TG400 User Manual
TG400 User Manual
USER MANUAL
Version: 2.0.1
0.3 Trademark....................................................................................................................... 5
Introduction .............................................................................................................................. 6
2
3.1.5 L2TP with WAN (Static, DHCP, PPPoE)......................................................................... 19
SIP Setting............................................................................................................................... 23
3
5.2 Region Tone Setting ...................................................................................................... 36
Maintenance........................................................................................................................... 38
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Preface
0.1 About this manual
This manual is intended to help users to the proper use of TG400/TG800 series VOIP voice
gateway.
The manual will be different based on the different firmware version, the latest content
(specifications) subject to the original notice, if any content in this document is supposed to
change without notice.
0.2 Copyright
All rights reserved for the 2009 Telephony Corporation. The information contained in this
publication is protected by copyright. Without written permission, no part of the copy,
transmitted, transcribed, or translated into any language, will be stored in a retrieval system,
ownership of copyright owners.
0.3 Trademark
Appear in this manual product and company names may be related to the company's
registered trademark that appears in this manual product and company names used only for
identification or explanation.
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1
Introduction
TG400/TG800 Series VoIP Gateway (VOIP Gateway) and Call Manager (Call Manage) include
low-to high-end Internet telephony overall solution. This article describes how to use the Voice
Gateway (VOIP Gateway) and Call Manager (Call Manage).
1.1 Overview
Internet Telephony Gateway (VOIP Gateway) is a communication device that can be
integrated analog phone, PSTN, or PBX trunk card/extension card.
TG400/TG800 series voip gateway can be connected with personal computers, such as IP
sharing / router, it may be a separate device, such as telecommunications and Internet
service providers (VPN).
Provide a direct analog port. For computer modulation decoders, fax machines, analog
phones and other devices, the analog port is necessary.
Support for standard Internet services such as IP sharing, NAT, Dynamic DNS (DDNS),
Quality of Service (QoS), port filtering, IP filtering.
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Protocol
RTP Real-Time Transport Protocol RTCP Real-Time Transport Control
Protocol (also known as RTP
control protocol)
SIP Session Initiation Protocol SLIC Subscriber Line Interface Circuit
STUN Simple Traversal of UDP through URI Uniform Resource Identifier
NATs
TCP Transmission Control Protocol UDP User Gateway gram Protocol
UPnP Universal Plug and Play VoIP Voice Over Internet Protocol
1.3 Introduction
combines VoIP and traditional analog phone
Support SIP VoIP protocol.
TG400 series: allow 4 SIP call
TG800 series: allow 8 SIP call
equipped with basic 4-port IP sharing function
Model FXS Port FXO port WAN Port LAN Port RJ-11 port SIP
TG400 4 1 4 4
TG402 2 2 1 4 4
TG404 4 1 4 4
TG800 8 1 4 8
TG804 4 4 1 4 8
TG808 8 1 4 8
TG400 Series:
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Green light flashing registering
Red light on SIP account registered failed
If any account registered failed, it
shows red light.
P1 / P2 Green light on VoIP call using
P3 / P4 Green light flashing VoIP / PSTN line rings
Orange light on PSTN call using
WAN Off Waiting for network connection
Green light on network connection established
Green light flashing Data traffic on cable network
LAN Off Ethernet not connected to PC
Green light on LAN is connected successfully
Green light flashing Data is transmitting
TG800 Series:
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Specifications
TG400/TG800 series VOIP Gateway provides many functions such as built-in server and
software, providing a convenient interface linking to your VoIP network.
1.7 IP Specifications
SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP
STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound
Proxy Support
WAN: PPPoE client, DHCP client, Static IP Address, DDNS client
Support PPTP/L2TP VPN Client
Network Address Translation: Providing build-in NAT router function
QOS
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Web-based Graphical User Interface
Remote management over the IP Network.
Backup and Restore Configuration file.
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2
Installation and setting
2.1 packing contents
Please check the closed packing of products and accessories before installation. The following
are the default contents of the product. The contents of the actual product may be varied
according to different versions.
TG400/TG800 series VOIP Gateway packing contents
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2.2 Hardware Installation
TG400:
TG800:
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TG400/TG800 series voip gateway internet setting
1. Time zone setting
2. LAN IP, subnet mask IP setting (please keep default setting if not necessary to change the
IP))
3. WAN internet setting
WAN connection type, please choose your networking type
Static IP (fixed IP, IP information provided by ISP)
DHCP Client (distributing IP automatically)
PPPoE (account / password provided by ISP)
5. Click “end” and finish the internet setting, then, go to “VOIP Setup” to configure the SIP
account.
6. Connect TG400/TG800 series voip gateway to your internet environment
(Connect RJ45 cable to WAN port of TG400/TG800 series voip gateway).
7. The device will set up voip information automatically if devices setting finished
8. The TG400/TG800 series voip gateway LED light showed green light on: "REG" port after 2
sec, which means the device registered automatically and successfully
9. Start to call voip call
Notice: If REG is red light, it means the wrong account registration or not connected to internet.
If there is no sign on REG light, it means the account doesn’t open yet, please contact your
supplier.
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3
Network setting
3.1 Internet
WAN (Wide Area Network) is a network connection connecting one or more LANs together
over some distance. For example, the means of connecting two office buildings separated by
several kilometers would be referred to as a WAN connection. The size of a WAN and the
number of distinct LANs connected to a WAN is not limited by any definition. Therefore, the
Internet may be called a WAN.
WAN Settings are settings that are used to connect to your ISP (Internet Service Provider). The
WAN settings are provided to you by your ISP and often times referred to as "public settings".
Please select the appropriate option for your specific ISP.
For most users, Internet access is the primary application. TG400/TG800 series voip gateway
supports the WAN interface for internet access and remote access. The following sections will
explain more details of WAN Port Internet access and broadband access setup. When you click
“WAN Setting”, the following setup page will be shown. Three methods are available for
Internet Access.
1. Static IP
2. DHCP
3. PPPoE
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3.1.1 Static IP
If you are a user with static IP address, please enter the IP address, subnet mask, default
gateway and DNS servers, which are provided by your ISP (Internet Service Provider). Each IP
address must be entered in the field of IP in the appropriate forms, which are four IP octets
separated by points (XXXX). Router will not be accepted if IP address not like in this format. For
example: [Link]
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□IP Address: Check with your ISP provider.
□Subnet Mask: Check with your ISP provider.
□Default Gateway: Check with your ISP provider.
□MTU size: MTU stands for Maximum Transmission Unit, the largest physical packet size,
measured in bytes that a network can transmit. Any messages larger than the MTU are divided
into smaller packets before being sent.
The key is to be deciding how big your bandwidth pipe is and select the best MTU for your
configuration. For example, you have a 33.6 modem, you use a MTU of 576, and if you have a
larger pipe you may want to try 1500.
□DNS server: enter DNS server (suggest default setting)
□Enable Keep Alive
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□Name of service: it may be set by your own or blank
□MTU size: enter MTU (Maximum transmission unit) size (suggest set to default)
□Assign DNS Dynamically: get DNS IP address WAN interface automatically (suggest use it)
□First DNA server, Second DNS server: Manually enter the domain name server WAN interface
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□USER name: the user name provided by your ISP
□user password: enter password provided by your ISP
□Name of service: you may set up the name of service or blank
□Type of connection:
Continuous: will keep trying to connect internet while disconnect
Connect on demand: will be set according to the idle time when connection
Manual: connect or disconnect by manual
□Idle time: when the connection type is: Connect on demand, can set up the interval time to
reconnect internet
□MTU size: enter MTU (max transmission unit) size (suggest set to default)
□Server 1, 2: Manually enter the domain name server WAN interface
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□PPTP Server IP Address: check with your VPN PPTP Server provider
□PPTP User Name: PPTP Dial-in account
□PPTP Password: PPTP Dial-in Password
□PPTP MTU Size
□Request MPPE Encryption
□Remote LAN setting: select SIP Call by LAN IP Address or not
□L2TP Server IP Address: check with your VPN L2TP Server provider
□L2TP User Name: L2TP Dial-in account
□L2TP Password: L2TP Dial-in Password
□L2TP MTU Size
□Request MPPE Encryption
□Remote LAN setting: select SIP Call by LAN IP Address or not
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3.2 LAN interface
The IP settings of the LAN (Local Area Network) interface for the device. These settings may be
referred to as "private settings". You may change the LAN IP address if needed. The LAN IP
address is private to your internal network and cannot be seen on the Internet. The default IP
address is [Link] with a subnet mask of [Link].
LAN is a network of computers or other devices that are in relatively close range of each other.
For example, devices in a house or office building would be considered part of a local area
network
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3.3 Dynamic DDNS
Maintenance of dynamic database of domain names, and the corresponding "Internet
Protocol"(IP) address
DNS: is a core Internet services, as domain names and IP addresses can be mapped to a
distributed database, can make people more convenient access to the Internet, without having
to remember that the machine can be directly read a string of IP
How to use dynamic domain name service?
From this web site you can register a new DDNS service account:
[Link]
Note that if you are using a fixed IP address, do not set the gateway dynamic domain name.
DDNS and use both a fixed IP, DynDNS DDNS service will stop your DDNS service.
□Enable DDNS: Enable/Disable the DDNS service. The default setting is disable.
□DDNS Server Type: Support two types of DDNS, [Link] or [Link]
□Domain Name: The domain which you register in [Link] or [Link] website.
□DDNS Username: The username which you register in [Link] or [Link] website.
□DDNS Password: The password which you register in [Link] or [Link] website.
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3.4 QOS setting
When making a VoIP call, in order to ensure the voice bandwidth, the other connection will
automatically reduce the transmission capacity
You can set your Qos upload speed
□NAT Mode: This mode allows LAN users to share a single IP address and Internet connection.
□Bridge Mode: This mode allows the Internet interface and local area network interface
bridging, Notice: the bridge mode failure once you start NAT and firewall
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4
SIP Setting
4.1 SIP setting
SIP is a request-response protocol, dealing with requests from clients and responses from
servers. Participants are identified by SIP URLs. Requests can be sent through any transport
protocol. SIP determines the end system to be used for the session, the communication media
and media parameters, and the called party's desire to engage in the communication. Once
these are assured, SIP establishes call parameters at either end of the communication, and
handles call transfer and termination.
□Individual Account:
Each port register different accounts. Accounting can be independent. Need to apply account
from System operators
Key in Display Name, User name, Authorization user name, password, Proxy Server IP and SIP
domain, and enable the Register items. Then, click the OK button and check the Registration
Status to see if the gateway registers to sip account successfully or not.
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□Represent Account:
Representative number is used for PBX integration, one representative number for 4 port
gateway or 8 port gateway is enough, but the account of these 4 port or 8 port can’t be
independent. Need to apply account from System operators
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□Ring type:
Ring by order: Each time starting from the line 1 and sequentially looking unoccupied line for
ring
Round Robin: Turn calls increased each cycle, ringing every user
□SIP port number: Local SIP port number setting defaults: 5060
□Media Port Start: The starting range of RTP port. Port number for initial of sending RTP
packet, default setting is 9000.
□RTP Packetization interval (Ptime), default is 20 ms
Value less: Increase network load and improve the sound quality
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Value gain: Reduce the network load, reduce the sound quality.
□DTMF Transmit method: Select DTMF transmission format RFC2833, Sip info, In band.
Default is RFC2833
□DTMF detection Sensitivity:
Value gain: Sensitivity enhancement, more digit issue may happen, suitable for enlarge the
item when dial short length number
Value less: Lower sensitivity, less digit issue may happen, suitable for enlarge the item when
dial the long length number.
Default is -1
□DTMF volume: Sometimes no action when dial, you may adjust the function.
For example: when you make call to a company, you get an IVR, but no action when dial the
extension number, you may increase the value. Default is -2dB
□RFC2833 Payload Type: Sending the DTMF tone as a RTP payload signal. The RFC2833
signaling
□SIP INFO Duration (ms): Modify SIP INFO duration time.
□FAX type:
G711 fax pass through (not compressed by using a 64Kbps channel to transmit voice signals)
T.38 fax relay (fax encoded by IP protocol)
□SIP Account Pooling (for Registered Accounts): SIP account sharing
□Local Port call directly: the telephony port of the same gateway call each other, not send
invite packet message.
□SIP Invite timeout: after send invite packet message, server not response the busy tone
within the seconds you set.
□SIP 100red support: if enable the item, when device receive SIP : 183, will reply PACK.
4.1.3 Codec
A CODEC is an algorithm for taking voice or video and compressing the information. This type
of codec combines analog-to-digital conversion and digital-to-analog conversion functions in a
single chip. The Codec is used to compress the voice signal into data packets. Each Codec has
different bandwidth requirement. There are 9 kinds of codec, G.711/Ulaw, G.711/Alaw, G.729,
G.723 (5.3k / 6.3k bps), G.726 (16K bps), G.726 (24K bps), G.726 (32K bps), G.726 (40K bps),
GSM-FR
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4.2 Call Feature
4.2.1 Call Forward Setting
You can setup the phone number you want to forward in this page. There are three type of
Forward mode. You can choose Immediate Forward, Busy Forward, and No Answer Forward
by click the icon.
Immediate Forward: All incoming call will immediately forward to the number you entered.
Busy Forward: When you are on phone, the new incoming call will forward to the number you
entered.
No Answer Forward: If no one picks up the phone, the incoming call will forward to the number
you entered.
After you finished the setting, please click the OK button.
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4.2.2 Do Not Disturb
DND Setting: you can setup the DND Setting item to keep the device silence. You can choose
Always or Enable or Disable.
DND Always: All incoming call will be blocked until disable this feature.
DND Enable: Set Enable and the device will be blocked during the time period. If the “From”
time is large than the “To” time, the Block time will from 00:00 to 23:59
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□“Lead Number” is the leading digits of the call out dialing number.
□“Min-Max Digits” has two text fields need filled: “Min Length” and “Max Length” is the
min/max allowed length you can dial.
□“Strip Digits Length” is the number of digits that will be stripped from beginning of the dialed
number.
□“Prefix Number” is the digits that will be added to the beginning of the dialed number.
□“Destination IP/ URL” is the IP address / Domain Name of the destination TG400/TG800 series
voip gateway that owns this phone number.
□“Destination SIP Port” is port of the destination TG400/TG800 series voip gateway use.
(Default is 5060)
□“Destination Telephone Port” is port of the destination TG400/TG800 series voip gateway use.
Ex. 1,2,3,4
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4.4 NAT Traversal
STUN means Simple Traversal of UDP through NATs (Network Address Translation), it’s a
protocol for assisting devices behind a NAT firewall or router with their packet routing.
STUN enables a device to find out its public IP address and the type of NAT service its sitting
behind.
When you enable the STUN function, you must input the STUN server address.
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5
Advance setting
5.1 Phone setting
5.1.1 Common
□Call transfer: Set your call control keys, it starts to call transfer when phone hang up, default
is “* 1”
Dialing Parameter
□Auto Dial Time: If no other number is being dialed within this interval, the TG400/TG800
series voip gateway will terminate this call. Assign the time interval from 3 to 9 seconds.
□Off-Hook Alarm Time: Set has been off-hook, after this time, user will hear alarm.
□FXS off-Hook Debounce: For example, if the value is 10ms, user needs to push the flash key
longer than 10ms to hang up the call. For any reason, user may dial the flash key wrongly, if
user push the flash key under 10ms, the device will not hang up the call.
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is DSP, another is SLIC
□Phone Out:You may adjust DSP chip to control output volume, rang from -32~31dB
□Phone In:You may adjust DSP chip to control input volume, range from -32~31dB
□FXS Tx:YOU may adjust SLIC chip to control output volume, range from 1~ 9
□FXS Rx:You may adjust SLIC chip to control the input volume, range from 1~9
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Telephone interface for each group can do loop current control. Adjust your current control for
PBX or phone
□Min delay (ms): Select min delay buffer time. (40ms – 100 ms)
□Max delay (ms): Select Max delay Buffer time. (130ms – 300ms)
□Optimization factor: Controls quickly the length of the Jitter Buffer is increased when voice
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RTP on the network. Default is 6
□Flash Time Setting: Set Min / max storage Flash Time Setting
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5.1.7 Caller ID Setting
FSK Date and time synchronization: Enable FSK single sending date and time to show on phone
screen
FSK Date & Time Sync: Send FSK Date & Time to display device.
Short Ring before Caller ID: Send short ring before caller ID.
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Dual Tone before Caller ID: Send Dual Tone before Caller ID.
Caller ID prior First Ring: Send Caller ID before first ring.
Caller ID DTMF Start Digit: Set Caller ID DTMF start digit.
Caller ID DTMF END Digit: Set Caller ID DTMF end digit.
□SIP DSCP: enter the priority for SIP voice transmission. The device creates type of service
priority tags; with this priority to voice traffic that is transmits.
□RTP DSCP: enter the priority for RTP voice transmission. The device creates type of service
priority tags; with this priority to RTP traffic that is transmits
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When SIP register failed or Network failed, playing the Voice Prompt.
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6
Maintenance
6.1 System
6.1.1 Account Setting
Users can log in via the web interface by entering the username and password to view
administration pages. Only the administrator account has permissions to change user account
and password
Default value:
User name: guest, Password: guest
Administrator name: admin, password: admin
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□Time Zone Select: Choose your time zone
□NTP Server: Select NTP server.
6.2 Tool
6.2.1 Backup / Restore
This page allows you save current settings to a file or reload the settings from the file which
was saved previously. Besides, you could reset the current configuration to factory default.
□Backup: click “Backup” button to save the current configuration of your system to your
computer.
□Restore: To restore a previously saved configuration file to your system, “browser” to the
location of the configuration file and click “Upload”
□Back to Factory Defaults: click “Default” button to clear all user-entered configuration
information and return to factory defaults. After resetting,
-User name will be admin, Password will be admin
-Lan IP Address: [Link]
-DHCP will be reset to server
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6.2.2 Firmware Upgrade
This page allows you upgrade the Access Point firmware to new version. Please note, do not
power off the device during the upload because it may crash the system.
Notice: It won’t change the system parameter you set when upgrading firmware, but better
you save the system setting before upgrade firmware.
Ex. we key in Ping Times as 5 and Ping Destination as [Link] ip address, the result is as
attachment.
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6.2.4 Restart
Click “Restart” button to restart the device. For some reasons, the device is not responding
correctly, if you want to restart the TG400/TG800 series voip gateway, you can just click the
Restart button, than the VoIP equipment will reboot automatically.
6.3 Langue
The TG400/TG800 series voip gateway support three language included English, Traditional
Chinese and Simplified Chinese. Default is English. You can select the language version you
prefer.
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7
Installation structure
7.1 Installation structure
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