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MBR L13
User’s
Guide
61
Gamer
Game-related traffic will receive priority.
Priority By Host
This entry provides the option to configure which computer in your
LAN will receive the highest priority and which the lowest. If you have
additional computers, they will receive medium priority.
High Priority Host
- Enter the host name or IP address of the
computer to which you would like to grant the highest bandwidth
priority.
Low Priority Host
- Enter the host name or IP address of the computer
to which you would like to grant the lowest bandwidth priority.
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MBR L13
User’s
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3.7.3
Voice Service
There are two modes of work. One is with a single analog extension and no PBX functionality. The second is with the
internal PBX and up to five extensions including some PBX features: call routing, hunt group, pickup call, Class Of
service, etc.
The following table describes some of the calling operations which can be performed by the extensions.
Note
:
Call transfer, call switching, and 3-way conference calls are only available if the cellular provider provides call-
waiting functionality.
Calling operation
VoIP Extension
Analog Extension
Placing/Recalling a call
to/from hold
Using “hold” function
Hitting on hook/flash button.
Supervised call transfer -
Where the call is placed on
hold, a call is placed to
another party, a
conversation can take place
privately before the caller on
hold is connected to the new
destination
Use a “Transfer” function on the
SIP phone. Call the wanted
extension. Once the person on
the destination extension
answered and agreed to accept
the call, hook on.
Use a “Flash” or “Hook flash” function
on the analog phone to put the current
call on hold. After hearing the dial tone,
establish another call. Once the person
on the destination extension answered
and agreed to accept the call, hook on.
Blind call transfer - Where
the call is transferred to the
other destination with no
intervention (the other
destination could ring out
and not be answered for
instance)
Use a “Transfer” function on the
SIP phone. Call the wanted
extension and hook on once you
hear the dial tone.
Use a “Flash” or “Hook flash” function
on the analog phone to put the current
call on hold. After hearing the dial tone,
establish another call. Once you hear a
dial tone, hook on.
Switching between waiting
calls
While hearing a waiting call tone,
use an appropriate function on
the SIP phone to put the current
call on hold and switch to a new
call.
While hearing a waiting call tone, use
“Flash” or “Hook Flash” to put the
current call on hold and switch to a new
call.
3-way conference
Put the current call on hold by
choosing an appropriate function
on the SIP phone. After hearing
the dial tone, establish another
call. Once the second call
established, use the appropriate
function on your SIP phone for 3-
way conference.
Use “Flash” or “Hook Flash” to put the
current call on hold and to hear a dial
tone. Establish another call, and use
“Flash” or “Hook Flash” for 3
-way
conference.
Pickup group membership
pick up a ringing call in a
group.
*8
*8
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MBR L13
User’s
Guide
63
3.7.3.1
Extensions
Analog Extension
- There is one analog FXS port extension in the system.
SIP Extensions
- A VoIP SIP local extension is a device that supports VoIP (Voice over IP) services using the Session
Initialization Protocol (SIP). It can be a hardware SIP phone unit, a PC with software SIP phone application (SIP soft-
phone) installed, a wireless Wi-Fi phone with SIP client firmware installed, or a cellular phone with SIP client software
installed.
A SIP device can be connected to the 3L13 unit directly using a wired LAN connection or a Wi-Fi wireless connection.
L13 can handle up to 4 VoIP SIP extensions.
To view existing PBX phone numbers:
Click the
Services
tab, and select
Voice
service (or navigate to
Home
Voice
). The
Extensions
screen
appears.
Figure 84: Voice
Extensions
To open the analog extension settings screen:
1.
Navigate to Services
Voice (or Home
Phones).
The
Analog Extensions
section shows L13 physical telephone port, for which the L13 PBX functions as an
Analog Telephone Adaptor (ATA) device.
Figure 85: Analog Extension
To modify the analog port extensions and other settings:
1.
Click the extension number or its
action icon. The
Edit Extension
screen appears.
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MBR L13
User’s
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Figure 86: Edit Extension Screen
2.
Configure the parameters as follows:
Parameter
Description
Extension Number
Specify the extension number as it was set on the device (the SIP
phone). The PBX supports extension numbers containing 3 to 9
digits.
Last Name, First Name
You can specify the last name and first name of the extension's
owner.
Class of Service
Select the classes of service to apply to the extension. Selecting a
Class of Service applies the functionality associated with that Class
of Service to the extension. For example, a Class of Service can
specify a particular dial plan for outgoing calls.
Enable Do Not Disturb
Select this checkbox to prevent calls from reaching your extension.
Require Authentication
Select this checkbox to set a specified user name and password for
the extension. The SIP device will only be able to use the extension
if it supplies these values to the L13.
When you select this option, the screen refreshes, and
Authentication User Name
and
Authentication Password
fields are
added. Enter the required user name and password in these fields.
Optimize RTP Path Using re-
INVITE
Select this option to enable the LAN device and the SIP proxy to
exchange Real Time Protocol (RTP) traffic directly. Note that in
order for this feature to work, it must also be enabled for the VoIP
line through which the call is routed.
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MBR L13
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65
To enable/disable the VoIP extensions:
1.
Click the extension number or its
action icon. The
Edit Extension
screen appears.
2.
Check or uncheck the
Enable VoIP Extension.
Figure 87: Enable VOIP Extension
To Configure a SIP Phone
This section provides general information about how to configure a SIP phone to work with MBR. It also includes
information about SIP phone devices.
Configure the phone with the following settings. Refer to the device's documentation if necessary.
It is highly recommended to configure a SIP phone as described below:
1.
The SIP phone must be configured to get an IP address automatically when it is connected directly to the L13
LAN port. In this case, the appropriate IP address will be of the form 192.168.1.XXX, where XXX is an address
provided by the L13
’s internal DHCP s
erver.
2.
The SIP phone must be configured to perform the SIP registration procedure with the L13 PBX. For this
purpose, the SIP phone must create SIP registration parameters (“User name” and “Password”), which must
correspond to the SIP extension’s account i
n L13. (See the next paragraph.) The IP address of the L13 unit
must be set as the SIP server registration point for the SIP phone. For a SIP phone that is connected directly
to the LAN, the SIP server IP address is 192.168.1.1.
3.
A wireless (Wi-Fi) SIP device must first establish a Wi-Fi connection with the L13 wireless network, and then
run the SIP registration procedure.
To enter the SIP VoIP extension’s parameters:
1.
Navigate to
Services
Voice
. The Extensions screen
opens. The
VoIP Extensions
section displays a list of the
VoIP extensions that are available in your L13. SIP devices connected directly to the gateway's LAN must be
configured to use 192.168.1.1 as their SIP server IP address and an extension account number.
2.
Click
New VoIP Extension
. The Edit Extension screen opens.

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