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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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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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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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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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

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.