DSG InterPBX Blaze5000, InterPBX Blaze1200, InterPBX Savanna8000 Installation And Configuration Manual

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InterPBX Communication System

Blaze5000/Blaze1200/Savanna8000

Administrator Installation and Configuration Guide

08IPBXM.BZ/BA/BC2.00h.EN16

DSG, DSG logo, InterPBX, Blaze, Savanna, VG5000, VG6000, VG7000, IP590, IP580, IP500, IP510, InterConsole,

DSG SoftPhone, DSG SoftConsole, DSG SIP Proxy Server, DSG NAT Proxy Server, SIP1200, Blaze Logger,

BlazeLink, and S300X are trademarks of DSG Technology. Windows and Outlook Express are trademarks of

Microsoft Inc. Other names used here are trademarks of their respective owners.

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Table of Contents 3

Table of Contents

Chapter 1 InterPBX Communication System Overview...............................................................7

IP-Based Business Telephone System.......................................................................................8

Advantages of InterPBX..............................................................................................................8

Key Components .........................................................................................................................9

InterServer...........................................................................................................................9

VMS Server..........................................................................................................................9

Conference Server ..............................................................................................................9

Recording Servers ..............................................................................................................9

Voice Gateways ..................................................................................................................9

SIP Proxy Server...............................................................................................................10

NAT Proxy Server............................................................................................................10

CTI Solutions ....................................................................................................................10

Extension Types................................................................................................................10

Web-based Management Tools ......................................................................................11

SH2500 PoE Switching Hub............................................................................................12

Chapter 2 Installing InterPBX Communication System.................................................................13

Before You Start .........................................................................................................................14

Prepare Your Telephone Numbering Plan ...................................................................14

Installing PBX Server.................................................................................................................15

Installing and Configuring PBX Server.........................................................................16

Connecting PBX Server via Console Port......................................................................19

Installing VG5000 Voice Gateway...........................................................................................20

Connecting VG5000 via Telnet.......................................................................................24

Connecting VG5000 via Console Port ...........................................................................25

Installing IP Phones...................................................................................................................26

Create IP Phones via Auto Discovery ...........................................................................27

Create IP Phones Manually.............................................................................................32

Chapter 3 System Configuration.......................................................................................................35

General Parameters ...................................................................................................................36

Basic IP Settings................................................................................................................36

Email Settings ...................................................................................................................37

NAT Settings.....................................................................................................................37

Set QoS...............................................................................................................................38

Music on Hold ..................................................................................................................38

Ringing Patterns ...............................................................................................................39

Company Information...............................................................................................................39

Business Hours...........................................................................................................................39

Holidays......................................................................................................................................41

System Speed Dialing................................................................................................................41

Call Restriction ...........................................................................................................................42

Route............................................................................................................................................43

Function Code ............................................................................................................................44

Password Management.............................................................................................................45

Set System Date/Time ..............................................................................................................46

Miscellaneous.............................................................................................................................46

Timeout Settings...............................................................................................................46

Authorization Code .........................................................................................................47

Chapter 4 Gateway Configuration....................................................................................................49

Analog Gateways.......................................................................................................................50

Creating Analog Gateway List .......................................................................................50

Editing Analog Gateways ...............................................................................................51

4 Table of Contents

Configuring CO Line Ports.............................................................................................52

Configuring SLT Ports.....................................................................................................55

Off-Premises Gateways.............................................................................................................56

Creating Off-Premises Gateway List .............................................................................57

Configuring Off-Premises CO Line Ports .....................................................................58

Configuring Off-Premises SLT Ports.............................................................................59

Recording System ......................................................................................................................60

Setting Store-on-Demand (Available on Blaze5000 Series) ........................................60

Setting Record-on-Demand (Available on Blaze5000 and Blaze1200 Series)...........63

Setting Professional Recording System.........................................................................65

Digital Line Gateways...............................................................................................................66

Adding Digital Gateway .................................................................................................66

Modifying a Digital Gateway.........................................................................................68

Setting Trunk Parameters................................................................................................68

Setting Trunk Port Parameters .......................................................................................69

CTI Gateway...............................................................................................................................72

NAT Proxy..................................................................................................................................73

SIP Proxy.....................................................................................................................................74

Add SIP Proxy ..................................................................................................................74

Add SIP Trunks ................................................................................................................75

Create SIP Trunk ARS .....................................................................................................77

Add SIP Extensions..........................................................................................................79

Chapter 5 Extension Management....................................................................................................80

Configuring IP Phone, SoftPhone and SoftConsole..............................................................81

Extension No., Phone Type and MAC Address...........................................................81

User Information ..............................................................................................................81

User Password..................................................................................................................81

Off-Hook Access to ..........................................................................................................82

Class of Service .................................................................................................................82

Button Mapping Group...................................................................................................82

CODEC ..............................................................................................................................82

Jitter Buffer Depth ............................................................................................................83

Silence Suppression .........................................................................................................83

Conference Disabled........................................................................................................83

Paging Disabled................................................................................................................84

Enable/Disable Extension...............................................................................................84

Apply Settings ..................................................................................................................84

Button Mapping ...............................................................................................................85

Station Speed Dialing ......................................................................................................89

Answer Option .................................................................................................................90

Mailbox..............................................................................................................................91

Notification .......................................................................................................................93

Distribution List ...............................................................................................................95

Virtual Extensions............................................................................................................95

Analog Extensions .....................................................................................................................97

Off-Premises Extensions ...........................................................................................................98

SIP Phone List...........................................................................................................................100

InterConsole List......................................................................................................................101

Chapter 6 Group Management........................................................................................................103

CO Line Groups .......................................................................................................................104

Creating CO Line Groups .............................................................................................104

Assigning Members to CO Line Groups.....................................................................105

Extension Groups ....................................................................................................................105

Creating Extension Groups...........................................................................................106

Hunting Method.............................................................................................................107

Wrap Up Time ................................................................................................................107

Group Administrator Password...................................................................................107

Table of Contents 5

Set Sec. of Rings..............................................................................................................107

Queuing...........................................................................................................................108

Login to Group ...............................................................................................................108

Assigning Members to Extension Groups ..................................................................109

Group Answering Option.............................................................................................110

Group Mailbox ...............................................................................................................111

Message Notification for Extension Groups...............................................................113

Button Mapping Groups.........................................................................................................115

Class of Service.........................................................................................................................119

Creating a Class of Service............................................................................................119

Setting Allow and Disallow Table ...............................................................................121

Setting CO Priority.........................................................................................................122

Automatic Route Selection (ARS) ................................................................................123

Automatic Alternate Route (AAR) ..............................................................................124

Operator....................................................................................................................................125

Authorization Code.................................................................................................................126

Boss and Secretary...................................................................................................................127

Chapter 7 Voice Mail Configuration...............................................................................................129

Voice Mail Parameters ............................................................................................................130

Language .........................................................................................................................130

Name Directory ..............................................................................................................130

Inter-Digit and AA Timeout .........................................................................................130

Messages Storage Settings.............................................................................................131

Ring Notification ............................................................................................................131

Adjust Gain Level...........................................................................................................132

Transfer Options Settings .......................................................................................................132

Transfer Announcement ...............................................................................................132

Leave Message Directly.................................................................................................133

Transfer Options.............................................................................................................133

Before Leaving Message Timeout................................................................................135

AA Management......................................................................................................................135

Planning Your Auto Attendant Tree ...........................................................................135

Planning an AA Menu...................................................................................................136

Recording AA Menu Greetings....................................................................................138

CO Priority of Notification .....................................................................................................140

Tel. Programming Mode.........................................................................................................140

Chapter 8 Operation Management .................................................................................................141

General Information ................................................................................................................142

Check Software Version ................................................................................................142

License Key .....................................................................................................................143

Auto Discovery ........................................................................................................................143

Setting Default Class of Service....................................................................................144

Enable/Disable Auto Discovery ..................................................................................144

Backup System Files................................................................................................................144

Restore System Files................................................................................................................145

Application Upgrades.............................................................................................................146

CDR ...........................................................................................................................................146

Billing System .................................................................................................................148

Reboot........................................................................................................................................148

Reset to Default........................................................................................................................149

Chapter 9 Report Management .......................................................................................................151

System Summary Report ........................................................................................................152

System Log................................................................................................................................152

Extension Report......................................................................................................................153

Chapter 10 Multi-Server Management...........................................................................................155

Server Information...................................................................................................................156

6 Table of Contents

Redundant Server....................................................................................................................156

Assigning Slave PBX Server..........................................................................................157

Installing and Setting Slave PBX Server......................................................................157

Joint Server ...............................................................................................................................158

Creating Joint Server......................................................................................................158

Setting Joint Server CO Lines Access Control............................................................160

Setting Links with Joint Server.....................................................................................160

Setting Joint Server Extension List ...............................................................................161

Joint Server Status ..........................................................................................................162

Appendix A: Function Code List ....................................................................................................163

Appendix B: Retrieving Voice Messages and Recordings...........................................................167

Appendix C: InterPBX Management Website...............................................................................171

Appendix D: Terminologies ............................................................................................................173

Appendix E: Default Values ............................................................................................................175

Appendix F: DTMF Programming..................................................................................................177

Appendix G: System Capacity.........................................................................................................183

Index ...................................................................................................................................................185

Chapter 1 InterPBX Communication System Overview 7

Chapter 1 InterPBX Communication System Overview

InterPBX Communication System is a new generation business phone system

that employs a data network for terminal connectivity and voice transport.

Contrary to traditional business phone systems that carry voice over legacy

TDM networks, InterPBX delivers voice in digitized packets over LAN or

Internet networks.

8 Chapter 1 InterPBX Communication System Overview

IP-Based Business Telephone System

Packet switching telephony technology allows enterprises to work upon their

existing data network infrastructure for implementing advanced applications

and to save costs on long distance and international calls. As a nature of

IP-based communication systems, the InterPBX brings significant savings on

call tolls, simplifies wiring, improves management and maintenance, and

provides advanced voice applications to enterprises’ needs.

Advantages of InterPBX

y Feature-rich – InterPBX supports an abundant number of features, including

Auto Attendant, built-in Messaging System, Automated Call Distribution

(ACD), Conference, Unified Messaging, Recording System, Auto discovery,

Contact Center applications, and other new generation business

applications.

y Flexibility – The capacity of InterPBX is not limited to a fixed system chassis.

InterPBX provides flexible structure for growing businesses. It can be set up

to meet the needs of enterprises from small to large or single to multiple

sites. Enterprises can connect each location by VPN or Internet.

y Reliability – InterPBX employs a unique distributed NeuralServer

architecture that allows multiple servers to coexist and communicate with

one another in a single system. Through synchronization and backup

mechanisms, InterPBX provides redundancy in the event of server failure

and therefore drastically enhances the reliability of the system.

y Management – Browse-based management tool provides administrators and

users a user-friendly interface to set, control, and maintain the system.

Chapter 1 InterPBX Communication System Overview 9

Key Components

DSG’s InterPBX Communication System supports flexible components. They

could either be integrated as an embedded system or work individually.

y Blaze 5000 Series: It contains InterServer, VMS Server, Conference Server,

Voice Gateway, and Recording Server.

y Blaze 1200 Series: It contains InterServer, VMS Server, Conference Server,

Recording Server and Voice Gateway.

y Savanna 8000 Series: It contains InterServer, VMS Server and Conference

Server.

InterServer

InterServer is the brain (central administration) of InterPBX Communication

System. It is a software-based switching solution that handles call signals, call

control and voice processing activities and manages all the extensions, voice

gateway and other applications within the system.

VMS Server

VMS Server works seamlessly with PBX Server and provides Auto Attendant,

Voice Mail, Automated Call Distribution (ACD) and Unified Messaging

features.

Conference Server

Conference Server improves communications among employees by providing

conference function.

Recording Servers

Recording Server helps enterprises record, monitor and search recording files.

It can be integrated seamlessly with users’ IP phones bringing more efficiency

for customer service representatives, banking officers or attorneys.

Note: Blaze 5000 provides built-in Record-on-Demand and Store-on-Demand

functions and Blaze200 provides built-in Record-on-Demand. Savanna 8000

Series requires a stand-alone Recording Server.

Voice Gateways

Voice Gateway connects PSTN and IP networks and allows voice packets to be

transferred between Internet and traditional PSTN network. VG5000 Voice

10 Chapter 1 InterPBX Communication System Overview

Gateway supports CO Line and/or SLT interfaces, which are able to connect

CO lines or analog phones. VG6000 and VG7000 Voice Gateway supports

T1/E1 ISDN network.

Note: The Savanna 8000 Series doesn’t provide built-in Voice Gateway.

SIP Proxy Server

DSG's SIP Proxy is a full-featured proxy server. It allows your Blaze or Savanna

Series IP-PBX to get connected to operators' softswitch and to enjoy the

provided SIP Trunk services. DSG SIP Proxy can register the IP-PBX to your

service provider. All the extensions under the system, including DSG IP

extensions or third parties' SIP phones, can access the SIP Trunks to make

long-distance or international calls.

NAT Proxy Server

DSG NAT Proxy is a complete solution to solve complicated network problems

that may happen when enterprises use the IP PBX Communication System.

With DSG NAT Proxy, Blaze or Savanna IP PBX Communication Systems and

Off-Premises Phones can traverse more than 95% of NAT or firewall

successfully without creating other service ports, and therefore enterprises can

communicate via IP-PBX Communication System with pure and stable voice

quality.

CTI Solutions

DSG BlazeLink is a CTI solution enables business to develop applications to

optimize workforce in a call center or contact center. Applications developed on

BlazeLink for call control, monitoring, or managing are able to connecting DSG

IP-PBX System to the data processing environment. With the supported tools

and resources, developers are able to deliver a comprehensive solution for

effective communications and better interactive experiences.

Extension Types

InterPBX Communication System supports various types of extensions – IP

phones, analog phones, software phones and SIP phones. Administrators may

set up any types of the phones as users’ needs.

IP Phone: IP phones offer various business telephone features. They could be

installed on LAN or on remote sites.

Chapter 1 InterPBX Communication System Overview 11

SoftPhone: DSG’s SoftPhone is an application software installed on PCs

offering telecommuters or business travelers full-featured extension functions.

Users can adopt their own headsets or use DSG S300X USB Phones when

utilizing the SoftPhone.

Off-premises Phone: Off-premises IP phones, SoftPhones or analog phones on

off-premises gateways are all able to connect to a company’s InterPBX

Communication System via Internet or VPN and perform the same way as the

extensions in the head office.

Analog Phone: Analog extensions are connected to the FXS ports of Voice

Gateway like VG5000.

Attendant Console: Both software attendant console and hardware DSS

console are supported.

Web-based Management Tools

InterPBX provides administrators and extension users a management tool with

a GUI interface that allows administrators and extension users to set the

preferences of InterPBX or extension via browser. The suggested browser is

Internet Explorer 5.0 or later.

Administrator: The Administrator web site provides a web-based GUI interface

allowing administrators to configure the InterPBX Communication System

through the web browser. It helps administrators to manage and maintain the

system easily.

12 Chapter 1 InterPBX Communication System Overview

Extension User: Extension users log in to user’s section to customize personal

settings on phones like Button Mapping, Speed Dial, Message Option, Answer

Option, Password and Phone Book.

SH2500 PoE Switching Hub

SH2500 is the recommended switching hub for InterPBX Communication

System. It supports 802.3af PoE (Power-over-Ethernet), which offers IP

extensions the current power over network cables. Users do not need additional

power socket when using IP phones.

Chapter 2 Installing InterPBX Communication System 13

Chapter 2 Installing InterPBX Communication System

This chapter guides you through the preparation, installation and basic

configuration of InterPBX Communication System.

14 Chapter 2 Installing InterPBX Communication System

Before You Start

The component of InterPBX Communication System contains servers, gateways

and IP phones. Please follow the recommendations below when you install or

operate your system in order to avoid any injury and damage.

Safety Recommendation

Always use ESD-preventive tools when you plug the power cord. Do not

disassemble or remove chassis cover of any components of your InterPBX

Communication System. If there is any problem of your system, please contact

our service representatives.

Environmental Prerequisite

InterPBX Communication System needs to be installed in clean, dry, adequately

ventilated areas. The server, gateway, and switching hub can be placed in a

control room or on a rack. Please remain the control room in a suitable

temperature and adequately ventilated environment.

Local Telecommunications Service

You have to apply for the local phone service in your area. The trunk lines of

the central office need to be connected to Voice Gateways. DSG VG5000 Voice

Gateway supports CO Line interface. Please make sure if your trunk lines

support analog interface.

Prepare Your Telephone Numbering Plan

If you are going to replace your traditional telephony system with InterPBX

Communication System, you can retain the old dialing plan. InterPBX

Communication System supports flexible extension number lengths. The

maximum extension number length is “5 digits”. The telephone numbering

plan must not overlap or conflict with other numbers that are on your system or

on Joint Server’s system.

You need to prepare the followings before you install InterPBX Communication

System.

y CO Line Extension Number: Prepare an extension number for each CO Line

port so that the associated extension number can reach each trunk line that

is connected to it.

Chapter 2 Installing InterPBX Communication System 15

y SLT Extension Number: Prepare an extension number for each SLT port so

that the associated extension number can reach each analog phone that is

connected to it.

y IP Phone: Prepare an extension number for each IP phone. y Group Number: If you plan to group specific trunk lines or extensions as a

group, prepare a Group Number for each CO Line Group and Extension

Group.

y AA Menu: Plan your AA procedure and prepare an access code for each AA

Menu.

y Routes: Plan your Routes, ARS and Class of Service to be uses when making

calls.

Installing PBX Server

Figure: Blaze 5000 Series Connection Diagram

Figure: Blaze 1200 Series Connection Diagram

Figure: Savanna 8000 Series Connection Diagram

16 Chapter 2 Installing InterPBX Communication System

Installing and Configuring PBX Server

1. Connecting to Power Cord

Plug one end of the power cord to the power connector on the rear panel of

PBX Server. Plug the other end of the power cord into a power outlet.

2. Connecting to LAN

You need to connect PBX Server to your existing Ethernet network. Connect an

Ethernet cable from the “Network” RJ45 port on PBX Server to any

10/100BaseT RJ45 port on a switching hub.

3. Connecting to CO Lines or Extension Lines (For Blaze Series Only)

With Blaze Series, the Server is integrated with Voice Gateway module. Voice

Gateway supports CO line or analog extension modules. Please plug a CO line

connector to a CO Line port or plug an analog extension to a SLT port. If you

would like to install other embedded analog or digital gateways, please refer to

the related manual.

Note: If you accidentally plug the equipment with CO Line interface to SLT

port, it may damage your equipment. Please make sure that your equipment

connects to an appropriate interface.

4. Connecting to External Audio Source (For Blaze Series Only)

The MOH (Music on Hold) port can connect to a radio or CD player for playing

music or your customized greetings for callers placed on hold.

You can also choose the pre-recorded audio files as the MOH on the PBX Server.

If you choose to use the pre-recorded audio files from the system, do not

connect the external audio source to the MOH port.

5. Connecting to External Paging Facility (For Blaze Series Only)

The “Paging” port allowing you to connect external paging equipment for

broadcasting.

6. Setting Power Failure Transfer (For Blaze Series Only)

When power failure occurs and there is no backup power, the C.O. lines

connected to port 1 will be switched to port 24. You can connect single line

phones to port 24 to pick up calls. The PFT function only works on port 1 with

Chapter 2 Installing InterPBX Communication System 17

FXO interface supported. Port 24 can be FXO, FXS or empty ports. When power

is supplied, the PFT function is not available.

7. Preparing a Computer

Prepare a computer with web browser and be sure it is on the same subnet as of

your PBX Server. Change the computer’s IP settings if necessary. The suggested

web browser is Internet Explorer version 5.0 or later.

The default values of PBX Server are as follows:

y Default IP Address: 192.168.1.200:88 (The service port is 88) y Default Gateway: 192.168.1.254 y Default Subnet Mask: 255.255.0.0

8. Login PBX Server

y Launch the web browser. On the address bar, type in the PBX Server’s

default IP address 192.168.1.200:88 to access PBX Server.

y You will be connected to InterPBX Administration Website. Please click the

Administrator icon.

y When you login the InterPBX Administration for the first time, you will see

the following picture. Please enter your user name and password and then

save the settings. Please remember your user name and password in order

to login again in the future. The password shall not exceed 8-character

alphanumeric. The user name and password are case sensitive.

18 Chapter 2 Installing InterPBX Communication System

When you login the InterPBX Administration next time, you will see the

following picture. Please enter your user name and password to login.

y After you login the InterPBX Administration successfully, the screen will

display the main menu of InterPBX Administration website as follows:

9. Basic Settings

There are some basic items you need to configure allowing the system to work

properly.

y PBX Server IP Settings: Go to Main Menu>System Configuration>General

Parameters to set your PBX’s IP. If you have more than one PBX server, go

to Main Menu>Multi-Server Management to assign each an ID and set the

Joint Server connection.

Chapter 2 Installing InterPBX Communication System 19

y License Keys: Go to Main Menu>Operation Management>General

Information. Input the license keys you purchased.

y Trunks: Go to Main Menu>Gateway Configuration. Depending on the voice

gateway you purchased, add your gateways and assign CO line extension

numbers. In addition, you many go to Main Menu>Group

Management>CO Line Group to group CO lines for better managing your

trunks.

y Extensions: Go to Group Management>Class of Service to set CoS for

providing different call permissions to extensions applied. Go to Main

Menu>Extension Management to add your extensions or turn on the Auto

Discovery allowing the system to detect connected extensions automatically.

y AA Menu: Go to Main Menu>System Configuration>Business Hours to set

your company operating hours. Then go to Main Menu>VM

Configuration>AA Management to set AA trees. And make sure the Ring

Assignment of CO lines is set to the appropriate AA Menu Access Code.

y Accessing CO Lines: Each CoS will have its CO priority. Or you may go to

Main Menu>System Configuration>Route to set routes and go to Main

Menu>Group Management>Class of Service to further assign ARS for better

using your trunk resource.

You may change your computer IP setting to the original values after you finish

setting up the Blaze/Savanna Server. You will need to enter the new IP address

to connect to the Blaze/Savanna Server next time.

If you encounter any problem or have questions during installation or

operation, click on the Help icon from the web setting page for assistance.

Connecting PBX Server via Console Port

When you forget the IP address of the PBX Server, you can connect a computer

to the console port of the PBX Server in order to check the current IP address,

gateway address, and subnet mask.

1. Connect one end of the RS232 transmission line to the computer’s COM

port and the other end to the Console port of PBX Server.

2. Open the Hyper Terminal program. Enter the connection name and set the

baud rate to 9600 bps.

3. Turn off the power of PBX Server. Wait for a while and then turn on the

power again. The computer can start to connect to the PBX Server.

20 Chapter 2 Installing InterPBX Communication System

4. You can see the IP address from the “Hyper Terminal” page. You can also

change the IP address here. If you are going to change the IP address,

please press any key within 3 seconds to setup.

Note: The RS-232 DB-9 connector enclosed with the package is with pin 2/3,

pin 4/6, and pin 7/8 shorted.

Installing VG5000 Voice Gateway

Figure: VG5000 Voice Gateway Connection Diagram

This section will guide you through the installation and configuration of

VG5000. For more details, please refer to VG5000 Operation Manual. For other

model Voice Gateways, please refer to their respective manuals.

1. Connecting to Power Cord

Plug one end of the power cord to the power connector on the rear panel of the

VG5000. Plug the other end of the power cord into a power outlet. After you

turn on the power switch on the rear panel, you can verify the function by

checking if the LED labeled Power is on.

Chapter 2 Installing InterPBX Communication System 21

2. Connecting to LAN

You can connect VG5000 to your existing Ethernet network. Please connect one

end of the Ethernet cable to the network RJ45 port of VG5000 and connect the

other end of the Ethernet cable to any 10/100BaseT RJ45 port on DSG SH2500

Switching Hub or your existing switching hub.

3. Connecting to Trunk Lines or Analog Extensions

VG5000 supports trunk lines and analog extensions. Please plug the trunk line

from the Central Office to the CO Line port or plug the analog extension to the

SLT port.

Note: Misplacing FXO equipment into FXS interface and vice versa may

damage your VG5000 Voice Gateway. Be sure that the FXO/FXS interfaces of

VG5000 are connected to suitable telephone lines and equipment.

4. Connecting to External Audio Source (Optional)

InterPBX Communication System supports external and internal MOH (Music

on Hold). VG5000 Voice Gateway provides a “MOH” port for connecting to the

radio or CD player. When callers are placed on hold, the system will play the

music from the radio or CD player. Please insert the plug of your music source

to the MOH port to enable this function.

You can also choose the pre-recorded audio files as the MOH on the PBX Server.

If you choose to use the pre-recorded audio files from the system, do not

connect the external audio source to the MOH port.

5. Connecting to External Paging Facility (Optional)

VG5000 Voice Gateway provides a “Paging” port allowing you to connect

external paging equipment for broadcasting to co-workers. Each VG5000

supports a paging port. If you have more than one VG5000, you can separate

them into different paging zones.

6. Setting Power Failure Transfer (Optional)

VG5000 Voice Gateway provides PFT (Power Failure Transfer) function. When

power failure occurs and there is no backup power, the C.O. lines connected to

port 1 will be switched to port 24. You can connect single line phones to port 24

to pick up calls. The PFT function only works on port 1 with FXO interface

22 Chapter 2 Installing InterPBX Communication System

supported. Port 24 can be FXO, FXS or empty ports. When power is supplied,

the PFT function is not available.

7. Prepare a Computer

Prepare a computer with web browser and be sure it is on the same subnet as of

VG5000. Change the computer’s IP settings if necessary. The suggested web

browser is Internet Explorer version 5.0 or later.

The default settings of VG5000 are as follows:

y Default IP Address: 192.168.1.201:89 (The service port is 89) y Default Gateway: 192.168.1.254 y Default Subnet Mask: 255.255.0.0

8. Login VG5000

y Launch the web browser. On the address bar, enter the VG5000 default IP

address http://192.168.1.201:89 to access the VG5000.

y After you access to the web page of the VG5000, click on the Login icon. On

the login page, please enter the password. The default password is “1234”.

y After you log into the system successfully, the screen will display the System

Information page of DSG VG5000 as below:

Chapter 2 Installing InterPBX Communication System 23

y Basic IP Settings: On the System Information page, please enter the IP

address, gateway address and subnet mask assigned for VG5000. And then

enter the IP address of the PBX Server in “PBX Server.”

y Upgrade VG5000 Software (Optional)

If you need to upgrade the software version of the VG5000 in the future,

please click “Upload Pack” item after you access to VG5000.

9. Settings on InterServer

After you install the VG5000 Voice Gateway, you need to set the VG5000

on the PBX Server.

y Enter License Key : Please login the PBX Server and go to Main

Menu>Operation Management>General Information. Click “Add” and

then enter your license key to make sure the system capacity fits your

needs.

y Creating Analog Gateway List: After you install VG5000, you need to

set VG5000 on the PBX Server. Each VG5000 has a MAC address. You

can see the MAC address from the label on the VG5000 or from the

VG5000 web or through Telnet. Please record your MAC address in

order to register to the PBX Server. Access the PBX Server and login to

the InterPBX Administration Website. Go to Main Menu>Gateway

Configuration>Analog Gateways. Click the Add button to create an

24 Chapter 2 Installing InterPBX Communication System

Analog Gateway.

y Editing Basic Gateway Data

On the “Add Analog Gateway” page, enter the name and the MAC

address of your gateway, and the range of CO Line/SLT ports. The

port number range starts from 1. For example, the CO Line port range

of a 24-port VG5000 with FXO interface is from 1 to 24.

y External Paging Setup: If you connect a paging amplifier to the

“PAGING” port on VG5000, select the “External Paging Enabled”

option to enable this function and assign a specific paging code at

“Paging Code” box. For example, if you assign “111” as the paging

code of your VG5000, please dial “111” to broadcast.

y Music On Hold (MOH) Setup: If the VG5000 Voice Gateway connects

to the external music source to play the Music On Hold, please go to

Main Menu>System Configuration>General Parameters and then

enable the “Music On Hold” function. For more details about MOH,

please refer to Chapter 3 System Configuration/System

Parameters/Music On Hold.

Connecting VG5000 via Telnet

Besides connecting VG5000 via the web browser, you can also connect VG5000

via Telnet.

1. Prepare a computer with Telnet program and be sure it is on the same

subnet as of VG5000.

2. Open the Telnet program and connect to the IP address of VG5000, or enter

C:\Telnet 192.168.1.201 90 (IP addresss+space+90) under DOS. In which

“192.168.1.201” is the default IP address of VG5000 and the communication

port is 90.

Chapter 2 Installing InterPBX Communication System 25

3. Enter the default password “1234” to login VG5000.

4. After you have logged into the VG5000 successfully, a window will be

shown as below. Please follow the instruction to edit the IP address,

Gateway IP, Subnet Mask, PBX IP, Off-premises items and Telnet password

of VG5000. After the editing, please save and exit.

Connecting VG5000 via Console Port

When you forgot the IP address for VG5000, you could connect a computer to

the console port of VG5000 to look up the current IP address, Gateway IP, and

Subnet Mask.

1. Please plug one end of the RS232 cable into the COM port of your

computer and the other end into the Console port of VG5000.

2. Open the Hyper Terminal program. Enter the profile name for the

connection and set the baud rate to 9600 bps.

3. Turn off the power of VG5000. Wait for a while and turn on the power

again. The computer will create a connection with VG5000.

4. You can see the IP address of the VG5000 from the window of Hyper

Terminal. You may also change the IP address in this window. If you want

26 Chapter 2 Installing InterPBX Communication System

to change the IP address, please press any key within 3 seconds.

Note: The RS-232 DB-9 connector enclosed with the package is with pin 2/3,

pin 4/6, and pin 7/8 shorted.

Installing IP Phones

InterPBX Communication System supports IP phones, such as IP590, IP580 or

IP500. IP phones provide various functions like Voice Mailbox, Message

indicator light, Auto-Answer, Mute, Replay, Hold, Transfer, DND, and

Speaker.

You may allow the system to automatically discover and register all the IP

phones connected LAN via Auto Discovery function. Strongly recommend you

to install IP phones via Auto Discovery at the first time. You can also create an

IP Phone manually.

Note: Please do not connect your IP phone to LAN before you start Auto

Discovery.

Chapter 2 Installing InterPBX Communication System 27

Figure: IP580 IP Phone

Create IP Phones via Auto Discovery

InterPBX Communication System provides Auto Discovery function that allows

PBX Server to automatically search and register IP phones. When you install

your IP phones at the first time, we recommend you to start Auto Discovery in

order to simplify the installation procedure. Please make sure the relevant

License Key has been added.

This section will guide you through the installation of IP580. For other types of

phones, please refer to the respective operation manuals.

1. Creating Default Class of Service

Specify a commonly used call limitation as the default Class of Service (CoS).

The default CoS will be applied to all IP extensions when registering with PBX

Server using Auto Discovery.

y Access the PBX Server and login to InterPBX Administration Website. y Go to Main Menu>Group Management>Class of Service. And then click the

Add button to create a new Class of Service (CoS).

y Assign a name and select the call permissions. Click the Submit button to

save and exit. You can modify the CoS or create more entries later. See

28 Chapter 2 Installing InterPBX Communication System

Chapter 6 Group Management/Class of Service for more details.

2. Turn on Auto Discovery

y Go to Main Menu> Operation Management>Auto Discovery. y Click the Turn On Auto Discovery button to enable the Auto Discovery

function. The Current Mode will display “Auto Discovery On.” After

the Auto Discovery is turned on, if you don’t turn off this function

manually, the system would automatically turn off the Auto Discovery

within 2 hours.

y You will find the Class of Service you have created from the list of “Set

Default CoS for Auto Discovery.” Select the one you would like to set

as the default value and click the Set Default button. You will see an

arrow sign pointed to the default CoS. Click the Submit button to save

and exit.

Chapter 2 Installing InterPBX Communication System 29

3. Enter License Key

Before installing the IP phones, be sure to enter the License Key on the PBX

Server set up page. Please login to the PBX Server and then go to Main

Menu>Operation Management>General Information. Click “Add” and then

enter your license key.

4. Settings on IP Phones

You will need to assign the Extension Number, IP Address, Gateway IP

Address, Subnet Mask and PBX Server IP to each IP phone. Please DO NOT

connect your IP phones to LAN before completing the above setups. Press the

middle

button to enter the setup menu. Press the downward button

next to the middle button to move to the next setting item or the upward

button next to the middle button to move to the previous setting item. Use the

keypad to input digits.

y Plug in the power cord of the IP phone and please DO NOT connect the IP

phone to LAN.

y After the system check, press and hold the middle button for 3

seconds to enter the setup menu.

y You will see “EXT NUMBER” shown on the LCD screen. Press the

button to start editing and use the keypad to input the extension number

assigned to the IP phone. Press the

button again to save.

y Press the downward button and the LCD screen will display “IP

ADDRESS.” Press the

button to start editing and use the keypad to

30 Chapter 2 Installing InterPBX Communication System

input the IP address assigned to the IP phone. Press the

button to

save.

y Follow the above procedures to assign the “GATEWAY IP” and “SUBNET

MASK” to the IP phone

y Keep pressing the downward button until the LCD displays “PBX

SERVER IP.” Press the

button to start editing and use the keypad to

input the IP address of your PBX Server. Press the

button to save.

y Keep pressing the downward button until the LCD displays “EXIT

SETUP”. Press the

button to exit.

The IP Phone will automatically search and download the latest version of

software from the PBX Server and then reboot with the latest version.

In DHCP Environment

If your LAN is under DHCP environment, you can skip the settings of IP

address, Gateway IP and Subnet Mask. Instead, you can move to “DHCP”

setup item and enable this function.

y Plug in the power cord of the IP phone and please DO NOT connect the IP

phone to LAN.

y After the system check, press and hold the middle button for 3

seconds to enter the setup menu.

y You will see “EXT NUMBER” shown on the LCD screen. Press the

button to start editing and use the keypad to input the extension number

assigned to the IP phone. Press the

button again to save.

y Keep pressing the downward button until the LCD displays “DHCP.”

Press the

button to start editing and then press the downward

Chapter 2 Installing InterPBX Communication System 31

button to change the selection to “On” Press the

button again to save.

y Keep pressing the downward button until the LCD displays “PBX

SERVER IP.” Press the

button to start editing and use the keypad to

input the IP address of your PBX Server. Press the

button to save.

y Keep pressing the downward button until the LCD displays “EXIT

SETUP”. Press the

button to exit.

Other Setting Items:

y MAC Address: MAC address shows the hardware address of an IP phone. It

is not editable.

y Image Version: Display the current software version of IP phones. y Set Password: You may change the password for logging in to the phone. If

it is blank, the password will not be required when entering the setting

mode. The default password is “1234”.

y Echo Utility: You may input the PBX Server IP here and press Test. The

phone set will send testing packets to the assigned IP address and reply the

round trip time. This can help you test the connection status with the PBX

server.

5. Connect Your IP Phone to LAN

y Connect your IP phone to the Ethernet Network. Please connect one end of

the network cable to the “LAN” port on an IP phone and connect the other

end to the RF45 port of 10/100 BaseT switching hub or Hub on LAN. PBX

Server will automatically search the IP phones on LAN via Auto Discovery

and register them to the phone list.

y During the registration, the LCD screen will display “System Checking”,

“Server Searching”, “Authorizing”, “Load Setting”, and then “Extension

Number and Current Time.”

If the LCD on the phone displays “Server Searching” or “Server Not Found”, it

means there is a connection problem between this phone and PBX Server.

Please check the network settings of your phone or your network environment

32 Chapter 2 Installing InterPBX Communication System

again. Please also make sure the IP address and extension number are not used

by other extension accidently.

6. Turning Off Auto Discovery

Access PBX Server and login InterPBX Administration Website. Go to Main

Menu>Operation Management>Auto Discovery. On the “Auto Discovery”

page, click the “Turn Off Auto Discovery” icon to disable the Auto Discovery

function. This will prevent your InterPBX Communication System from being

accessed by unauthorized users.

After the Auto Discover was turned on, if you didn’t turn it off manually, the

system will automatically turn off the Auto Discovery function within 2 hours

in order to avoid unauthorized users using the system.

Create IP Phones Manually

Besides creating IP phones via Auto Discovery, you can also create IP phones

one by one manually. Please make sure the relevant License Key has been

added.

1. The Settings on the Administration Website

y Before installing the IP phones, be sure to enter the License Key on the

PBX Server set up page. Please login to the PBX Server and then go to

Main Menu>Operation Management>General Information. Click

“Add” and then enter your license key.

y Go to Main Menu>Extension Management>InterPhone List. Click the

Add button to create IP phones.

y Enter the IP phone’s extension number on “Extension No.”

Chapter 2 Installing InterPBX Communication System 33

y Select the phone type from the “Phone Type” list. y Enter the password of the extension in order to edit the personal

options from the InterPBX Management Website. The password must

only be numbers. If you create IP phones via Auto Discovery, the

phone password is blank. You can set the password later.

y Enter the MAC address of the phone in “MAC Address.” MAC is the

hardware address of the IP phone. Each IP phone has a unique MAC

address. You can see the MAC address of each IP phone by the

following procedure. Press and hold

on the phone for 3 seconds

to enter the setup menu. And then keep pressing

until the LCD

screen displays “MAC ADDRESS.”

You may set the other items later. For other settings, please refer to Chapter

5 Extension Management/Set IP Phones, SoftPhones, and Virtual Phones.

2. Settings on IP Phones

Press and hold

button on the IP phone for 3 seconds. The LCD screen

will display “EXT NUMBER.” Please refer to the above IP phones settings to

enter the relevant Extension Number, IP Address, Gateway IP, Subnet Mask,

and the IP Address of PBX Server.

3. Connect IP Phones to LAN

y After you finish the above procedure, please connect your IP phone to LAN. y During the registration, the LCD screen of IP phone will display “System

Checking”, “Server Searching”, Authorizing”, “Load Setting”, and finally

show the Extension Number and the Current Time.

y Login to InterPBX Administration website. Go to Main Menu>Extension

Management>InterPhone List. You will see the list of all the IP phones. You

can start to set the phone functions one by one.

Chapter 3 System Configuration 35 Chapter 3 System Configuration 35

Chapter 3 System Configuration

This chapter guides you through the initial and basic configuration of InterPBX

Communication System. The basic system settings include PBX Server IP

settings, your company information, your business hours, system-wide speed

dialing, call restriction, password management, and system date and time

settings.

36 Chapter 3 System Configuration

General Parameters

Please login to the InterPBX Administration Website. Go to Main Menu>System

Configuration>General Parameters. In this section, you can edit the basic IP

settings of your PBX Server.

Basic IP Settings

IP Address, Default Gateway, Subnet Mask

The PBX Server is shipped with default IP settings as below for your initial

configuration. You may edit the IP address, Gateway IP and Subnet Mask in

this section.

y Default IP Address: 192.168.1.200 (service port 88) y Default Gateway IP: 192.168.1.254 y Default Subnet Mask: 255.255.0.0

The PBX Server requires a static IP address. Either real IP or pseudo IP can do.

The Gateway IP is for the destination host to route the IP packets addressed to a

host outside the local subnet. Notice that all the on-site IP extensions need to be

set in the same subnet of your PBX Server.

Host Name

Enter the Host Name of your PBX Server. If you have added the host name to

your name resolution systems (DNS), you could connect to your InterPBX

Communication Server by typing its name on a web browser. If you haven’t

added it to your DNS, you can still access PBX Server using its IP address.

Chapter 3 System Configuration 37

DNS 1 and DNS 2

Enter the Domain Name Server (DNS) IP address offered by your ISP. You can

enter up to 2 DNS IP addresses.

Email Settings

Please enter the SMTP Server IP address in the field SMTP Server and the

e-mail address in the field E-mail Account. When the Unified Messaging

function is enabled, the system can send the voice message file via e-mail

through the SMTP server to the extension user.

You may also enable the SMTP Authorization function if the SMTP Server

needs to verity the e-mail account and password. If you enable SMTP

Authorization, please be sure to enter the Account Name and Password.

NAT Settings

If your network environment is behind NAT, you can assign a virtual IP to the

PBX Server. If you have off-premises extensions or Joint Servers, you need to

enter the NAT information of the PBX Server to allow the off-premises

extensions to connect to your PBX Server which is behind NAT. Please check

the “Behind NAT” box and enter your real IP address of NAT equipment on

the “NAT IP Address” box. If there are no off-premises extensions or Joint

Servers in your InterPBX Communication System, you don’t need to edit the

settings of NAT.

Open a Communication Port: When there are off-premises extensions or Joint

Servers in your PBX Server that is behind NAT, besides the above settings, you

also need to open a service port on your NAT Equipment or Firewall in order to

allow the off-premises equipments to connect to your PBX Server.

Off-Premises Extensions: When you have off-premises extensions, please open

the UDP6046 port on your NAT equipment. If the off-premises extensions are

also behind NAT equipment, you also need to open the UDP6046 port for them.

Please refer to Chapter 5 Extension Management/ Off-Premises Extensions for

more details.

Joint Servers: When you have Joint Servers, please open the UDP6055 port on

your NAT equipment. If the Joint Servers are behind NAT, you need to turn on

38 Chapter 3 System Configuration

the NAT settings of the PBX Server and open the UDP6055 port on NAT

equipment.

NAT Proxy: If you have DSG NAT Proxy, you only need to register the PBX

Server to the NAT Proxy without opening the communication port for PBX

Server or off-premises extensions.

Set QoS

The InterPBX Communication System supports 802.1p/Q. You can put a check

on “QoS Enabled” box to activate the system’s QoS. You may set VLAN ID and

802.1p Priority. If you activate the QoS function, your settings need to be the

same as the settings of Switch.

The QoS of system only covers local IP extensions and Voice Gateway. You can

also set the individual QoS on Off-Premises Extensions, Off-Premise Voice

Gateway and Joint Server. For more information about QoS settings, please

refer to Chapter 4 Gateway Configuration/Off-Premises Gateways, Chapter 5

Extension Management/Off-Premises IP Extensions, and Chapter 10

Multi-Server Management/Joint Server.

Note: If the values of VLAN ID and 802.1p Priority are both set as “0”, the QoS

will be disabled.

Music on Hold

When a call is on hold, the system will provide music or prerecorded

announcements for callers. Select the check box of “Music on Hold” to enable

this function. You can select the Music on Hold (MOH) source from an external

audio source connected to a gateway or from a prerecorded file.

y You can play the music by selecting the prerecorded audio file. Please click

the check box of “From Files” and select a preferred file from the list.

y If you connect the system with a radio or CD player, you can click the check

box “From Gateways” and select the specific gateway from the gateway list.

Make sure the gateway is properly connected to the audio source before you

select the specific gateway.

You may also amplify the MOH volume if necessary.

Chapter 3 System Configuration 39

MOH Converter: DSG provides a MOH Converter program allowing you

convert your own music on hold or queue announcement files from WAV or

MP3 format to the format compatible to the PBX Server. To know to how

convert your own MOH files, please consult our dealers or sales

representatives.

Ringing Patterns

You can assign different ringing patterns to identify internal or external calls.

From the “Internal” list, select your preferable ringing pattern for Intercom calls.

From the “External” list, select one for calls from CO lines.

Company Information

Go to Main Menu>System Configuration>Company Information. In this section,

you can edit your company’s contact information.

Business Hours

Your office hour settings will affect the behaviors of Automated Attendant,

Class of Service (CoS) or Voice Mail System. Automated Attendant may play

different greetings when the office is open, close or during lunch breaks.

Permissions of making calls after office hour may be limited by the settings of

CoS.

40 Chapter 3 System Configuration

1. Go to Main Menu>System Configuration>Business Hours.

2. From the “Operating Mode” list, select your company’s operating mode. y Auto: It will switch automatically according to the schedule you set

below.

y Business Hours: The open hours on workdays. y Break Hours: The lunch breaks on workdays. y After Hours: The hours after open hours on workdays. y Closed: The hours of non-workdays.

3. When setting the business hours, select the check box of the workday and

enter the opening hours and lunch breaks. If your company doesn’t offer

lunch breaks or days off, you may enter “0”.

Ring Assignment: The Ring Assignment will follow the operating hours you

set to play different greetings for incoming calls. For more information about

Ring Assignment, please refer to Chapter 4 Gateway Configurations/Analog

Gateways.

Holidays: For holidays, you may go to Holiday to set hours and dates. For

more information about Holiday, please refer to Chapter 3 System

Configuration/Holiday.

Change Current Operating Mode: You may manually change your current

Operating Mode to be Business Hours, Break Hours, After Hours or Closed.

Select the preferable item from the “Operating Mode” list. Notice that your

operating mode will be permanently in the mode you selected. To let the

operating mode be switched automatically according to the business hours you

set, please select “Auto” again.

Chapter 3 System Configuration 41

Night Service: You may assign a specific programmable button as Night

Service. And press to switch the Operation Mode to Closed Hours. The AA

menu and operators will be changed accordingly. This function is available for

Extension Group Button Mapping. Please refer to Chapter 6 Group

Management/Button Mapping Groups for more details.

Holidays

In addition to the regular business hour schedule, you can set the holiday list so

that the system will play different greetings on holidays. For more information

about Business Hours, refer to Chapter 3/System Configuration/Business

Hours.

1. Go to Main Menu>System Configuration>Holiday.

2. Click the Add button to create an entry in the Holiday List.

3. Set the date range (month/day) of the holiday. If there is only one day off,

enter the same date on the Date range boxes.

4. Input the name of the holiday.

5. Select an AA Menu from the list. If you select the “Enable Holiday”

checkbox in CO Lines or the Ring Assignment setting of CO Line groups,

the system will play the greeting of AA Menu you assigned here during

holidays.

For more information about Ring Assignment, please refer to Chapter 4

Gateway Configuration/Analog Gateways.

System Speed Dialing

You can edit frequently dialed phone numbers in this section to be used

throughout the system.

42 Chapter 3 System Configuration

1. Go to Main Menu>System Configuration>System Speed Dialing.

2. On the “System Speed Dialing” list, select one entry.

3. Enter the destination phone number or extension number on the “New

Number” box.

4. Enter the name of the destination or other description on the “Comment”

box.

5. Click the Assign button to save.

Note: You could set up to 1000 sets of system speed dialing. If you would like

to set a Speed Dialing Number for an outgoing call, please remember to add the

CO line access code (e.g. 0). For long distance calls, please enter your long

distance access code (e.g. 1) followed by the area code and phone number. For

international calls, please enter your international access code (e.g. 011) and the

country code, area code, and phone number.

To make a call with System Speed Dialing:

1. Lift the handset or press the Speakerphone button.

2. Press the System Speed Dialing function code #20.

3. Press the System Speed Dialing number (e.g. 000) to call.

Call Restriction

You can set the long distance and international access codes in the “Call

Restriction” in order to help the PBX Server to identify the types of outgoing

calls (e.g. local, long distance or international calls). Call Restriction allows you

to create call restrictions, exceptions, and routing tables in Class of Service.

Chapter 3 System Configuration 43

1. Go to Main Menu>System Configuration> Call Restriction.

2. Set the “Country Code (e.g. 1 for US)” and “Area Code (e.g. 213 for L.A.)”

where PBX Server is located.

3. On the list of “Long Distance Call Prefixes”, select one entry and assign the

long distance call prefix plus the area code (e.g. 1310 for Beverly Hills) in

the “New Number” box. Press the Assign button to save. Repeat this

procedure to edit all the area codes of your country.

4. Follow the above-mentioned procedure to set the “International Call

Prefixes”.(e.g. 011)

Route

If you want to activate the ARS function, please create the available routes for

selection. The route types for selection are CO Group, SIP Trunk, or Joint

Server.

44 Chapter 3 System Configuration

1. Go to Main Menu>System Configuration> Route. Click “Add” to create a

route.

2. Enter the name of the route.

3. Set the route in different time sections: Business Hour, Break Hour, After

Hour, Closed Hour, and Holiday. The route types for selection are as

follows:

● CO Group: Select the CO group from the list. For more information

about CO Group settings, please go to Main Menu>CO Line Groups.

● Account: Please select your SIP Account. For more information about

SIP Account settings, please refer to Set SIP Proxy.

● Joint Server: You may also make the phone call through another

InterPBX’s CO line. Please select your Joint Server here.

Function Code

The System provides Function Code for administrators to configure so that the

extension users can use multiple functions by entering the specified function

code directly on their extensions.

Chapter 3 System Configuration 45

Go to Main Menu>System Configuration>Function Code. You will see all the

function items available for configuration. Please enter a code for each function

and click the save button.

Password Management

This section allows the administrator to change Administrator’s Login Name

and Password and reset User Password to login the InterPBX Administration

website. The administrator will be authorized to edit all the administrative

features after login. For security reason, please change the Administrator

Password from time to time.

Go to Main Menu>System Configuration>Password Management. Input your

current password and new password. Re-enter the new password for

confirmation. Your password shall not exceed 8 characters. The password is

case sensitive.

Each user will be authorized to edit the user features after login. When users

forgot their passwords, you may reset their passwords to the default setting

(The default extension password is blank if created by Auto Discovery).

46 Chapter 3 System Configuration

You can connect VMS Server from a telephone remotely and edit the relevant

settings. The default remote Tel. Programming Password of VMS Server is

“1234”. For more information about DTMF remote control settings, please refer

to Appendix F.

Set System Date/Time

Go to Main Menu>System Configuration>Set System Date/Time in order to set

the current date and time of your time zone for your InterPBX Communication

System. The system date and time affect the play contents of business hours

and greetings. If you change the system Date and Time, the time display on the

phones will be updated after 8 seconds to 2 minutes.

Miscellaneous

Go to Main Menu>System Configuration>Miscellaneous. You could edit

system timeout settings and authorization code.

Timeout Settings

Chapter 3 System Configuration 47

y Forward Voice Mail Timeout: If an incoming call is not answered, the call

will be transferred to the voice mail system after the timeout.

y Call Park Timeout: When the call is parked, if no one picks up the call, it

will be bounced back to the original extension parked the call after the

timeout.

y Call Hold Timeout: When a call is placed on hold without any further action,

the call will ring back to the original extension after the timeout.

y Transfer Timeout : When a call is blind-transferred to a target extension but

no one answered or the extension is busy, the call will be transferred to the

Auto Attendant after the timeout. The Auto Attendant will guide the caller

to other selections according to the settings of Transfer Options. If the

personal forwarding is set on the target extension, the call will be forwarded

according to the personal forwarding setting.

y First-Digit Timeout: When a line is accessed, if the first digit is not

received within the period, the system will disconnect the call and release

the line.

y Inter-Digit Timeout : When entering digits, if the next number is not

received within the period, the system will see it as a typing ending signal

and use the numbers been input to make calls.

y Auth. Code Timeout: When entering authorization code, if the next digit is

not received before the timeout, the system will deem the input ends and

send out the numbers received.

y Max Trunk-to-Trunk Duration : The function allows CO line incoming calls

to be transferred out through the other CO lines. Please enter the call

duration limit time. When the period expires, the system will cut off the call

and release the CO line.

Authorization Code

Authorization Code function allows callers to make calls, such as long distance

or international calls, from an extension with limited call permissions.

Note: The Authorization Code function needs to be implemented with ARS.

1. Select the “Authorization Code Enable” check box.

2. Input the length of Authorization Code.

3. Input the Cancel Symbol allowing users to clear the authorization code

been input. This will help the user not to input the phone number again.

48 Chapter 3 System Configuration

Auth. Code Timeout: Please input the Authorization Code timeout at the above

Timeout Settings.

Assign Authorization Code: Please go to Main Menu>Group

Management>Authorization Code to assign Authorization Codes.

Chapter 4 Gateway Configuration 49

Chapter 4 Gateway Configuration

Voice Gateways are bridges between IP and traditional (PSTN) telephony

networks. Gateways connect the InterPBX Communication System to CO trunk

lines or analog extensions. InterPBX supports various gateway types including

analog gateways, digital gateways, and off-premises gateways. If you have

included Digital Gateway, Recording System, CTI Gateway, SIP Proxy or NAT

Proxy in your purchase, please consult their specific documentation for further

details on the settings.

Note: Savanna 8000 Series doesn’t include the Voice Gateway. If you would like

to add the Voice Gateway into the Savanna 8000 Series IP-PBX system, we offer

the VG5000 Voice Gateway or VG6000 Digital Gateway for your options.

50 Chapter 4 Gateway Configuration

Analog Gateways

InterPBX Communication System supports analog gateways such as VG5000.

VG5000 Voice Gateway provides up to 24 ports of FXO (for CO Lines), FXS (for

SLT), or mixture of both. Every unit of VG5000 also provides interfaces for

external music-on-hold and paging equipment.

Creating Analog Gateway List

After the installation of your gateway, you will need to incorporate it into the

PBX Server.

1. On the main screen go to Main Menu>Gateway Configuration>Analog

Gateways.

2. Click the Add button to add a new gateway to PBX Server’s “Analog

Gateway List”.

3. Name this gateway and enter its MAC address. MAC address is the

hardware address of the gateway, you will find this information printed in

a label at the back of the gateway or alternatively, you may obtain this

information from the VG5000 Voice Gateway’s web page or Telnet. Please

see Chapter 2 “ Installing InterPBX Communication System “ for more

details.

4. Enter the range of CO Line and/or SLT ports of your gateway. Please note

the port number starts from 1, for example, if the VG5000 you purchased is

a 24-port FXO gateway, in the “CO Line Ports (From/To)” fields enter “1”

and “24” respectively.

5. At “CO Line Extension Base” input the starting number of CO Line

extension number. The CO Line extension numbers will be assigned

automatically. For example if you set it as 4000 for an 8-port FXO gateway,

the CO Line extension number of this gateway will be 4000-4007.

Chapter 4 Gateway Configuration 51

6. At “CO Line Index Base” input the starting number of the CO line display

number. For example if you set it as 1, when there is an incoming call from

port 1, the IP phones will display “Incoming Call CO 1” instead of the

extension number of the port, helping you to identify the CO lines being

used.

7. From “Jitter Buffer Depth” drop down menu, depending on your

bandwidth and CODEC, start with a Jitter Buffer with the minimum value.

Jitter Buffer will dynamically adjust its value according to the party you are

talking to and the bandwidth of the call but never below the value you

have set. The higher value of Jitter Buffer will reduce the chance of packet

loss during a call but might cause delays and the lower value will help to

transmit the voice packets faster but might cause packet loss problems.

8. Min. Loop Current Drop Time: Normally, when a telephone call terminates,

the telephone company sends a momentary drop in loop current to signal

the disconnect event. Please enter the minimum Loop Current Drop time in

this field to avoid the IP-PBX system detecting the disconnect signal

incorrectly. The value should not be too long or too short. Recommended

value is 400 ms.

Note: If later on your needs grow and you need to add more CO lines or

channels into an existing gateway, you will need to delete the existing gateway

from the list and create a new one with new trunk or channel numbers.

External Paging & MOH Functions

If you connect a paging facility to your gateway, click the “External Paging

Enabled” check box to enable the Paging function. Assign a specific paging

code to be used for announcement via this gateway.

Using External Paging

If you assign “111” as the paging code of your VG5000, you need to dial “111”

to page a person via the external paging facility on this VG5000. Refer to

Chapter 2 Installing InterPBX Communication System/Configuring VG5000

Voice Gateway for more details.

Editing Analog Gateways

After the addition of the new gateway, you may edit the settings to control the

input and output volume of the CO Line and SLT ports.

1. Go to Main Menu>Gateway Configuration>Analog Gateways.

52 Chapter 4 Gateway Configuration

2. From the “Analog Gateway List” choose the gateway to edit then click on

the Modify button.

3. You will see the gateway type, software version running as well as the

ports that correspond to CO Line or SLT.

4. If needed, you may change in this screen the name of the gateway, MAC

address, CO Line Index Base, Jitter Buffer as well as to enable the Paging

feature. For more detailed information please see above section “Creating

Analog Gateway List”.

Configuring CO Line Ports

1. Go to Main Menu>Gateway Configuration>Analog Gateways.

2. Select a gateway from the “Analog Gateway List” and click the Port Setting

button.

3. Select a CO Line port from the “Port List” and click the Port Parameters

button to configure the settings for this port.

Chapter 4 Gateway Configuration 53

Setting CO Line Extension Number

An extension number in InterPBX Communication System might represent a

CO line, an analog extension, an IP extension, a CO line group or an extension

group.

In “Extension Number” it will display the extension number of this CO Line

port automatically based on your settings under “CO Line Extension Base”, you

may also change the extension number from here. Please select “Enabled” to

activate this port. For maintenance purpose, you may click again on “Enabled”

to disable this port. In “Description” input the telephone number or other

information for this port.

CODEC

Choose the preferred CODEC from PCM, G.723.1 or G.729 supported by

InterPBX Communication System. PCM CODEC compresses and decompresses

voice conversation to 64 Kbps providing the best speech quality but consumes

larger bandwidths. G.723.1 CODEC uses 6.3 Kbps, the less bandwidth from the

three choices but may result in poorer call quality and finally G.729, which uses

8 Kbps.

Fax

Enter the extension number of the SLT port where the fax machine is connected.

When detecting fax tones, the fax will be automatically transported to the fax

machine. You may set all the CO Lines to the same fax extension or to different

fax extensions, as you need.

T.38 Fax: InterPBX system also support T.38 protocol allowing fax to be

transmitted across the IP networks. When selected, please also select the T.38

setting at SLT port. If you adopt T.38 fax, at the above Fax Ext Number, you

may set a local fax extension number, an off-premises fax machine or a fax of

Joint Server.

Ring Assignment

Ring Assignment allows you to assign a specific extension, extension group or

AA Menu to answer incoming calls from a specific CO line or CO Line Group.

In the Ring Assignment field, choose the option of “Local” or “From CO

Group”.

54 Chapter 4 Gateway Configuration

y If you select “Local”, input a specific extension number, extension group or

AA Menu access code to answer incoming calls for this port in Business

Hours, Break Hours, After Hours and Closed. The most usual setting is to

assign an AA Menu to answer phone calls, taking advantage of the

InterPBX’s VMS server. To learn more about AA Menu settings, please refer

to Chapter 7 Voice Mail Configuration. If you would like to play specific

greetings for holidays, select the “Enable Holiday” checkbox so that the

system will play the holiday greetings you assigned on the Holiday List. For

more details about Holiday settings, please refer to Chapter 3 System

Configuration/Holidays.

y If you select “From CO Group”, incoming calls will be answered according

to the Ring Assignment of the CO Line Group you have assigned. For more

details about CO Group’s Ring Assignment, please refer to Chapter 6 Group

Management/CO Group.

Silence Compression

The silence compression feature reduces network traffic whenever a period of

silence is detected during a conversation. When there is silence in a

conversation, silence indicator packets will be sent by the system instead of full

voice packets to reduce traffic. Select the check box of “Silence Compression

Enabled” to enable this function or uncheck to disable it.

Caller ID

InterPBX’s CO Line ports have the ability to detect either DTMF or FSK Caller

ID modes. You may select this item to enable this function. Please note that

your local telephone company must provide Caller ID service in order to

display the incoming party’s number.

Pickup After Rings

On the “Pickup After Rings” column, input the number of the ring count. If you

set the ring count as “2”, the system will pickup the incoming call after 2 rings.

Flash Timing

Under the application of Centrex, you can set the Flash Timing on the IP

phones through the system. The Flash function was set on the MENU button of

the IP phones. After you complete the Flash Timing setting, you can start to use

some of the Centrex functions. Please make sure the IP phone’s Flash Timing

must be within system’s Flash Timing.

Chapter 4 Gateway Configuration 55

Voice Gain Level

You may edit the input and output volume of:

y Transmit Gain: this field edits the TX Gain of the CO Line ports, or how loud

the external party hears your voice.

y Receive Gain: this field edits the RX Gain of the CO Line ports, or the volume

of incoming calls.

y DTMF Volume: this field edits the intensity of the DTMF tones in the CO

Line ports.

Apply Settings

After finalizing the settings in this port, if you want them to be identical as this

one, you may apply the settings to other CO Line ports present in the system,

saving you the hassles of configuring each port individually.

1. Click on “Apply Setting to” and a new window will appear displaying the

parameters you have entered.

2. Apply or change the settings, as you desire.

3. Enter in the field “Apply to Port” the first and the last CO Line port you

would like to be identical to the present settings. If the range selected

includes this port, the new settings, if any, will also be applied to this one.

Configuring SLT Ports

1. Go to Main Menu>Gateway Configuration>Analog Gateways.

2. Select a gateway from the list then click the Port Setting button.

3. Select a SLT port and click the Port Parameters button.

Flash Time

You may set the Flash Timing of the SLT ports to check whether the Hook

action (quick press and release of the phone’s hook) on the analog phones

56 Chapter 4 Gateway Configuration

works fine. Analog extensions require the Hook action to transfer, answer call

waiting, or create a conference call. Make sure the phone set’s Flash Timing

must be within the system’s Flash Timing.

Caller ID

The SLT ports have the ability to detect either DTMF or FSK Caller ID modes.

You may this item to enable this function. Note that your local telephone

company must provide Caller ID service in order to display the incoming

party’s number.

T.38 Fax

T.38 is a fax relay protocol. It allows fax to be transported across IP networks

between fax machines. When selected, please be sure the T.38 is also enabled at

the CO line port setting.

Note: You may set the SLT port to get a CO line when off-hooked, without the

need to pressing the CO line access code first. Go to Main>Extension

Management>Analog Port list for more details.

Voice Gain Level

You may edit the input and output volume of:

y Transmit Gain: this field edits the TX Gain of the CO Line ports, or how loud

the external party hears your voice.

y Receive Gain: this field edits the RX Gain of the CO Line ports, or the volume

of incoming calls.

y DTMF Volume: this field edits the intensity of the DTMF tones in the CO

Line ports.

y Dial Tone Volume: this field edits the dialing volume of the SLT ports, or the

dial tone volume of analog phones.

Setting Analog Extension Numbers

Each of the SLT port in the InterPBX can be connected to an analog phone. After

setting up the Analog Gateway, go to Main Menu>Extension

Management>Analog Port List. You will see all your available SLT ports in this

window. Select a SLT port and click the Modify button to configure your analog

extension. For more details, please refer to Chapter 5 Extension Management.

Off-Premises Gateways

Chapter 4 Gateway Configuration 57

InterPBX Communication System supports Off-Premises Voice Gateways. By

installing Voice Gateways in remote offices or satellite offices, enterprises can

enjoy great savings on inter-office communication. Also, more savings can be

achieved through the remote gateways by utilizing it to place outgoing local

calls to local customers, saving this way the long distance charges since through

the remote gateway this call is now same as a local call.

Creating Off-Premises Gateway List

After installing your gateway in the remote office, you will need to add it to the

PBX Server.

1. Go to Main Menu>Gateway Configuration>Off-Premises PSTNGW.

2. Click the Add button to add a new gateway to PBX Server’s “Off-Premises

Analog Gateway List”.

3. Refer to the section above “Analog Gateways” to configure the “Name”,

“MAC Address”, “Gateway Type”, “CO Line Ports”, “SLT Ports”, “CO

Line Index Base” and “Jitter Buffer Depth”.

4. For Off-Premises Gateways, you will need to enter its “IP Address”. The IP

Address assigned to this Off-Premises Gateways has to be a REAL

(PUBLIC) IP address. In the Signal Port, InterPBX will automatically preset

the port therefore you do not have to make any changes here.

5. Min. Loop Current Drop Time: Normally, when a telephone call terminates,

the telephone company sends a momentary drop in loop current to signal

the disconnect event. Please enter the minimum Loop Current Drop time in

this field to avoid the IP-PBX system detecting the disconnect signal

incorrectly. The value should not be too long or too short. Recommended

value is 400 ms.

58 Chapter 4 Gateway Configuration

6. If you have connected any type of paging device, select the “External

Paging Enabled” to activate this feature, then create and enter a code in the

Paging Code field to use this feature.

7. Lastly, if desired, you may activate the QoS feature from your Off-Premises

Gateway by enabling it here.

After finishing creating the gateways, you may edit the ports incoming and

outgoing volume as described in “Editing Analog Gateways” section above.

Configuring Off-Premises CO Line Ports

1. Go to Main Menu>Gateway Configuration>Off-Premises PSTNGW.

2. Select a gateway from the list then click the Port Setting button.

3. Choose a CO Line port and click the Port Parameters button to edit the

settings to this port.

4. Refer to the sections above Editing Analog Gateway/Configuring CO Line

Ports to configure the “Extension Number”, “CODEC”, “Fax”, “Ring

Assignment”, “Silence Compression”, “Caller ID Detection” and “Voice

Gain Level” as well as how to duplicate the settings.

5. The system will automatically preset a port to the Media Port. Please do not

change this port number.

Make a call via off-premises CO Line port

To make a call via your off-premises CO Line port, pick up the phone, dial the

remote CO Line extension number then dial the phone number you need to call.

For long distance calls, please dial remote site’s long distance access code

followed by the phone number to be dialed.

Chapter 4 Gateway Configuration 59

Configuring Off-Premises SLT Ports

1. Go to Main Menu>Gateway Configuration>Off-Premises PSTNGW.

2. Select a gateway from the list then click the Port Setting button.

3. Choose a SLT port and click the Port Parameters button to edit the settings

to this port.

4. Refer to the sections above Editing Analog Gateways/Analog Port to

configure this port.

5. Leave the Media Port as it is since it is assigned by the system

automatically.

6. Each off-premises SLT port can connect to an off-premises analog phone.

Go to Main Menu>Extension Management>Analog Port List and you will

see all your available SLT ports in this window. Select a SLT port and click

on Modify to configure your off-premises analog extension. For more

details, please refer to Chapter 5 Extension Management/Analog

Extensions.

Make a call to off-premises SLT port

To make a call to extensions connected on off-premises SLT ports, simply pick

up the phone and dial the remote SLT extension number normally as you

would for calling extensions in your office. You can make calls to remote

extensions as the way you do for calling extensions in the same office.

60 Chapter 4 Gateway Configuration

Recording System

The InterPBX System supports recording system to improve enterprise

productivity and enhance efficiency of business communication. InterPBX

Communication System supports two types of recording systems: the built-in

recording system which provides Record-on-Demand and/or

Store-on-Demand functions and the stand alone professional recording system.

y Blaze5000 Series: It provides built-in Record-on-Demand and

Store-on-Demand recording functions. When there is no one on the

Store-on-Demand list, Record-on-Demand will still be available. Up to 10

concurrent recording channels will be provided.

y Blaze1200 Series: It supports built-in Record-on-Demand with maximum

10 concurrent recording channels.

y Savanna8000 Series: It doesn’t support built-in recording function and an

additional recording system is required.

Note: The recording function does not apply to analog and off-premises

extensions.

Setting Store-on-Demand (Available on Blaze5000 Series)

Chapter 4 Gateway Configuration 61

The built-in Store on Demand function allows the system to record calls for

extensions on the Store on Demand list when conversation starts. The extension

user can save the recording at any time during a call by pressing the Record

button. Recordings will not be automatically saved for the concern of storage

space. User may also retrieve the recording when needed.

1. Go to Main Menu>Gateway Configuration> Recording System.

2. Enter the IP address of the PBX Server in the field “IP Address”.

3. The default value of “Port Num” is “10”, which means the system can

record up to 10 extensions at the same time. You need to assign up to 10

extensions on the Store-on-Demand List.

4. Define the ”Storage Limitation” of the hard disk and the “Auto Purge

Days”. When exceeding the hard disk limit, the system will delete the

recordings of days as set on “Auto Purge Days”. For example, if the

Storage Limitation is “80%” and the Auto Purge Days is “7”, the system

will automatically delete previous 7-days recording data when it reaches

the storage limitation.

5. Put a check on the item of “Recording Beep” according to your company’s

requirement or the Government’s regulation. When the Recording Beep is

activated, the caller and the receiver will hear the beep sound every 15

seconds during a call.

6. Recording Flash: You can enable the Recording Flash function. After the

Recording Flash is enabled, the recording lamp on the recorded extension

will keep flashing until the call is hung up. If you need to record to an

extension confidentially, please do not enable this function.

Store on Demand List

62 Chapter 4 Gateway Configuration

You can add up to 10 extensions in the Store on Demand list. The system will

continuously record to these extensions during a call.

1. Go to Main Menu>Gateway Configuration> Recording System.

2. From the ”Extension” list, move the extension number to the “Store on

Demand” list. The system can record to up to 10 extensions at once.

Activate the Recording

1. During a call, press the specific programmable button or Rec/Play button

to active the function.

2. After the call is hung up, the system will stop recording.

Note: The recording function does not apply to analog and off-premises

extensions.

Playing the Recording

The extension users can play the recording from their IP phones directly.

1. Press the specific programmable button or REC/Play button or function

code #42 to play recordings

2. Enter the password and then press #. (The password for the recording

system is same as your mailbox password. The default password is blank.)

3. Follow the system prompts, press 1 to play the latest recording or press 2 to

check the recording by entering date and time. Please follow the procedure

below to assist you in listening to the recording.

y In the recording system, you can press * anytime to return to the previous

menu or back to the main menu.

y When playing recordings, you can press 0 (zero) to skip the time stamp

announcement of the recording.

Chapter 4 Gateway Configuration 63

y If your UMS is enabled and the e-mail is set, after playing a recording, you

will be prompted to send the recording you just played to your email.

y For Store on Demand recording, you can only play the latest, previous or

next recording for the current day.

y When you input the time for searching recording, the system will play the

recent recording after the time.

Setting Record-on-Demand (Available on Blaze5000 and Blaze1200 Series)

The built-in Record on Demand allows the system to record calls after users

pressing the Record button. Calls will be recorded and saved after pressing the

Record button. User may also retrieve the recording when needed.

1. Go to Main Menu>Gateway Configuration> Recording System.

2. Enter the IP address of the PBX Server in the field “IP Address”.

3. The default value of “Port Num” is “10”, which means the system can

record up to 10 extensions at the same time. You may set the permission

from extensions’ Class of Service.

4. Define the ”Storage Limitation” of the hard disk and the “Auto Purge

Days”. When exceeding the hard disk limit, the system will delete the

recordings of days as set on “Auto Purge Days”. For example, if the

Storage Limitation is “80%” and the Auto Purge Days is “7”, the system

will automatically delete previous 7-days recording data when it reaches

the storage limitation.

5. Select the “Recording Beep” checkbox according to your company’s

requirement or the government’s regulation. When the Recording Beep is

activated, the caller and the receiver will hear the beep sound every 15

seconds during a call.

6. Recording Flash: You can enable the Recording Flash function. After the

Recording Flash is enabled, the recording lamp on the recorded extension

will keep flashing until the call is hung up. If you need to record to an

64 Chapter 4 Gateway Configuration

extension confidentially, please do not enable this function.

Activate the Recording

1. During a call, press the specific programmable button or Rec/Play button

to active the function.

2. After the call is hung up, the system will stop recording.

Note: The recording function does not apply to analog and off-premises

extensions.

Playing the Recording

The extension users can play the recording from their IP phones directly.

1. Press the Play button to play the recording. On programmable button

supported phones, you can press a specific programmable button that is

assigned via Button Mapping to play the recording. Or press the

REC/PLAY button, if applicable.

2. Enter the password and then press #. (The password for the recording

system is same as your mailbox password. The default password is blank.)

3. Follow the system prompts, press 1 to play the latest recording or press 2 to

check the recording by entering date and time. Please follow the procedure

below to assist you in listening to the recording.

Chapter 4 Gateway Configuration 65

y In the recording system, you can press * anytime to return to the previous

menu or back to the main menu.

y When playing recordings, you can press 0 (zero) to skip the time stamp

announcement of the recording.

y If your UMS is enabled and the e-mail is set, after playing a recording, you

will be prompted to send the recording you just played to your email.

y For Store on Demand recording, you can only play the latest, previous or

next recording for the current day.

y When you input the time for searching recording, the system will play the

recent recording after the time.

Setting Professional Recording System

InterPBX Communication System can also work with the professional recording

system Blaze Logger that provides more powerful recording and searching

functions.

1. Go to Main Menu>Gateway Configuration>Recording System.

2. On the “IP Address” field, enter the IP address of the recording system.

You don’t need to set the other items.

The Port Num will display the license numbers of your Blaze Logger. Please

refer to the Blaze Logger Administration Guide for more information.

66 Chapter 4 Gateway Configuration

Digital Line Gateways

When adopting VG6000 or VG7000 Digital Gateway, your InterPBX IP

Communication System, including Blaze, Savanna series IP-PBX, needs to

enable the relevant function as well.

Adding Digital Gateway

You need to set the MAC address of Digital Gateway to the PBX Server.

1. Go to Main Menu>Gateway Configuration>Digital Line Gateways,

the ”Digital Gateway” list will be displayed.

2. Click the Add button to create a Digital Gateway. Before you add a new

gateway, make sure the relevant License Keys has been added.

Chapter 4 Gateway Configuration 67

3. Input the required data of this gateway.

Name: Enter a name to describe this Digital Gateway.

y MAC Address: Enter the MAC address of the Digital Gateway. You

may find the MAC address from the device label.

y Trunk Number: Enter the available trunk numbers you may use or

purchase.

y CO Line per Trunk: Input the available channels of each trunk. For

example, input 32 for E1 or 24 for T1.

y CO Line Extension Base: Input the starting number of CO Line

extension number. The CO Line extension numbers will be assigned

automatically. For example if you set it as 4000 for an 8-port FXO

gateway, the CO Line extension number of this gateway will be

4000-4007.

y Jitter Buffer Depth: From “Jitter Buffer Depth” drop down menu,

depending on your bandwidth and CODEC, start with a Jitter Buffer

with the minimum value. Jitter Buffer will dynamically adjust its value

according to the party you are talking to and the bandwidth of the call

but never below the value you have set. The higher value of Jitter

Buffer will reduce the chance of packet loss during a call but might

cause delays and the lower value will help to transmit the voice

packets faster but might cause packet loss problems.

y Off-Premises and IP address: For Off-Premises Gateways, you will

need to select the check box and enter its “IP Address”. The IP Address

assigned to this Off-Premises Gateways needs to be a REAL (PUBLIC)

IP address.

68 Chapter 4 Gateway Configuration

Note: If later on your needs grow and you need to add more trunks or channels

into an existing gateway, you will need to delete the existing gateway from the

list and create a new one with new trunk or channel numbers.

Modifying a Digital Gateway

To modify, select a gateway from the list and click the Modify button. You may

change the name, MAC address or Jitter Buffer Depth. The software version of

the gateway will also be displayed here.

Setting Trunk Parameters

You can set the parameters of each trunk to adjust the voice quality and other

items.

1. Go to Main Menu>Gateway Configuration>Digital Line Gateways.

2. Select a gateway you would like to edit from the list and click the Trunk

button to edit the trunk parameters.

y Trunk: Select a trunk you would like to edit from the list.

Chapter 4 Gateway Configuration 69

y CO Line Index Bass: Input the starting number of the CO line display

number. For example if you set it as 1, when there is an incoming call

from port 1, the IP phones will display “Incoming Call CO 1” instead

of the extension number of the port, helping you to identify the CO

lines being used.

y Transmit Gain: This field edits the TX Gain of the trunk, or how loud

the external party hears your voice. The value here should conform to

the one on Digital Gateway. The default value is 5.

y Receive Gain: This field edits the RX Gain of the trunk, or the volume

of incoming calls. The value here should conform to the one on Digital

Gateway. The default value is 5.

y DTMF Volume: This field edits the intensity of the DTMF tones in the

trunk. The value here should conform to the one on Digital Gateway.

The default value is 5.

y Tie Line: Select Tie Line check box if you would like to establish a tie to

connect the Digital Gateway to other PBX using this trunk.

Setting Trunk Port Parameters

You can edit the ring assignment and other settings for individual ports or

channels of a trunk.

1. Go to Main Menu>Gateway Configuration>Digital Line Gateways.

2. Select a gateway from the list and click the Port Setting button.

3. Select a port from the port list and click the Port Parameters button to edit.

y Extension Number: The extension number will be displayed

automatically based on your settings under “CO Line Extension Base”.

You may also change the extension number here.

70 Chapter 4 Gateway Configuration

y Description: Input the telephone number or other information for this

port.

y Enable: For maintenance purpose, you may need to temporarily

disable some channels. Click the “Enable” check box to enable the

channel or uncheck to disable it.

y CODEC: Choose the preferred CODEC from PCM, G.723.1 or G.729

supported by InterPBX Communication System. PCM CODEC

compresses and decompresses voice conversation to 64 Kbps providing

the best speech quality but consumes larger bandwidths. G.723.1

CODEC uses 6.3 Kbps, the less bandwidth from the three choices but

may result in poorer call quality and finally G.729, which uses 8 Kbps.

y DID (Direct Inward Dialing): When selected, incoming calls will be

connected to extensions directly in stead of been picked up by Auto

Attendant. This function is available when you have subscribed the

DID service provided by your carrier. When you would like to set a

direct line between two PBXs connected by Tie Line, you may also

enable this function.

y Strip Prefix Digit: When the length of the number sent by DID is

longer than your extension numbers, set the digit that needs to be

removed. For example, if the DID prefix and adding number is

660x-xxxx, but you have 4-digit extension number, set “1” to delete the

first digit sent by your carrier. The default value is 0.

y Fax Ext. Number: The InterPBX System can detect the fax signal of CO

Line. Please enter the extension number of your fax machine here.

When the InterPBX System detects a fax signal, the fax will be

transferred to the fax machine.

y Ring Assignment: Ring Assignment allows you to assign a specific

extension, extension group or AA Menu to answer incoming calls from

a specific CO line or CO Line Group. In the Ring Assignment field,

choose the option of “Local” or “From CO Group”.

- Local: If you select “Local”, input a specific extension number,

extension group or AA Menu access code to answer incoming calls for

this port in Business Hours, Break Hours, After Hours and Closed. The

most common setting is to assign an AA Menu to answer phone calls.

If you would like to play special greetings for holidays, select the

“Enable Holiday” checkbox and the system will play the assigned

greeting from Holiday List. For more details about AA Menu, Voice

Mail and Holiday settings, please refer to InterPBX Administrator

Chapter 4 Gateway Configuration 71

Installation and Configuration Guide.

- From CO Group: If you select “From CO Group”, incoming calls will

be answered according to the Ring Assignment of the CO Line Group

you have assigned. For more details about CO Group’s Ring

Assignment, please refer to Administrator Installation and

Configuration Guide.

y Silence Compression Enable: Silence Compression reduces network

traffic whenever silence is detected during a conversation for a specific

amount of time. Silence indicator packets will be sent out instead of full

voice packets to reduce traffic.

y Caller ID Enabled: InterPBX’s CO Line ports can detect both DTMF

and FSK Caller ID. You may select or deselect it per your preference.

Please note that before using this function, the Caller ID service must

be provided by your local telephone company.

y Pickup After Rings: Please enter the Pickup After Rings times for the

CO Line port. If you set the ring times as 2, the system will pick up a

call after it rings twice and then transfer the call as settings.

y Flash Timing: You may set the Flash Timing of the SLT ports to check

whether the Hook action (quick press and release of the phone’s hook)

on the analog phones works fine. Analog extensions require the Hook

action to transfer, answer call waiting, or create a conference call. Make

sure the phone set’s Flash Timing must be within the system’s Flash

Timing.

Apply Settings

After finalizing the settings of this port, if you want the other ports to be

identical as this one, you may apply the settings to the others, saving you the

hassles of configuring each port individually.

1. Click on “Apply Setting to” and a new window will appear displaying the

parameters you have entered.

2. Apply or change the settings as you need.

3. Enter in the field “Apply to Port” the first and the last port you would like

to be identical to the present settings. If the range selected includes this port,

72 Chapter 4 Gateway Configuration

the new settings, if any, will also be applied to this one.

After you complete the installation of a new digital gateway, you need to set up

the associated Route and ARS. For more details, please refer to System

Configuration>Route and Group Management>Class of Service>ARS.

CTI Gateway

DSG BlazeLink is a CTI solution software enables business to develop

applications to optimize workforce in a call center or contact center.

Applications developed on BlazeLink for call control, monitoring, or managing

are able to connecting DSG IP-PBX System to the data processing environment.

1. Go to Main Menu>Gateway Configuration> CTI Gateway.

2. In the field ”IP Address”, please enter the IP address of your CTI gateway.

3. Agent License: It displays the Agent License numbers you purchased. If

you purchase 100 Agent Licenses, the system will allow the 100 agents to

Chapter 4 Gateway Configuration 73

currently set at BlazeLink.

NAT Proxy

With DSG NAT Proxy, Blaze or Savanna IP PBX Communication Systems and

Off-Premises Phones can traverse more than 95% of NAT or firewall

successfully without creating other service ports, and therefore enterprises can

communicate via IP-PBX Communication System with pure and stable voice

quality.

1. Go to Main Menu>Gateway Configuration> NAT Proxy.

2. On the field “IP Address of Master Proxy”, please enter the IP address of

the main NAT Proxy Server.

3. If you have the other NAT Proxy server, please also enter its IP address on

the field “IP Address of Slave Proxy”.

Note: When the NAT Proxy Server is activated successfully, a symbol

will be

shown beside the “IP Address of Master Proxy” field to represent that the NAT

Proxy Server is working.

74 Chapter 4 Gateway Configuration

For more information about the NAT Proxy Server’s settings and operation,

please refer to the NAT Proxy Administration Guide.

SIP Proxy

SIP (Session Initiation Protocol) is a new generation multimedia communication

protocol. It enables voice and other media types such as video to be transmitted

over the IP networks. With DSG SIP Proxy, your Blaze or Savanna series IP-PBX

will be able to adopt the SIP Trunk services provided by ITSPs or deploy SIP

phones as your extensions.

Add SIP Proxy

You need to enroll the SIP Proxy IP address to the PBX Server.

1. Launch the web browser. On the address bar, enter your PBX Server’s IP

address. (Please use service port 88.)

2. Click on the Administrator icon and use your user name and password to

3. Go to Main Menu>Gateway Configuration>SIP Proxy, the ”SIP Proxy” list

will be displayed.

Chapter 4 Gateway Configuration 75

4. Click the Add button to create a SIP Proxy.

y Name: Enter a name to describe this SIP Proxy. y MAC Address: Enter the MAC address of the SIP Proxy. You may find

the MAC address from SIP Proxy’s administration web site or from the

device.

y Port Number: Enter the available SIP channels you may use or

purchased.

y Extension Base: Input the starting number of SIP Trunks. The number

shall be a unique one from others on the system such as extension

numbers, group numbers or pickup codes. Notice that the SIP Trunk

numbers will not be displayed on the IP phone when accessed. Instead,

the associated SIP account number will be represented.

y Off-Premises: When SIP Proxy is not located in the LAN where PBX

Server is located and is deployed on the Internet, select the

“Off-Premises” check box and input its real IP address assigned to SIP

Proxy.

Add SIP Trunks

If you have subscribed the SIP Trunks service provided by your carrier, you can

input your account information at this section. Before you start, please prepare

the followings.

y Account and Password y Registrar Server IP address y Domain Name (In some cases, it is the same as the registrar Server) y Specific service port used by your carrier

1. Go to Main Menu>Gateway Configuration>SIP Proxy.

76 Chapter 4 Gateway Configuration

2. Select a SIP Proxy from the list and click the Account button.

3. Click the Add button to create a SIP Account.

y Account Name: Assign a name for this SIP Trunk. When accessed, the

name will be displayed on IP phone.

y Phone Number: The account provided by your carrier. y Registrar Server IP: The IP address of your carrier’s Registrar Server. y Port: The service port provided by your carrier. The default setting is

5060.

y Domain Name: The domain name provided by your carrier. In some

cases, it is the same as the Registrar Server.

y User ID: The account provided by your carrier. In some cases, it is the

same as Phone Number.

y Password: The password provided by your carrier. y Registrar Server Type: Depending on your carrier’s equipment, select

a type of Registrar Server from the list.

y Register Expire Time: Please input the expiring time of registration.

Once registration succeeds, the system will follow the required time of

your carrier to reset this item.

y CODEC: Select a CODEC supported by the SIP Trunk. y Enabled: Click this item to enable the SIP Proxy. y RTP Relay: When the network is behind NAT, select this item to

enable RTP Relay allowing voice packets to traverse NAT and firewall

automatically.

y DID and Strip Prefix Digit: If you have applied DID service provided

by your carrier, select the DID check box and input the digit of prefix

that needs to be deleted.

Chapter 4 Gateway Configuration 77

y Fax Ext Number: If you carrier supports SIP fax, you may input the

extension number connected to your fax machine. When a fax comes

from SIP Trunks, it will be transferred to the fax machine.

y Ring Assignment: You could assign a specific extension, extension

group or AA menu to answer incoming calls. In ring assignment, select

“Local” to set individual trunk line or select “From CO Group” to

allow this trunk referring to a CO Line Group. For more details about

ring assignment, please refer to InterPBX Administrator Installation

and Configuration Guide.

Create SIP Trunk ARS

After you add SIP Trunks to your system, you need to edit the associated ARS

(Automatic Route Selection) policies and procedures allowing extensions to

access those SIP Trunks for making calls. Notice that extensions are not able to

access SIP Trunk using the assigned extension number.

Creating Routes:

1. Go to Main Menu>System Configuration>Route.

2. Click the Add button to increase a route.

3. Input the description of this route.

4. In different business hours, click “Account” and select a SIP Proxy you

would like to use from the list.

78 Chapter 4 Gateway Configuration

Setting ARS:

1. Go to Main Menu>Group Management>Class of Service.

2. Select a Class of Service from the list and click the ARS button to apply the

route you just created to the selected CoS.

3. In “Dialed String”, define your policy and procedure. You may input the

commonly used international access code and country code, such as

“01144” for calls to UK, or long distance access code and area code, such as

“1212” for calls to New York. You may also set a specific number

representing the SIP Trunk code such as “7”. The digit length shall not

exceed 28 and the string shall be a unique one.

4. Set the Minimum and Maximum Digits length of dialed numbers.

5. From the “Route” list, select a SIP Trunk route you would like to adopt.

6. When editing SIP Trunk ARS, skip the ”Delete” and ”Insert” items.

7. Click the Add button to increase this policy and procedure to ARS list. To

remove, select one from the list and click the Remove button.

8. You may also click the “Copy from” button and increase one by modifying

others.

Making Calls via SIP Trunks

1. Lift the handset. Please make sure your Class of Service is allowed to access

SIP Trunk ARS.

2. Depending on the ARS policies and procedures you created, input the SIP

Trunk access code followed by the phone number, or simply input the

phone number. Press the # key to start dialing.

Chapter 4 Gateway Configuration 79

For example, if you set “7” in “Dialed String” as the SIP Trunk access code,

when you dial 7-44-2055551234#, the call to UK will get through your SIP

Trunk. If you set “01186” in “Dialed String”, when you dial

011-86-xxxxxxxxxxx#, all calls to China will get through the SIP Trunks.

Add SIP Extensions

SIP Proxy helps you to register third parties’ SIP phones to your InterPBX

System and they will be one of your extensions.

1. Go to Main Menu>Extension Management >SIP Phone List.

2. Click the Add button to increase SIP phones.

3. Assign an extension number in “Extension No.”.

4. Assign a password for this extension. Extension users can change the

password from InterClient. Password must be numbers.

5. From “SIP Proxy” list, select one for this SIP phone.

For more information about the SIP Proxy Server’s settings and operation,

please refer to the SIP Proxy Server Administration Guide.

80 Chapter 5 Extension Management 80 Chapter 5 Extension Management

Chapter 5 Extension Management

This chapter describes the installation and configuration of the various types of

extensions supported by InterPBX Communication System. These include IP

phones, soft-phones, virtual extensions, analog phones, and off-premises

extensions. You may at your own evaluation provide the appropriate type of

extension according to the user’s needs, whether they are at the head office,

branch offices or business travelers.

Chapter 5 Extension Management 81

Configuring IP Phone, SoftPhone and SoftConsole

You may create extensions via Auto Discovery or manually. For more

information about settings, please refer to Chapter 2 Installing InterPBX

Communication System.

Go to Main Menu>Extension Management>InterPhone List. Select one of the

extensions from the list and then click the Modify button to edit.

Extension No., Phone Type and MAC Address

If the extension is registered to the PBX server using Auto Discovery, these info

will be recorded automatically. You do not need to change them.

If you need to increase extensions manually, please set these info. For software

phone and console, the MAC address is the one of user’s PC.

User Information

Fill in the information about extension users, including First Name, Last Name,

Title, Location and Department. The user’s name will be displayed on the IP

phone’s LCD and the extension list. The information about extension users will

be applied to Caller ID of internal calls.

User Password

With User Password, administrators are allowed to modify the password for

logging in InterPBX Management Website. With Auto Discovery, the extension

password is blank. In case password is forgotten, the system allows setting

password back to default through Password Management. This is located at

Main Menu>System Configuration>Password Management.

82 Chapter 5 Extension Management

Off-Hook Access to

InterPBX provides extensions different Off-Hook selections that enable users

directly off-hook access to extension, CO line, operator, or hotline. Please select

a direct access port for your extension from the “Off-Hook Access to” list.

y Intercom: When you lift the headset, you can dial to an extension directly. To

dial the CO line number, you have to press the CO line access code first.

y CO Line: When you lift the headset, it will retrieve the CO line access code

automatically. To dial the extension number, you have to press the Intercom

button first.

y Operator: When you lift the headset, it will directly connect to the operator.

For the settings of operator, please refer to Chapter 6 Group Management.

y Hotline: When you lift the headset, it will automatically connect to a specific

extension or extension group. Please enter the hotline number in the field of

Hotline.

Class of Service

Assign one Class of Service to restrict extensions to make unnecessary calls.

With Auto Discovery, the settings in Class of Service will be applied to the

extension automatically.

Calling permission on Class of Service will be performed with top priority.

Please refer to Chapter 6 Group Management/Class of Service for more details

about call permission.

Button Mapping Group

Select a “Button Mapping Group” which supports programmable keys on IP

phone as feature access buttons to enable speed dialing, call park and to access

CO lines. You can set Button Mapping Group as a template to be adopted by

the IP phones.

For more information about Button Mapping, please refer to Chapter 6 Group

Management/IP Phone/Button Mapping and Chapter 5 Extension

Management/Button Mapping Groups for more details about Button Mapping

Extension.

CODEC

Chapter 5 Extension Management 83

From the CODEC list, select PCM format G.711 that compresses a call to 64

Kbps, providing higher call quality but using higher bandwidths. Other

alternatives include G.723.1 compressing to 6.3 Kbps consuming smaller

bandwidth, but with poorer voice quality and G.729 compressing to 8 Kbps.

Jitter Buffer Depth

From “Jitter Buffer Depth”, select “Low” based on the bandwidth and CODEC.

Jitter Buffer will be adjusted to accommodate the callers and bandwidth, but the

Depth will never be lower than the “Low” standard. The Depth of Jitter Buffer

can be specified as None, Low, Middle and High. The None buffer corresponds

to 0 packet, Low to 2 packets, Middle to 4 packets and High to 6 packets.

Higher Jitter Buffer Depth has more storage capacity when storing temporarily

the packets avoiding packets to be discarded, but additional delay may be

bothersome at times. On the contrary, smaller Depth offers faster transmission

but excessive number of packets might be lost during transmission. For

example, G.711 format transmits packets every 20 ms which probably leads to

40ms additional delay if you set the Jitter Buffer as Low.

Silence Suppression

Silence Suppression reduces network traffic whenever silence is detected

during a conversation for a specific amount of time. Furthermore, silence

indicator packets will be sent out instead of full voice packets to reduce traffic.

Select “Silence Suppression” to enable this function or uncheck to disable it.

Conference Disabled

Conference allows the extension making an outgoing call to initiate a

conference between three or more parties. Select the “Conference Disabled”

check box if the user does not need this function.

To initiate a conference call

1. During a call, press the Hold button to hold a party.

2. Call the second party where this can be an internal extension or an external

line. More parties can join this conference by pressing the Hold button to

hold the conversation and dialing to the next party. You can hold up to 4

parties.

3. After the call is connected, press the Transfer button and “#40” to start a

Conference.

84 Chapter 5 Extension Management

4. To invite more parties, hold this conference and dial to the new party.

When connected, press the Transfer button and “#40” to start.

Each conference allows for up to 18 parties in different sessions. But each

session requires one party to be a direct extension under the InterPBX.

Note: Off-premises extensions are not allowed to initiate a conference but can

be invited.

Paging Disabled

You may select “Paging Disabled” when this is not needed.

How to activate External Paging

1. Pick up the handset.

2. Dial the Paging Code assigned on the gateway where the Paging

equipment is connected.

3. Broadcast through the handset or the speakerphone on the phone.

4. After broadcasting, hang up the phone.

How to activate Internal Paging

1. Pick up the handset.

2. Dial #38.

3. Dial the pre-defined extension number or Extension Group Code for

internal paging.

4. Broadcast through the handset or the speakerphone on the phone.

5. After broadcasting, hang up the phone.

Enable/Disable Extension

For maintenance purpose, you may need to temporarily disable some

extensions. Click the “Enable” check box to enable the extension or uncheck to

disable it.

Apply Settings

After configuring one extension, apply the settings to other extensions to save

time on initial setups. Press “Apply Setting to” and a window will pop up

showing all your pre-defined settings. Apply or change the settings, as you

need. Enter the range of extensions in “Apply to Extension”. If the range

Chapter 5 Extension Management 85

selected includes the extension you are using, the new settings will also be

applied to this extension.

Button Mapping

Under Button Mapping, set programmable keys on IP phones as feature access

buttons to enable Speed Dialing, Call Park, Access CO Lines or Conference.

Depends on phone types, the available number of Programmable Buttons are

different.

1. Go to Main Menu>Extension Management>InterPhone List.

2. Select one of the extensions from the list and press Button Mapping to edit

the programmable keys. Please refer to Action List below, which lists

details about the action, numbers and functions required.

3. Select one programmable button and scroll down to the preferred Action.

4. After the action is selected, enter the correspondent number in the

“Number” field. For example, if you select “System Speed Dial” from the

“Action” drop down menu and enter a speed dial number. When you press

this programmable key, the system will dial the phone number assigned on

86 Chapter 5 Extension Management

this system speed dial number. If the “Number” field is left blank, press the

programmable key and followed by any speed dial number you desire.

Action Designated Number Description

Access Ext. An extension number

of a station, extension

group, CO line, CO

line group, or an

outside phone number

To make calls to the

assigned number. Please

note to add CO line access

code, e.g. 05551234 for an

outbound call. Notice that if

Trunk Group is set on a

button, when parts of

trunks are available, the

LED displays available

status for other people.

After Call Work - To allow agents to have a

period of after call work

time between two calls.

Ask Member Login An Extension Group

Number

Check the login status of

the extension group.

Hearing a dial tone means

members are logged-in

successfully and a busy

tone means not logged-in.

Auto Line Access - To get a CO Line

Auto-In - Agents’ extensions will pick

up the next call

automatically.

Auxiliary Time - Agents can enable this

function to notify the

system stop assigning

incoming calls to their

extensions so that the

agents can leave their seats

temporarily without

logging out the system.

Chapter 5 Extension Management 87

Action Designated Number Description

Call Appearance

Press to make or receive

calls. It functions similar to

intercom. When more than

one button is assigned,

buttons with lowest

number will take the first

call.

Call Hold Retrieve CO An extension number

of a CO line

To retrieve an incoming call

from a specific CO line

placed on hold. (You

should be able to see the

CO line extension number.

for using this function.) If

this CO line is set on the

programmable button,

users can directly press the

flashing button to retrieve

the incoming calls

Call Hold Retrieve Ext An extension number

of a station

To retrieve an internal call

placed on hold. If this

extension number is set on

the programmable button,

users can directly press the

flashing button to retrieve

the call.

Call Park A slot number (0-9) To park a call to a specific

slot.

Call Pickup CO Line - To answer the least recent

incoming call ringing on

the system.

Call Pickup Directed An extension number To answer a call ringing at

another extension.

Conference Call - Start Conference with the

callers placed on Hold.

88 Chapter 5 Extension Management

Action Designated Number Description

Forward All Calls An extension number

+ #

To forward all the incomin

calls to a specific extension

or an external phone

number automatically.

Press again to disable.

When setting external

number, please add the CO

line access code such as 0.

Headset - Press to allow voice been

transmitted from the

attached headset, instead of

the handset.

Internal Paging An extension number

or extension group

number

Broadcast for members

through the extension or

the extension group.

Manual-In - To allow agents to pick up

the next incoming call

manually by pressing the

specific button of

Manual-In function.

Member Login An Extension Group

Number

members in the Extension

Group.

Member Logoff An Extension Group

Number

Logout from the Extension

Group.

Personal Speed Dial

A personal speed dial

number(e.g. 00)

To dial a number defined

on the Personal Speed Dial

Number.

Record on Demand - Press to save the recording

or start recording. When

recording, the LED will be

flashing.

Retrieve Msg - To access mailbox.

Retrieve Record - To play the saved

recordings.

System Speed Dial A system speed dial

number(e.g. 000)

To dial a number defined

on the speed dial number.

Chapter 5 Extension Management 89

Action Designated Number Description

Transfer to Ext VM An extension number To transfer a call to the

extension’s voice mailbox.

Transfer to AA Tree An AA menu access

code

To transfer a call to the AA

menu.

Virtual Extension A virtual extension no. Press to act as the assigned

virtual extension to make

calls.

Note: A symbol “Lock” will show on the screen when the programmable keys

are assigned as Group’s Button Mapping and only the unlocked ones can be

used. Please refer to Chapter 6 Extension Management/Button Mapping for

more details about Button Mapping and Appendix A for Function Codes List.

Station Speed Dialing

InterPBX Communication System provides System Speed Dialing and Station

Speed Dialing functions. You can assign frequently dialed phone numbers at

“Station Speed Dialing” for individual use. For more details about System

Speed Dialing, please refer to Chapter 3 System Configuration/System Speed

Dialing.

1. Go to Main Menu>Extension Management>InterPhone List.

2. Select one of the extensions from the list and press Speed Dialing button.

3. From the "Station Speed Dialing" list, select one entry.

4. Enter the targeted phone number or extension number in the "New

Number" box.

5. Give details about New Number in the “Comment” box.

6. Click the Assign button to save.

90 Chapter 5 Extension Management

You can create up to 100 entries. If you would like to set an outside phone

number, please remember to add the CO line access code such as “0” in front of

the phone number (e.g. 05551234). For long distance calls, please add the long

distance code (e.g. 02125551234). For international calls, please add the

international code and country code (e.g. 001188628861558).

Making Calls using Personal Speed Dialing

1. Lift the handset or press the Speakerphone button.

2. Press #21.

3. Press the specific personal speed dialing number, such as “00”.

Answer Option

You can better manage the incoming calls by setting the forward targets for all

calls or when not available.

1. Go to Main Menu>Extension Management>InterPhone List, select an

extension from the list and click the Answer Option button to select Call

Forward.

2. Select All Calls, Busy or Ring-No-Answer as per your preference.

3. Select one of the targets to answer incoming calls.

y Extension: When selected, calls will be forwarded to the other

extension or an external phone number. Enter the target extension

number or phone number in the “Number” box.

Note: When setting an external phone number, remember to input

your CO line access code.

y Voice Mail: When selected, incoming calls will be sent to your voice

mailbox. Note: Notice that your administrator needs to set a DTMF

button in the Transfer Option menu allowing callers to leave messages.

Chapter 5 Extension Management 91

y Auto Attendant: When selected, incoming calls will be forwarded to

the voice mail system, but callers cannot leave messages.

4. Ring-No-Answer Timeout: When ring-no-answer, the incoming calls will

be redirected to the Ring-No-Answer forward target after the defined

timeout. The unit used here is in seconds.

5. Call Waiting: You may enable the “Call waiting” function to pick up the

second incoming call during a call. After Call Waiting is enabled, if your

line is busy, the second incoming caller will hear the normal ringing tone,

and you will hear the “beep” tone indicating that you have a call waiting.

The “Call Waiting” function can allow only one more incoming call to stay

on queue.

Note: If the forward destination is busy or no answer, the forwarded calls will

be redirected to the original destination’s busy or no answer setting.

If Personal Answering Option is not configured, the calls will be directed to

Auto Attendant when the requested extension is busy or not available. The

administrator needs to configure the Transfer Option of VMS for taking calls.

Please refer to Chapter 7 Voice Mail Configuration/Transfer Options for more

details.

Enable Call Forward from Phone: Users may also press #44 + forward

number +# from IP phone directly to enable Forward All Call function.

External Call Forward: To allow the External Call forward functioning well,

both internal callers and receivers’ COS need to support external calls. To allow

external calls been forwarded to external numbers, the administrator needs to

proper set the Trunk to Trunk Class of Service. For more details, please go to

Chapter 6 Group Management/Class of Service.

Mailbox

In Mailbox, users can select language options, Message Play Priority and

Message Forward.

92 Chapter 5 Extension Management

1. Go to Main Menu>Extension Management>InterPhone List, select an

extension from the list and click the Mailbox button to edit the mailbox

preferences.

2. Select a language from the “Language” list and the system prompts will be

announced in the language you have selected. InterPBX can support up to

4 languages, please consult your distributor regarding multiple language

options.

3. Set Message Play Priority which allows users to select “Last-In-First-Out”

for playback of the latest message first or “First-In-First-Out” to play the

oldest one first.

4. Enter the extension number in “Message Forward to” to allow the system

to forward messages to pre-defined mailboxes.

5. Play Time Stamp: If “Play Time Stamp” is enabled, the system will play the

message recorded time when you play the message.

6. Call Screen: This function allows you to filter callers before answering calls.

When enabled, the voice mail system will require callers to say the name,

hold the call, and then make calls to the receiver and announce the name of

the caller. Receivers will be offered options to answer or not to answer the

call. If the receiver chooses to answer, the call will be connected. If not, the

voice mail system will reply the caller that the receiver is not available.

7. External Forward: When you are not available, incoming calls will be

bounced back to the VMS. Depending on the Transfer Options settings,

callers can connect to the operator, voice mail or a desired external phone

number if this option is enabled by the Administrator. Set your external

forward number here allowing the VMS to forward your incoming calls.

You may also set the external forward number from your IP phone.

How to retrieve messages

Chapter 5 Extension Management 93

1. From a phone set, press the “Message” button or “##” if messages are

retrieved via an analog phone.

2. Enter the password of the voice mailbox followed by the “#” sign (on its

first initial use, no password is required as the default password is null).

3. To retrieve messages or modify personal options simply follow the voice

guidance prompts.

Notification

InterPBX provides options for message notification. Go to Main

Menu>Extension Management>InterPhone List, select an extension from the

list then click the Notification button.

Internal Notification

The Internal Notification is Ring Notification. When the extension receives new

messages, the system will notify the receiver by ringing the extension. This

function can be used on different kinds of extensions such as assistant

extensions, analog phones, and virtual extensions.

1. Internal Notification Method: Select “Extension” from the drop-down

menu to enable the internal notification.

2. Setting the Notification Target: If you enable the internal notification, input

the desired extension number. The system will ring the extension number

you assigned when receiving new messages. If you want to enable your

station’s notification function, please input your extension number or

simply leave it blank.

94 Chapter 5 Extension Management

External Notification

1. Click the “Enabled External Notification” check box to enable the External

Notification function, or click the “Urgent Messages Only” check box to

notify you only when receiving urgent messages. The notification of urgent

messages will be delivered even when you disable the external notification.

2. Set the “Notification Schedule” to deliver notification during business

hours or specified hours to avoid interruptions.

3. Set the notification sequence. Set the destination type and its associated

phone number or pager ID. You don’t need to add the CO line access code

such as “0” here.

4. Set the interval and trial times. The system will repeat the notification, if

failed, after the interval period.

The system will send notification to the destination’s phone/pager numbers

from 1 to 5. If the number in sequence 1 is busy or unavailable, the system will

retry after the interval time you set. If failed, the system will start from

sequence 2 and repeat until it reaches the “Try Times.”

Note: If the Answer Options of the destination’s phone or extension is set, the

notification will also follow the settings.

Note: Sending a notification via phone or pager is limited by your call

permission in the Class of Service. Please consult your administrator for your

Class of Service.

Unified Messaging

You may enable the unified messaging function allowing the system to forward

the voice messages to your e-mail. The new voice messages will be as saved as a

WAV file and sent to your e-mail address.

1. Select “Enable UMS”

2. Select “Keep as new” or “Save as old” after messages are forwarded to

e-mail.

3. Input the e-mail address in “E-mail”. You can input up to 3 different e-mail

addresses.

When the “Unified Messaging” is enabled, if you select “Save as old”, the

internal and external notifications won’t be delivered due to all new messages

will be switched to old messages.

Chapter 5 Extension Management 95

Note: If you are on the recording list, when Unified Messaging is enabled, you

will be prompted for asking if you want to send recordings to e-mail when

finish listening recordings.

Note: Set the SMTP Server when Unified Messaging is enabled. For more

details about SMTP, please refer to Chapter 3 System Configuration/General

Parameters/SMTP Server.

Distribution List

Users can record or forward voice messages to a personal voice mailbox or

members in Distribution List. Users can create the Distribution Lists based on

his preferences:

1. Go to Main Menu>Extension Management>InterPhone List.

2. Select an extension from the list and then click the Distribution List button.

3. Select a List ID from the list.

4. Select extensions from the Non-Member List and move to Members List.

InterPBX allows editing up to 9 groups for each extension in Distribution List

and each can hold up to 15 members.

Virtual Extensions

You can associate Virtual Extensions with regular extensions offering one

extension with multiple numbers to be used for different purpose. The regular

extensions to be applied need to have programmable buttons.

1. Creating Virtual Extension Number. Go to Main Menu>Extension

Management>InterPhone List. Click the Add button and assign the

96 Chapter 5 Extension Management

extension number and password. From the Phone Type drop down menu,

select “Virtual”. You do not need to set the MAC address. Refer to the

above Chapter 3 Extension Management/InterPhone to set other items.

2. Associating with Extensions. Go to Main Menu>Extension

Management>InterPhone List. Select one extension from the list and click

the Virtual Ext button. You will see available virtual extension number list.

Select one or more to be associated with the regular extension to the

Allowed Virtual Ext list. The amount of Virtual Extensions you could

assign should not exceed the number of programmable buttons on the

phone.

3. Button Mapping. Go to Main Menu>Extension Management>InterPhone

List. Select the extension you just edit and click the Button Mapping button.

Pick one programmable button and set its action as “Virtual Ext” and input

the correspondent virtual extension number. To access the Virtual

Extension, simply press the specific programmable button.

Chapter 5 Extension Management 97

Making Calls: To place calls, simply press the specific button and dial the

number.

Answering Calls: When the Virtual Extension receives calls, your phone will

ring and the specific programmable button will be flashing. You may simply

pick up the handset to answer calls.

Retrieving Voice Messages

You could listen to voice messages for Virtual Extensions in many ways.

y Set Virtual Ext message forward to a regular extension. When receiving new

message, the regular extension’s message lamp will be lit and a running

message will appear on the LCD display.

y Set Virtual Ext message UMS to an e-mail. You will be able to listen to the

new messages from e-mail.

y Accessing Voice Mail System. You can also access voice mail system to listen

to messages.

- Press the Message button or press ##, then press * to go back to the main

menu of VMS. Follow the prompt to enter the extension number and

password to listen to voice messages.

- Or you could access a specific AA Menu using the AA menu access code or

make calls to the company phone number. Select a pre-defined DTMF key to

access voice mailbox. Follow the prompt to enter the extension number to

retrieve messages. (The administrator need to assign a key functioning as

“Access Mailboxes”. Please refer to Chapter 7 Voice Mail

Configuration/Auto Attendant for more details.)

Analog Extensions

The analog extensions are connected to the analog ports of the gateway like

VG5000 Voice Gateway. When you install an Analog Gateway, if it provides

FXS interface, you have already created the Analog Extensions. For more details

about installing Analog Gateway, see Chapter 4 Analog Gateway/Creating

Analog Gateway List.

1. Go to Main Menu>Extension Management>Analog Port List.

2. Select a SLT port from the list and click Modify button.

3. You will find “Gateway Name”, “Gateway Port Number” and “MAC

Address” have already been entered when administrators configured the

gateway.

98 Chapter 5 Extension Management

4. Input the extension number of the SLT port in “Extension No.”.

5. Fill in information including Name, Title, Location, Department, Change

Password, Class of Service, CODEC, Silence Compression, Conference,

Paging, Off-Hook Access to CO Directly, Apply Setting to, Station Speed

Dialing, Answer Option, Mailbox, Notification and Distribution List. For

more details, please refer to Chapter 5 Extension Management/InterPhone.

Off-Premises Extensions

InterPBX is also applied to branches or the staff who work at home. Branches

can reach Headquarters via Internet, VPN or Ethernet instead of leased lines as

Off-Premises Extensions have the same functions as any other extension in the

Headquarters.

Make sure the remote site has a permanent Internet connection using xDSL,

Cable Modem or Leased Line. Ideally it is recommended that each of the

Off-Premises phones can be assigned with a Real IP address but this is often not

the case. If you are not able to assign a Real IP address to each of the

Off-Premises phone, you can assign it a virtual IP address and open specific

ports on the Off-Premises NAT device. Before you create extensions, make sure

the relevant License Keys has been added.

Chapter 5 Extension Management 99

1. Go to Main Menu>Extension Management>Off-Premises List. Click the

Add button to create a new Off-Premises extension.

2. Assign an “Extension No.” to this phone.

3. Create a password for this extension.

4. Input the “MAC Address” of this extension. Every IP phone or PC running

the SoftPhone have a unique MAC address to be entered into this field.

5. From the “Phone Type” drop down menu select a phone type.

6. Do not change the default Signal Port 6045 and Media Port 5000 provided

by the system. If more than one Off-Premises IP phone is used in a single

remote site, set the second phone’s Signal Port to 6046 and Media Port to

5001 and so on.

7. Select the “Static IP” will reinforce the system’s security and the system can

only be connected by the specified IP address. If a real IP is used on

off-premises phone, enter it in “IP Address”. If a virtual IP is used on

off-premises phone, enter the NAT equipment’s real IP of the remote site in

“IP Address”. If a fixed IP cannot be obtained (e.g. PPPoE) do not select the

“Static IP” option.

8. Make sure “Enabled” check box is selected

9. Activate QoS if applicable. All other settings you may leave to be set at a

later time. For other setting items, please refer to Chapter 3 Extension

Management/InterPhone.

Note: If the Off-Premises extension uses virtual IP, on remote NAT equipment,

you will have to open the Signal and Media Ports as well as setup their Port

Mapping.

100 Chapter 5 Extension Management

Note: If the PBX Server is also under NAT, please activate PBX Server’s NAT

setting then open the NAT equipment’s UDP port 6046. Please refer to Chapter

3 System Configuration/General Parameters/Behind NAT, NAT IP Address

for more detailed information. On the Off-Premises phone, input the NAT

device’s Real IP address in the “PBX Server IP” item.

Note: When the off-premises extension is registered to the NAT Proxy, input

the NAT Proxy’s IP address on phone’s “PBX Server IP” setting item.

Configuring Off-Premises Analog Phones

If the Off-Premises gateway you have installed supports FXS ports, you will see

under “Analog Port List” the Off-Premises SLT ports. Please refer to above

sections for information on how to install analog extensions. For Off-Premises

gateway information, please see Chapter 4 Gateway

Configuration/Off-Premises Gateways.

SIP Phone List

SIP Proxy helps you register third parties’ SIP phones to your InterPBX System

and they will be one of your extensions.

1. Go to Main Menu>Extension Management >SIP Phone List.

2. Click the Add button to increase SIP phones.

3. Assign an extension number in “Extension No.”.

4. Assign a password for this extension. Extension users can change the

password later. Password must be numbers.

5. From “SIP Proxy” list, select one applied for this SIP phone. For other

settings, please refer to Extension Management>InterPhone.

DSG InterPBX Blaze5000, InterPBX Blaze1200, InterPBX Savanna8000 Installation And Configuration Manual

Specifications and Main Features

Frequently Asked Questions

User Manual