DHCP;
DHCP optionspaths to configuration and firmware files are defined by the DHCP Option
43, 66, and 67 (to do this, you should select DHCP for the Internet service); for detailed
algorithm operation, see Section 6 DEVICE AUTOMATIC UPDATE ALGORITHM BASED ON
DHCP.
Configuration filefull path to configuration filedefined in URL format (at this time, you may
upload configuration files via TFTP and HTTP protocols):
tftp://<server address>/<full path to cfg file>
http://<server address>/<full path to cfg file>
where< server address > is HTTP or TFTP server address (domain name or IPv4).
<full path to cfg file >full path to configuration file on server.
Configuration update interval, secondstime period in seconds that will be used for periodic
device configuration update; if 0 is selected, device will be updated only onceimmediately after
startup.
Firmware filefull path to firmware filedefined in URL format (at this time, you may upload
firmware files via TFTP and HTTP protocols):
tftp://<server address>/<full path to firmware file>
http://<server address>/<full path to firmware file>
where< server address > is HTTP or TFTP server address (domain name or IPv4).
<full path to firmware file >full path to firmware file on server.
Firmware update interval, secondstime period in seconds that will be used for periodic device
firmware update; if 0 is selected, device will be updated only onceimmediately after startup.
For detailed DHCP-based automatic update algorithm, see Section 6 DEVICE AUTOMATIC UPDATE
ALGORITHM BASED ON DHCP.
TR-069 protocol autoconfiguration:
Common:
Enable TR-069 clientwhen seleced, integrated TR-069 protocol client will be enabled;
Interfaceselect the interface for TR-069 protocol operation. If 'Management VLAN' interface is
enabled on the gateway, this VLAN will be used for TR-069 protocol operation automatically.
Interface selection setting will be disabled;
Interfacean interface, that will be used for the device autoconfiguration;
ACS server addressautoconfiguration server address. Enter address in the following format:
http://<address>:<port> or https://<address>:<port> (<address> ACS server IP address or domain
name, <port>ACS server port, default value is 80). Alternatively, the client will exchange the data
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with ACS server via the secure protocolHTTPS. By default, ACS server produced by Eltex utilizes port
9595 for communication;
Enable periodic informwhen selected, integrated TR-069 client performs periodic ACS server polling
at intervals equal to 'Polling period' value, in seconds. Goal of the polling is to identify possible
changes in the device configuration.
ACS connection request:
Username, passwordusername and password used by client to access ACS server.
Client connection request:

Username, passwordusername and password used by ACS server to access TR-069 client.
NAT settings
If there is a NAT (network address translation) between the client and ACS server, ACS server may not be
able to establish the connection to client without specific technologies intended to prevent such situations.
These technologies allow the client to identify its so called public address (NAT address or in other words
external address of a gateway, that covers the client). When public address is identified, the client reports it to
the server that uses this public address for establishing connection to the client in the future.
NAT modeidentifies the method, that will be used by client for obtaining its public address
information. The following modes are possible:
STUNuse STUN protocol for public NAT address discovery;

Manualmanual mode, when public address is explicit in configuration; in this mode, you
should add a forwarding rule on a device that acts as a NAT for TCP port used by TR-069
client;
OffNAT will not be usedthis mode is recommended only when the device is directly
connected to ACS server without network address translation. In this case public address
will match local client address.
When choosing STUN mode, you should define the following settings:
STUN server addressSTUN server IP address or domain name;

STUN server portSTUN server UDP port (default value is 3478);
Minimum keep alive period, seconds and Maximum keep alive period, secondsdefine the time
interval in seconds for periodic transmission of messages to STUN server in order to identify
public address modification.
When Manual mode is selected, client public address should be entered manually with NAT address
setting (in IPv4 format).
For correct ACS server operation behind the NAT, STUN server minimum polling period should be
less than maximum session time provided by the NAT device.
TR-069 protocol allows for comprehensive device configuration, software updates, reading device
information (software version, model, serial number, etc.), complete configuration file downloading/uploading,
remote device restart (TR-069, TR-098, TR-104 specifications are supported).
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3.6.5.9 'Management interface' submenu
Use this menu to configure the network interface and establish the device network management via
HTTP, HTTPS, Telnet or SSH protocols
Enable management interfacewhen seleced, device management will be performed via this
interface.
Access typedefines interface operation mode.
Taggeddata is transferred by the interface with the defined VLAN ID.
Untaggeddata is transferred by the interface without VLAN.
VLAN IDidentifier for interface definition into virtual local area network.
802.1P802.1P marker (another name: CoS Class of Service), assigned to the outgoing frames
from this interface. It may take values from 0 (the lowest priority) to 7 (the highest priority).
Protocolselect address assigning protocol for the interface:
Staticoperation mode where IP address and all the necessary settings for WAN
interface are assigned manually. When 'Static' type is selected, the following parameters
IP addressspecify the IP address for the management interface;

Subnet masksubnet mask for the management interface;
Default gatewaydefault gateway IP address for the management interface;
Primary DNS, Secondary DNSDNS server IP addresses required for the gateway
autoconfiguration protocols' operation; to configure protocols, use System
Autoconfiguration page.
DHCPoperation mode where IP address, subnet mask, DNS address and other necessary
settings for the interface operation (e.g. static routes) are automatically obtained from
DHCP server. If you are unable to obtain DNS server addresses from the provider, you
may specify them manually using 'Primary DNS' and 'Secondary DNS' fields. Manually
defined addresses will have a priority over DNS addresses obtained via DHCP.
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Alternative Vendor ID (Option 60)when seleced, the device transmits Vendor ID
(Option 60) field value in Option 60 DHCP messages (Vendor class ID). If the field
is empty, Option 60 will not be transmitted in DHCP messages.
address][VERSION:software version]
Example:
[VENDOR:Eltex][DEVICE:TAU-1M.IP][HW:1.0][SN:VI23000118][WAN:A8:F9:4B:03:2A:D0]
[LAN:02:20:80:a8:f9:4b][VERSION:#1.12.0]
3.6.5.10 Advanced submenu

In 'Advanced' submenu, you may enable UPnP.
Enable UPnPwhen seleced, UPnP will be active. UPnP is used by some applications (for example,
DC clients, such as FlylinkDC++) for automatic creation of forwarding rules for TCP/UDP ports used
by these applications on the uplink router. We recommend turning UPnP on in order to enable file
sharing services within the network.
Reserved VLAN ID
Reserved VLAN IDs are required for solving intrasystem tasks of the gateway and may be changed
depending on the VLAN ID being used for the system:
Start VLAN IDstarting VLAN ID value in the reserved range, may take values in range [1-4090];
End VLAN IDending VLAN ID value in the reserved range. This setting is unavailable for editing
and calculated automatically.
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3.7 System monitoring

To enter the 'System monitoring' mode, select 'Monitoring' from the left-hand side panel.
Some pages does not feature automatic update of the device monitoring data. To obtain the
current information from the device, click
button.
3.7.1 Internet submenu

In the 'Internet' submenu, you may view basic network settings of the device.
Internet connection
Network connectionmethod of connection to data network. To configure the connection, use

NetworkInternet section.
Wiredconnection to the provider network is established through the WAN port using
copper-wire or optical patch cord.
3G/4G USB modemconnection to the provider network is established using 3G/4G USB
modem connected to USB port on the device rear panel.
Access protocolprotocol used for the Internet access.

IP addressdevice IP address in the external network.
IP address in the internal provider networkIP address used within the internal provider network
for access to local network resources of the provider.
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3.7.2 'VoIP' submenu
In 'VoIP' submenu, you may view VoIP network interface status, monitor subscriber units, test lines, and
monitor IMS.
Status of VoIP network interface
IP addressIP address for VoIP service network interface.
FXS status
Linedevice subscriber unit number.

Local numbersubscriber phone number assigned to this subscriber port.
Registrationstate of registration on proxy server for the group phone number:
DisabledSIP server registration function is disabled in SIP profile settings,

Errorregistration was unsuccessful,
Completedregistration on SIP server successfully completed.
Expires inexpiration time of subscriber port registration on SIP server.

Server addressaddress of the server that the subscriber line has been registered at for the last
time.
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Line statusphysical line status. The line may have one of the following states:
Call state 1, 2each subscriber port may support up to 2 communication sessions simultaneously.
In this field you may see the state of the call with the respective remote subscriber. The call may
have one of the following states:
Inactivethe phone handset is on-hook (or subscriber port is disabled), normal operation,
Activethe phone handset is off-hook; 'PBX response' tone, ringback tone, or error tone
is transmitted into the line, or the line is in the call state,
Ringing toneringing tone is supplied to the phone headset (during the incoming call),
Testline testing is started.
Dialling numbercall is being dialled from the phone unit,

Busycall has cleared back for some reason, busy tone is transmitted into the line,
Outgoing callremote subscriber is being dialled, ringback tone is transmitted into the
line,
Incoming callincoming call is being received at the phone port, ringing tone is
transmitted into the line,
Conversationvoice connection is established with the remote subscriber,
Opposite on holdremote subscriber is on hold,
Local on holdlocal subscriber has been put on hold by the remote subscriber,
Error, put phone onhookerror tone is transmitted into the line. As a rule, error tone is
played on the busy signal timeout expiration (configured separately for each line), when
the subscriber doesn't put the phone onhook.
Remote user 1, 2remote subscriber phone number for each communication session.
Line testclick the 'Test' button to launch the subscriber line testing. Process status is
represented by a countdown timer (in the 'Line state' column) indicating the remaining test time.
You cannon run the test for multiple lines simultaneously. Test duration80 seconds. Subscriber
unit is blocked for the time of the testit will not be able to make or receive calls.
To see the results when the test finishes, click

button in the 'Line test' column. Results are
represented in the table mode and contain the following data:
Test date
TIP wire constant extraneous voltage
RING wire constant extraneous voltage
Line power voltage
Voltage between TIP and RING wires
Voltage between TIP wire and ground
Voltage between RING wire and ground
Capacity between TIP and RING wires
Capacity between TIP wire and ground
Capacity between RING wire and ground
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Line 1 test result example:
Under the Subscriber set monitoring table, you may find buttons for forced registration or de-registration
of the selected lines.
Hunt groups status
Group name call group name;

State call group status: enable or disable;
Phone phone number assigned to a call group;
Line list (), ;
Registrationstate of registration on proxy server for the group phone number:
DisabledSIP server registration function is disabled in SIP profile settings,

Errorregistration was unsuccessful,
Completedregistration on SIP server successfully completed.
Expires inexpiration time of call group registration on SIP server.

Server addressaddress of the server that the call group has been registered at for the last time.
IMS monitoring
IMS monitoring shows the state (active or inactive) for some services on the subscriber line provided that
the remote control from IMS server is enabled for this line (IP Multimedia Subsystem).
IMSmanagementshows whether the subscriber line service remote control from IMS server is
enabled (configured in SIP profile, see 'Profiles' submenu).
Three-party conferenceshows whether the 'Three-party conference' service activation
command is received from IMS server.
Call holdshows whether the 'Call hold' service activation command is received from IMS server.
Call waitingshows whether the 'Call waiting' service activation command is received from IMS
server.
Hotlineshows whether the 'Hotline' service activation command is received from IMS server.
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Hotline numbershow the 'Hotline' service phone number in the activation command from IMS
server.
Hotline timeout, secondsshow the dialling timeout for the 'Hotline' service in the activation
command from IMS server.
XCAP name for call transfershows the defined name of the 'Call transfer' service.
service is active.
service is inactive.
3.7.3 'Ethernet ports' submenu

In the 'Ethernet ports' submenu, you may view the device Ethernet port state..
Ethernet port status
Portname of the port.
Connectionstatus of the connection to this port:
Enablednetwork device is connected to the port (link is active),

Disablednetwork device is not connected to the port (link is inactive).
Speeddata rate of the external network device connected to the port (10/100/1000Mbps).
Modedata transfer mode:
WANexternal network port.

LAN 1..2local area network port.
Full-duplexfull duplex,
Half-duplexhalf-duplex.
Transmittedquantity of bytes sent from the port.

Receivedquantity of bytes received by the port.
To obtain actual information on the Ethernet port state, click 'Update' button.
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3.7.4 DHCP submenu
In the 'DHCP' submenu, you may view the list of network devices connected to the LAN interface, that
were assigned IP addresses by a local DHCP server, and also the IP address lease expiration time.
List of DHCP clients
MAC addressconnected device MAC address.

Client nameconnected device network name.
IP addressIP address assigned to the client from the address pool.
Lease expiresremaining time of the assigned address lease.
To obtain actual information on DHCP clients, click 'Update' button.
3.7.5 ARP submenu

In the 'ARP' submenu, you may view an ARP table. In ARP table, you may find information on IP and MAC
address correspondence for neighbouring network devices. .
ARP table
IP addressdevice IP address.
MAC addressdevice MAC address.
Client nameconnected device network name.
Interfaceinterface of the device active side: WAN, LAN, Bridge.
To obtain actual information, click 'Update' button.
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3.7.6 Device submenu
In the 'Device' submenu, you may find general device information.
Device Information
Productdevice model name.

Firmware versiondevice firmware version.
Factory addressdevice WAN interface address defined by the manufacturer.
Serial numberdevice serial number defined by the manufacturer.
System timecurrent date and time defined in the system.
Uptimetime of operation since the last startup or reboot of the device.
3.7.7 Conntrack submenu

In the 'Conntrack' submenu, you may find the current active network connections of the device.
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Active NAT session
Count of active connectiontotal number of active network connections.

Count of shown connection number of connections shown in the WEB interface. In order to
maintain high performance of the WEB interface, maximum number of connections shown is
limited to 1024. You may view other connections with the device command console (cat
/proc/net/nf_conntrack).
List of connections
Protocolprotocol that the connection is being established through.

Source addressconnection initiator IP address and port number.
Destination IPconnection destination IP address and port number.
Timeouttime period until the connection termination.
3.7.8 Static routes submenu

In the 'Static routes' submenu, you may view the device routing table.
DestinationIP address of destination host or subnet that the route is established to.
Gatewaygateway IP address that allows for the access to the 'Destination'.
Genmask subnet mask.
Flagsspecific route attributes. The following flag values exist:
Umeans that the route is created and passable;

Hidentifies the route to the specific host;
Gmeans that the route lies through the external gateway. System network interface
provides routes in the network with direct connection. All other routes lie through the
external gateways. 'G' flag is user for all routes except for the routes in the direct
connection networks;
Rmeans that the route most likely was created by a dynamic routing protocol running
on a local system with the 'reinstate' parameter;
Dmeans that the route was added on reception of the ICMP Redirect Message. When
the system learns the route from the ICMP Redirect message, the route will be added into
the routing table in order to exclude redirection of the following packets intended for the
same destination. Such routes are marked with the 'D' flag;
means that the route was modifiedlikely by a dynamic routing protocol running on
a local system with the 'mod' parameter applied;
means buffered route with corresponding record in the ARP table;
means that the route source in the core routing buffer;
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Lmeans that the route destination is an address of this PC. Such 'local routes' exist in
the routing buffer only;
means that the route destination is a broadcasting address. Such 'broadcast routes'
exist in the routing buffer only;
Imeans that the route is related to the loopback interface with a goal that differs from
the access to the ring network. Such 'internal routes' exist in the routing buffer only;
! means that datagrams sent to this address will be rejected by the system.
Metricdefines route cost. Metrics allows you to sort the duplicate routes, if they are exist in the
table.
Refidentified number of references to the route for connection establishment (not used by the
system).
Usenumber of route detections performed by IP protocol.
Interfacename of the network interface that the route lies through.
3.7.9 'Call history' submenu

In the 'Call history' submenu, you may view the list of performed phone calls and the summary for each
call.
The device RAM may store up to 10,000 records for performed calls. If the record number exceeds 10,000,
the oldest records (at the top of the table) will be removed, and new ones will be added at the end of the file.
Log statistics will not be collected, when the history size is zero.
'Call history' table field description:
sequence number of the record in the table,

Linedevice subscriber port number,
Localsubscriber number assigned to this subscriber port,
Remoteremote subscriber number that the phone connection has been established with,
Remote hostremote subscriber IP address that the phone connection has been established with,
Start call timecall received/performed time and date,
Call talk timeCall start time and date,
Talk durationcall duration in seconds,
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Statetransient state or reason for call clearing; description becomes available, when you hover
the cursor over the call state record,
typecall type: outgoing or incoming,
Packets sentnumber of RTP packets sent during the call.,
Bytes sentnumber of bytes sent during the call,
Packets receivednumber of RTP packets received during the call,
Bytes receivednumber of bytes received during the call.
In the call history table, you may search records by different parameters; to do this, click the 'Filter
(Show)' link. Filtering may by performed by the subscriber line address, local or remote number, opposite side IP
address, call received time, call start time, call state and call type. For filtering parameter description, see call
history table field description above.
Call received time from/to or Call start time from/tocall received/performed time period or call start
time period in the 'hh:mm:ss dd.mm.yyyy' format.
To hide the table record filtration parameter settings, click the 'Hide filter' link.
To configure call history parameters, click 'Configure call history parameters' link. For detailed parameter
configuration description, see Section 3.6.3.9 Call history menu .
Click
button to proceed to the table showing the first record.
Click
button to proceed to the previous page with the call history table.
Click
button to proceed to the next page with the call history table.
Click
button to proceed to the table showing the last record.
Use 'Records on page' selector to configure the number of table records displayed on a single page.
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3.8 Configuration example

1. Connect a PC to one of the device LAN ports, connect provider network cable to the WAN port.
2. Enter the gateway IP address in the browser address bar (default value: 192.168.1.1).
3. When the device connection is successfully established, you will see the login and password request
window. Fill in the fields and click 'Sign in' button (default login: admin, default password: password).
If this field is not visible, make sure that the automatic IP address obtaining feature is enabled in the
network connection settings on your PC.
4. In the 'Internet' tile, you may configure an external connection. If TAU-2M.IPis intended to be used as a
routerselect the 'Router' value in the 'Operation mode' field of the 'Internet' tile. In 'Protocol' field, select the
protocol used by your ISP and enter all the necessary data according to the provider's instructions. If you are
required to use static settings in order to connect to the provider network, select 'Static' value in the 'Protocol'
field and fill the 'Device external IP address', 'Subnet mask', 'Default gateway', 'Primary DNS', and 'Secondary
DNS' fieldsparameter values should be obtained from the provider. To save and apply settings, click
button.
To specify additional parameters, go to advanced settings mode by clicking the 'Details' link (see Section
3.6.2.1. Internet submenu ).
5. When your ISP network employs MAC address tethering, click 'Details' button in the 'Internet' tile and
open the 'MAC address configuration' submenu. In the 'WAN MAC address configuration' section, select the
'Redefine ' checkbox and enter MAC address of the device previously connected to the Internet into the
'MAC' field. If you have connected a PC that is used for device configuration at the moment, just click 'Clone'
button to assign MAC address of your PC to the gateway. To save and apply settings, click 'Apply' button.
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If TAU-2M.IP is intended to be used as a 2-port switchselect the 'Bridge' value in the 'Work mode' field
of the 'Internet' tile. In 'IP address' field, specify an address that will be used for the device access. Define subnet
mask (default: 255.255.255.0). To save and apply settings, click
button.
In the 'Bridge' mode, the gateway will not assign IP addresses automatically using DHCP to devices
connected to the LAN interface.
6. In the 'VoIP' tile, you may perform quick configuration of the subscriber line for operation via SIP. To do
this, select 'Line 1' tab. Select 'Enable' checkbox, enter the phone number for this line, and login and password
for SIP server authorization. To save and apply settings, click
button.
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In the same way, you may configure subscriber line in the next Line tab.
7. Select 'SIP' tab in the 'VoIP' tile to configure SIP parameters. Specify IP address or domain name of the
SIP server and registration server (if necessary) in the corresponding fields. If ports used by servers differ from
5060, specify alternative ports after the colon. Specify SIP domain, if necessary. Select 'Registration' checkbox, if
the subscriber is required to register on SIP server for VoIP operation (as a rule, registration is required). To save
and apply settings, click
button.
To specify additional parameters, go to advanced settings mode by clicking the 'Details' link (see Section
3.6.3 VoIP Menu).
8. If you are planning to use IPTV, select 'Enable IPTV' in the 'IPTV' tile. To enable transmission of IPTV
streams via HTTP, select 'Enable HTTP proxy' checkbox. In the 'HTTP port' field, specify the port that will be used
for connection of external devices to a local HTTP proxy. To save and apply settings, click
button.
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If IPTV service require a dedicated VLAN, go to advanced settings mode by clicking the 'Details' link and
specify VLAN ID in the respective field.
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4 VALUE ADDED SERVICE USAGE

4.1 Call transfer
Call transfer service may be performed locally using gateway resources or using resources of a
communicating device. If the service is performed using resources of a communicating device, the access to 'Call
transfer' service is established via subscriber port settings menu'VoIP' -> 'Line configuration'by selecting
'Transmit Flash' value in 'Flash operation mode' field. Service process logics in this case will be defined by the
communicating device.
When 'Call transfer' service is performed locally using gateway resources, the access to this service is
established via subscriber port settings menu'VoIP' -> 'Line configuration'by selecting 'Attended calltransfer'
or 'Unattended calltransfer' in 'Flash operation mode' field.
'Attended calltransfer' service allows you to temporarily disconnect an online subscriber (Subscriber A),
establish connection with another subscriber (Subscriber C) and return to the previous connection without
dialling or transfer the call while disconnecting Subscriber B.
'Attended calltransfer' service usage:
While being in a call state with the subscriber A, put him on hold with short clearback flash (R), wait for
station response signal and dial subscriber C number. When Subscriber C answers, the following operations will
be possible:
R 0disconnect a subscriber on hold, connect to online subscriber;

R 1disconnect an online subscriber, connect to subscriber on hold;
R 2switch to another subscriber (change a subscriber);
R 3conference;
R 4call transfer; voice connection will be established between Subscribers A and C;
R 4call transfer; voice connection will be established between Subscribers A and C;
R clearbackcall transfer; voice connection will be established between Subscribers A and C.
Figure below shows 'Attended calltransfer' service operation algorithm:
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'Unattended calltransfer' service allows you to put an online subscriber (Subscriber ) on hold with a short
clearback flash and dial another subscriber's number (Subscriber ). Call will be transferred automatically when
Subscriber B finishes dialling the number.
Figure below shows 'Unattended calltransfer' service operation algorithm.
Local calltransfer service allows you to transfer call through the gateway without sending the external
REFER message if C subscriber is a local TAU-2M.IP client and its call was directly perfomed with ignoring of the
proxy server. If C subscriber is external or local client which was called via proxy server, Local calltransfer
service operates like Attended calltransfer and call transfer is performed by sending REFER message to the B
subscriber.
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4.2 Call Waiting

This service allows you to inform "busy" subscribers about new incoming calls with a special signal.
Upon receiving this notification, user can answer or reject waiting call.
Access to this service is established via subscriber line settings menu by selecting 'Attended calltransfer',
or 'Unattended calltransfer' in the 'flash operation mode' field and selecting 'Call waiting' checkbox.
Service usage:
If you receive a new call while being in a call state, you may do the following:
R 0reject a new call;

R 1answer the waiting call;
R 2switch to a new call (change a subscriber);
Rshort clearback (flash).
4.3 Three-way conference call

Three-way conference is a service, that enables simultaneous phone communication for 3 subscribers. Press
R 3 keys to enter the conference mode (see Section 4.1 Call transfer ).
Subscriber that started the conference is deemed to be it's initiator, two other subscribers are the
participants.
There are two operation modes for a three-way conference: local mode and remote mode. In the first
mode, the conference is assembled locally by the initiating subscriber; in the second mode, the conference is
established remotely by a remote server, also known as the conference server.
4.3.1 Local conference

In the conference mode, short clearback 'flash' pressed by the initiator is ignored. Signalling protocol
messages, received from the participants and intended to put the initiator side into hold mode, force this
participant to leave the conference. At that, the initiator and the second participant will switch into the ordinary
two-party call mode.
The conference terminates, when initiator leaves; in this case, both participants will receive clearback
message. If one of the participants leaves the conference, the initiator and the second participant will switch
into a standard two party call. Short flash clearback is processed as described in Sections 4.1 Call transfer 4.2
Call Waiting.
Figure below shows an algorithm of '3-way conference' service performed locally by the Subscriber B via SIP
protocol.
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4.3.2 Remote conference

Conference operation complies with the algorithm described in RFC4579. The special aspect of the
algorithm is that the initiating subscriber should press flash+3 in order to establish connection with the
conference server (also called 'focus'), and request the focus to connect to remaining conference
participants. The figure below shows detailed operation algorithm.
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5 CONNECTION ESTABLISHMENT ALGORITHM

5.1 Algorithm of a Successful Call via SIP Protocol
SIP (Session Initiation Protocol) is a session initiation protocol, that performs basic call management tasks
such as starting and finishing session.
SIP defines 3 basic connection initiation scenarios: between users, involving proxy server, involving
forwarding server. Basic connection initiation algorithms are described in IETF RFC 3665. This section describes
an example of a connection initiation scenario via SIP between two gateways, that know each other IP addresses
in advance.
Algorithm description:
1. Subscriber A rings up Subscriber B.
2. Subscriber B gateway receives the command for processing.
3. Subscriber B is free. In this moment, 'ringing' tone is sent to the Subscriber B phone, and
'ringback' tone to Subscriber A phone.
4. Subscriber B answers the call.
5. Subscriber A gateway confirms session establishment.
6. Subscriber A clears back, 'busy' audio tone is sent to the Subscriber B.
7. Subscriber B gateway confirms received clearback command.
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5.2 Call Algorithm involving SIP Proxy server

This section describes a connection initiation scenario between two gateways involving SIP proxy server.
In this case, caller gateway (Subscriber A) should know subscriber's permanent address and proxy server IP
address. SIP proxy server processes messages received from Subscriber A, discovers Subscriber B, prompts the
communication session and performs router functions for two gateways.
Registration at the SIP server.
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.
16.
Subscriber A and Subscriber B register at SIP server.

SIP server prompts for authorization.
Subscriber A and Subscriber B register at SIP server with authorization.
SIP server responses on successful registration.
Subscriber A rings up Subscriber B.
SIP server requests authentication.
Subscriber A gateway confirms received authorization request command.
Subscriber A rings up Subscriber B.
SIP server receives the command for processing.
SIP server translates Subscriber A call request directed at Subscriber B.
Subscriber B gateway receives the command for processing.
Subscriber B is free. In this moment, 'ringing' tone is sent to the Subscriber B phone, and
Subscriber B answers the call.
Subscriber A gateway confirms session establishment.
Subscriber A clears back, 'busy' audio tone is sent to the Subscriber B.
Subscriber B gateway confirms received clearback command.
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5.3 Call Algorithm Involving Forwarding Server

This section describes a connection initiation scenario between two gateways involving forwarding server.
In this case, caller gateway (Subscriber A) establishes connection unassisted, and the forwarding server only
translates callee permanent address into its current address. Subscriber obtains forwarding server address from
the network administrator.
1. Subscriber A rings up Subscriber B. Call is sent to the forwarding server with the callee address
information.
2. Forwarding server receives the command for processing.
3. Forwarding server requests the information on the Subscriber B current address from the location
server. Received information (the callee current address and the list of callee registered
addresses) is sent to Subscriber A in '302 moved temporarily' message.
4. Subscriber A gateway confirms the reception of reply from the forwarding server.
5. Subscriber A rings up Subscriber B directly.
6. Subscriber B gateway receives the command for processing.
7. Subscriber B is free. In this moment, 'ringing' tone is sent to the Subscriber B phone, and
8. Subscriber B answers the call.
9. Subscriber A gateway confirms session establishment.
10. Subscriber A clears back, 'busy' audio tone is sent to the Subscriber B.
11. Subscriber B gateway confirms received clearback command.
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6 DEVICE AUTOMATIC UPDATE ALGORITHM BASED ON DHCP
Device automatic update algorithm is defined by the 'Parameter priority from' value.
1. If the 'Static settings' value is selected, then the full path (including access protocol and server address)
to configuration file and firmware file will be defined by 'Configuration file' and 'Firmware file'
parameters. Full path should be specified in URL format (HTTP and TFTP are supported):
<protocol>://<server address>/<path to file>, where
<protocol>protocol used for downloading corresponding files from the server (HTTP and TFTP
are supported).
<server address>address of the server with a file to be downloaded (domain name or IPv4).
<path to file>path to file on the server.
You may use the following wildcards in URL (reserved words substituted with the specific values):
$MAMAC addressthis wildcard in file URL is substituted by the native device MAC address;
$SNSerial numberthis wildcard in file URL is substituted by the native device serial number;
$PNProduct namethis wildcard in file URL is substituted by the model name (e.g, TAU-2M.IP);
$SWVERSoftware versionthis wildcard in file URL is substituted by the firmware version
number;
$HWVERHardware versionthis wildcard in file URL is substituted by the device hardware
version number.
For MAC address, serial number and model name, see 'Device' section on the monitoring page.
URL examples:
tftp://download.server.loc/firmware.file,
http://192.168.25.34/configs/tau2m/my.cfg,
tftp://server.tftp/$PN/config/$SN.cfg,
http://server.http/$PN/firmware/$MA.frm, etc.
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At that, some URL parameters may be omitted. For example, configuration file may be specified in the
following format:
http://192.168.18.6
or
config_tau2m.cfg
If the system is unable to extract the necessary file downloading parameters (protocol, server address
or path to file on server) from configuration file or firmware file URL, it will attempt to extract an unknown
parameter from DHCP Option 43 (Vendor specific info) or 66 (TFTP server) and 67 (Boot file name), when
address obtaining via DHCP is enabled for the Internet service (DHCP option format and analysis will be
provided below). If the system is unable to extract missing parameter from DHCP options, default value will
be used:
Protocol: tftp
Server address: update.local
Configuration file name: tau2m.cfg
Firmware file name: tau2m.fw
Thus, if you leave 'Configuration file' and 'Firmware file' fields empty, and Options 43 or 66, 67 with
file locations are not obtained via DHCP, configuration file URL will be as follows:
tftp://update.local/tau2m.cfg,
and the firmware file URL:
tftp://update.local/tau2m.fw.
2. If 'DHCP options' value is selected, configuration file and firmware file URLs will be extracted from DHCP
Option 43 (Vendor specific info) or 66 (TFTP server) and 67 (Boot file name), wherefore address
obtaining via DHCP should be enabled for the Internet service (DHCP option format and analysis will be
provided below). If DHCP options fail to provide some of the URL parameters, default parameter value
will be used.
Protocol: tftp
Server address: update.local
Configuration file name: tau1m.cfg
Firmware file name: tau1m.fw
Option 43 format (Vendor specific info)

1|<acs_url>|2|<pcode>|3|<username>|4|<password>|5|<server_url>|6|<config.file>|7|<firmware.file
>|8|<vlan_tag>
1TR-069 autoconfiguration server address code.
2'Provisioning code' parameter specification code.
3code of the username for TR-069 server authorization.
4code of the password for TR-069 server authorization.
5server address code; server address URL should be specified in the following format: tftp://address
or http://address. The first version represents TFTP server address, the second versionHTTP server address.
6configuration file name code.
7firmware file name code.
8VLAN tag code for management.
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'|'mandatory separator used between codes and suboption values.
Algorithm of identification for configuration file and firmware file URL parameters from DHCP Options
43 and 66.
1.
DHCP exchange initialization
Device initializes DHCP exchange after the startup.

2.
Option 43 analysis
When Option 43 is received, codes 5, 6, and 7 suboptions are analyzed in order to identify the server
address and the configuration and firmware file names.
3.
Option 66 analysis
If Option 43 is not received from DHCP server or it is received but the system fails to extract the server
address, Option 66 will be discovered. If the system fails to obtain the firmware file name, Option 67 will be
discovered. They are used for TFTP server address and the firmware file path extraction respectively. Next,
configuration and firmware files will be downloaded from Option 66 address via TFTP.
Special aspects of configuration updates.
Configuration file should be in.tar.gz format (this format is used when configuration is saved from the
web interface in the 'System' - 'Configuration management' tab). Configuration downloaded from the server
will be applied automatically and does not require device reboot.
Special aspects of firmware updates.
Firmware file should be in.tar.gz format. When the firmware file is loaded, the device unpacks it and
checks its version (using 'version' file in tar.gz archive).
If the current firmware version matches the version of the file obtained via DHCP, firmware will not
be updated. Update is performed only when firmware versions are mismatched. When the firmware image
is written into the device flash memory, the Power indicator will flash green, orange and red in succession.
Do not power off or reboot the device, when the firmware image is written into the flash memory.
These actions will interrupt the firmware update, that will lead to the device boot partition
corruption. The device will become inoperable. To restore the device operation, use the instruction
provided in Section 7.
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7 SYSTEM RECOVERY AFTER FIRMWARE UPDATE FAILURE

If the failure occurred during the firmware update (via Web interface or DHCP-based automatic update)
for example, you have pressed power button by accidentand the device became inoperable (Power LED is
solid red), use the following device recovery algorithm:
Extract the contents of the firmware archive.

Connect your PC to the device WAN port and specify the address for the network interface from
192.168.1.0/24 subnet.
Launch TFTP client on the PC (for Windows, we recommend using Tftpd32), specify 192.168.1.6 as
the remote host address and select linux.bin file from the extracted firmware archive.
Run the command to send the file to the remote host (Put command). File transfer to TAU-2M.IP
should start.
If the file transfer process is started, wait until it finishes, after that TAU-2M.IP will write the
firmware into the memory and launch the system automatically. Write time is approximately 5
minutes. When the device is successfully restored, the Power LED will be orange or green. Device
will retain the configuration that was used before the failure. If the device is unreachable, reset
the device to default settings.
If the file transfer is not initiated, check the PC network settings for errors and try again. If you
are unable to restore the device, send it to the service centre for repairs or connect it to the
device via COM port using special adapter (if available).
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APPENDIX . CALCULATION OF PHONE LINE LENGTH

Table Electrical resistance/cable type relationship for 1km of DC subscriber cable lines at 20 ambient
temperature.
Cable grade for subscriber
lines of local exchange
network
TPP, TPPep, TPPZ, TPPepZ,
TPPB,TPPepB, TPPZB, TPPBG,
TPPepBG, TPPBbShp,
TPPepBbShp, TPPZBbShp,
TPPZepBbShp, TPPt
TPV, TPZBG
TG, TB, TBG, TK
TStShp, TAShp
TSV
KSPZP
KSPP, KSPZP, KSPPB,
KSPZPB, KSPPt, KSPZPt,
KSPZPK
Core diameter,
mm
Electrical
resistance
of 1km circuit, ,
max.
Line length
(other phone
units), km
Line length
('Rus' phone
units), km
0,32
0,40
0,50
0,64
0,70
0,32
0,40
0,50
458,0
296,0
192,0
116,0
96,0
458,0
296,0
192,0
3,537
5,473
8,438
13,966
16,875
3,537
5,473
1,528
2,365
3,646
6,034
7,292
1,528
2,365
8,438
3,646
0,64
116,0
0,70
0,40
96,0
296,0
13,966
16,875
6,034
7,292
0,50
0,64
0,70
0,50
0,70
0,40
0,50
0,64
0,90
192,0
116,0
96,0
192,0
96,0
296,0
192,0
116,0
56,8
5,473
8,438
13,966
16,875
8,438
16,875
5,473
8,438
13,966
28,521
2,365
3,646
6,034
7,292
3,646
7,292
2,365
3,646
6,034
12,324
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APPENDIX B. RUNNING USER-DEFINED SCRIPT UPON SYSTEM STARTUP

Sometimes, it is necessary to perform specific actions on the device startup, that may not be specified in
the configuration file settings. For this purpose, TAU-2M.IP allows you to set the user-defined script in the
configuration file. This script may feature any desired sequence of commands.
For user-defined script execution, use the following settings section in the configuration file:
UserScript:
Enable: "0"
URL: ""
'Enable' option allows (if the value is 1) or denies (if the value is 0) execution of the script which path is
specified in the URL parameter.
Executed script may be located on the remote server or on the device itself. The script may be
downloaded from the remote server via HTTP or TFTP. Consider configuration file examples for user-defined
script execution from various sources.
1. Execution from HTTP server
To execute the script from HTTP server, you should specify full path to file in HTTP-URL format
within URL parameter:
URL: http://192.168.0.250/user-script/script.sh
After the device startup, script.sh file located in the 'user-script' folder at 192.168.0.250 will be
downloaded automatically via HTTP from the server and executed afterwards.
2. Execution from TFTP server
To execute the script from TFTP server, you should specify full path to file in TFTP-URL format within
URL parameter:
URL: tftp://192.168.0.250/user-script/script.sh
After the device startup, script.sh file located in the 'user-script' folder at 192.168.0.250 will be
downloaded automatically via TFTP from the server and executed afterwards.
3. Local script execution
Due to file system specifics, local script should be located in the /etc/config folder only, as the
contents of this folder are the only one that remain after the device reboot. Script in /etc/config
folder may be created either with vi editor, or downloaded from the external TFTP server (using 'tftp
gluser.sh<TFTP-server address>' command). After creation of the script, you should set execution
permissions with 'chmod 777 /etc/config/user.sh' command.
In the configuration file, local script execution URL should be as follows:
URL: File://etc/config/user.sh
It is important to note, that the user script should begin with the '#!/bin/sh' directive.
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TECHNICAL SUPPORT
Contact Eltex Service Centre to receive technical support regarding our products:
29v Okruzhnaya Street, Novosibirsk, Russian Federation, 630020

Technical support phone numbers:
+7(383) 274-47-87,
+7(383) 272-83-31,
E-mail: [email protected]
Visit Eltex official website to get the relevant technical documentation and software, benefit from our
knowledge base, send us online request or consult a Service Centre Specialist in our technical forum.
http://www.eltex.nsk.ru/en/support/downloads/
http://www.eltex.nsk.ru/en/search/
http://www.eltex.nsk.ru/en/support/knowledge/
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Notes and warnings

2.2 Device specification

Gateway is powered by external 12V DC adapter for 220V electrical networks.

Operation in bridgeor router mode;

IPTV function support;

Wi-Fi adapters are not supported in the current firmware version.

Operartion w/ and w/o SIP-server;

Firmware update via web-interface;

Fig. 1 TAU-2M.IP operation diagram

2.3 Device design and Operating Principle

Realtek RTL8972C highly-integrated System-on-a-Chip (SoC), including a CPU, 100 Mbits

2 x SLIC subscriber units;

Design diagram for device is depicted in Fig. 2.

Fig. 2 TAU-2M.IP Design diagram

Support block of the device network functions;

Fig. 3 TAU-2M.IP series functional diagram

2.4 Main specifications

G.729, annex A, annex B

Ethernet WAN interface specifications

Ethernet LAN-interface specification

Analogue user ports example

Web-interface, Telnet, SSH, SNMP, TR-069

power adapter 12 V DC, 1.5

2.5.1 Top panel of the device

Fig. 4 TAU-2M.IP top panel layout

Front panel element

Analog phone indicator

2.5.2 Rear panel of the device

Connector for power adapter

2 x RJ-11 port for analogue unit connection

10/100BASE-T Ethernet (RJ-45) port for local device network connection

10/100BASE-T (RJ-45) port for external network connection

2.6 Light indication

Green, double short flashes in 3

2.7 Reset to factory defaults

2.8 Delivery Package

3 DEVICE MANAGEMENT VIA WEB CONFIGURATOR

Factory settings: login: admin, password: password.

3.2 Changing Users

3.3 WEB-interface operation mode

Monitoring system monitoring modeallows you to view various device operation

3.4 Applying and Discarding Changes Made to Configuration

3.4.2 Discarding changes

3.5 Quick Configuration Menu

Fig. 6 Quick configuration menu

Internet quick Internet access configuration;

Work mode the device operation mode:

External IP addressspecify device WAN interface IP address in the provider

PPTP-Server PPTP server address (domain name or IP address in IPv4 format);

IPSec is not supported in the current firmware version.

Enablewhen selested, the current line is active;

3.6 Advanced settings

3.6.1 WEB interface basic element

Fig. 7 Web configurator navigation elements

3.6.2.1 Internet submenu

Hostnamedevice network name.