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Page 96 / 130 Scroll up to view Page 91 - 95
Billion 810VGTX Router
Page | 95
Configuring L2TP VPN in the Branch Office
The IP address 69.1.121.30 is the Public IP address of the router located in head office. If you registered the DDNS
(please refer to the DDNS section of this manual), you can also use the domain name instead of the IP address to reach
the router.
Function
Description
Name
Head Office
Give a name to the L2TP Connection
Connection Type
LAN to LAN
Select LAN to LAN from the Connection Type drop-down
menu
Type
Dial in
Select Dial in from the Type drop down menu
IP Address
69.121.1.33
IP address assigned to the branch office network
Peer Network IP
192.168.1.0
Head office network
Netmask
255.255.255.0
Username
username
A username and password assigned to
authenticate the branch office network
Password
123456
Auth. Type
Chap (Auto)
Keep this as the default value in most cases
IPSec
Enable
Enable this to enhance your L2TP VPN security
Authentication
MD5
Both sides should use the same value
Encryption
3DES
Prefer Forward
Security
None
Pre-Shared Key
12345678
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Billion 810VGTX Router
Page | 96
°
Remember to apply changes, SAVE CONFIG and restart the router after
completing your VoIP configuration. This is to ensure that your VoIP is
activated.
VoIP - Voice over Internet Protocol
VoIP enables telephone calls through existing Internet connection instead of going through the PSTN (Public
Switched Telephone Network).
It is not only cost-effective, especially for a long distance telephone charges, but also
toll-quality voice calls over the Internet.
Attention
Here are the items within the VoIP section:
SIP Device Parameters, SIP Accounts, Phone Port, PSTN Dial Plan,
VoIP Dial Plan, Call Features, Speed Dial
and
Ring &Tone.
SIP Device Parameters
This section provides easy setup for your VoIP service. Phone port 1 and 2 can be registered to different SIP
Service Providers.
SIP Device Parameters
SIP:
To use VoIP SIP as VoIP call signaling protocol. Default is set to ‘Disable’.
Silence Suppression (VAD):
Voice Activation Detection (VAD) prevents transmitting the nature silence to consume
the bandwidth. It is also known as Silence Suppression which is a software application that ensures the bandwidth is
reserved only when voice activity is activated.
Default is set to ‘Enable’.
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Billion 810VGTX Router
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Echo Cancellation:
G.168 echo canceller is an ITU-T standard.
It is used to isolate the echo while you are on the
phone. This helps you not hear your own voice on the phone while you talk. Default is set to ‘Enable’.
RTP Port:
Provides the base value from the media (RTP) ports that are assigned for various endpoints and the
different call sessions that may exist within an end-point. (Range from 5100 to 65535, default value is 5100)
Region:
This selection is a drop-down box, which allows a user to select the country in which the VoIP device is
active. When a country is selected, the country parameters are automatically loaded.
Voice QoS, DSCP Marking:
Differentiated Services Code Point (DSCP), it is the first 6 bits in the ToS byte. DSCP
Marking allows users to assign specific application traffic to be executed in priority by the next Router based on the
DSCP value.
See Table 4. The DSCP Mapping Table:
Note: Be sure that the router(s) in the backbone network have the ability to execute and check the DSCP
markings
through-out the QoS network.
Advanced – Parameters
VoIP through IP Interface:
IP Interface decides where to send/receive the VoIP traffic; it includes: ipwan and iplan.
Easy way to select the interface is to check the location of the SIP server.
If it is located somewhere on the Internet
then select
ipwan.
If the VoIP SIP server is on the local Network then select
iplan.
Voice Frame Size:
Frame size is available from 10ms to 60ms.
Frame size meaning how many milliseconds the
Voice packets will be queued and sent out.
It is ideal to have the same frame size on both Caller and Receiver.
Dial Plan Priority: Define the priority between VoIP and PSTN dial plan.
PSTN Auto-fallback:
Whenever VoIP SIP responds with an error or an error code matching the codes in the
Edit
section, the VoIP calls will automatically fallback to PSTN.
In other words, the call will be made via the PSTN when
VoIP SIP returns an error code.
Click ‘Edit’ to add or remove response codes. Be sure that the codes are separated by a comma (,).
For more information about SIP response codes, please go to this link
http://voip-info.
org/wiki/view/sip+response+codes
. You will get the meaning of all the SIP responses here.
T.38 Fax Relay:
It allows the transfer of facsimile documents in real-time between two standard Group 3 facsimile
terminals over the Internet or other networks using IP protocols. It will only function when both sites support this
feature and have it enabled.
Page 99 / 130
Billion 810VGTX Router
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Advanced – PSTN Environment Adjustment
PSTN Environment Adjustment options will help you to adjust the onhook and offhook voltage detection values for
your environment.
You should use these if the default values are incorrect and result in PSTN calls not being
detected properly, e.g. calls being terminated within 5 seconds of being answered. The actual levels are determined
by your environment including the number and type of telephones used.
Note: ONHOOK means hung up.
To take your phone OFFHOOK, lift the receiver then press Hook/Flash until you hear your normal PSTN dial tone,
and not your VoIP dial tone. Wait several seconds and then press Check Level.
You should check the OFFHOOK value for each telephone you have connected to this device. Set the OFFHOOK
voltage to the lowest setting registered for all your telephones, e.g. if your telephones return values of 4, 5 and 7 then
you should set your OFFHOOK voltage to 4.
Note: The detected values will not automatically be set by the Check Level function; you must enter the
lowest level detected after testing all your telephones.
SIP Accounts
This section contains the basic settings of the VoIP module from the selected provider in the Wizard section.
Providing the incorrect information will cause it to stop making calls through the Internet.
Profile Name:
Assign a name for profile identification.
Registrar Address (or Hostname):
Indicate the VoIP SIP registrar IP address.
Registrar Port:
Specify the port of the VoIP SIP registrar on which it will listen for register requests from VoIP
devices.
Expire:
This is the duration for the registration message being sent.
User Domain/Realm:
Set a different domain name for the VoIP SIP proxy server.
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Billion 810VGTX Router
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Outbound Proxy Address:
Indicate the VoIP SIP outbound proxy server IP address. This parameter is very useful
when VoIP devices are behind a NAT.
Outbound Proxy Port:
Specify the port of the VoIP SIP outbound proxy on which it will listen for messages.
Phone Number:
This parameter holds the registration ID of the user within the VoIP SIP registrar.
Username:
Same as Phone Number.
Password:
This parameter holds the password used for authentication within the VoIP SIP registrar.
Display Name:
This parameter will appear on the Caller ID.
Direct in Dial:
Select the ringing port when getting an incoming VoIP call.
Phone Port
This section displays the status and allows for further editing of the account information of the Phones. Click Edit to
update your phone information.
Port:
It allows you to change the phone port setting for a specific FXS port.
*69 (Return Call):
Dial *69 to return the last missed call. It is only available for VoIP call(s).
*20 (Do not Disturb ON):
Dial *20 to enable the No Disturb feature. Your phone will not ring if someone calls.
*90x (Blind Call Transfer):
Dial *90 + phone-number to transfer a call to a third party. This feature is enabled by
default.
x# Speed Dial (x:2..9):
Refer to the Phone Port section in the Web GUI. Set up your Speed Dial phone book first
before accessing the Speed Dial feature. This feature is enabled by default.
## Redial:
Press ## to redial the last phone number. This feature is enabled by default.
*74<x><number>#:
Use your phone key pad to insert a phone number to the Speed Dial phone book. Or you can
update your Speed Dial phone number manually. Refer to the Phone Port section in the Web GUI for details.

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