Billion BiPAC-7404VGO Router Manual PDF (Setup & Configuration Guide)

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120

a traditional phone number (PSTN) to an Internet addresses/ SIP URL. The ENUM number must be

registered via a public ENUM site or your VoIP Service Provider.

<@ SIPgateway>:

It is used for the

Intelligent Call Routing

feature where you need to set up your SIP

account on the

VoIP User-defined Profiles

link on the VoIP Wizard page. Go to the

VoIP Wizard

in this

manual for more information.

xxx

Any 3 digit number only between 0 and 9.

Total

length is 3.

Note: No period is needed (.)

123x

.

Any number (0-9) starting with 123. Maximum length

is 16.

[x-x]x

.

For example: [1-3]x

.

Any number (0-9) starting with number 1 to 3.

Maximum length is 16.

*

xx

Starting with ‘* sign’ + any 2 digit numbers between 0

and 9. Total length including the * is 3.

Note: No period is needed (.)

#

xx

.

Starting with ‘# sign’ + any digit number (0-9) in

variable length but no shorter than 1 digits. Maximum

length is 16.

<@SIPgateway> / Intelligent Call Routing Example:

VoIP Gateway let you use 3 VoIP/SIP providers at the same time.

VoIP/SIP providers are

localcheap.com, longdischeap.com

and

mobilecheap.com

.

Each provider has its price for different type

of calls and I can set the following rule for each providers.

1) Phone 1: For Local calls: I use

localcheap.com

that charges $0.01 per minute to all local calls.

I set

a dial rule,

<:03>[123]x.T

,

on my phone port 1.

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Localcheap.com is the default VoIP provider I set on phone port 1.

When I call out any number start with

1 or 2 or 3 and plus rest of the phone number for local call, 03 is always prepended in front of these

number.

If

2

3295 are dialed, 03-2-32935 is the actual phone number called out via localcheap.com

provider.

2) Phone 1: For International calls: I use

longdischeap.com

that charges $0.05 per minute to all

International long distance calls.

I set a dial rule,

0[2456]x.T<@LongdisCheap>

,

on my phone port

1.

Longislcheap.com is one of the VoIP providers I set in the User-defined profile.

No prefix is attached to

the dialed number when I call out number 0 plus any following number 2 or 4 or 5 or 6 and plus rest of the

phone number for an international call.

If

02

016148513295 are dialed, 0-2-016148513295 is the actual

phone number called out via longdischeap.com provider.

3) Phone 2: For Mobile calls: I use

mobilecheap.com

that charges $0.25 per minute to all local calls.

I

set a dial rule,

<123:09>39x.T

,

on my phone port 2.

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Chapter 4: Configuration

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Mobilecheap.com is the default VoIP provider I set on phone port 2.

When I call out 123-39-45678 for a

mobile call, 123 is replaced with 09.

Therefore, 09-39-45678 is the actual phone number called out via

Mobilecheap.com provider.

The Intelligent Call Gateway not only saves time from changing VoIP settings to different provider to

make call get routed to specific gateway(s) automatically but also taking advantage of different call rate.

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Ring & Tone

This section allows advanced user to change the existing or newly defined parameters for the

various ring tones (dial tone, busy tone, answer tone and etc.)

Country Specific Ring & Tone

Region:

Select a country ring-tone, from the drop-down list, where you are located. This VoIP router

provides default parameter of ring tones according to different countries.

The ring-tone parameters are

automatically displayed after entering a specific country.

If your country is not in the list, you may

manually create ring-tone parameters.

Ring Parameters

Ring Cadence (in ms):

Ring cadence is defined by three fields, Frequency: On Time1, Off Time1, On

Time2, Off Time2 and On Time3, Off Time3. Frequency is specified in Hertz. Time is given in

milliseconds.

Tone Parameters

You may need to check with your local telephone service provider for such information. Also, it is

recommended that this option be configured by an advanced user, unless you are instructed to do so.

Click

Apply

to apply the settings.

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Chapter 4: Configuration

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Special Dial Codes

The following table lists the special dial codes that are built-in to the system:

Flash-hook

(Wireless

Router

with

LINE port only)

Switch to PSTN line

Note: A quick press of the hook. On some phones a button is

provided which provides Flash-hook functionality. The button is

marked "FLASH" or "RECALL".

*69

Return the last missed call for SIP service only

Note: Entering this on a phone will call the last number which

made a call to the phone. For example A makes a call to B, but

hangs up before B answers. If B enters *69, A will be called.

##

Last number redial

*20

Set do not disturb on

Note: It is possible to set a Do-Not-Disturb feature on a phone

such that any phone which calls the phone will receive an engaged

tone and the phone called will not ring. For example, B enters *20

and hangs up. A makes a call to B, and receives the engaged tone

and phone B does not ring.

*80

Set do not disturb off

*74<x><number>#

Set the number for Speed dial code ‘x’, where ‘x’ is a number

between 2 and 9.

Note: Where <x> is a number between 2 and 9, and <number> is

the number to dial. The code needed to dial a speed dial from a

phone connected to a VoIP Router is:

<

x

>#, where

<x>

is a number between 2 and 9.

The settings will infect to your setting in Speed Dial on WEB GUI.

*90<phone-number>

Set the number for performing

Blind Call Transfer

, where

<phone-number> is the number that you wish to transfer the call

to. It’s for SIP service only.

Note: In

Blind Call Transfer

, you have a call in progress

(incoming or outgoing) and decide you wish to transfer the call to

another phone. To transfer the call, perform the following steps:

1. Hook-flash to get a dial tone.

2. Dial *90<phone-number> (e.g. *907401), there will be a

confirmation tone then hang up. The other end will hear ring back

and the called third-party phone will ring. When the third-party

phone is picked up the two calls will be connected.

If the third-party phone does not answer then the caller being

transferred can hang up to cancel the connect attempt.

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