This publication may not be reproduced, in whole or in part, without prior expressed written permission from MultiTech Systems, Inc. All rights reserved.
Multi-Tech Systems, Inc. makes no representations or warranties with respect to the contents hereof and
specifically disclaims any implied warranties of merchantability or fitness for any particular purpose. Furthermore,
Multi-Tech Systems, Inc. reserves the right to revise this publication and to make changes from time to time in the
content hereof without obligation of Multi-Tech Systems, Inc. to notify any person or organization of such
revisions or changes.
Record of Revisions
RevisionDescription
AManual released; covers software version 2.01. All pages at revision A.
(5/14/99)
Patents
This Product is covered by one or more of the following U.S. Patent Numbers:
5.355.653; 5.452.289; 5.453.986
. Other Patents Pending.
TRADEMARK
Trademark of Multi-Tech Systems, Inc. is the Multi-Tech logo.
Windows is a registered trademark of Microsoft.
Preview of this Guide ................................................................................................................................. 7
Installing Y our MultiVOIP.......................................................................................................................... 16
Phase 1: Configure and Install Your Master MultiVOIP ..................................................................... 16
Phase 2: Configure Your Slave MultiVOIP(s) .................................................................................... 16
Phase 3: Deploy the VOIP Network................................................................................................... 16
IP Statistics ........................................................................................................................................ 52
T elnet ................................................................................................................................................. 64
WEB Management............................................................................................................................. 65
Service ..................................................................................................................................................... 69
Tech Support ............................................................................................................................................ 70
Recording MultiVOIP Information ...................................................................................................... 70
The Multi-Tech BBS ................................................................................................................................. 71
To Log on to the Multi-Tech BBS........................................................................................................ 71
To Download a File ............................................................................................................................ 71
About the Internet..................................................................................................................................... 72
Appendixes
Appendix A - TCP/IP (Transmission Control Protocol/Internet Protocol) Description............................... 74
Appendix B - Cabling Diagrams ............................................................................................................... 77
Appendix C - Regulatory Information ....................................................................................................... 79
Class A Statement .............................................................................................................. ............... 79
FCC Part 68 Telecom......................................................................................................................... 79
Canadian Limitations Notice .............................................................................................................. 80
EMC, Safety and Terminal Directive Compliance .............................................................................. 81
Glossary
Index
iv
Page 5
Voice / Fax over IP Networks
Chapter 1 - Introduction and Description
Page 6
MultiVOIP 200 User Guide
Introduction
Welcome to Multi-Tech's new voice/fax gateway, the MultiVOIP, model MVP200. The MultiVOIP
allows analog voice and fax communication over a traditional data communications/data networking
digital Internet. Multi-Tech’s new voice/fax gateway technology allows voice and fax communication
to ride, with no additional expense, over your existing communications Internet, which has
traditionally been data-only . To access this free voice and fax communication, all you have to do is
connect the MultiVOIP to a phone or to your existing in-house phone switch, and then to your existing
Internet connection. Once configured, the MultiVOIP then allows voice and fax to travel down the
same path as your traditional data communications.
The MVP200 is designed with two voice/fax channels (which offer three voice/fax interfaces on each
channel), a 10 Mbps Ethernet LAN interface, and a command port for configuration. System
management is provided through the command port using bundled Windows® software which
provides easy-to-use configuration menus and a comprehensive Help system.
Figure 1-1. MultiVOIP
6
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Preview of this Guide
This guide describes the MultiVOIP and tells you how to install and configure the unit. The
information contained in each chapter is as follows:
Chapter 1 - Introduction and Description
Chapter 1 describes the MultiVOIP. Front panel indicators, and back panel connector descriptions are
provided. In addition, a list of relevant specifications is provided at the end of the chapter.
Chapter 2 - Installation
Chapter 2 provides information on unpacking and cabling your MultiVOIP. The installation procedure
describes each cable connection.
Chapter 3 - Software Loading and Configuration
Chapter 3 provides instructions for software loading and initial configuration. The MultiVOIP software
disks are Windows® based. Later chapters, as well as the on-line Help, describe the MultiVOIP
software in more detail.
Chapter 4 - MultiVOIP Software
Chapter 1 - Introduction and Description
Chapter 4 describes the MultiVOIP software package designed for the Windows ® environment. This
chapter describes the software from an applications standpoint, and in so doing, not every screen is
shown, nor is each field within a screen defined. For explanations and parameters of each field within
a dialog box, refer to the Help.
Chapter 5 - Remote Configuration and Management
Chapter 5 provides procedures for changing the configuration of a remote MultiVOIP. Remote
configuration enables you to change the configuration of a unit by simply connecting two modems
between the two MultiVOIPs and remotely controlling the unit. Chapter 5 also describes typical client
applications (i.e., Telnet and Web-based management) used for remote configuration of the
MultiVOIP.
Chapter 6 - Warranty, Service and Tech Support
Chapter 6 provides instructions on getting service for your MultiVOIP at the factory, a statement of
the limited warranty , information about our Internet presence and user bulletin board service, and
space for recording information about your MultiVOIP prior to calling Multi-Tech’s Technical Support.
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MultiVOIP 200 User Guide
Typical Application
Before Voice Over IP (VOIP), voice over the Internet, a corporate of fice had a data connection to the
Internet and a voice connection to the public telephone network. With VOIP, the two networks can be
tied together. To accomplish this, a MultiVOIP is connected between the public telephone network
and the data network. A typical application for a MultiVOIP is shown in Figure 1-2.
Corporate Office
512-4122
Workstation
LAN
HUB
Web Server
Analog Connections
Channel 1: FXO
Channel 2: FXO
102
Workstation
MultiVOIP
IP Address
204.22.122.118
Mask 255.255.255.128
512-4123
101
4124
PSTN Connection
(T1/E1, PRI, etc.)
Router
IP Address 204.22.122.1
Mask 255.255.255.128
Router
4125
P
B
X
ISP
Internet/Intranet
IP Network
PSTN
Remote Branch
Office
Workstation
ProxyServer Static
IP Address 209.96.211.90
Workstation
HUB
LAN
Proxy
Server
MultiVOIP
IP Address 202.54.39.100
Mask 255.255.255.240
ProxyServer
IP Address 202.54.39.97
Mask 255.255.255.240
#301
#302
Figure 1-2. T ypical VOIP Application
Now, to set up the VOIP network, a MultiVOIP at the corporate office is connected between the data
network and the corporate telephone switch (PBX). To connect the MultiVOIP to the data network, an
Ethernet cable is connected to the Ethernet port on the MultiVOIP and the other end of the cable is
plugged into a hub on the data network. On the phone side, two special adapter cables are
connected to the FXO jacks on the back of the MultiVOIP and run to two station lines on the phone
switch. These two lines on the PBX occupy phone extensions 4124 and 4125.
To set up a MultiVOIP at the remote branch office, the Ethernet jack on the MultiVOIP is connected to
the hub and two special adapter cables are connected between the FXS jacks on the MultiVOIP and
two analog phones.
To configure a MultiVOIP , the COM port of a PC is connected to the Command port on the MultiVOIP.
Configuration software is loaded on to your PC and your unique LAN parameters are established.
The configuration software is based on a standard Windows Graphical User Interface (GUI) which
simplifies your selection process to a single parameter group within a dialog box. For example, your
LAN IP parameters are contained on a single dialog box. You can configure the IP address and
mask for the MultiVOIP, and the gateway address for the corporate router, on the same dialog box.
Once the LAN parameters are established, you can set up the voice channel parameters.
The unique feature here is that both channels do not have to be configured the same way. One
channel could be connected to an extension line off the phone switch and the other channel
connected directly to your fax machine. In this situation, you would use the FXS interface which has
two options -- loop start and ground start. Loop start is generally the correct interface to use.
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Chapter 1 - Introduction and Description
If you want a user to be able to call into a facility and then use the local phone capability to dial out to
the public phone network, you can choose either the FXO or E&M interface. If you chose the FXO
interface, you can connect to the station side of your phone switch and be able to use all the features
of your local phone system. If you want to call out on the local phone network while you are at a
remote branch office, you would dial the MultiVOIP at the corporate facility and then dial an outside
line to access the local phone network.
If you use the E&M interface, you have more control over call management. The E&M connection
uses the trunk side of the phone switch; therefore, to access a trunk line, you would dial the trunk’s
access number instead of an extension number. Also, the E&M interface has five signaling types that
are supported by the MultiVOIP.
Once you have mapped out your phone and Internet/Intranet connections, you can build your VOIP
phone directory database that connects your MultiVOIPs together. From the phone directory
database you can construct a VOIP phone book and a VOIP dialing directory. One phone directory
database is all that is needed for a VOIP network. This database can be built at one site and then
distributed to other sites. The database ties the voice/fax channels to phone equipment and the
ethernet port to your local area network.
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MultiVOIP 200 User Guide
Before you construct the phone directory database, you must establish the Master-Slave relationship
between MultiVOIPs, which enables one multiVOIP (the “Master”) to control the database. The
database defines each phone number that can be dialed on the VOIP network. Each phone number
entry in the database identifies a phone connection. For example, Station Phone Number 101 is at
the corporate office. This phone number is connected to voice/fax channel 1 on the corporate office
MultiVOIP. Voice/fax channel 1 is connected to extension 4124 on the corporate phone switch. So if
a person in the corporate facility wants to access the VOIP network, they would merely dial extension
4124. However, if someone at the remote branch of fice wants to access the corporate phone system,
they have to dial Station Phone Number 101.
From the phone directory database you can build your VOIP dialing directory. For example, a
corporate user picks up a phone in his/her office and dials extension 4124. This connects them to
channel 1 on the corporate MultiVOIP. A second dial tone is heard, and they then dial VOIP
extension 301 which connects them to the remote branch office and enables a voice conversation to
take place over the VOIP network.
Similarly, a remote branch office employee can pick up the phone at 301 and dial VOIP extension
102. This routes the call to the corporate office MultiVOIP where a second dial tone is heard; the
remote branch office employee then dials extension 4122 and a conversation can take place between
the two phones.
To call from
Corporate Office
to Remote
Branch Office
Remote Branch
Office to
Corporate
Office
Call Process
Pick up any phone and dial an extension
number (e.g., 4124 or 4125). Upon hearing
the second dial tone, dial the Remote Branch
Office MultiVOIP (301).
Pick up telephone and dial 101 or 102.
Upon hearing the second dial tone,
dial 4122 or, for the fax machine, 4123.
VOIP Dialing Directory
Dialing
Sequence
4124
301
101
4122
Figure 1-3. VOIP Dialing Directory
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Front Panel Description
The front panel has three groups of LEDs that provide the status of the Ethernet connection
(Ethernet), Voice/Fax channels (Voice/Fax 1 and 2), and general status of the MultiVOIP (Boot and
Power). The front panel is shown in Figure 1-4, and a description of each LED follows.
Figure 1-4. Front Panel
Ethernet
RCVReceive Data indicator blinks when packets are being received from the local area network.
XMTTransmit Data indicator blinks when packets are being transmitted to the local area network.
LNKLink indicator lights when the Ethernet link senses voltage from a concentrator or external
device.
COLCollision indicator lights when a collision is detected on the Ethernet link.
Chapter 1 - Introduction and Description
Voice/Fax 1 and 2
FXSForeign Exchange Station indicator lights when the voice/fax channel is configured for FXS
operation.
FXOForeign Exchange Office indicator lights when the voice/fax channel is configured for FXO
operation.
E&MEar and Mouth indicator lights when the voice/fax channel is configured for E&M operation.
FAXFax indicator lights when there is fax traffic on the voice/fax channel.
XMTTransmit indicator blinks when voice packets are being transmitted to the local area network.
RCVReceive indicator blinks when voice packets are being received from the local area network.
XSGTransmit Signal indicator lights when the FXS-configured channel is off-hook, the FXO-
configured channel is receiving a ring from the Telco, or the M lead is active on the E&M
configured channel.
RSGReceive Signal indicator lights when the FXS-configured channel is ringing, the FXO-
configured channel has taken the line off-hook, or the E lead is active on the E&M-configured
channel.
Boot
The Boot indicator lights when the MultiVOIP is booting or downloading setup.
Power
The Power indicator lights when power is applied to the MultiVOIP.
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MultiVOIP 200 User Guide
Back Panel Description
The cable connections for the MultiVOIP are made at the back panel. Connectors include Power ,
Command Port (RS232), Ethernet (10BASE-T), Voice/Fax Channels 1 and 2 (E&M, FXO and FXS).
The cable connectors are shown in Figure 1-5 and defined in the following groups.
Voice/Fax Channel 1
E&MFXSFXO
Voice/Fax Channel 2
FXOFXSE&M
Ethernet RS232
Command
10Base-T
Power
1
0
Figure 1-5. Back Panel
Power Connector
The Power connector is used to connect the external power supply to the MultiVOIP. The Power
connector is a 6-pin circular DIN connector. A standard computer power cord connects the power
supply to a live AC grounded outlet.
Command Connector
The Command connector is used to configure the MultiVOIP using a PC with an available serial port
and running Windows® software. The Command connector is an RJ-45 jack, and an adapter cable is
provided to convert to a standard serial port DB9 female connector.
10Base-T (Ethernet) Connector
The Ethernet 10Base-T connector is used to connect the MultiVOIP to a LAN using unshielded
twisted cable. This connector is an RJ-45 jack.
Voice/Fax Channel 1 and 2
The Voice/Fax Channel connectors include three options per channel: E&M, FXS and FXO.
E&M - This connector is used for connecting Voice/Fax Channel 1 or 2 to the E&M trunk on a PBX.
This connector is an RJ-45 jack.
FXS - This connector is used for connecting Voice/Fax Channel 1 or 2 to a station device; e.g., an
analog phone, a KTS phone system, or a fax machine. This connector is an RJ-11 jack.
FXO - This connector is used for connecting V oice/Fax Channel 1 or 2 to the station side of a PBX.
This connector is an RJ-11 jack.
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Specifications
•One 1 Meg by 32 byte at 70 nanosecond SIMM is 4 Mb DRAM
Caution: SIMM speed and size cannot be mixed
•Single 19.2K bps asynchronous Command Port using an RJ-45 to DB9 cable with a DB9
female connector
Voice/Fax Channel 1 and 2
•Two RJ-11 jacks (FXO and FXS)
•One RJ-45 jack (E&M)
Electrical/Physical
Chapter 1 - Introduction and Description
•Voltage - 115 VAC (Standard), 240 Volts AC (Optional)
•Frequency - 47 to 63 Hz
•Power Consumption - 18 Watts
•Dimensions - 1.625" high x 6.175" wide x 9" deep
(4.13 cm x 15.68 cm x 22.86 cm)
•Weight - 2 pounds (0.9 kg)
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MultiVOIP 200 User Guide
14
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Voice / Fax over IP Networks
Chapter 2 - Installation
Page 16
MultiVOIP 200 User Guide
Installing Your MultiVOIP
The basic steps of installing your MultiVOIP network involve unpacking the units, connecting the
cables, and configuring the units using the included management software (MultiVOIP Configuration).
The recommended installation process includes three phases that, when completed, result in a fully
functional Voice Over IP network. A general description of each phase is provided below , and detailed
instructions are provided in Chapter 3, Software Loading and Configuration.
Phase 1: Configure and Install Your Master MultiVOIP
As the first step, the VOIP administrator configures the MultiVOIP designated as the “Master” unit.
This includes the assignment of a unique LAN IP address, subnet mask, and Gateway IP address; as
well as the selection of appropriate channel interface type for each of the Voice/Fax channels. Once
all connections have been made, the VOIP administrator configures the unit and builds the Phone
Directory Database that will reside with the Master unit.
Phase 2: Configure Your Slave MultiVOIP(s)
Once Phase 1 has been completed, the administrator moves on to configure the MultiVOIP(s)
designated as “Slave” units. Again, unique LAN IP addresses, subnet masks, and Gateway IP
addresses are assigned, and each Voice/Fax channel is configured for the appropriate channel
interface type. When this is done, the Phone Directory Database option is set to Slave, and the IP
address of the Master MultiVOIP is entered. Once all Slave units are configured, the process moves
on to Phase 3.
Phase 3: Deploy the VOIP Network
The final phase of the installation is deployment of the network. Through the first two phases, the
VOIP administrator controls configuration, so when the Slave MultiVOIPs are sent to their remote
sites, the remote site administrators need only to connect the units to their LAN and telephone
equipment. A full Phone Directory Database (supplied by the Master MultiVOIP) will be loaded into
their unit within minutes of being connected and powered up.
The final task of the VOIP Administrator is to develop the VOIP Dialing Directory based on the Phone
Directory Database and telephone numbers of the interfacing telephone equipment; at which point, a
VOIP user can call any person on the VOIP network.
Safety Warning Telecom
1.Never install phone wiring during a lightning storm.
2.Never install phone jacks in wet locations unless the jacks are specifically designed for wet
locations.
3.This product is to be used with UL and cUL listed computers.
4.Never touch uninsulated phone wires or terminals unless the phone line has been
disconnected at the network interface.
5.Use caution when installing or modifying phone lines.
6.Avoid using a phone (other than a cordless type) during an electrical storm. There may be a
remote risk of electrical shock from lightning.
7.Do not use the phone to report a gas leak in the vicinity of the leak.
8.To reduce the risk of fire, use only No. 26 AWG or larger Telecommunication line Cord.
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Unpacking Your MultiVOIP
Remove all items from the box. (See Figure 2-1.)
Chapter 2 - Installation
MADE IN U.S.A
MADE IN U.S.A
Figure 2-1. Unpacking
Safety Warnings
Caution: Danger of explosion if battery is incorrectly replaced.
A lithium battery on the circuit board provides backup power for the time keeping capability. The battery has an estimated life
expectancy of ten years.
When the battery starts to weaken, the date and time may be incorrect. If the battery fails, the board must be sent back to
Multi-Tech Systems for battery replacement.
The E&M, FXS, and Ethernet ports are not designed to be connected to a Public Telecommunication Network.
Valid VOIP Network Connections
The following VOIP network connections can be made at this time:
• FXS to FXS
• FXS to E&M
• FXS to FXO
• E&M to E&M
also,
• FXO to FXO (New)
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MultiVOIP 200 User Guide
Cabling Your MultiVOIP
Cabling your MultiVOIP involves making the proper Power , Command Port, and Internet connections.
Figure 2-2 shows the back panel connectors and the associated cable connections. The Cabling
Procedure has step-by-step instructions for cabling your MultiVOIP.
PSTN
Voice/Fax Channel 1
E&MFXSFXO
Voice/Fax Channel
1 & 2 Connections
E&MFXO
PBX
Voice/Fax Channel 2
FXOFXSE&M
FXS
Ethernet RS232
Command
10Base-T
Power
1
0
Hub
Power Connection
Command Port Connection
Network Connection
Figure 2-2. Cable Connections
Note: Before starting to cable your MultiVOIP, perform the E&M Jumper Block Positioning
Procedure if either voice/fax channel (1 or 2) will be connected to an E&M trunk that is a Type
1,3,4, or 5 rather than a Type 2 (the default).
Cabling Procedure
1Using the supplied cable, connect the power supply to a live AC outlet, then plug the power
supply into the MultiVOIP as shown in Figure 2-2. The power connector is a 6-pin circular
DIN connector.
2Connect the MultiVOIP to a PC using the RJ-45 to DB9 (female) cable provided with your
unit. Plug the RJ-45 end of the cable into the Command port of the MultiVOIP and connect
the other end to the PC serial port you’re using. See Figure 2-2.
3Connect a network cable to the Ethernet 10Base-T connector on the back of the MultiVOIP.
Connect the other end of the cable to your network.
4If you are connecting a station device; e.g., analog telephone, fax machine, or Key Telephone
System (KTS); to your MultiVOIP, connect the smaller end of a special adapter cable (supplied) to the Voice/Fax Channel 1 FXS connector on the back of the MultiVOIP and the other
end to the station device.
If you are connecting a PBX extension to your MultiVOIP, connect the smaller end of a special
adapter cable (supplied) to the Voice/Fax Channel 1 FXO connector on the back of the
MultiVOIP and the other end to the PBX extension.
If you are connecting an E&M trunk from a telephone switch to your MultiVOIP, connect one
end of an RJ-45 phone cord to the Voice/Fax Channel 1 E&M connector on the back of the
MultiVOIP and the other end to the trunk. Note: Appendix B has a pinout diagram for the E&M
back panel connector on the MultiVOIP.
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Chapter 2 - Installation
5Repeat the above step to connect the remaining telephone equipment to each Voice/Fax
Channel on your MultiVOIP.
6Turn on power to the MultiVOIP by setting the power switch on the back panel to the 1 (up,
On) position. Wait for the Boot LED on the MultiVOIP to go Off before proceeding. This may
take a couple of minutes.
Proceed to the Chapter 3, Software Loading and Configuration, to load the MultiVOIP software.
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MultiVOIP 200 User Guide
E&M Jumper Block Positioning Procedure
Each voice/fax channel on the MultiVOIP has a separate E&M jumper block, located near the jacks on the back panel of the
MultiVOIP. Each jumper block has 8 pairs of pins with a jumper plug on three adjacent pairs of pins. The jumper plug must be
centered on the E&M type number (see Figure 2-3) that matches the E&M connection for that channel. Perform the following procedure if you need to move the E&M jumper block from its default (Type 2) position.
1Ensure that the external power supply is disconnected from the MultiVOIP.
2Turn the MultiVOIP upside down and remove the cabinet mounting screw at the center back of
the cabinet.
3Turn the MultiVOIP right side up, then slide the base out the rear of the cabinet.
Note: To change a jumper position, lift the jumper plug up off the jumper block, then move it to
the new position, ensuring that the middle jumper of the jumper block is centered on the E&M
type number (1,3; 4; or 5) as shown on Figure 2-3. (Note: Numbers are
Back Panel Connectors
not
on the board.)
2
Channel 2
2
Channel 1
Jumper Blocks
In Position 2
(Default)
1,3
4
Alternate Positions
Note: Markings do not appear on board.
5
Figure 2-3. E&M Jumper Block Positions
4Change the jumper block position for any voice/fax channel to be connected to an E&M trunk
that is not a Type 2 (the default position).
5Slide the base all the way into the cabinet until it stops.
6Turn the MultiVOIP upside down and replace the cabinet mounting screw that was removed in
step 2.
7Turn the MultiVOIP right side up, then perform the cabling procedure.
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Voice / Fax over IP Networks
Chapter 3 - Software Loading and Configuration
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MultiVOIP 200 User Guide
Installing Your MultiVOIP Software
The following installation procedures do not provide every screen or option in the process of installing
the MultiVOIP software. It is assumed that a technical person with a thorough knowledge of Windows
and the software loading process is doing the installation. Additional information on the MultiVOIP
software is provided in Chapter 4, MultiVOIP Software, and in the on-line Helps.
Phase 1: Configuring Your Master MultiVOIP
Configuring your Master MultiVOIP involves software loading and configuration.
Software Loading
The software loading procedure does not provide every screen or option in the loading process. It is
assumed that a technical person with a thorough knowledge of Windows and the software loading
process is performing the installation. Additional information on the MultiVOIP software is provided in
the User Guide supplied with your MultiVOIP.
If you are installing a MultiVOIP behind a firewall, you need to add the following UDP ports to your
firewall.
Ch2 RTP [5006]Ch2 RTCP [5007]
Refer to your firewall user documentation to enter and open these ports.
1Run Windows on the PC connected to the MultiVOIP.
2Insert the MultiVOIP diskette labeled Disk 1 into the disk drive on the PC connected to the
MultiVOIP.
3
4The MultiVOIP Setup welcome screen is displayed.
Win3.1 users
b:\setup (depending on the location of your floppy disk drive) in the Command Line field and
then click OK.
Win95/98/NT users
choose a:\setup or b:\setup (depending on the location of your floppy disk drive), then click
OK.
- in Program Manager, click File | Run. In the Run dialog box, type a:\setup or
- click Start | Run. In the Run dialog box click the down arrow and
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Chapter 3 - Software Loading and Configuration
Press Enter or click Next> to continue.
5Follow the on-screen instructions to install your MultiVOIP software.
6The following dialog box selects the COM port of your PC connected to the Command port of
the MultiVOIP. From the Select Port drop-down list, choose the COM port of your PC.
Click OK to continue.
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MultiVOIP 200 User Guide
7The Setup Complete dialog is displayed.
Click Finish to continue.
8The following message is displayed:
Click Yes to continue.
9The following dialog box is displayed.
Click YES to continue.
10The IP Protocol Default Setup dialog box is displayed.
The default Frame Type is TYPE_II. If this does not match your IP network, change the Frame
Type by clicking the drop-down arrow and selecting SNAP. The available Frame Type choices
are TYPE_II and SNAP.
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Chapter 3 - Software Loading and Configuration
11In the Ethernet group, enter the IP Address, Subnet Mask, and Gateway Address, unique to
your IP LAN, in the corresponding fields.
The IP address is your unique LAN IP address, and the Gateway address is the IP address of
the device connecting your MultiVOIP to the Internet.
Click OK when you are finished.
12The Channel Setup dialog box is displayed. Its three tabs are used to define the voice/fax
channel interface, voice coder, fax parameters, and regional phone parameters (tone pairs)
for each channel.
Configure each channel for the type of interface you are connecting to. The Interface tab
defaults to Channel 1 in the Select Channel field. To change the channel number, click the
drop-down arrow and the list of channels is displayed. Highlight the channel you want to
configure.
13The Interface group defaults to FXS (Loop Start). Select the interface option that corre-
sponds to the interface type being connected to the Voice/Fax Channel 1 jack on the back
panel of the MultiVOIP.
If you are connected to a station device; e.g., an analog telephone, fax machine, or KTS
telephone system to the Voice/Fax connector on the back of the unit, FXS (Loop Start) will
likely be the correct Interface option.
If the station device uses ground start, then choose the FXS (Ground Start) option. Refer to
the device’s user documentation.
If you are using an analog extension from your PBX, then choose the FXO option. Check with
your in-house phone personnel to verify the connection type.
If you are connecting to an analog E&M trunk on your PBX, then choose the E&M option.
If you choose the FXO interface, the Dialing Options Regeneration group is enabled. Check
with your local in-house phone personnel to verify whether your local PBX dial signaling is
Pulse or tone (DTMF). Then, set the Regeneration option accordingly.
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MultiVOIP 200 User Guide
For FXO-to-FXO communications, you can enable a specific type of FXO Disconnect -- either
current loss, tone detection, or silence detection. (Check with your in-house phone
personnel to verify the preferred type of disconnect to use.) For tone detection, you can
select from drop-down lists either one or two tones that will cause the line to be disconnected;
the person hanging up a call must then hit the key(s) that will produce those tones. For
silence detection, select One Way or Two Way, then set the timer for the number of sec-
onds of silence before disconnect. Note: the default value of 15 seconds may be shorter than
desired for your application.
If you choose the E&M interface, then the E&M Options group is enabled. Check with your
local in-house phone personnel to determine if the signaling is Dial Tone or Wink and if the
connection is 2-wire or 4-wire. If Wink signaling is used, then the Wink Timer is enabled with
a default of 250 milliseconds. The range of the Wink Timer is from 100 to 350 milliseconds.
Consult with your local in-house phone personnel for this timer setting.
If you want to dedicate a local voice/fax channel to a remote voice/fax channel (so you will not
have to dial the remote channel), click the Auto Call Enable option in the Auto Call group.
Then enter the phone number of the remote MultiVOIP in the Phone Number field.
Note: After configuring a given channel (1 or 2), you can copy that channel’s configuration to
the other channel by clicking the Copy button. Everything on the Interface tab will be copied
to the other channel.
14Repeat the above step to configure the interface type for voice/fax channel 2.
15The Voice/Fax tab displays the parameters for the voice coder, faxing, DTMF gain, billing
charges for inbound and/or outbound calls, password authentication on inbound and/or
outbound calls, and auto disconnect which limits call duration.
16To change the voice coder, first select the channel by clicking the Select Channel down arrow
and highlighting the channel number, then click the Voice Coder down arrow and highlight
your new voice coder entry.
If you changed the voice coder, ensure that the same voice coder is used on the voice/fax
channel you are calling; otherwise, you will always get a busy signal.
17If you selected the FXO interface and are using touchtone dialing, you can set up the DTMF
gain (or output level in decibels - dB) for the higher and lower frequency groups of the DTMF
tone pair. Make your selections in the drop-down lists in the DTMF Gain group.
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Chapter 3 - Software Loading and Configuration
Note: Only change the DTMF gain under the direction of Multi-Tech Technical Support
supervision.
18The Fax group enables you to send and receive faxes on the selected voice/fax channel. You
can set the maximum baud rate for faxes in the drop-down list in the Fax group. If you do not
plan to send or receive faxes on a given voice/fax channel, you can disable faxes in the Fax
group.
19You can set up billing options for inbound and/or outbound calls by checking them in the
Billing Options group and then typing in the charge in cents per x seconds.
20Password protection can be enabled for outbound and/or inbound calls on the selected voice/
fax channel. If you enable password Authentication on inbound or outbound calls, you need
to also enter a password of up to 14 numeric characters in the Password field.
A user attempting to access the voice/fax channel will be repeatedly prompted with a series of
two short tones. At this point they must enter the password from their dtmf keypad to access
the voice/fax channel.
21The Automatic Disconnect option limits call duration to the number of seconds entered in the
Timer: (sec) field. The default value of 15 seconds can be changed to any other value up to
65,535 (roughly 18.2 hours).
Note: After configuring a given channel (1 or 2), you can copy that channel’s configuration to
the other channel by clicking the Copy button. Everything on the Voice/Fax tab will be copied
to the other channel.
If your country/region is not the default USA, click the Regional tab and proceed to step 22;
otherwise, proceed to step 23 to begin building your phone directory database.
22To change the Tone Pairs on the Regional tab, click the Country/Region down arrow and
highlight your specific country or region.
The Tone Pairs group parameters change per your choice. Click OK when finished. Proceed
to step 23 to begin building your phone directory database.
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23The Phone Directory Database dialog box is displayed. You will build your personalized
MultiVOIP Phone Directory in the following steps.
The MultiVOIP configured as a “Master” will contain the master database. The master data-
base has the phone numbers of all the MultiVOIP’s available for communication on an IP
network. This database is downloaded to each Slave MultiVOIP as it comes on-line.
Click Add (+) to begin building your phone directory database.
24The Add/Edit Phone Entry dialog is displayed.
In the Station Information group, enter the unique phone number of the local device con-
nected to Channel 1 in the Phone Number field (for example phone number 101).
25The Description field is optional, but can be useful in associating the channel to the exten-
sion. If you wish, enter a description of your local phone number. This description serves to
identify the phone number you entered in the previous step (for example, “Jerry’s Desk”).
26The Permit Hunting option enables the answering unit to roll over to a second channel if the
first channel is busy. Click Permit Hunting if you want the calls to roll over to a second voice/
fax channel.
Note: The Master MultiVOIP
reference to obtain a downloaded copy of the master phone directory database.
28
must
have a static IP address that the remote MultiVOIP can
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Chapter 3 - Software Loading and Configuration
27In the MultiVOIP Identification group, enter the IP address of the Master MultiVOIP in the IP
Address field. For example, 204.22.122.118. Then obtain the 12-digit Node ID#
(0008005xxxxx) from the ID plate on the back panel of the MultiVOIP and enter this number
in the Ethernet Node ID field. If the ID plate is missing or damaged, you can also Telnet to the
MultiVOIP and, on the MultiVOIP Telnet Server menu enter 1 to advance to the Main Menu,
then enter 3 for System Information where item 1 is the Ethernet Port Address you want to
enter in the Ethernet Node ID field.
28Click OK and you are returned to the Phone Directory Database dialog box, which now
includes phone number 101 with its IP address, channel number and description.
29Click Add (+) and the Add/Edit Phone Entry dialog box is displayed again.
30Enter the phone number for the remote MultiVOIP in the Station Information group Phone
Number field. For example, 201.
31Click the Description field and enter a description for the remote MultiVOIP phone number for
Channel 1. For example, “Hari’s Office.”
Note: If the remote MultiVOIP is located behind a proxy server that uses a dynamically
assigned IP address, select Dynamic (
field blank. The Master MultiVOIP will
MultiVOIP.
disabling
learn
Static IP Address) and leave the IP Address
the IP address when it is contacted by the remote
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MultiVOIP 200 User Guide
32Enter the IP address of the remote MultiVOIP in the IP Address field in the MultiVOIP Identi-
fication group. For example, 202.56.39.100.
33Click OK and you are returned to the Phone Directory Database dialog box, which now
includes the second number and related information in the Phone Number list.
Note: If only Channel 1 is active, you must enter two phone numbers. The first number will be
the local MultiVOIP phone number for Channel 1, and the second number will be the remote
MultiVOIP phone number for Channel 1.
If both Channels 1 and 2 are active, four phone numbers will have to entered.
34When you have finished, click OK to download the setup configuration to the MultiVOIP.
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Chapter 3 - Software Loading and Configuration
35The Checking MultiVOIP dialog box is displayed.
Click OK to proceed.
36The Writing Setup dialog box is displayed as the setup configuration is written to the Multi-
VOIP.
37After the setup is written to the MultiVOIP, the unit is rebooted.
38Check to ensure that the BOOT LED on the MultiVOIP is Off after the download is complete.
This may take several minutes as the MultiVOIP reboots.
39Win3.1 users - you are returned to your Program Manager where the MultiVOIP Program
Group and Program Items (Windows icons) are displayed.
Win95/98/NT users - you are returned to your MultiVOIP folder which is open and visible on
your desktop.
At this time, your master MultiVOIP is configured. Proceed to Phase 2 to configure the slave
MultiVOIP(s).
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Phase 2: Configure Your Slave MultiVOIPs
If you are installing a MultiVOIP behind a firewall, you need to add the following UDP ports to your
firewall.
Ch2 RTP [5006]Ch2 RTCP [5007]
Refer to your firewall user documentation to enter and open these ports.
1Disconnect the PC from the command port of the Master MultiVOIP and connect it to the com-
mand port on the Slave MultiVOIP.
2Win 3.1 users - from the Program Manager, click the MultiVOIP Configuration icon in the
MultiVOIP Program Group. The main menu is displayed.
Win95/98/NT users - from your desktop, click Programs I MultiVOIP I MultiVOIP Configura-
tion. The main menu is displayed.
3Click IP and the IP Setup dialog box is displayed.
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The default Frame Type is TYPE_II. If this does not match your IP network, change the
Frame Type by clicking on the drop-down arrow. The Frame Type choices are TYPE_II and
SNAP.
4In the Port Address group, enter the IP Address and IP Mask. In the Gateway Address
group, enter the gateway IP address for the slave unit.
The IP address is your unique LAN IP address, and the Gateway address is the IP address of
the device connected to the Internet/Intranet.
Click OK when you are finished. The main menu is displayed.
5From the main menu, click Voice Channels and the Channel Setup dialog box is displayed.
The three tabs in this dialog box define the channel interface, voice coder, fax parameters,
and regional telephone parameters for each channel.
Configure each channel for the type of interface you are connecting to. The Interface tab
defaults to Channel 1 in the Select Channel group. To change the channel number, click on
the down arrow for the Select Channel and a drop-down menu appears with all the channels
displayed. Highlight the channel number you want to configure.
6The Interface group defaults to FXS (Loop Start). Select the interface option to correspond to
the interface type being connected to the Voice/Fax connector on the back panel of the
MultiVOIP.
If you are connecting a station device, e.g., analog telephone, fax machine, or Key Telephone
System (KTS) to the Voice/Fax connector on the back of the unit, then the FXS (Loop Start)
will likely be the correct Interface option.
If the station device uses ground start, then chose FXS (Ground Start) option. Refer to the
device’s user documentation.
If you are using an extension from your PBX, then chose FXO option. Check with your in-
house telephone personnel to verify connection type.
If you chose an FXO interface, then the Dialing Options Regeneration group is enabled.
Check with your local in-house telephone personnel to verify whether your local PBX dial
signaling is Pulse or tone (DTMF). Set the Regeneration option accordingly.
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For FXO-to-FXO communications, you can enable a specific type of FXO Disconnect -- either
current loss, tone detection, or silence detection (Check with your in-house phone personnel to verify the preferred type of disconnect to use). For tone detection, you can select from
drop-down lists either one or two tones that will cause the line to be disconnected; the person
hanging up a call must then hit the key(s) that will produce those tones. For silence detec-tion, select One Way or Two Way, then set the timer for the number of seconds of silence
before disconnect.
Note: the default value of 15 seconds may be shorter than desired for your application.
If you are connecting to an analog E&M trunk on your PBX, then choose the E&M option.
If you choose the E&M interface, then the E&M Options group is enabled. Check with your
local in-house telephone personnel to determine if the signaling is Dial Tone or Wink and if the
connection is 2-wire or 4-wire. If Wink signaling is used, then the Wink Timer is enabled with a
default of 250 milliseconds. The range of the Wink Timer is from 100 to 350 milliseconds.
Consult with your local in-house telephone personnel for this timer setting.
If you want to dedicate a voice/fax channel to a point-to-point configuration, i.e., the device on
a local channel can call only a device on a remote MultiVOIP channel, click on the Auto CallEnable option in the Auto Call group. Then enter the phone number of the remote MultiVOIP
in the Phone Number field.
7Repeat the above step to configure the interface type for voice/fax Channel 2.
8The Voice/Fax tab displays the parameters for the voice coder, faxing, DTMF gain for FXO
interface, billing charges for inbound and/or outbound calls, password authentication on
inbound and/or outbound calls, and auto disconnect which limits call duration.
9To change the voice coder, first select the channel by clicking the Select Channel down arrow
and highlighting the channel number, then click the Voice Coder down arrow and highlight
your new voice coder entry.
If you changed the voice coder, ensure that the same voice coder is used on the voice/fax
channel you are calling; otherwise, you will always get a busy signal.
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Chapter 3 - Software Loading and Configuration
10If you selected the FXO interface and are using Touchtone dialing, you can set up the DTMF
gain (or output level in decibels - dB) for the higher and lower frequency groups of the DTMF
tone pair. Make your selections in the drop-down lists in the DTMF Gain group. Note: Only
change the DTMF gain under the direction of Multi-Tech Technical Support supervision.
11The Fax option enables you to send and receive faxes on the selected voice/fax channel. You
can set the maximum baud rate for faxes in the drop-down list in the Fax group. If you do not
plan to send or receive faxes on a given voice/fax channel, you can disable faxes in the Fax
group.
12You can set up billing options for inbound and/or outbound calls by checking them in the
Billing Options group and then typing in the charge in cents per x seconds.
13Password protection can be enabled for outbound and/or inbound calls on the selected voice/
fax channel. If you enable password Authentication on inbound or outbound calls, you need
to also enter a password of up to 14 numeric characters in the Password field.
A user attempting to access the voice/fax channel will be repeatedly prompted with a series of
two short tones. At this point they must enter the password from their dtmf keypad to access
the voice/fax channel.
14The Automatic Disconnect option limits call duration to the number of seconds entered in the
Timer: (sec) field. The default value of 15 seconds can be changed to any other value up to
65,535 (roughly 18.2 hours).
Note: After configuring a given channel (1 or 2), you can copy that channel’s configuration to
the other channel by clicking the Copy button. Everything on the Voice/Fax tab will be copied
to the other channel.
If your country/region is not the default USA, click the Regional tab and proceed to step 15;
otherwise, proceed to step 16 to setup your phone directory database.
15To change the call progress signaling for your Country/Region, click on the down arrow and
highlight your specific country or region.
The Tone Pairs group parameters change per your choice. Click OK and you are returned to
the main menu.
16From the main menu, click Phone Book to display the Phone Directory Database dialog
box.
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In the Database Type group, click on the Slave option. The Update Database From group
becomes active.
17Enter the IP address of the master MultiVOIP in the Master IP Address field.
18Click OK and you are returned to the main menu.
19Click Download Setup to write the new configuration to the slave unit. The Save Setup dialog
box is displayed.
20Select (check) the Save Current Setup as User Default Configuration and click OK. The
Writing Setup dialog box is displayed as the setup configuration is written to the MultiVOIP.
After the setup is written to the MultiVOIP, the unit reboots.
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Chapter 3 - Software Loading and Configuration
21Check that the BTG LED on the MultiVOIP is Off after the download is complete. This may
take several minutes as the MultiVOIP reboots.
22You are returned to the main menu.
Your MultiVOIP is operational at this time.
Repeat Phase 2 for each of the slave units. When all slaves have been configured, proceed with
Phase 3.
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Phase 3: Deploy the VOIP Network
Phase 3 involves the VOIP Administrator developing the VOIP Dialing Directory and deploying the
pre-configured slave MultiVOIPs to their remote sites. The remote site administrators need only
connect power to the pre-configured MultiVOIP, connect it to their Ethernet LAN and predefined
telephone equipment, and then wait for the phone directory database to be downloaded.
Perform the following procedure to deploy your VOIP network.
VOIP Administrator
1Establish your VOIP Dialing Directory based on your Phone Directory Database for the
numbers to connect the MultiVOIP’s to your VOIP network and the telephone extension
number you need to connect to the Voice/Fax channels. A sample VOIP Dialing Directory is
provided below for your consideration and use.
To call from
VOIP Dialing Directory
Call Process
Dialing
Sequence
2Send the slave MultiVOIPs to their remote sites.
Remote Site Administrator
3Unpack your MultiVOIP.
4Connect one end of the power supply to a live AC outlet and connect the other end to the
Power connection on your MultiVOIP.
Voice/Fax Channel
Connections
FXO
E&M
FXS
FXSE&M
FXO
PSTN
Figure 5. Remote Site Cable Connection
5Connect a network cable to the ETHERNET 10Base-T (RJ-45) connector on the back of your
MultiVOIP.
10BASET
ETHERNET
POWER
Power Connection
Ethernet Connection
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Chapter 3 - Software Loading and Configuration
6If you are connecting a station device (e.g., analog telephone, fax machine, or Key Telephone
System (KTS) to your MultiVOIP, connect the smaller end of a special adapter cable (sup-
plied) to the Voice/Fax Channel 1 FXS connector on the back of the MultiVOIP and the other
end to the station device.
If you are connecting a PBX extension to your MultiVOIP, connect the smaller end of a special
adapter cable (supplied) to the Voice/Fax Channel 1 FXO connector on the back of the
MultiVOIP and the other end to the PBX extension.
If you are connecting an E&M trunk from a telephone switch to your MultiVOIP, connect one
end of an RJ-45 phone cord to the Voice/Fax Channel 1 E&M connector on the back of the
MultiVOIP and the other end to the trunk phone jack. See connector drawing in Appendix B.
If you are connecting to an E&M trunk, you need to ensure that the E&M trunk jumper is in the
correct position for the E&M type trunk. The default E&M jumper position is E&M type 2. To
change the E&M jumper position, perform the E&M jumper block positioning procedure.
7Repeat the above step to connect the remaining telephone equipment to each Voice/Fax
Channel on your MultiVOIP.
8Turn on power to the MultiVOIP by placing the ON/OFF switch on the back panel to the ON
position. Wait for the BOOT LED on the MultiVOIP to go OFF before proceeding. This may
take a couple of minutes to go OFF.
9At this time your VOIP network should be fully operational, dial one of the sites in your network
using the dialing directory supplied by your network Administrator.
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Voice / Fax over IP Networks
Chapter 4 - MultiVOIP Software
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MultiVOIP 200 User Guide
Introduction
This chapter describes various features of the MultiVOIP software that enable you to change
(update) the configuration of your MultiVOIP. The basic configuration parameters were established
during the loading of the software (Chapter 3). The MultiVOIP software and configuration utilities
described in this chapter enable you to change that initial configuration as necessary.
The primary interface to the MultiVOIP software is a main menu (with MultiVOIP 200 Setup in the
title bar) with individual buttons that enable you to quickly and easily select a desired function. These
features are discussed in detail in the MultiVOIP Configuration section later in this chapter .
The other seven configuration utilities in the MultiVOIP 200 software provide additional functionality.
The
Configuration port setup
MultiVOIP (i.e., through a direct connection of a PC to the Command Port on the MultiVOIP, or via
your Internet or LAN connection to the LAN port on the MultiVOIP). The
enables you to easily set the date and time used for data logging in the MultiVOIP.
Factory Defaults
Firmware
Download User Defaults
software installation) and update the MultiVOIP configuration with any necessary changes.
Download Voice Coders
upgrade. The
software from your PC.
enables you to download new versions of firmware as enhancements become available.
utility enables you to change the method by which you access the
Date and Time setup
enables you to return the configuration to the original factory settings.
enables you to repeat the download user defaults process (part of
enables you to download voice coders to the MultiVOIP after repair or
Uninstall MultiVOIP Configuration
utility removes most of the MultiVOIP 200
utility
Download
Download
The MultiVOIP software includes a context-sensitive Help system. Clicking a Help [ ? ] button
anywhere in the graphical user interface (GUI) will display definitions and recommended values for
the buttons, options, and fields on that dialog box or menu. Clicking the green underlined text in the
Helps displays a popup box of related supplementary information for that topic. Clicking the Search
button (just below the menu bar) displays an Index tab with a list of 16 different topics. Click a topic,
then click the Display button to display the text associated with that topic.
Before You Begin
The MultiVOIP software operates in a Microsoft Windows® environment. The MultiVOIP 200 program
group contains icons for all the utilities described above. In Windows 95/98/NT, you can access the
individual utility programs either by clicking Start | Programs | MultiVOIP 200 |
double-clicking the utility icon in the MultiVOIP 200 program group shown here:
(utility)
, or by
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MultiVOIP Configuration
The MultiVOIP Setup menu consists of 10 buttons in which you can point and click, an Events
window in the middle of the menu, and a status bar at the bottom of the menu. The 10 buttons allow
you to display and change the voice channels and IP protocol parameters, display and manage the
Phone Book listing, view statistics and call progress, and change features such as SNMP Agent,
Telnet Server, WEB Server, and assign a MultiVOIP password.
Chapter 4 - MultiVOIP Software
The Events window in the lower third of the Setup menu provides information about the boot process.
The status bar at the bottom of the Setup menu displays the current status of the unit and shows, for
example, if it is Running, the most recent date the unit was configured, the type of connection you
have to the unit, and your rights. It shows if your PC is connected directly to the command port of the
MultiVOIP or is communicating with the Ethernet port. The last field on the status bar is the Rights
field which displays either Read/Write or Read only rights. The first user to communicate with the
MultiVOIP has Read/Write rights that enable the user to view and/or change the configuration of the
MultiVOIP. Any additional users have only Read Only rights and can only display the configuration of
the MultiVOIP but are prohibited from changing the configuration.
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Changing Channel Parameters
The channel parameters include the interface type and its options, voice and fax settings, and voice
communications for the country and region in which the MultiVOIP is operating. The Channel Setup
dialog box, accessed by clicking the Voice Channels button on the Setup men, has three tabs that
display the following categories of channel information -- Interface, Voice/Fax, and Regional.
Interface
The Interface tab defines the parameters related to the physical interface of the voice/fax channel.
Depending on the interface type selected (FXS, FXO, or E&M), other options on the interface tab will
turn grey (become inactive) indicating that they do not apply to the selected interface. Max Dial
Digits, Inter Digit Time and Autocall features apply to all interface types.
The Max Dial Digits indicates the maximum number of digits the MultiVOIP will allow you to enter
when dialing one of the numbers in the Phone Directory Database. As soon as you have entered this
number of digits, the MultiVOIP will immediately attempt to match the digits you have dialed with an
entry in the database. The range for the Max Dial Digits is from zero to 16 digits with a default of five.
The Inter Digit Time (in milliseconds) option in the Dialing Options group defines the amount of time
the MultiVOIP waits between digits as they are entered by the user. If this timer expires, the
MultiVOIP will immediately attempt to match the digits entered to an entry in the Phone Directory
Database. The range for this option is 200 to 10,000 with a default of 2,000.
The Auto Call option allows the local MultiVOIP to call a remote MultiVOIP without the user having to
dial a Phone Directory Database number. As soon as you access the local MultiVOIP voice/fax
channel, the MultiVOIP immediately connects to the remote MultiVOIP that you identified in the
Remote MultiVOIP Phone Number field of this option.
FXS Interface
The FXS Interface is used to connect telephones, fax machines, key telephone systems, etc., to the
MultiVOIP. In addition, you need to select either Loop Start or Ground Start. Most of the equipment
mentioned will use Loop Start which is the default.
FXO Interface
The FXO Interface is used to connect PBX extensions or central office telephone lines. You also,
need to select DTMF or Pulse dialing in the Regeneration field of the Dialing Options group. If you
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Chapter 4 - MultiVOIP Software
are unsure of the correct selection, contact the personnel in charge of your PBX or your local
telephone company to determine whether pulse or DTMF should be used.
E&M Interface
The E&M Interface is used to connect PBX E&M trunks. You will need to select between Dial T one or
Wink signaling and also between 2-wire and 4-wire mode. If wink signaling is selected, the wink
timer field becomes active with a range from 100 to 350 milliseconds. Contact the personnel in
charge of your PBX to determine the proper configuration of these settings.
FXO Disconnect On
The FXO Disconnect On option applies when two MultiVOIPs are used in an FXO-to-FXO
configuration. When you have an FXO-to-FXO configuration, you need to determine the method of
terminating the call. Three methods of terminating the call are provided: Current Loss, Tone
Detection, or Silence Detection. Current Loss is the preferred method. Current Loss has to be
supported by your PBX or local telephone company. Current Loss terminates the call when the PBX
or local telephone company switch detects a person hanging up the phone and opens the local circuit
for a minimum of 600 milliseconds.
T one Detection disconnect method terminates the call when the party who wishes to disconnect
enters a one or two digit sequence on the telephone keypad. V alid digits are zero to nine, *, #, and A
thru D.
Silence Detection can be silence in one direction or silence in both directions for a specified amount
of time. The amount of time is defined by the entry in the Silence Timer. The range of the Timer is
from one to 65535 seconds (roughly 18 hours). The default is 15 seconds.
Voice/Fax
The Voice/Fax tab controls the voice coder, Fax settings, DTMF gain, and some miscellaneous
options.
The MultiVOIP supports many state-of-the art ITU (International Telecommunications Union) voice
coders. The V oice Coder drop-down menu enables you to select from a range of coders with specific
bandwidths. The higher the bps rate, the more bandwidth is used. The channel that you are calling
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MultiVOIP 200 User Guide
has to have the same voice coder selected; otherwise, you will always get a Busy signal.
The Fax group enables a fax machine to transmit and receive faxes through the MultiVOIP. If a fax
machine is connected to one of the voice/fax channels, the Max Baud Rate should be set to match
the baud rate of the fax machine (refer to user documentation). The Fax V olume setting controls the
output level of the fax tones, and this setting should be changed only under the direction of MultiTech’s Technical Support personnel (see Chapter 6 - Warranty, Service and Tech Support).
The DTMF Gain group controls the volume level of the digital tones sent out for Touchtone dialing.
The Gain High and Gain Low fields control the gain in dB (decibels) of the High and Low tones in the
tone pairs; the default gain values are -4 dB and -7 dB, respectively . DTMF Gain should not be
changed except under supervision of MultiTech’s Technical Support.
Billing Options can be used to track the cost of Inbound and/or Outbound calls on any of the three
interfaces (FXO, FXS, or E&M). The amount to be charged in cents is entered in the Charge ( )Cents field together with the associated time duration in the Per ( )Seconds field. While a given
call is active, the accumulated charges can then be viewed on the Call Progress dialog box. When
the call ends, the charges are transferred to a Log File that can be viewed by highlighting the call
event in the Log Entries dialog box and selecting Details.
The Authentication Option enables you to provide Password Protection on Inbound and/or
Outbound calls on any of the three interfaces (FXO, FXS, and E&M). A password of up to 14
numeric characters can be assigned to either or both voice/fax channels. The required password
must then be entered from the device initiating a call over the protected voice/fax channel.
The Automatic Disconnect Option enables you to limit the duration of a call on any of the three
interfaces (FXO, FXS, or E&M). This function will hang up the call when a timer expires. The default
timer value of 15 seconds can be increased to any value up to 65535 seconds (roughly 18 hours).
Regional
The regional tab controls the voice communications for the country or region in which the MultiVOIP
is being used.
From the Country/Region drop-down list you can select the country or region for which you are
configuring the MultiVOIP. The Tone Pairs group always displays the tones used in the country or
region currently selected. In addition to Australia, Central America, Chile, Europe, France, Japan, UK,
and USA, there is a Custom selection (with defaults identical to USA) that will make the Custom
button active. Clicking the Custom button enables you to edit the Tone Pairs and establish custom
sets of tone pairs for Dial Tone, Ring, and Busy on a Custom Tone Pair Settings dialog box.
The Pulse Generation Ratio group contains two ratios: the 60/40 is for the USA, and the 67/33 ratio
is for international applications.
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Changing the Phone Directory Database
The Phone Directory Database dialog box displays all the phone numbers in your MultiVOIP network.
The database displays the phone numbers in numerical order with the IP Address, Channel
assignment, and Description.
Chapter 4 - MultiVOIP Software
Access this database by clicking the Phone Book button on the Main MultiVOIP menu. You can add,
delete, or edit any entry in the database and you can change the master - slave relationship of the
database. The Slave Status displays status of all the slave units in your VOIP network. The Phone
Number of each slave is displayed with its IP Address, current line status, and the description of the
phone number.
The phone number does not have to be a conventional phone number; for example, it does not have
to be 717-5565. It can be a single digit or several digits, except it cannot be longer than the entry in
the Max Dial Digits field in the Dialing Options group of the Channel Setup dialog box. For example,
you could enter a phone number of 101 with a description of Jerry’s Desk, the phone number is
assigned to channel 1. If you want the call to be rolled over to a second channel, you can enable
Permit Hunting. If the assigned channel is busy , then the call is rolled over to the next channel.
The MultiVOIP Identification group defines the type of addressing (Dynamic or Static) for the
master and slave units for their respective Phone Numbers. The Phone Numbers assigned to the
master MultiVOIP have to be Static addressing and the Phone Numbers assigned to the slave
MultiVOIP can be either dynamic or static depending on whether a Proxy Server is providing the
connection to the Internet. If a Proxy Server is in front of the MultiVOIP providing the Internet
connection and the Proxy Server is using dynamic addressing (i.e., the ISP is assigning the Proxy
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Server IP address), then this slave MultiVOIP will be defined as using Dynamic addressing and the
IP Address field in the Identified By group will grey out (be inactive).
If a Proxy Server with a static IP address is in front of the slave MultiVOIP, then the Identified By IP
Address field must contain the address of the Proxy Server.
If the slave MultiVOIP is connected directly to the Internet, then its addressing mode must be Static.
If the slave unit is using Static addressing, then the IP Address field in the Identified By group has to
contain the Static IP address of the slave MultiVOIP.
The Ethernet Node ID is a 12-digit Identification Number assigned to each unit. This Ethernet Node
ID number is a hardware identification number that is affixed to each unit during the manufacturing
process and cannot be changed. This ID number (for example, 0008005xxxxx) is located on an ID
plate attached to every unit. This ID number has to be entered in the Ethernet Node ID field for the
telephone number entered in the Phone Number field. If you are assigning a Phone Number for a
slave unit, the Ethernet Node ID has to be for that slave unit.
If this plate is damaged or missing, you can also obtain the ID number by Telneting to the unit. From
the MultiVOIP Telnet Server menu, choose the Voice over IP Configuration option which takes you to
the Main Menu. In the Main Menu, choose System Information and the ID number is presented in the
Ethernet Port Address of the System Information menu.
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When you enter this information and click OK, the information is loaded into the phone directory
database.
To add a second entry, click Add(+) and the Add/Edit Phone Entry dialog box is again displayed.
The same data needs to be added for channel 2. After the two local entries are added to the
database, then you need to turn your attention to the entries for the remote MultiVOIPs. The same
data has to be added for each remote MultiVOIP.
To establish the phone directory database for a remote MultiVOIP , you do not have to enter phone
numbers, but you have to check the Slave option in the Database Type group. When you click the
Slave option, the Update Database From group becomes active. You need to enter the IP address of
the MultiVOIP that you established as the Master .
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Changing IP Parameters
The IP Setup dialog box establishes the IP addressing for the local Ethernet LAN, defines the Internet
gateway address, and for a remote MultiVOIP the global-to-local IP address translation is defined on
the Proxy Setup tab. The IP Setup dialog box is accessed by clicking the IP button on the MultiVOIP
200 Setup menu.
When the IP Setup dialog box is displayed, the IP address of your MultiVOIP is displayed with its IP
Mask. The Gateway Address is the IP address of the device connected to the Internet. This
completes the fields in the IP Protocol Default Setup dialog box.
Chapter 4 - MultiVOIP Software
ProxyServer
The Proxy Setup tab is used when a ProxyServer is used to connect the LAN to the Internet. The
Proxy Setup dialog box is displayed by clicking the Proxy Setup tab in the IP Setup dialog box.
If a ProxyServer is used in a MultiVOIP network, an address translation takes place within the
ProxyServer to direct the phone call to the correct MultiVOIP.
When a MultiVOIP’s connection to the Internet is through a ProxyServer , the WAN port on the
ProxyServer must have a static registered IP address. Remote MultiVOIPs will only be able to
access a MultiVOIP located behind a ProxyServer at the static IP address. This static IP address will
be used in the Phone Directory Database when assigning directory numbers to this MultiVOIP.
The Global IP Address field in the Proxy Setup dialog box must contain the static IP address of the
WAN port of the ProxyServer. The Local IP Address field in the Proxy Setup dialog box must contain
the local IP address of the MultiVOIP. In this case the local IP address is not used in the Phone
Directory Database. There must be a unique static IP address on the Wan side of the proxy server for
each MultiVOIP located behind the proxy server .
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Viewing Call Progress
The Call Progress dialog box displays the status of a call in progress. This dialog box is accessed
from the MultiVOIP 200 Setup menu by clicking the Call Progress button.
The ratio of Packets Lost versus Packets Received provides a general indication of the integrity of
the Internet connection. To reduce the frequency of lost packets, select a low-bit-rate coder, such as,
G.723 or Netcoder. In addition, enabling the Forward Error Correction option on the Voice/Fax tab on
the Channel Setup dialog box will enable the MultiVOIP to recover many of the lost packets.
The Jitter (ms) value indicates the mean deviation of the difference in packet spacing at the receiver
compared to the sender for a pair of packets.
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Applications Setup
Clicking the Others button on the Setup menu displays the Applications Setup dialog box. This
dialog box lets you to enable SNMP Agent (the default is
parameters; enable or disable various remote configuration methods such as TFTP (T rivial File
Transfer Protocol) Server, Web Server, Dumb Terminal Management, and Telnet Server; and assign a
Password to the MultiVOIP for Internet security. These applications enable remote viewing and
changing of the MultiVOIP configuration, or updating firmware, from anywhere on the connected
internetwork.
Verify that the desired applications are enabled (checked). The default condition is all applications are
checked. To disable a given application, click to uncheck the check box and disable support.
SNMP related operations can be performed only when the SNMP Agent is enabled (checked) on this
dialog box. The IP address of the system (i.e., SNMP Manager) that will receive the Traps from the
MultiVOIP should be entered in the IP Address field in the Trap Manager group. The CommunityName of the SNMP Manager receiving the Traps can be a maximum of 19 characters and is case
sensitive. The default Port Number of the SNMP Manager receiving the Traps is 162. The MultiVOIP
currently supports a maximum of two community users at a time, and they can be assigned either
Read/Write or Read Only rights.
The Password group enables you to enter a password, up to 13 alphanumeric characters, to be used
for Internet Security. Once the password is entered (in the MultiVOIP Password field) and confirmed
(in the Confirm Password field), remote users will be queried to enter the password before gaining
access to the MultiVOIP.
disabled
Chapter 4 - MultiVOIP Software
) and set up all the necessary
For more information on using these applications, click the on-line Help button or refer to Chapter 5,
Remote Configuration and Management.
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Viewing Statistics
The Statistics dialog box enables you to view statistics for major events of the MultiVOIP operation.
This dialog box is accessed by clicking the Statistics button on the MultiVOIP 200 Setup menu.
Statistics can be a helpful troubleshooting tool. For example, viewing the Voice Channel statistics you
can see the attempted and completed calls, call duration, average call length, bytes/packets sent and
received, etc.
IP Statistics
IP is a connection-less network protocol residing in the network layer of a conventional OSI layered
model (for more information on this model, refer to Appendix A). Depending on what is going on at
the application layer, IP will typically use one of two transport layer protocols. User Datagram
Protocol (UDP), a connection-less transport layer protocol used with TFTP or SNMP; and Transport
Control Protocol (TCP) is a connection-oriented transport layer protocol used with FTP, Telnet, and
SNMP.
UDP makes use of the port concept and has no measures for flow control, reliability, or error
recovery. It is used when the full services of TCP are not required, and the reliability measures must
be assumed by another layer.
TCP works well in environments where the reliability measures are not assumed by other layers. It is
connection-oriented and has a full range of services.
For the most part these statistics are informational, and their use as a troubleshooting tool will be
contingent on the applications running in the upper layers. For example, if you were having problems
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connecting to the MultiVOIP’s web server , you would look under the TCP section to see if any
connections are being established. If not, that may indicate the web server is not enabled. Or, if you
were having problems establishing a remote connection through TFTP, you could look in the UDP
section to see if any packets are being received. If not, you may need to review your network
addressing.
SNMP Statistics
The SNMP Statistics dialog box provides statistical information on Simple Network Management
Protocol (SNMP).
SNMP is an application layer protocol that facilitates the exchange of management information
between network devices. There are three key components in SNMP: the devices that are to be
managed, agents, and the network management systems. The managed device is the network
device, like a router. The agent is the software module residing in the managed device pertaining to
network management. The network management system runs the SNMP application that controls the
managed devices and monitors their status. Four primary operations -- Set, Get, Get Next, and Trap -
- are performed using SNMP.
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Viewing Logs
The Log Entries dialog box displays the a chronological history of all calls into and out of this unit.
This dialog box is accessed from the Logs button on the Statistics dialog box.
The Log Entries dialog box displays each call as a sequentially numbered Event with the date, time,
duration of the call, the status of the call (Successful or Unsuccessful), Mode (Voice or Fax), and the
from and to numbers.
Viewing Log Entry Details
The Log Entry Details dialog box displays the status of a completed call. This dialog box displays
the same details as the Call Progress dialog box after a call is completed.
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Viewing Channel Totals
The Channel Totals dialog box displays Outgoing and Incoming calls with their Attempted and
Completed numbers for each channel on this MultiVOIP. The Total Connected Time for the channel
is also displayed. This can provide you with a sense of successful call completions on each channel
of the unit.
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Reports
A report of the contents of the Log Entries dialog box can be generated using the Windows Notepad
accessory and then printed from your local PC. The report is generated by entering the To and From
dates in the Report Generation dialog box and then clicking the Generate button. This function
provides a hard copy of the Log Entries dialog box.
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Upgrade Procedures
Whenever you upgrade your version of the MultiVOIP software, you must first install the new
software on your PC. Then, download the Firmware, the Factory Defaults, and/or the V oice Coders to
upgrade the MultiVOIP itself.
Before starting the upgrade process, view the current configuration and write down important data
such as your IP address, phone book contents, and voice channel configurations; these settings
must be put back in place after the software has been upgraded.
Two utility programs included in the MultiVOIP software are to be used only after the unit has been
repaired or upgraded. They are
Download Firmware
If you have obtained a new firmware version from the Multi-Tech Web site, the Multi-Tech FTP site,
or another source, do the following:
Win3.1 users - In the Program Manager, double-click the Download Firmware icon in the
MultiVOIP 200 program group. The Open dialog box appears and the file list contains a single file,
mtvoip.bin.
Win95/98/NT users - Click Programs | MultiVOIP 200 | Download Firmware. The Open dialog box
appears and the file list contains a single file, mtvoip.bin.
Download Firmware
Chapter 4 - MultiVOIP Software
and
Download Voice Coders
.
After you select the mtvoip.bin file and click OK, the Downloading Code dialog box appears with a
progress bar so you can monitor the download process.
After the new firmware file is downloaded to the MultiVOIP, the unit reboots and is then ready for use
with upgraded firmware.
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Download Coders
If you have obtained a new coders file, do the following:
Win3.1 users - In the Program Manager, double-click the Download Coders icon in the MultiVOIP
200 program group. The Open dialog box appears and the file list contains a single file, coders.hst.
Win95/98/NT users - Click Programs | MultiVOIP 200 | Download Coders. The Open dialog box
appears and the file list contains a single file, coders.hst.
After you select the coders.hst file and click OK, the Downloading Voice Coder dialog box appears
with a progress bar so you can monitor the download process.
After the file is downloaded to the MultiVOIP, the unit reboots and is then ready for use with upgraded
voice coders.
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Chapter 5 - Remote Configuration and Management
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Introduction
This chapter provides procedures for viewing or changing the configuration of a remote unit. T wo
methods are provided to access a remote unit; the first method is modem based and the second
method is using IP. Within the IP method, three applications can be used: 1) LAN-Based using TFTP
(Trivial lFile Transfer Protocol), 2)Telnet as a client application, or 3) a standard web browser on the
Internet.
Remote Configuration
Remote configuration requires the MultiVOIP software to be loaded on the local PC. The local PC
then controls the remote MultiVOIP either via the modem connection or the LAN.
Modem-Based
To remotely configure a MultiVOIP , a local PC needs to be connected to a dial-up line and the
MultiVOIP software configured to call the remote MultiVOIP. The remote MultiVOIP needs to have a
modem connected to a dial-up line and the Command Port. Once the connection to the remote unit is
made, you can change the configuration as you see fit. Once the configuration is changed, you can
down load the new configuration to the remote MultiVOIP. Refer to the Modem-Based Remote
Configuration Procedure in this chapter to remotely configure a MultiVOIP.
1At the remote site, remove the serial cable from the PC to the Command Port connector on
the back panel of the MultiVOIP.
2At the remote site, connect a special cable (Remote Configuration Cable) from the Command
Port connector on the back panel of the MultiVOIP to the RS232 connector on the modem.
The special cable is a serial cable with male connectors on both ends. Refer to Appendix B
for cable details.
Connect the modem to your local telephone line.
Provide your telephone number to the person verifying your configuration.
Configure the remote modem for 19200 baud and turn on Force DTR.
3At the main site, connect your local PC to a modem that is connected to a dial-up line.
4Install the MultiVOIP software on the local PC. When installed, click Start | Programs |
MultiVOIP 200 | Configuration Port Setup, or double click on the Configuration Port icon
in the MultiVOIP 200 program group.
5The MultiVOIP 200 Setup dialog box is displayed.
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Verify that the Communication Type is set for COM Port and the Select Port field is set for
the COM port of your local PC.
In the Dial String field, enter the AT command for dialing (ATDT) plus the phone number of
the remote MultiVOIP.
If your Modem Initialization String, Initialization Response, or Connect Response values are
different than the defaults in the dialog box, refer to your modem user documentation and
change the default values to match your modem.
Click OK when you are satisfied with your selections.
6Run the MultiVOIP Configuration program. Click Start | Programs | MultiVOIP 200 |
MultiVOIP Configuration, or double click on the MultiVOIP Configuration icon in the
MultiVOIP 200 program group.
7The Dialing dialog box is displayed while software is dialing the remote MultiVOIP .
8The Reading Setup dialog box is displayed.
9The MultiVOIP 200 Setup menu is displayed. This is the dialog box of the remote MultiVOIP.
Refer to the online Help provided with your software for a description of each dialog box and
field within a dialog box.
10After you have changed the configuration of the remote MultiVOIP, click Download Setup to
update the configuration. The remote MultiVOIP will be brought down, the new configuration
written to the unit, and the unit will reboot.
1 1Click Exit when the downloading is complete.
12The Hangup connection? dialog box is displayed
Click Y es to disconnect the phone connection to the remote site.
13If the same telephone number is not going to be used again in the immediate future, you may
want to remove it from the Port Setup dialog box.
14At the remote site, reconnect the MultiVOIP to the serial port of the PC and from the
MultiVOIP 200 program group double click on the MultiVOIP Configuration icon to verify
that the MultiVOIP is running.
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LAN-Based
The LAN-based remote configuration requires a Windows Sockets compliant TCP/IP stack. TCP/IP
protocol software must be installed and functional before the configuration program can be used.
1You must assign an Internet (IP) address for the PC and for each node that will be managed
by the configuration program. Refer to the protocol software documentation for instructions
on how to set the IP addresses.
Once you have completed this step, you should be able to use the protocol Ping command
for the PC host name. You should also test the network interface configuration by Pinging
another TCP/IP device that is connected to the network.
2Install the MultiVOIP software on the local PC. When installed click Start | Programs |
MultiVOIP 200 | Configuration Port Setup, or double-click on the Configuration Port
Setup icon in the MultiVOIP 200 program group.
3The MultiVOIP Port Setup dialog box is displayed.
Verify that the Communication Type field is set IP.
In the MultiVOIP IP Address field, enter the IP Address of the remote MultiVOIP.
4Click OK when you are satisfied with your selections.
5Run the MultiVOIP Configuration program. Click Start | Programs | MultiVOIP 200 |
MultiVOIP Configuration, or double click on the MultiVOIP Configuration icon in the
MultiVOIP 200 program group.
The Reading Setup dialog box is displayed.
6The MultiVOIP 200 Setup dialog box is displayed. This is the dialog box of the remote
MultiVOIP. Refer to the online Help provided with your MultiVOIP for the definition of each
dialog box and field within a dialog box.
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7After you have changed the configuration of the remote MultiVOIP, click Download Setup to
update the configuration. The remote MultiVOIP will be brought down, the new configuration
written to the unit, and the unit will reboot.
8Click Exit when the downloading is complete.
9Double click on the MultiVOIP Configuration icon in the MultiVOIP 200 program group to
verify that the MultiVOIP is running.
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Remote Management
This section describes typical client applications that can be used to configure the MultiVOIP
remotely. It is important to note that although any subsequent changes to configuration can be made
using these applications, the initial setup and configuration of the MultiVOIP must be done on the
local PC, using the MultiVOIP software provided with your unit.
Although establishing access to the MultiVOIP varies between applications, the configuration
functions mirror those of the MultiVOIP software. For more information on MultiVOIP software, refer
to Chapter 4 - MultiVOIP Software.
Telnet
A typical Telnet client application is described next. The MultiVOIP has a built-in Telnet Server that
enables Telnet client PCs to access the MultiVOIP. A typical Telnet client is allowed to configure the
MultiVOIP. In addition, the MultiVOIP can be remotely accessed and configured from any where on
the Internet through its Web interface.
The TCP/IP stack has to be loaded before the Telnet client (a Windows application) will run. The
Telnet Server option has to be selected from the Applications Setup dialog box using the MultiVOIP
Configuration icon. Double click on the Telnet icon (or shortcut) and a blank Telnet screen is
displayed. Click Connect | Remote System and the Connect dialog box is displayed. Select (or
enter) a Host Name (the IP address of the MultiVOIP).In this example, the Host Name is
192.168.2.8.
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When you enter a valid Host Name (IP address) and click on the Connect button, you are
immediately connected to the target MultiVOIP and the MultiVOIP Telnet Server screen is displayed.
MultiVOIP Telnet Server Menu
The MultiVOIP Telnet Server menu provides three basic options: Voice over IP Configuration, Phone
Directory Database, and Phone Directory Configuration. A further option enables you to close the
Telnet session.
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Chapter 5 - Remote Configuration and Management
Voice over IP Configuration
Selecting Option 1 displays the main menu, which allows further configuration options. These
options include Protocol Stacks (option 1), Applications (option 2), System Information (option 3), and
Voice Channels (option 4). For further descriptions of these options, refer to Chapter 4 - MultiVOIP
Software.
Phone Directory Database
Selecting Option 2 allows you to add entries to the Phone Directory Database. Refer to Chapter 4 -
MultiVOIP Software, for more details on the database.
Phone Directory Configuration
Selecting Option 3 allows you to configure and manage the Phone Directory. The various options
are described in detail in Chapter 4 - MultiVOIP Software.
WEB Management
The MultiVOIP can be accessed, via a standard Web browser, from anywhere on the connected
Internet. In order to provide this support, the WEB Server option has to be enabled from the Others
button in the main menu which displays the Applications Setup dialog box (see Chapter 4 -
MultiVOIP Software).
Once enabled, users can access the MultiVOIP by entering its IP address in the Address field of their
web browser. The following screen appears.
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If a Password was entered in the Applications Setup dialog box, then enter the password and click on
the Enter button.
From this screen you can access all the configuration options. Refer to Chapter 4 - MultiVOIP
Software, for a description of the various options.
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Voice / Fax over IP Networks
Chapter 6 - Warranty, Service and Tech Support
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Introduction
This chapter starts out with statements about your MultiVOIP 2-year warranty. The next section, Tech
Support, should be read carefully if you have questions or problems with your MultiVOIP. It includes
the technical support phone numbers, space for recording your product information, and an
explanation of how to send in your MultiVOIP should you require service. The final two sections
explain how to use our bulletin board service (BBS), and get support the Internet.
Limited Warranty
Multi-Tech Systems, Inc. (“MTS”) warrants that its products will be free from defects in material or
workmanship for a period of two years from the date of purchase, or if proof of purchase is not
provided, two years from date of shipment. MTS MAKES NO OTHER WARRANTY, EXPRESSED
OR IMPLIED, AND ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A
PARTICULAR PURPOSE ARE HEREBY DISCLAIMED. This warranty does not apply to any
products which have been damaged by lightning storms, water, or power surges or which have been
neglected, altered, abused, used for a purpose other than the one for which they were manufactured,
repaired by the customer or any party without MTS’s written authorization, or used in any manner
inconsistent with MTS’s instructions.
MTS’s entire obligation under this warranty shall be limited (at MTS’s option) to repair or replacement
of any products which prove to be defective within the warranty period, or, at MTS’ s option, issuance
of a refund of the purchase price. Defective products must be returned by Customer to MTS’s factory
transportation prepaid.
MTS WILL NOT BE LIABLE FOR CONSEQUENTIAL DAMAGES AND UNDER NO
CIRCUMSTANCES WILL ITS LIABILITY EXCEED THE PURCHASE PRICE FOR DEFECTIVE
PRODUCTS.
On-line Warranty Registration
To register your MultiVOIP on-line, click the following link:
http://www.multitech.com/register/
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Service
If your tech support specialist decides that service is required, your MultiVOIP may be sent (freight
prepaid) to our factory . Return shipping charges will be paid by Multi-Tech Systems.
Include the following with your MultiVOIP:
•a description of the problem.
•return billing and return shipping addresses.
•contact name and phone number.
•check or purchase order number for payment if the MultiVOIP is out of warranty. (Check with
•if possible, note the name of the technical support specialist with whom you spoke.
If you need to inquire about the status of the returned product, be prepared to provide the serial
number of the product sent.
Send your MultiVOIP to this address:
Chapter 6 - Warranty, Service and Tech Support
your technical support specialist for the standard repair charge for your MultiVOIP).
MULTI-TECH SYSTEMS, INC.
2205 WOODALE DRIVE
MOUNDS VIEW, MINNESOTA 55112
ATTN: SERVICE OR REPAIRS
You should also check with the supplier of your MultiVOIP on the availability of local service and/or
loaner units in your part of the country .
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Tech Support
Multi-Tech has an excellent staff of technical support personnel available to help you get the most out
of your Multi-Tech product. If you have any questions about the operation of this unit, call 1-800-972-
2439. Please fill out the MultiVOIP information (below), and have it available when you call. If your
MultiVOIP requires service, the tech support specialist will guide you on how to send in your
MultiVOIP (refer to the next section).
Recording MultiVOIP Information
Please fill in the following information on your Multi-Tech MultiVOIP . This will help tech support in
answering your questions. (The same information is requested on the Warranty Registration Card.)
Model No.: _________________________
Serial No.: _________________________
Software Version: ____________________
The model and serial numbers are on the bottom of your MultiVOIP.
Please note status of your MultiVOIP including LED indicators, screen messages, diagnostic test
results, problems with a specific application, etc. Use the space below to note the MultiVOIP status:
________________________________________________________________________________________________________
________________________________________________________________________________________________________
________________________________________________________________________________________________________
________________________________________________________________________________________________________
______________________________________________________________________________________________________________
______________________________________________________________________________________________________
______________________________________________________________________________________________________________
______________________________________________________________________________________________________
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The Multi-Tech BBS
For customers who do not have Internet access, Multi-Tech maintains a bulletin board system (BBS).
Information available from the BBS includes new product information, product upgrade files, and
problem-solving tips. The phone number for the Multi-Tech BBS is (800) 392-2432 (USA and
Canada) or (612) 785-3702 (international and local).
The BBS can be accessed by any asynchronous modem operating at 1200 bps to 56K bps at a
setting of 8 bits, no parity , and 1 stop bit (8-N-1).
To Log on to the Multi-Tech BBS
1.Set your communications program to 8-N-1.
2.Dial our BBS at (800) 392-2432 (USA and Canada) or (612) 785-3702 (international and
local).
3.At the prompts, type your first name, last name, and password; then press ENTER. If you are
a first time caller, the BBS asks if your name is spelled correctly. If you answer yes, a
questionnaire appears. You must complete the questionnaire to use the BBS on your first
call.
4.Press ENTER until the Main Menu appears. From the Main Menu you have access to two
areas: the Files Menu and News. For help on menu commands, type ?.
Chapter 6 - Warranty, Service and Tech Support
To Download a File
If you know the file name
1.From the Main Menu, type F to access the Files Menu, then type D.
2.Enter the name of the file you wish to download from the BBS.
3.If a password is required, enter the password.
4.Answer Y or N to the automatic logoff question.
5.Select a file transfer protocol by typing the indicated letter, such as Z for Zmodem (the
recommended protocol).
6.If you select Zmodem, the transfer will begin automatically . If you select another protocol, you
may have to initiate the transfer yourself. (In most data communications programs, the P AGE
DOWN key initiates the download.)
7.When the download is complete, press ENTER to return to the File Menu.
8.To exit the BBS, type G and press ENTER.
If you don’t know the file name
1.From the Main Menu, type F to access the Files Menu. For a list of file areas, type L, press
ENTER, then type L and press ENTER again. (If you do not type the second L, you will list all
of the files on the BBS.)
2.Mark each file area you would like to examine by typing its list number and pressing ENTER.
3.Enter L to list all the files in the selected file areas. Enter C to go forward in the file list and P
to go back.
4.To mark one or more files for download, type M, press ENTER, type the list numbers of the
files, and press ENTER again.
5.Enter D. You will see a list of the files you have marked. Enter E if you would like to edit the
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list; otherwise enter D again to start the download process.
6.Select a file transfer protocol by typing the indicated letter, such as Z for Zmodem (the
recommended protocol).
7.If you select Zmodem, the file will transfer automatically . If you select another protocol, you
may have to initiate the transfer yourself. (In most data communications programs, the P AGE
DOWN key initiates the download.)
About the Internet
If you prefer to receive technical support via the Internet, you can contact Tech Support via e-mail at
the following address:
Appendix A - TCP/IP (Transmission Control Protocol/Internet Protocol) Description
TCP/IP is a protocol suite and related applications developed for the U.S. Department of Defense in
the 1970s and 1980s specifically to permit different types of computers to communicate and
exchange information with one another. TCP/IP is currently mandated as an of ficial U.S. Department
of Defense protocol and is also widely used in the UNIX community .
Before you install TCP/IP on your network, you need to establish your Internet addressing strategy.
First, choose a domain name for your company . A domain name is the unique Internet name, usually
the name of your business, that identifies your company. For example, Multi-Tech’s domain name is
multitech.com ( .com indicates this is a commercial organization; .edu denotes educational
organizations, .gov denotes government organizations). Next, determine how many IP addresses
you’ll need. This depends on how many individual network segments you have, and how many
systems on each segment need to be connected to the Internet. You’ll need an IP address for each
network interface on each computer and hardware device.
IP addresses are 32 bits long and come in two types: network and host. Network addresses come in
five classes: A, B, C, D, and E. Each class of network address is allocated a certain number of host
addresses. For example, a class B network can have a maximum of 65,534 hosts, while a class C
network can have only 254. The class A and B addresses have been exhausted, and the class D and
E addresses are reserved for special use. Consequently , companies now seeking an Internet
connection are limited to class C addresses.
Early IP implementations ran on hosts commonly interconnected by Ethernet local area networks
(LAN). Every transmission on the LAN contains the local network, or medium access control (MAC),
address of the source and destination nodes. The MAC address is 48-bits in length and is nonhierarchical; MAC addresses are never the same as IP addresses.
When a host needs to send a datagram to another host on the same network, the sending application
must know both the IP and MAC addresses of the intended receiver . Unfortunately, the IP process
may not know the MAC address of the receiver. The Address Resolution Protocol (ARP), described
in RFC 826 (located at ftp://ds.internic.net/rfc/rfc826.txt) provides a mechanism for a host to
determine a receiver’s MAC address from the IP address. In the process, the host sends an ARP
packet in a frame containing the MAC broadcast address; and then the ARP request advertises the
destination IP address and asks for the associated MAC address. The station on the LAN that
recognizes its own IP address will send an ARP response with its own MAC address. An ARP
message is carried directly in an IP datagram.
Other address resolution procedures have also been defined, including those which allow a diskless
processor to determine its IP address from its MAC address (Reverse ARP, or RARP), provides a
mapping between an IP address and a frame relay virtual circuit identifier (Inverse ARP, or InARP),
and provides a mapping between an IP address and ATM virtual path/channel identifiers (ATMARP).
The TCP/IP protocol suite comprises two protocols that correspond roughly to the OSI Transport and
Session Layers; these protocols are called the Transmission Control Protocol and the User Datagram
Protocol (UDP). Individual applications are referred to by a port identifier in TCP/UDP messages. The
port identifier and IP address together form a “socket”. Well-known port numbers on the server side
of a connection include 20 (FTP data transfer), 21 (FTP control), 23 (Telnet), 25 (SMTP), 43 (whois),
70 (Gopher), 79 (finger), and 80 (HTTP).
TCP, described in RFC 793 ( ftp://ds.internic.net/rfc/rfc793.txt) provides a virtual circuit (connectionoriented) communication service across the network. TCP includes rules for formatting messages,
establishing and terminating virtual circuits, sequencing, flow control, and error correction. Most of
the applications in the TCP/IP suite operate over the “reliable” transport service provided by TCP.
UDP, described in RFC 768 (ftp://ds.internic.net/rfc/rfc768.txt) provides an end-to-end datagram
(connectionless) service. Some applications, such as those that involve a simple query and
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response, are better suited to the datagram service of UDP because there is no time lost to virtual
circuit establishment and termination. UDP’s primary function is to add a port number to the IP
address to provide a socket for the application.
The Application Layer protocols are examples of common TCP/IP applications and utilities, which
include:
•Telnet (Telecommunication Network): a virtual terminal protocol allowing a user logged on to
one TCP/IP host to access other hosts on the network, described in RFC 854 ( ftp://
ds.internic.net/rfc/rfc854.txt).
•FTP: the File Transfer Protocol allows a user to transfer files between local and remote host
computers per IETF RFC 959 ( ftp://ds.internic.net/rfc/rfc959.txt).
•Archie: a utility that allows a user to search all registered anonymous FTP sites for files on a
specified topic.
•Gopher: a tool that allows users to search through data repositories using a menu-driven,
hierarchical interface, with links to other sites, per RFC 1436 ( ftp://ds.internic.net/rfc/
rfc1436.txt).
•SMTP: the Simple Mail Transfer Protocol is the standard protocol for the exchange of
electronic mail over the Internet, per IETF RFC 821 ( ftp://ds.internic.net/rfc/rfc821.txt).
•HTTP: the Hypertext Transfer Protocol is the basis for exchange of information over the
World Wide Web (WWW). Various versions of HTTP are in use over the Internet, with HTTP
version 1.0 (per RFC 1945) ( ftp://ds.internic.net/rfc/rfc1945.txt) being the most current.
•HTML: WWW pages are written in the Hypertext Markup Language (HTML), an ASCII-based,
platform-independent formatting language, per IETF RFC 1866 ( ftp://ds.internic.net/rfc/
rfc1866.txt).
•Finger: used to determine the status of other hosts and/or users, per IETF RFC 1288 ( ftp://
ds.internic.net/rfc/rfc1288.txt).
•POP: the Post Office Protocol defines a simple interface between a user’s mail reader
software and an electronic mail server; the current version is POP3, described in IETF RFC
1460 ( ftp://ds.internic.net/rfc/rfc1460.txt).
•DNS: the Domain Name System defines the structure of Internet names and their association
with IP addresses, as well as the association of mail, name, and other servers with domains.
•SNMP: the Simple Network Management Protocol defines procedures and management
information databases for managing TCP/IP-based network devices. SNMP, defined by RFC
1 157 ( ftp://ds.internic.net/rfc/rfc1157.txt) is widely deployed in local and wide area network.
SNMP V ersion 2 (SNMPv2), per RFC 1441< ftp://ds.internic.net/rfc/rfc1441.txt) adds security
mechanisms that are missing in SNMP, but is also more complex.
•Ping: a utility that allows a user at one system to determine the status of other hosts and the
latency in getting a message to that host. Ping uses ICMP Echo messages.
•Whois/NICNAME: Utilities that search databases for information about Internet domain and
domain contact information, per RFC 954 ( ftp://ds.internic.net/rfc/rfc954.txt).
•Traceroute: a tool that displays the route that packets will take when traveling to a remote
host.
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Internet Protocol (IP)
IP is the Internet standard protocol that tracks Internetwork node addresses, routes outgoing
messages and recognizes incoming messages, allowing a message to cross multiple networks on
the way to its final destination. The IPv6 Control Protocol (IPV6CP) is responsible for configuring,
enabling, and disabling the IPv6 protocol modules on both ends of the point-to-point link. IPV6CP
uses the same packet exchange mechanism as the Link Control Protocol (LCP). IPV6CP packets are
not exchanged until PPP has reached the Network-Layer Protocol phase. IPV6CP packets received
before this phase is reached are silently discarded. (See also TCP/IP.)
Before you install TCP/IP on your network, you need to establish your Internet addressing strategy.
You first choose a domain name for your company. A domain name is the unique Internet name,
usually the name of your business, that identifies your company . For example, Multi-Tech’s domain
name is multitech.com (where .com indicates this is a commercial organization; .edu denotes
educational organizations, .gov denotes government organizations). Next, you determine how many
IP addresses you’ll need. This depends on how many individual network segments you have, and
how many systems on each segment need to be connected to the Internet. You need an IP address
for each network interface on each computer and hardware device.
IP addresses are 32 bits long and come in two types: network and host. Network addresses come in
five classes: A, B, C, D, and E. Each class of network address is allocated a certain number of host
addresses. For example, a class B network can have a maximum of 65,534 hosts, while a class C
network can have only 254. The class A and B addresses have been exhausted, and the class D and
E addresses are reserved for special use. Consequently , companies now seeking an Internet
connection are limited to class C addresses. The current demand for Internet connections will
exhaust the current stock of 32-bit IP addresses. In response, Internet architects have proposed the
next generation of IP addresses, Ipng (IP Next Generation). It features 16-byte addressing,
surpassing the capacities of 32-bit IP.
An IP address can serve only a single physical network. Therefore, if your organization has multiple
physical networks, you must make them appear as one to external users. This is done via
“subnetting”, a complex procedure best left to ISPs and others experienced in IP addressing. Since
IP addresses and domain names have no inherent connection, they are mapped together in
databases stored on Domain Name Servers (DNS). If you decide to let an Internet Service Provider
(ISP) administer your DNS server, the ISP can assist you with the domain name and IP address
assignment necessary to configure your company’s site-specific system information. Domain names
and IP addresses are granted by the InterNIC. To check the availability of a specific name or to obtain
more information, call the InterNIC at (703)742-4777.
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Appendix B - Cabling Diagrams
Command Port Cable
1 2 3 4 5 6 7 8
Appendix B - Cabling Diagrams
PIN NO.
1
2
To Command
Port Connector
3
4
5
6
7
8
LAN Cable
1 2 3 4 5 6 7 8
PinCircuit Signal Name
1TD+ Data Transmit Positive
2TD- Data Transmit Negative
3RD+ Data Receive Positive
6RD- Data Receive Negative
NOTE: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant
to Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful
interference when the equipment is operated in a commercial environment. This equipment generates, uses and
can radiate radio frequency energy and, if not installed and used in accordance with the instruction manual, may
cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to
cause harmful interference, in which case the user will be required to correct the interference at his own expense.
This device complies with Part 15 of the FCC rules.
Operation is subject to the following two conditions:
(1)This device may not cause harmful interference.
(2)This device must accept any interference that may cause undesired operation.
Warning: Changes or modifications to this unit not expressly approved by the party responsible for compliance
could void the user’s authority to operate the equipment.
Industry Canada
This Class A digital apparatus meets all requirements of the Canadian Interference-Causing Equipment
Regulations.
Cet appareil numerique de la classe A respecte toutes les exigences du Reglement sur le materiel brouilleur du
Canada.
Fax Branding Statement
The Telephone Consumer Protection Act of 1991 makes it unlawful for any person to use a computer or other electronic
device, including fax machines, to send any message unless such message clearly contains the following information:
•Date and time the message is sent
•Identification of the business or other entity , or other individual sending the message
•Phone number of the sending machine or such business, other entity , or individual
This information is to appear in a margin at the top or bottom of each transmitted page or on the first page of the transmission.
(Adding this information in the margin is referred to as
Since any number of Fax software packages can be used with this product, the user must refer to the Fax software manual for
setup details. Typically, the Fax branding information must be entered via the configuration menu of the software.
fax branding
.)
FCC Part 68 Telecom
1. This equipment complies with Part 68 of the Federal Communications Commission (FCC) rules. On the outside
surface of this equipment is a label that contains, among other information, the FCC registration number. This
information must be provided to the telephone company .
2. As indicated below, the suitable jack (Universal Service Order Code connecting arrangement) for this
equipment is shown. If applicable, the facility interface codes (FIC) and service order codes (SOC) are shown.
3. An FCC-compliant telephone cord with modular plug is provided with this equipment. This equipment is
designed to be connected to the phone network or premises wiring using a compatible modular jack which is Part
68 compliant. See installation instructions for details.
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4. If this equipment causes harm to the phone network, the phone company will notify you in advance that
temporary discontinuance of service may be required. But if advance notice is not practical, the phone company
will notify the customer as soon as possible. Also, you will be advised of your right to file a complaint with the
FCC if you believe it is necessary.
5. The phone company may make changes in its facilities, equipment, operations, or procedures that could affect
the operation of the equipment. If this happens, the phone company will provide advance notice in order for you
to make necessary modifications in order to maintain uninterrupted service.
6. If trouble is experienced with this equipment (the model of which is indicated below) please contact Multi-Tech
Systems, Inc., at the address shown below for details of how to have repairs made. If the equipment is causing
harm to the network, the phone company may request that you remove the equipment from the network until the
problem is resolved.
7. No repairs are to be made by you. Repairs are to be made only by Multi-Tech Systems or its licensees.
Unauthorized repairs void registration and warranty .
8.Manufacturer:Multi-Tech Systems, Inc.
Trade name:MultiVOIP
Model Numbers:MVP200
FCC Registration Number:AU7USA-25715-DF-N
Modular Jack (USOC):RJ-11C or RJ-11W
Service Center in U.S.A.:Multi-Tech Systems Inc.
Notice: The ringer equivalence number (REN) assigned to each terminal device provides an indication of the maximum
number of terminals allowed to be connected to a phone interface. The termination on an interface may consist of any
combination of devices subject only to the requirement that the sum of the ringer equivalence numbers of all the devices does
not exceed 5.
Notice: The Industry Canada label identifies certified equipment. This certification means that the equipment meets certain
telecommunications network protective, operational and safety requirements. The Department does not guarantee the
equipment will operate to the user’s satisfaction.
Before installing this equipment, users should ensure that it is permissible to be connected to the facilities of the local telecommunications company. The equipment must also be installed using an acceptable method of connection. The customer
should be aware that compliance with the above conditions may not prevent degradation of service in some situations.
Repairs to certified equipment should be made by an authorized Canadian maintenance facility designated by the supplier.
Any repairs or alterations made by the user to this equipment, or equipment malfunctions, may give the telecommunications
company cause to request the user to disconnect the equipment.
Users should ensure for their own protection that the electrical ground connections of the power utility, phone lines and
internal metallic water pipe system, if present, are connected together. This precaution may be particularly important in rural
areas.
Caution: Users should not attempt to make such connections themselves, but should contact the appropriate electric inspection authority, or electrician, as appropriate.
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Appendix C - Regulatory Information
EMC, Safety and Terminal Directive Compliance
The CE mark is affixed to this product to confirm compliance with the following European Community Directives:
Council Directive 89/336/EEC of 3 May 1989 on the approximation of the laws of Member States relating to
electromagnetic compatibility .
and
Council Directive 73/23/EEC of 19 February 1973 on the harmonization of the laws of Member States relating to
electrical equipment designed for use within certain voltage limits:
and
Council Directive 98/13/EC of 12 March 1998 on the approximation of the laws of Member States concerning
telecommunications terminal and Satellite earth station equipment.
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Glossary
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A
Access: The T1 line element made up of two pairs of wire that the phone company brings to the customer premises. The Access portion
ends with a connection at the local telco (LEC or RBOC).
Accunet Spectrum of Digital Services (ASDS): The AT&T 56K bps leased (private) line service. Similar to services of MCI and Sprint.
ASDS is available in nx56/64K bps, where n=1, 2, 4, 6, 8, 12.
ACK (ACKnowledgement code) (pronounced "ack"): A communications code sent from a receiving modem to a transmitting modem to
indicate that it is ready to accept data. It is also used to acknowledge the error-free receipt of transmitted data. Contrast with NAK.
Adaptive Differential Pulse Code (ADCPM): In multimedia applications, a technique in which pulse code modulation samples are com-
pressed before they are stored on a disk. ADCPM, an extension of the PCM format, is a standard encoding format for storing audio information in a digital format. It reduced storage requirements by storing differences between successive digital samples rather than full values.
Address: A numbered location inside a computer. It's how the computer accesses its resources, like a video card, serial ports, memory, etc.
AMI line coding: One of two common methods of T1 line coding (with B8ZS). AMI line coding places restrictions on user data (B8ZS does
not).
Analog signal: A waveform which has amplitude, frequency and phase, and which takes on a range of values between its maximum and
minimum points.
Analog Transmission: One of two types of telecommunications which uses an analog signal as a carrier of voice, data, video, etc. An
analog signal becomes a carrier when it is modulated by altering its phase, amplitude and frequency to correspond with the source signal.
Compare with digital transmission.
Application Program Interface (API): A software module created to allow dissimilar, or incompatible applications programs to transfer
information over a communications link. APIs may be simple or complex; they are commonly required to link PC applications with mainframe
programs.
ASCII (American Standard Code for Information Interchange) (pronounced "askey"): A binary code for data that is used in communications and in many computers and terminals. The code is used to represent numbers, letters, punctuation and control characters. The basic
ASCII code is a 7-bit character set which defines 128 possible characters. The extended ASCII file provides 255 characters.
Asynchronous Transfer Mode (ATM): A very high-spped method of transmission that uses fixed-size cells of 53 bytes to transfer information over fiber; also known as cell relay.
AT Commands: A standard set of commands used to configure various modem parameters, establish connections and disconnect. The "AT"
is used to get the "attention" of the modem before the actual command is issued.
Availability: The measure of the time during which a circuit is ready for use; the complement of circuit "outage" (100% minus % outage =
% available).
B
B7ZS (Bipolar 7 Zero Suppression) line coding: One method of T1 line coding (see also "B8ZS" and "AMI"). B7ZS line coding does not
place restrictions on user data (AMI does).
B8ZS (Bipolar 8 Zero Suppression) line coding: One of two common methods of T1 line coding (with AMI). B8ZS line coding does not
place restrictions on user data (AMI does). A coding method used to produce 64K bps "clear" transmission. (See also "B7ZS" and "AMI" line
coding)
Backbone: 1. A set of nodes and their interconnecting links providing the primary data path across a network. 2. In a local area network
multiple-bridge ring configuration, a high-speed link to which the rings are connected by means of bridges. A backbone may be configured as
a bus or as a ring. 3. In a wide area network, a high-speed link to which nodes or data switching exchanges (DSEs) are connected. 4. A
common distibution core that provides all electrical power, gases, chemicals, and other services to the sectors of an automated wager
processing system.
Background: An activity that takes place in the PC while you are running another application. In other words, the active user interface does
not correspond to the 'background' task.
Bandwidth: The transmission capacity of a computer channel, communications line or bus. It is expressed in cycles per second (hertz), the
bandwidth being the difference between the lowest and highest frequencies transmitted. The range of usable frequencies that a transmission
medium will pass without unacceptable attenuation or distortion. Bandwidth is a factor in determining the amount of information and the
speed at which a medium can transmit data or other information.
Backward Explicit Congestion Notification (BECN): A bit that tells you that a certain frame on a particular logical connection has
encountered heavy traffic. The bit provides notification that congestion-avoidance procedures should be initiated in the opposite direction of
the received frame. See also FECN (Forward Explicit Congestion Notification).
Basic Rate Interface (BRI): An ISDN access interface type comprised of two B-channels each at 64K bps and one D-channel at 64K bps
(2B+D).
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Glossary
Bell Operating Companies (BOC): The family of corporations created during the divestiture of AT&T. BOCs are independent companies
which service a specific region of the US. Also called Regional Bell Operating Companies (RBOCs).
Bell Pub 41450: The Bell publication defining requirements for data format conversion, line conditioning, and termination for direct DDS
connection.
Bell Pub 62310: The Bell publication defining requirements for data format conversion, line conditioning, and termination for direct DDS
connection.
Binary Synchronous Communication (BSC): A form of telecommunication line control that uses a standard set of transmission control
characters and control character sequences, for binary synchronous transmission of binary-coded data between stations.
Bit (Binary digIT): A bit is the basis of the binary number system. It can take the value of 1 or 0. Bits are generally recognized as the
electrical charge generated or stored by a computer that represent some portion of usable information.
Bit Error Rate Test (BERT): A device or routine that measures the quality of data transmission. A known bit pattern is transmitted, and the
errors received are counted and a BER (bit error rate) is calculated. The BER is the ratio of received bits in error relative to the total number
of bits received, expressed in a power of 10.
Bit robbing: The use of the least significant bit per channel in every sixth frame for signaling. The line signal bits "robbed" from the speech
pat conveys sufficient pre-ISDN telephony signaling information with the remaining line signal bits providing sufficient line signaling bits for
recreating the original sound. See "robbed bit signaling".
Blue Alarm: An error indication signal consisting of all 1s indicating disconnection or attached device failure. Contrast "Red Alarm" and
"Yellow Alarm".
Bps (bits per second): A unit to measure the speed at which data bits can be transmitted or received. Bps differs from baud when more
than one bit is represented by a single cycle of the carrier.
Bridges: 1. A functional unit that interconnects two local area networks that use the same logical link protocol but may use different medium
access control protocols. 2. A functional unit that interconnects multiple LANs (locally or remotely) that use the same logical link control
protocol but that can use different medium access control protocols. A bridge forwards a frame to another bridge based on the medium
access control (MAC) address. 3. In the connection of local loops, channels, or rings, the equipment and techniques used to match circuits
and to facilitate accurate data transmission.
Buffer: A temporary storage register or Random Access Memory (RAM) used in all aspects of data communications which prevents data
from being lost due to differences in transmission speed. Keyboards, serial ports, muxes and printers are a few examples of the devices that
contain buffers.
Bus: A common channel between hardware devices either internally between components in a computer, or externally between stations in a
communications network.
Byte: The unit of information a computer can handle at one time. The most common understanding is that a byte consists of 8 binary digits
(bits), because that's what computers can handle. A byte holds the equivalent of a single character (such as the letter A).
C
Call Setup Time: The time to establish a circuit-switched call between two points. Includes dialing, wait time, and CO/long distance service
movement time.
Carrier Group Alarm (CGA): A T1 service alarm generated by a channel bank when an OOF condition occurs for a predefined length of
time (usually 300mS to 2.5 seconds). The CGA causes the calls using a trunk to be dropped and for trunk conditioning to be applied.
Carrier signal: An analog signal with known frequency, amplitude and phase characteristics used as a transport facility for useful informa-
tion. By knowing the original characteristics, a receiver can interpret any changes as modulations, and thereby recover the information.
CCITT (Consultative Committee for International Telephone and Telegraph): An advisory committee created and controlled by the
United Nations and headquartered in Geneva whose purpose is to develop and to publish recommendations for worldwide standardization of
telecommunications devices. CCITT has developed modem standards that are adapted primarily by PTT (post, telephone and telegraph)
organizations that operate telephone networks of countries outside of the U.S. See also ITU.
Central Office (CO): The lowest, or most basic level of switching in the PSTN (public switched telephone network). A business PABX or any
residential phone connects to the PSTN at a central office.
Centrex: A multi-line service offered by operating telcos which provides, from the telco CO, functions and features comparable to those of a
PBX for large business users. See also "Private Branch Exchange", "Exchange".
Channel: A data communications path between two computer devices. Can refer to a physical medium (e.g., UTP or coax), or to a specific
carrier frequency.
Channel bank: A device that acts as a converter, taking the digital signal from the T1 line into a phone system and converting it to the
analog signals used by the phone system. A channel bank acts as a multiplexer, placing many slow-speed voice or data transactions on a
single high-speed link.
Circuit-switched Network: A technology used by the PSTN that allocates a pair of conductors for the exclusive use of one communication
path. Circuit switching allows multiple conversations on one talk path only if the end-users multiplex the signals prior to transmission.
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Circuit switching: The temporary connection of two or more communications channels using a fixed, non-shareable path through the
network. Users have full use of the circuit until the connection is terminated.
Clear Channel: A transmission path where the full bandwidth is used (i.e., no bandwidth needed for signaling, carrier framing or control
bits). A 64K bps digital circuit usually has 8K bps used for signaling. ISDN has two 64K bps circuits, and a 16K bps packet service of which
part is used for signaling on the 64K channels.
Client-Server: In TCP/IP, the model of interaction in distributed data processing in which a program at one site sends a request to a program
at another site and awaits a response. The requesting program is called a client; the answering program is called a server.
Cluster Controller: A device that can control the input/output operations of more than one device connected to it. A cluster controller may be
controlled by a program stored and executed in the unit, or it may be entirely controlled by hardware.
Committed Burst Size: the maximum number of bits that the frame relay network agrees to transfer during any measurement interval
Committed Information Rate (CIR): An agreement a customer makes to use a certain minimum data transmission rate (in bps). The CIR is
part of the frame relay service monthly billing, along with actual usage, that users pay to their frame relay service provider.
Compression: 1. The process of eliminating gaps, empty fields, redundancies, and unnecessary data to shorten the length of records or
blocks. 2. In SNA, the replacement of a string of up to 64-repeated characters by an encoded control byte to reduce the length of the data
stream to the LU-LU session partner. The encoded control byte is followed by the character that was repeated (unless that character is the
prime compression character). 3. In Data Facility Hierarchical Storage Manager, the process of moving data instead of allocated space during
migration and recall in order to release unused space. 4. Contrast with decompression.
COMx Port: A serial communications port on a PC.
congestion: A network condition where there is too much data traffic. The ITU I.233 standard defines congestion management in terms of
speed and burstiness.
congestion notification: The function in frame relay that ensures that userdata transmitted at a rate higher than the CIR are allowed to
slow down to the rate of the available network bandwidth.
Consecutive Severely Errored Seconds (CSES): An error condition that occurs when from 3 to 9 SES (Severely Errored Seconds) are
logged consecutively.
Customer Premise Equipment (CPE): The generic term for data comm and/or terminal equipment that resides at the user site and is
owned by the user with the following exclusions: Over voltage protection equipment, inside wiring, coin operated or pay telephones, "company-official" equipment, mobile phone equipment, "911" equipment, equipment necessary for the provision of communications for national
defense, or multiplexing equipment used to deliver multiple channels to the customer.
D
D4: the T1 4th generation channel bank.
D4 channelization: Refers to the compliance with AT&T TR 62411 for DS1 frame layout.
D4 framing: The T1 format for framing in AT&T D-Series channel banks, in which there are 12 separate 193-bit frames in a super-frame. A
D4 framing bit is used to identify the channel and the signaling frame. Signalling for voice channels is carried in-band for every channel, along
with the encoded voice. See "robbed-bit signaling".
Data Communications Equipment (DCE): Any device which serves as the portal of entry from the user equipment to a telecommunications
facility. A modem is a DCE for the phone network (PSTN) that is commonly on site at the user’s premises. Packet Switched Networks have
another level of DCE which is most often located at a central office.
Data Link Connection Identifier (DLCI): One of the six components of a frame relay frame. Its purpose is to distinguish separate virtual
circuits across each access connection. Data coming into a frame relay node is thus allowed to be sent across the interface to the specified
"address". The DLCI is confirmed and relayed to its destination, or if the specification is in error, the frame is discarded.
Dataphone Digital Service (DDS): A private line digital service that offers 2400, 4800, 9600 and 56K bps data rates on an inter-LATA basis
by AT&T and on an intra-LATA basis by the BOCs.
Data Service Unit (DSU): A device that provides a digital data service interface directly to the data terminal equipment. The DSU provides
loop equalization, remote and local testing capabilities, and a standard EIA/CCITT interface.
Dedicated Line: A communication line that is not switched. The term leased line is more common.
Default: This is a preset value or option in software packages, or in hardware configuration, that is used unless you specify otherwise.
Device driver: Software that controls how a computer communicates with a device, such as a printer or mouse.
Digital Cross-connect System (DCS): The CO device which splits and redistributes the T1 bandwidth. the DCS takes time slots from
various T1 lines and alters them to provide the needed connectivity. DCS connections are made with software at an administrator's workstation.
Digital Data: Information represented by discrete values or conditions (contrast "Analog Data").
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Glossary
Digital Loopback: A technique used for testing the circuitry of a communications device. Can be initiated locally, or remotely (via a telecom-
munications device). The tested device decodes and encodes a received test message, then echoes the message back. The results are
compared with the original message to determine if corruption occurred en route.
Digital PBX: A Private Branch Exchange that operates internally on digital signals. See also "Exchange".
Digital Service, level 0 (DS0): The world-wide standard speed (64K bps) for digital voice conversation using PCM (pulse coded modula-
tion).
Digital Service, level 1 (DS1): The 1.544M bps voice standard (derived from an older Bell System standard) for digitized voice transmission
in North America. The 1.544M bps consists of 24 digitally-encoded 64K bps voice channels (north America) and 2.048M bps (30 channels)
elsewhere.
Digital Signal: A discrete or discontinuous signal (e.g., a sequence of voltage pulses). Digital devices, such as terminals and computers,
transmit data as a series of electrical pulses which have discrete jumps rather than gradual changes.
Digital Signaling Rates (DSn): A hierarchical system for transmission rates, where "DS0" is 64K bps (equivalent to ISDN B channel), and
DS1 is 1.5 Mbps (equivalent to ISDN PRI).
Digital Transmission: A method of electronic information transmission common between computers and other digital devices. Analog
signals are waveforms: a combination of many possible voltages. A computer's digital signal may be only "high" or "low" at any given time.
Therefore, digital signals may be "cleaned up" (noise and distortion removed) and amplified during transmission.
Digitize: To convert an analog signal to a digital signal.
DIP switch (pronounced "dip switch"): A set of tiny toggle switches, built into a DIP (dual in-line package), used for setting configurable
parameters on a PCB (printed circuit board).
Driver: A software module that interfaces between the Operating System and a specific hardware device (i.e. color monitors, printers, hard
disks, etc.). Also known as a device driver.
Drop and Insert: The process where a portion of information carried in a transmission system is demodulated ("Dropped") at an intermedi-
ate point and different information is included ("Inserted") for subsequent transmission.
DTE (Data Terminating Equipment): A term used to include any device in a network which generates, stores or displays user information.
DTE is a telecommunications term which usually refers to PCs, terminals, printers, etc.
DTMF (Dual-Tone MultiFrequency): A generic push-button concept made popular by AT&T TouchTone.
E
E&M: A telephony trunking system used for either switch-to-switch, or switch-to-network, or computer/telephone system-to-switch connec-
tion.
EIA: The Electronics Industries Association is a trade organization in Washington, DC that sets standard for use of its member companies.
(See RS-232, RS-422, RS530.)
Encapsulation: A technique used by network-layer protocols in which a layer adds header information to the protocol data unit from the
preceding layer. Also used in "enveloping" one protocol inside another for transmission. For example, IP inside IPX.
Errored Seconds (ES): Any second of operation that all 1.544M bits are not received exactly as transmitted. Contrast "Error Free Seconds".
Error Free Seconds (EFS): Any second of operation that all 1.544M bits are received exactly as transmitted. Contrast "Errored Seconds".
ESF Error Event: A T1 error condition that is logged when a CRC-6 error or an OOF error occurs.
Ethernet: A 10-megabit baseband local area network that allows multiple stations to access the transmission medium at will without prior
coordination, avoids contention by using carrier sense and deference, and resolves contention by using collision detection and transmission.
Ethernet uses carrier sense multiple access with collision detection (CSMA/CD).
Excess Zeros: A T1 error condition that is logged when more than 15 consecutive 0s or less than one 1 bit in 16 bits occurs.
Exchange: A unit (public or private) that can consist of one or more central offices established to serve a specified area. An exchange
typically has a single rate of charges (tariffs) that has previously been approved by a regulatory group.
Exchange Area: A geographical area with a single uniform set of charges (tariffs), approved by a regulatory group, for phone services. Calls
between any two points within an exchange area are local calls. See also "Digital PBX", "PBX".
Exchange Termination (ET): The carrier's local exchange switch. Contrast with "Loop Termination - LT".
Explicit Congestion Management: The method used in frame relay to notify the terminal equipment that the network is overly busy. The
use of FECN and BECN is called explicit congestion management. Some end-to-end protocols use FECN or BECN, but usually not both
options together. With this method, a congestion condition is identified and fixed before it becomes critical. Contrast with "implicit congesion".
Extended Super Frame (ESF): One of two popular formats for framing bits on a T1 line. ESF framing has a 24-frame super-frame, where
robbed bit signaling is inserted in the LSB (bit 8 of the DS-0 byte) of frames 6, 12, 18 and 24. ESF has more T1 error measurement capabilities than D4 framing. ESF and B8ZS are typically both offered to provide clear channel service.
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F
Failed Seconds: A test parameter where the circuit is unavailable for one full second.
Failed Signal: A T1 test parameter logged when there are more than 9 SES (Severely Errored Seconds).
Fax (facsimile): Refers to the bit-mapped rendition of a graphics-oriented document (fax) or to the electronic transmission of the image over
phone lines (faxing). Fax transmission differs from data transmission in that the former is a bit-mapped approximation of a graphical document and, therefore, cannot be accurately interpreted according to any character code.
Firmware: A category of memory chips that hold their content without electrical power, they include ROM, PROM, EPROM and EEPROM
technologies. Firmware becomes "hard software" when holding program code.
Foreground: The application program currently running on and in control of the PC screen and keyboard. The area of the screen that
occupies the active window. Compare with "background".
Fractional T1 (FT1): A digital data transmission rate between 56K bps (DS0 rate) and 1.544M bps (the full T1 rate - in North America). FT1
is typically provided on 4-wire (two copper pairs) UTP. Often used for video conferencing, imaging and LAN interconnection due to its low cost
and relatively high speed. FT1 rates are offered in 64K bps multiples, usually up to 768K bps.
Frequency: A characteristic of an electrical or electronic signal which describes the periodic recurrence of cycles. Frequency is inversely
proportional to the wavelength or pulse width of the signal (i.e., long wavelength signals have low frequencies and short wavelength signals
yield high frequencies).
Foreign Exchange (FX): A CO trunk with access to a distant CO, allowing ease of access and flat-rate calls anywhere in the foreign
exchange area.
Foreign Exchange Office (FXO): provides local phone service from a CO outside of ("foreign" to) the subscriber's exchange area. In simple
form, a user can pick up the phone in one city and receive a tone in the foreign city.
Connecting a POTS phone to a computer telephony system via a T1 link requires a channel bank configured for the FX connection. To
generate a call from the POTS set to the computer telephony system, a FXO connection must be configured.
Foreign Exchange Station (FXS): See FX, FXO. To generate a call from the computer telephony system to the POTS set, a FXS connection must be configured.
Forward Explicit Congestion Notification (FECN): A bit that tells you that a certain frame on a particular logical connection has encountered heavy traffic. The bit provides notification that congestion-avoidance procedures should be initiated in the same direction of the received
frame. See also BECN (Backward Explicit Congestion Notification).
Frame: A group of data bits in a specific format to help network equipment recognize what the bits mean and how to process them. The bits
are sent serially, with a flag at each end signifying the start and end of the frame.
Frame Relay: A form of packet switching that uses small packets and that requires less error checking than other forms of packet switching.
Frame relay is effective for sending "bursty" data at high speeds (56/64K, 256K, and 1024K bps) over wide area networks. Frame Relay
specifications are defined by ANSI documents ANSI T1.602, T1.606, T1S1/90-175, T1S1/90-213, and T1S1/90-214. In using frame relay,
blocks of information (frames) are passed across a digital network interface using a "connection number" that is applied to each frame to
distinguish between individual frames.
Frame Relay Forum: A non-profit organization of 300+ vendors and service providers, based in Foster City, CA, that are developing and
deploying frame relay equipment.
Frame Relay Implementors Forum: A group of companies supporting a common specification for frame relay connection to link customer
premises equipment to telco network equipment. Their specification supports ANSI frame relay specs and defines extensions such as local
management.
Frame Relay Access Device (FRAD): A piece of equipment that acts as a concentrator or frame assembler/dissassember that can support
multiple protocols and provide basic "routing" functions.
G
Gateway: 1. A functional unit that interconnects two computer networks with different network architectures. A gateway connects networks or
systems of different architectures. A bridge interconnects networks or systems with the same or similar architectures. 2. A network that
connects hosts.
Graphical User Interface (GUI): A type of computer interface consisting of a visual metaphor of a real-world scene, often of a desktop.
Within that scene are icons, representing actual objects, that the user can access and manipulate with a pointing device.
H
Handshaking: A process that two modems go through at the time of call setup to establish synchronization over the data communications
link. It is a synchronization and negotiation process accomplished by the exchange of predefined, mutually recognized control codes.
High-level Data Link Control (HDLC): An ISO standard, bit-oriented data communications protocol that provides nearly error-free data
transfers.
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I
Hexadecimal: A base 16 numbering system used to represent binary values. Hex uses the numbers 0-9 and the letters A-F: usually notated
by an "h" (e.g., "4CF h", read "four charley fox, hex"). The result is that one hex digit represents a 4-bit value.
Implicit congestion management: A method of informing the terminal that the network is busy. This method relies on the end-system
protocol to detect and fix the congestion problem. (TCP/IP is an example of a protocol using only implicit congestion management.) See also
"explicit congestion management".
In-band: Refers to the type of signalling over the conversion path on an ISDN call. Contrast "out-of-band".
Insufficient Ones: A T1 error condition that is logged when less than one 1 in 16 0s or less than 12.5 % average 1s density is received.Inter Exchange Carrier (IEC): The long distance company (LE) who's central office provides the point of reference for T1 access. Any
common carrier authorized by the FCC to carry customer transmissions between LATAs.
Internet: Refers to the computer network of many millions of university, government and private users around the world. Each user has a
unique Internet Address.
Internet Address (IP Address): A unique 32-bit address for a specific TCP/IP host on a network. Normally printed in dotted decimal format
(e.g., 129.128.44.227).
Internet Protocol (IP): A protocol used to route data from its source to its destination in an Internet environment. The Internet Protocol was
designed to connect to local area networks. Although there are many protocols that do this, IP refers to the global system of interconnecting
computers. It is a highly distributed protocol (each machine only worries about sending data to the next step in the route).
Internetwork Packet Exchange (IPX): A NetWare communications protocol used to route messages from one node to another. IPX packets
include network addresses and can be routed from one network to another. An IPX packet can occasionally get lost when crossing networks,
thus IPX does not guarantee delivery of a complete message. Either the application has to provide that control, or NetWare's SPX protocol
must be used.
Interoperable: Devices from different vendors that can exchange information using a standard's base protocol.
I/O Addresses: Locations within the I/Oaddress space of your computer used by a device, such as an expansion card, a serial port, or an
internal modem. The address is used for communication between software and a device.
IRQ Level (Interrupt Request Level): The notification a processor receives when another portion of the computer's hardware requires its
attention. IRQs are numbered so that the device issuing the IRQ can be identified, and so IRQs can be prioritized.
ISA (Industry Standards Architecture) (pronounced "ice a"): The classic 8 or 16-bit architecture introduced with IBM's PC-AT computer.
ISDN (Integrated Services Digital Network): An International telecommunications standard for transmitting voice, video and data over a
digital communications line. ISDN is a world-wide telecommunications service that uses digital transmission and switching technology to
support voice and digital data communications. Frame relay was partially based on ISDN's data link layer protocol (LAPD). Frame relay can
be used to transmit across ISDN services offering circuit-switched connection at 64K bps and higher speeds.Contrast Public Switched
Telephone Network (PSTN).
ITU-TSS (formerly CCITT): International Telecommunications Union-Telecommunications Sector; the United Nations organization that
prepares standards ("Recommendations") for resolving communications issues and problems.
K
Key Telephone System (KTS): Phone devices with multiple buttons that let you select incoming or outgoing CO phone lines directly. Similar
in operation to a PBX, except a KTS you don't have to dial a "9" for a call outside the building.
Key Service Unit (KSU): A small device containing the switching electronics for a business key telephone system (KTS).
Key Set: A phone set with several buttons for call holding, line pickup, intercom, autodialing, etc. Also called a touchtone phone (Ericsson)
and a KTS (Key Telephone Set).
L
LAPB: Link Access Procedure Balanced; based on the X.25 Layer 2 specification. A full-duplex point-to-point bit-synchronous protocol
commonly used as a data link control protocol to interface X.25 DTEs. LAPB is the link initialization procedure that establishes and maintains
communications between the DTE and the DCE.
LAPD: Link Access Protocol for the D-Channel; based on the ISDN Q.921 specification. A full-duplex point-to-point bit-synchronous linklevel protocol for ISDN connections; different from LAPB in its framing sequence. Transmission is in units called "frames", and a frame may
contain one or more X.25 packets.
Line Coding: The representation of 1s and 0s on a T1 line. The two methods of line coding commonly used, B8ZS and AMI, differ in the
restrictions placed on user data. T1 line coding ensures that sufficient timing information is sent with the digital signal to ensure recovery of all
the bits at the far end. Timing information on the T1 line is included in the form of 1s in the data stream; a long string of 0s in the data stream
could cause problems recovering the data.
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Line Termination (LT): The electronics at the ISDN network side of the user/network interface that complements the NT1 at the user side.
The LT and the NT1 together provide the high-speed digital line signals required for BRI access.
Listed Directory Number (LDN): The main number assigned by the telco; the number listed in the phone directory and also provided by
Directory Assistance. Some devices can have more than one LDN, such as ISDN devices that have one LDN for voice and another LDN for
data.
Local Area Network (LAN): 1. A computer network located on a user's premises within a limited geographical area. Communication within a
local area network is not subject to external regulations; however, communication across the LAN boundary may be subject to some form of
regulation. 2. A LAN does not use store and forward techniques. 3. A network in which a set of devices are connected to one another for a
communication and that can be connected to a larger network.
Local Access and Transport Area (LATA): A post-divestiture geographical area generally equivalent to a Standard Metropolitan Statistical
Area. At divestiture, the territory served by the Bell system was divided into approximately 161 LATAs. The Bell Operating Companies (BOCs)
provide Intra-LATA services.
Local Exchange Carrier (LEC): The local phone company which provides local (i.e., not long distance) transmission services. AKA "telco".
LECs provide T1 or FT1 access to LDCs (unless the T1 circuit is completely intra-LATA). Inter-LATA T1 circuits are made up of a combination
of Access and Long Haul facilities.
Local Management Interface (LMI): A specification for frame relay equipment that defines status information exchange.
Local Loop: A transmission path, typically twisted-pair wire, between an individual subscriber and the nearest public telecommunications
network switching center. The wires provide ISDN service, but require an NT1 at the user end and an LT at the network end. (AKA, "loop" or
"subscriber loop".)
Logical Link Control (LLC2): In a local area network, the protocol that governs the exchange of transmission frames between data stations
independently of how the transmission medium is shared. The LLC2 protocol was developed by the IEEE 802 commitee and is common to all
LAN standards.
Logical Unit (LU): A type of network accessible unit that enables end users to gain access to network resources and communicate with each
other.
Long Haul: The T1 element that connects to the Access portion of the long distance company's (LDC's) central office. The LDC is commonly
called the point of presence (POP). Each LDC has a number of POPs, located throughout the country. The LDC is also called an IEC (Inter
Exchange Carrier).
Long Haul Communications: The type of phone call reaching outside of a local exchange (LE).
M
Management Information Base (MIB): A database of network management information used by the Common Management Information
Protocol (CMIP) and the Simple Network Management Protocol (SNMP).
Megacom: An AT&T service with a normal WATS line (typically T1) between the customer premise and the AT&T serving class 4 CO are
the customer's responsibility.
MegaLink: BellSouth's leased T1 service.
Message: Associated with such terms as packet, frame, and segment. 1. In information theory, an ordered series of characters intended to
convey information. 2. An assembly of characters and sometimes control codes that is transferred as an entry from an originator to one or
more recipients.
Modem: A communications device that enables a computer to transmit information over a phone line. It converts the computer's digital
signals into analog signals to send over a phone line and converts them back to digital signals at the receiving end. Modems can be internal
and fit into an expansion slot, or external and connect to a serial port.
Multiplexer (Mux): 1. A device that takes several input signals and combines them into a single output signal in such a manner that each of
the input signals can be recovered. 2. A device capable of interleaving the events of two or more activities or capable of distributing the
events of an interleaved sequence to the respective activities. 3. Putting multiple signals on a single channel.
Multiprotocol: A device that can interoperate with devices utilizing different network protocols.
Multithreading: The ability of a software system to be able to handle more than one transaction concurrently. This is contrasted to the case
where a single transaction is accepted and completely processed before the next transaction processing is started.
N
Nailed Connection: A permanent or dedicated circuit of a previously switched circuit or circuits.
Nailed-up Circuit: A semipermanent circuit established through a circuit-switching facility for point-to-point connectivity.
NAK (Negative Acknowledgment): Communications code used to indicate that a message was not properly received, or that a terminal
does not wish to transmit. Contrast with ACK.
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Network: A group of computers connected by cables or other means and using software that enables them to share equipment, such as
printers and disk drives to exchange information.
Node: Any point within a network which has been assigned an address.
O
Object-Orientated: A method for structuring programs as hierarchically organized classes describing the data and operations of objects that
may interact with other objects.
Office Channel Unit - Data Port (OCU-DP): The CO channel bank used as the interface between the customer's DSU and the channel
bank.
Off-hook: The condition of a device which has accessed a phone line (with or without using the line). In modem use, this is equivalent to a
phone handset being picked up. Dialing and transmission are allowed, but incoming calls are not answered. Contrast "on-hook".
Off Premise Extension (OPX): An extension or phone that terminates in a location other than that of the PBX. Commonly used to provide a
corporate member with an extension of the PBX at home.
Ones Density: the measure of the number of logical 1s on a T1 line compared to a given total number of bits on that line; used for timing
information in data recovery in AMI and B8ZS.
On-Hook: The condition of a device which has not accessed a phone line. In modem use, this is equivalent to a phone handset that has not
been picked up. In other words, it can receive an incoming call. Contrast "off-hook".
Open Shortest Path First (OSPF): A hierarchical Interior Gateway Protocol (IGP) routing algorithm for IP that is a proposed standard for
Internet. OSPF incorporates least-cost routing, equal-cost routing, and load balancing.
Outage: The measure of the time during which a circuit is not available for use due to service interrupt. Outage is the complement of circuit
"availability" (100% minus % available = % outage).
Out-of-band: Signaling that is separated from the channel carrying the information (i.e., the voice/data/video signal is separate from the
carrier signal). Dialing and various other "supervisory" signals are included in the signaling element. Contrast "In-band" signaling.
Out of Frame (OOF): A T1 alarm condition that is logged on the loss of 2, 3 or 4 of 5 consecutive FT framing bits.
P
Packet: 1. In data communication, a sequence of binary digits, including data and control signals, that is transmitted and switched as a
composite whole. The data, control signals and, possibly, error control information are arranged in a specific format. 2. Synonymous with data
frame. 3. In TCP/IP, the unit of data passed across the interface between the Internet layer and the link layer. A packet includes an IP header
and data. A packet can be a complete IP datagram or a fragment of an IP diagram. 4. In X.25, a data transmission information unit. A group
of data and control characters, transferred as a unit, determined by the process of transmission. Commonly used data field lengths in packets
are 128 or 256 bytes. 5. The field structure and format defined in the CCITT X.25 recommendation.
Packet Assembler/Dissembler (PAD): Used by devices to communicate over X.25 networks by building or stripping X.25 information on or
from a packet.
Packet Data: The information format ("packetized") used for packet-mode calls.
Packet Mode: Refers to the switching of chunks of information for different users using statistical multiplexing to send them over the same
transmission facility.
Parity bit: An extra bit attached to each byte of synchronous data used to detect errors in transmission.
Permanent Virtual Circuit (PVC): A connection between two endpoints dedicated to a single user. In ISDN, PVCs are established by
network administration and are held for as long as the user subscribes to the service.
Physical Unit (PU): The component that manages and monitors the resources (such as attached links and adjacent link stations) associated
with a node, as requested by an SSCP via an SSCP-PU session. An SSCP activates a session with the physical unit in order to indirectly
manage, through the PU, resources of the node such as attached links. This term applies to type 2.0, type 4, and type 5 nodes only.
Point of Presence (POP): The central office's end points of the long distance carriers.
Point to Point Protocol (PPP): A protocol that lets a PC user access TCP/IP (Internet member) using an ISDN terminal adapter or a high-
speed modem over a standard phone line.
Port: A location for input or output data exchange. Computers, muxes, etc. have ports for various purposes.
Primary Rate Interface (PRI): Used on ISDN. In North America, and Japan, PRI is one 64Kbps D channel and 23 B channels. Elsewhere, it
is one D channel and 30 B channels.
Primitive: An abstract representation of interaction across the access points indicating that information is being passed between the service
user and the service provider. The OSI Reference Model defines four types of primitives: Request, Indication, Response and Confirm.
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Private Branch Exchange (PBX): A phone exchange located on the customer's premises. The PBX provides a circuit switching facility for
phone extension lines within the building, and access to the public phone network. See also "Exchange".
PROM (Programmable Read Only Memory - pronounced "prom"): A permanent memory chip that can be programmed or filled by the
customer after by the manufacturer has set initial values. Contrast with ROM.
Protocol: 1. A set of semantic and syntactic rules that determines the behavior of functional units in achieving communication. 2. In Open
Systems Interconnection architecture, a set of semantic and syntactic rules that determine the behavior of entities in the same layer in
performing communication functions. 3. In SNA, the meanings of and the sequencing rules for requests and responses used for managing the
network, transferring data, and synchronizing the states of network components. 4. Synonymous with line control discipline.
PSTN (Public Switched Telephone Network): A worldwide public voice phone network that is used as a telecommunications medium for
the transmission of voice, data and other information.
Public Data Network (PDN): A packet-switched network that is available to the public for individual ("subscriber") use. Typically, controlled
by a government or a national monopoly.
Public Switched Telephone Network (PSTN): The group of circuit-switching voice carriers, which are commonly used as analog data
communications services.
Pulse Code Modulation (PCM): 1. In data communication, variation of a digital signal to represent information; for example, by means of
pulse amplitude modulation (PAM), pulse duration modulation (PDM), or pulse position modulation (PPM). 2. Transmissions of analog
information in digital form through sampling and encoding the samples with a fixed number of bits.
Pulse dialing: One of two methods of dialing a phone, usually associated with rotary-dial phones. Compare with "tone dialing".
Q
Quantizing: The process of analog-to-digital conversion by assigning a range, from the contiguous analog values, to a discrete number.
R
Random Access Memory (RAM): A computer's primary workspace. All data must be stored in RAM (even for a short while), before
software can use the processor to manipulate the data. Before a PC can do anything useful it must move programs from disk to RAM. When
you turn it off, all information in RAM is lost.
Rate Enforcement: The concept in frame relay where frames sent faster than the CIR are to be carried only if the bandwidth is available,
otherwise they are to be discarded. (The frame relay network assumes that anything exceeding the CIR is of low priority.) Rate enforcement
makes sure that the network will not get so congested that it isn't able to meet the agreed on CIR
Recognized Private Operating Agency (RPOA): A corporation, private or government-controlled, that provides telecommunications
services. RPOAs, such as AT&T, participate as non-voting members in the CCITT.
Red Alarm: A T1 error condition generated when a local failure (e.g., loss of synchronization) exists for 2.5 seconds, causing a Carrier Group
Alarm (CGA). See also "Blue Alarm" and "Yellow Alarm".
Request for Comment (RFC): A set of papers in which Internet standards (published and proposed), along with generally-accepted ideas,
proposals, research results, etc. are published.
Ring Down Box: A device that emulates a CO by generating POTS calls for testing and product demos.
Ring Down Circuit: A tie line connecting phones where picking up one phone automatically rings another phone. A feature used for
emergencies to alert the person at the other phone of the incoming call.
RJ-11: An industry standard interface used for connecting a phone to a modular wall outlet; comes in 4-and 6-wire packages.
RJ-45: An 8-wire modular connector for voice and data circuits.
Robbed Bit Signaling: The popular T1 signaling mechanism where the A and B bits are sent by each side of the T1 termination and are
"buried" in the voice data of each voice channel in the T1 circuit. Since the bits are "robbed" infrequently, voice quality is remains relatively
uncompromised. See "bit robbing".
The robbed-bit signaling technique is used in D4 channel banks to convey signaling information. The eighth (least significant) bit of each of
the 24 8-bit time slots is "robbed" every sixth frame to convey voice-related signaling information such as on-hook, off-hook, etc., for each
channel.
Router: A device that connects two networks using the same networking protocol. It operates at the Network Layer (Layer 3) of the OSI
model for forwarding decisions.
Routing Information Protocol (RIP): A distance vector-based protocol that provides a measure of distance, or hops, from a transmitting
workstation to a receiving workstation.
RS232-C: An EIA standard for a serial interface between computers and peripheral devices (modem, mouse, etc.). It uses a 25-pin DB-25, or
a 9-pin DB-9 connector. The RS-232 standard defines the purposes, electrical characteristics and timing of the signals for each of the 25
lines.
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RS-422: The EIA standard for a balanced interface with no accompanying physical connector. RS-422 products can use screw terminals,
DB-9, various DB-25, and DB-37 connectors.
RS-530: The EIA standard for the mechanical/electrical interface between DCEs and DTEs transmitting synchronous or asynchronous serial
binary data. RS-530 provides for high data rates with the same connector used for RS-232; however, it is incompatible with RS-232.
S
Serial Port: The connector on a PC used to attach serial devices (those that need to receive data one bit after another), such as a mouse, a
printer or a modem. This consists of a 9- or 25-pin connector that sends data in sequence (bit by bit). Serial ports are referred to as "COMx"
ports, where x is 1 to 4 (i.e., COM1 through COM4). A serial port contains a conversion chip called a "UART" which translates between
internal parallel and external serial formats.
Service: The requirements offered by an RPOA to its customers to satisfy specific telecommunications needs.
Severely Errored Seconds (SES): Refers to a typical T1 error event where an error burst occurs (a short term, high bit-error rate that is
self-clearing). Per the ITU-T (CCITT) G.821: any second in which the BER is less than 1x10-3.
Signaling: The process of establishing, maintaining, accounting for, and terminating a connection between two endpoints (e.g., the user
premises and the telco CO). Central office signals to the user premises can include ringing, dial tone, speech signals, etc. Signals from the
user's phone can include off-hook, dialing, speech to far-end party, and on-hook signals.
In-band signaling techniques include pulse and tone dialing. With common channel signaling, information is carried out-of-band.
Simple Network Management Protocol (SNMP): TCP/IP protocol that allows network management.
Simultaneous Voice Data (SVD): A technology for letting a user send data via a modem, and use a handset to talk to another user at the
same time over the same connection. The alternative, making a second call, can be expensive or even impossible. The uses for SVD are
telecommuting, videoconferencing, distant learning, tech support, etc.
Stop Bit: One of the variables used for timing in asynchronous data transmission. Depending on the devices, each character may be trailed
by 1, 1.5, or 2 stop bits.
Superframe (D4): A T1 transmission format that consists of 12 DS1 frames, or 2316 bits. A DS1 frame consists of 193 bit positions. A frame
overhead bit is in the first position, and it is used for frame and signaling phase alignment only.
Subscriber Loop: See "Local loop".
Switched 56: A circuit-switched (full duplex digital synchronous data transmission) service that lets you dial a number and transmit data to it
at 56K bps. It is a relatively low cost service, widely used in North America for telecommuting, videoconferencing and high speed data
transfers. Many phone companies are (or will be) phasing out Switched 56 in favor of ISDN service.
Switched Virtual Circuit (SVC): A type of data transmission where the connection is maintained only until the call is cleared.
Switched Line: In communications, a physical channel established by dynamically connecting one or more discreet segments. This
connection lasts for the duration of the call after which each segment may be used as part of a different channel. Contrast with leased line.
Switched Network: A network in which a temporary connection is established from one point via one or more segments.
Synchronous Data Link Control (SDLC): A discipline conforming to subsets of the Advanced Data Communications Control Procedures
(ADCCP) of the American National Standards Institute (ANSI) and High-level Data Link Control (HDLC) of the International Organization for
Standardization, for managing synchronous, code-transparent, serial-by-bit information transfer over a link connection. Transmission
exchanges may be duplex, or half-duplex over switched or nonswitched links. The configuration of the link connection may be point-to-point,
multipoint, or loop.
Synchronous Transmission: The transmission of data which involves sending a group of characters in a packet. This is a common method
of transmission between computers on a network or between modems. One or more synchronous characters are transmitted to confirm
clocking before each packet of data is transmitted. Compare to Asynchronous Transmission.
Systems Network Architecture (SNA): The description of the logical structure, formats, protocols, and operational sequences for transmitting information units through, and controlling the configuration and operation of of, networks.
T
Tariff: The rate/availability schedule for telephone and ISDN services from a regulated service provider.
TCP/IP: A set of communication protocols that support peer-to-peer connectivity functions for both local and wide area networks.
T Carrier: The generic name for a digitally multiplexed carrier system. In the North American digital hierarchy, a T is used to designate a DS
(digital signal) level hierarchy. Examples: T1 (DS1) is a 1.544 M bps 24-channel designation. In Europe, T1 is called E1. The T Carrier
system was originally designed for transmitting digitized voice signals, but has since been adapted for digital data applications.
T1: A digital transmission link capable of 1.544M bps. T1 uses two pairs of normal UTP, and can handle 24 voice conversations, each
digitized at 64K bps. T1 is a standard for digital transmission in the U.S., Canada, Japan and Hong Kong. T1 is the access method for highspeed services such as ATM, frame relay, and SMDS. See also T Carrier, T1 line and FT1.
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T1 Channel Tests: A set of diagnostics that vary by carrier, used to verify a T1 channel operation. Can include Tone, Noise Level, Impulse
Noise Level, Echo Cancellors, Gain, and Crosstalk testing.
T1 Framing: To digitize and encode analog voice signals requires 8000 samples per second (twice the highest voice frequency of 4000
Hz). Encoding in an 8-bit word provides the basic T1 block of 64K bps for voice transmission. This "Level 0 Signal, as its called, is represented by "DS-0", or Digital Signal at Level 0. 24 of these voice channels are combined into a serial bit stream (using TDM), on a frame-byframe basis. A frame is a sample of all 24 channels; so adding in a framing bit gives a block of 193 bits (24x8+1=193). Frames are transmitted at 8000 per second (the required sample rate), creating a 1.544M (8000x193=1.544M) transmission rate.
T1 Line: A digital communications facility that functions as a 24-channel pathway for data or voice. A T1 line is composed of two separate
elements: the Access element and the Long Haul element.
T1 Mux: A device used to carry many sources of data on a T1 line. The T1 mux assigns each data source to distinct DS0 time slots within
the T1 signal. Wide bandwidth signals take more than one time slot. Normal voice traffic or 56/64K bps data channels take one time slot. The
T1 mux may use an internal or external T1 DSU; a "channel bank" device typically uses an external T1 CSU.
Transmission Control Protocol / Internet Program (TCP/IP): A multilayer set of protocols developed by the US Department of Defense to
link dissimilar computers across dissimilar and unreliable LANs.
Terminal: The screen and keyboard device used in a mainframe environment for interactive data entry. Terminals have no "box", which is to
say they have no file storage or processing capabilities.
Terminal Adapter (TA): An ISDN DTE device for connecting a non-ISDN terminal device to the ISDN network. Similar to a protocol
converter or an interface converter, a TA connects a non-ISDN device between the R and S interfaces. Typically a PC card.
Tie line: A dedicated circuit linking two points without having to dial a phone number (i.e., the line may be accessed by lifting the phone
handset or by pushing a button).
Time-Division Multiplexing (TDM): Division of a transmission facility into two or more channels by allotting the common channel to several
different information channels, one at a time.
Time Slot: One of 24 channels within a T1 line. Each channel has a 64K bps maximum bandwidth. "Time slot" implies the time division
multiplexing organization of the T1 signal.
Toll Call: A call to a location outside of your local service area (i.e., a long distance call).
Tone dialing: One of two methods of dialing a phone, usually associated with Touch-Tone® (push button) phones. Compare with pulse
dialing.
Topology: Physical layout of network components (cables, stations, gateways, and hubs). Three basic interconnection topologies are star,
ring, and bus networks.
Transmission Control Protocol (TCP): A communications protocol used in Internet and in any network that follows the US Department of
Defense standards for internetwork protocol. TCP provides a reliable host-to-host protocol between hosts in packet-switched communications
networks and in interconnected systems of such networks. It assumes that the Internet protocol is the underlying protocol.
Transport Layer: Layer 4 of the Open Systems Interconnection (OSI) model; provides reliable, end-to-end delivery of data, and detects
transmission sequential errors.
Transport Protocol Data Unit (TPDU): A transport header, which is added to every message, contains destination and source addressing
information that allows the end-to-end routing of messages in multi-layer NAC networks of high complexity. They are automatically added to
messages as they enter the network and can be stripped off before being passed to the host or another device that does not support TPDU's.
Trunk: Transmission links that interconnect switching offices.
TSR (terminate and stay resident): A software program that remains active and in memory after its user interface is closed. Similar to a
daemon in UNIX environments.
Tunneling: Encapsulation data in an IP packet for transport across the internet.
Twisted pair wiring: A type of cabling with one or more pairs of insulated wires wrapped around each other. An inexpensive wiring method
used for LAN and telephone applications, also called UTP wiring.
U
UART (Universal Asynchronous Receiver/Transmitter) (pronounced "you art"): A chip that transmits and receives data on the serial
port. It converts bytes into serial bits for transmission, and vice versa, and generates and strips the start and stop bits appended to each
character.
UNIX: An operating system developed by Bell Laboratories that features multiprogramming in a multi-user environment.
Unshielded Twisted Pair (UTP): Telephone-type wiring. Transmission media for 10Base-T.
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Glossary
V.25bis: An ITU-T standard for synchronous communications between a mainframe or host and a modem using HDLC or other character-
oriented protocol.
V.54: The ITU-T standard for local and remote loopback tests in modems, DCEs and DTEs. The four basic tests are:
• local digital loopback (tests DTE send and receive circuits),
• local analog loopback (tests local modem operation),
• remote analog loopback (tests comm link to the remote modem), and
• remote digital loopback (tests remote modem operation).
Virtual Circuit: A logical connection. Used in packet switching wherein a logical connection is established between two devices at the start
of transmission. All information packets follow the same route and arrive in sequence (but do not necessarily carry a complete address).