Multitech | MultiVOIP 100 MVP120 | User guide | Multitech MultiVOIP 100 MVP120 User guide

Voice / Fax over IP Networks
100
Voice/Fax Over IP Networks
Model MVP120
H.323 Mode
User Guide
User Guide
MultiVOIP Series 100 - Model MVP120
S0000223 Revision A
Copyright
This publication may not be reproduced, in whole or in part, without prior expressed written permission from MultiTech Systems, Inc. All rights reserved.
Copyright © 2001, by Multi-Tech Systems, Inc.
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.
Revision
A
Date
Description
09/27/01
Manual released. Software at Version 7.51x.
Patents
This Product is covered by one or more of the following U.S. Patent Numbers: 5.682.386; 5.757.801; 6.151.333,
6,219,708. Other Patents Pending.
Trademarks
Multi-Tech and the Multi-Tech logo are registered trademarks and MultiVOIP is a trademark of Multi-Tech Systems,
Inc.
Adobe Acrobat is a trademark of Adobe Systems Incorporated.
Microsoft Windows, Windows 2000, Windows 98, Windows 95, Windows NT, and NetMeeting are either registered
trademarks or trademarks of Microsoft Corporation in the United States and/or other countries.
World Headquarters
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, Minnesota 55112
(763) 785-3500 or (800) 328-9717
Fax 763-785-9874
Technical Support (800) 972-2439
Internet Address: http://www.multitech.com
Contents
Chapter 1 - Introduction to the MultiVOIP ................................................................................ 6
Preview of This Guide ............................................................................................................. 7
Typical Application................................................................................................................... 8
Front Panel Description ........................................................................................................ 12
Back Panel Description ......................................................................................................... 13
Specifications ........................................................................................................................ 14
Chapter 2 - Installation ............................................................................................................. 16
Overview of the Installation Process ..................................................................................... 16
Unpacking Your MVP120 ...................................................................................................... 17
Telecom Safety Warning ....................................................................................................... 17
Other Safety Warnings .......................................................................................................... 17
Cabling Your MVP120 ........................................................................................................... 18
Chapter 3 - Installing the MVP120 Software ........................................................................... 20
Chapter 4 - Configuring the Master MultiVOIP ....................................................................... 24
Registering with a Gatekeeper Phone Directory ................................................................... 28
Building a Proprietary Phonebook Directory ......................................................................... 31
Chapter 5 - Configuring the Slave MultiVOIPs ....................................................................... 37
Chapter 6 - Deploying the VOIP Network ................................................................................ 44
Remote Site Administrator .................................................................................................... 44
Chapter 7 - Using the MultiVOIP Software .............................................................................. 46
MVP120 Configuration .......................................................................................................... 47
Changing Channel Parameters ............................................................................................. 48
Interface Tab.................................................................................................................... 48
Voice/Fax Tab .................................................................................................................. 49
Billing/Misc Tab ................................................................................................................ 50
Regional Tab ................................................................................................................... 51
Changing the Phone Directory Database ............................................................................. 52
Proprietary Phone Directory Database ........................................................................... 53
Gatekeeper Phone Directory Database .......................................................................... 55
Changing IP Parameters ....................................................................................................... 57
Viewing Call Progress ........................................................................................................... 58
Applications Setup ................................................................................................................ 58
Viewing Statistics .................................................................................................................. 60
IP Statistics ...................................................................................................................... 60
SNMP Statistics ............................................................................................................... 61
Viewing Logs ................................................................................................................... 62
Viewing Log Entry Details ............................................................................................... 62
Viewing Channel Totals ................................................................................................... 63
Reports ............................................................................................................................ 63
Upgrade Procedures ............................................................................................................. 64
Upgrade Software ........................................................................................................... 64
Manual Upgrade Procedure ............................................................................................ 66
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Chapter 8 - Remote Configuration and Management ............................................................ 68
Modem-Based ....................................................................................................................... 68
LAN-Based ........................................................................................................................... 70
Telnet ..................................................................................................................................... 72
Web Management ................................................................................................................. 73
Chapter 9 - Warranty, Service, and Technical Support ......................................................... 75
Multi-Tech Systems, Inc. Warranty & Repairs Policies ......................................................... 75
Service .................................................................................................................................. 77
Ordering Accessories ...................................................................................................... 77
Technical Support.................................................................................................................. 78
Appendix A - TCP/IP Description ............................................................................................. 80
Appendix B - Cabling Diagrams .............................................................................................. 83
Appendix C - Regulatory Information ..................................................................................... 85
Glossary ..................................................................................................................................... 89
Index ......................................................................................................................................... 102
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Voice / Fax over IP Networks
100
Chapter 1
Introduction to the MultiVOIP
Chapter 1 - Introduction to the MultiVOIP
Chapter 1 - Introduction to the MultiVOIP
The MultiVOIP 100 (Model MVP120) allows analog voice and fax communication over a
traditional IP network. Multi-Tech’s new voice/fax gateway technology allows voice and fax
communication to be transmitted, 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 MVP120 to an FXO PBX port or
PSTN and then to your existing Internet connection. Once configured, the MVP120 allows
voice and fax to travel down the same path as your traditional data communications.
The MVP120 supports the H.323 standards-based protocol enabling your MVP120 to
participate in real-time conferencing with other third-party VOIP Gateways or endpoints that
support the H.323 protocol (such as Microsoft NetMeeting® ). The H.323 standard defines
how endpoints make and receive calls, how endpoints negotiate a common set of audio and
data capabilities, how information is formatted and sent over the network, and how endpoints
communicate with their respective Gatekeepers. Gatekeeper software is optional and if
present in a network, it typically resides on a designated PC. It acts as the central point for all
calls within its zone and provides call control services to all registered endpoints. In addition,
Gatekeepers can perform bandwidth management through support for Bandwidth Request,
Confirm, and Reject messages.
Note: A zone consists of all H.323 endpoints that are under the Gatekeeper’s control.
The MVP120 is designed with one FXO voice/fax 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.
CL
MultiVOIP 100 Series - Model MVP120
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Chapter 1 - Introduction to the MultiVOIP
Preview of This Guide
This guide describes the MultiVOIP and tells you how to make the cable connections, install
the software, and configure it. The information contained in each chapter is as follows:
Chapter 1 - Introduction to the MultiVOIP
This chapter describes the MultiVOIP features, a typical application, and the front and back
panels.
Chapter 2 - Installation
This chapter provides an overview of the installation process, describes the contents of the
MultiVOIP package, diagrams the cabling instructions, and lists Telecom safety warnings.
Chapter 3 - Installing the MVP120 Software
This chapter describes how to load the MultiVOIP software.
Chapter 4 - Configuring the Master MultiVOIP
This chapter describes how to configure your Master.
Chapter 5- Configuring the Slave MultiVOIPs
This chapter describes how to configure your Slave.
Chapter 6 - Deploying the VOIP Network.
Chapter 7 - Using the MultiVOIP Software.
This chapter describes how to use the software to make changes and to the configuration of
your MultiVOIP.
Chapter 8 - Remote Configuration and Management
This chapter describes how to change the configuration of a remote MultiVOIP. The chapter
also describes typical client applications, such as Telnet and Web-based management.
Chapter 9 - Warranty, Service, and Technical Support
This chapter provides a statement of limited warranty, tells you how to get service for your
MultiVOIP, and gives you information about the Multi Tech Systems, Inc. Web site.
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Chapter 1 - Introduction to the MultiVOIP
Typical Application
Before VOIP (Voice Over IP) technology existed, a sales office used a data connection to the
Internet and a voice connection to the public telephone network. Now, with VOIP, the two
networks can be tied together. To accomplish this a Sales Office MultiVOIP is connected
between the public telephone network and the data network as shown in the typical
application below. The Sales Office MultiVOIP is going to be set up as the master MultiVOIP
and the MultiVOIP at the Remote Sales Office is going to be set up as the slave MultiVOIP.
With this approach, the person at the Remote Sales Office can pick up the telephone and dial
the Sales Office MultiVOIP and, after a second dial tone, can call anyone in the Sales Office,
or dial 9 for outside line and call anyone in the calling area as if they were at the Sales Office.
Sales Office
Workstation
Web Server
Remote Sales
Office
Workstation
Router
IP Address 201.22.122.1
Mask 255.255.255.128
LAN
HUB
Analog Connections
Channel 1: FXO
MultiVOIP MVP120
IP Address
201.22.122.118
Mask 255.255.255.128
Workstation
Workstation
MVP110
IP Address 206.25.124.120
Mask 255.255.255.240
#201
Internet/Intranet
IP Network
101
LAN
4124
HUB
512-4122
P
B
X
ISP
ProxyServer
IP Address 206.25.124.110
Mask 255.255.255.240
512-4123
Fax
PSTN
PSTN Connection
(T1/E1, PRI, etc.)
ProxyServer Static IP
Address 209.96.211.90
Typical VOIP Application
To set up this VOIP network, an MVP120 at the Sales Office is connected between the data
network and the sales office telephone switch (PBX). To connect the MVP120 to the data
network, an Ethernet cable is connected to the Ethernet port on the unit and the other end is
plugged into a hub on the data network. On the PBX side, one phone cord is connected to
the FXO jack on the back of the MVP120 and attached to an FXO port on the phone switch.
The line on the PBX occupies phone extension 4124.
To set up a remote sales office, connect the Ethernet jack on an MVP110 to the hub, and
connect (by a phone cord) a telephone to the FXS jack on the MVP110.
To configure the MVP120, the COM port of a PC is connected to the Command port on the
MVP120. 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 your network IP address and mask for the MVP120 and the gateway address
for the corporate router on the same dialog box.
After your network configuration is complete, then you can develop your VOIP phone
directory database. This database will be developed on the Sales Office MultiVOIP and sent
out to the Remote Sales Office.
You will need to add the phone numbers to the phone directory database. Before you set up
the phone directory database, you need to consider how the database is going to be used:
Are you going to have an H.323 Gatekeeper setup your call sessions,
or
Are you going to control your call sessions using the proprietary phone book?
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Chapter 1 - Introduction to the MultiVOIP
The H.323 Gatekeeper acts as the central point for all calls within its zone and provides call
control services to registered endpoints. If you choose the proprietary phone book, you
establish a master-slave relationship where the master MultiVOIP maintains the phone
directory and downloads the directory to each slave unit.
The Gatekeeper/Proprietary Phone Book selection is chosen from the Phone Directory
Database dialog box. Before you choose how the data base is going to be used, here are a
couple of things to keep in mind:
1. If a Gatekeeper is employed in the network, you need to choose the Gatekeeper option.
You can not mix the Proprietary Phone Book with the Gatekeeper. If you choose the
Gatekeeper option you can communicate with other third party endpoints that support
H.323 (such as Microsoft Netmeeting).
2. If you choose the Proprietary Phone Book, you establish a master-slave relationship in
which the master MultiVOIP maintains the phone directory database. All of the phone
numbers are listed in the data base so that if you want to communicate with someone in
your VOIP network, you can see the phone number in your data base. Everytime you
bring up your MultiVOIP the current phone directory is downloaded to your MultiVOIP.
Note: You can still call Netmeeting using the proprietary phone book.
The Gatekeeper
The GateKeeper is a separate application that can operate on a network pc and provides all
the controls needed to create, control, and manage an H.323 network zone. The H.323
network zone is all the endpoints (terminals and gateways (MultiVOIPs)) that are registered
with the gatekeeper. The gatekeeper function is address translation from LAN aliases for
terminals and gateways to IP addresses as defined in the RAS (Registration/Admission/
Status) specification. The RAS Protocol defines the communication with a gatekeeper and
support for RTP/RTCP for sequencing audio packets. The H.323 Gatekeeper also provides
call-authorization for both accepting and placing calls in its zone, and certain monitoring
features such as call permissioning and address resolution.
If you choose the Gatekeeper option, initially you need to communicate with the administrator
of the Gatekeeper to register your MultiVOIP. The information you need from the Gatekeeper
administrator is the IP address of the Gatekeeper and its port number. Then you need to
establish your alias address which includes phone number, channel number, H323 ID which
can be a numeric or an alphanumeric value, and your MultiVOIP LAN IP address. The port
number is 1720, but if the Gatekeeper uses a different port number, you have to ensure that
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Chapter 1 - Introduction to the MultiVOIP
you use the same port number. The Gatekeeper administrator will then enter your information into
the Gatekeeper data base. This concludes the preregisteration.
You can enter your alias address information into the Add/Edit Phone Entry dialog box. For
example, if you were setting up the corporate MultiVOIP, you could enter the following information
for the Voice/Fax Channel: Channel 1 of the corporate MultiVOIP uses extension 101. The
Description is optional, but it may be helpful if it is assigned to an individual or department.
The H323 ID assigned to this phone number identifies the office using this extension. The IP
Address of the Corporate MultiVOIP is 201.022.122.118. The default port number 1720 is used.
Once the MultiVOIP goes online, the Gatekeeper registers it with the above H.323 ID. No other
H323 endpoint can use this H.323 ID. This is like your own telephone number.
After you have entered the information in the Add/Edit Phone Entry dialog box and clicked OK,
the new phone number is added to the Phone Directory Database.
Proprietary Phone Book
If you choose the Proprietary Phone Book option in the Phone Directory Database dialog box
instead of the Gatekeeper option, the Database Type group becomes active and the RAS
Parameters group becomes inactive.
When you elect to use the Proprietary Phone Book, you set up a master-slave relationship. This
relationship allows one MultiVOIP to maintain the Phone Directory Database and publish this data
base to all MultiVOIP participants in the network. This proprietary database allows you to see all
the participants in your network and provides you with their phone numbers.
To set up a database so that the corporate MultiVOIP can call the remote branch office and the
remote branch office can call the corporate MultiVOIP, the Phone Directory Database will have
one entry for the corporate office and one entry for the Remote Branch Office. Extension 101 at
the corporate office is tied to voice channel 1. The Description again is optional. The Hunt Group
in this example is set for No Hunt. However, if you want to activate a Hunt Group (if an extension
on the MultiVOIP is busy and you wanted to look for another extension), you can assign a hunt
group to those extensions. In other words, if extension 101 is busy, the corporate MultiVOIP
would roll over to extension 102, if you have multiple voice channels.
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Chapter 1 - Introduction to the MultiVOIP
Again, the corporate MultiVOIP IP Address needs to be added. The port number is 1720.
This adds phone number 101 of the corporate MultiVOIP to the proprietary database.
After you have added Channel 1, you need to include Channel 1 at the remote branch office.
The proprietary database would then appear as in the following dialog box and when the
remote branch office MultiVOIP is turned on, the current database would be downloaded to
the remote branch office MultiVOIP.
So, if a sales person in the Sales Office picked up the telephone and dialed extension 4124,
the extension routes the telephone call to the VOIP network. After the second dial tone, the
person would dial extension 201. The telephone at the Remote Sales office would ring and a
telephone conversation is initiated just as if the Remote Sales office had been called over the
regular telephone network.
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Chapter 1 - Introduction to the MultiVOIP
Similarly, if a person at the Remote Sales office picked up extension 201 and dialed
extension 101, the MultiVOIP at the Sales Office would generate a second dial tone and the
person at the Remote Sales office would then dial any extension at the Sales Office, or the
person could then dial a 9, for example, and get an outside line of the Sales Office Public
Switched Telephone network. This person could then dial any telephone number, as if calling
from the Sales Office.
Front Panel Description
The MVP120 front panel has three groups of LEDs that provide the status of the Ethernet
connection (Ethernet), the Voice/Fax channel, and an LED for boot status. The front panel is
shown below and a description of each LED follows.
MVP120 Front Panel
Boot
The Boot indicator lights up when the MVP120 is booting during the downloading of the
setup.
Ethernet
RD Receive Data indicator blinks when packets are being received from the local area
network.
TD Transmit Data indicator blinks when packets are being transmitted to the local area
network.
LK Link indicator lights when the Ethernet link senses voltage from a concentrator or
external device.
CL Collision indicator lights when a collision is detected on the Ethernet link.
Voice/Fax
FX Fax indicator lights when there is fax traffic on the voice/fax channel.
TX Transmit indicator blinks when voice packets are being transmitted to another H.323
endpoint.
RX Receive indicator blinks when voice packets are being received from another H.323
endpoint.
XS Transmit Signal indicator lights when the Voice/Fax channel is off-hook.
RS Receive Signal indicator lights when the Voice/Fax channel is ringing.
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Chapter 1 - Introduction to the MultiVOIP
Back Panel Description
The cable connections for the MVP120 are made at the back panel. Connectors include
Power, Ethernet, Command Port (RJ-45), and FXO Voice/Fax Channel. The cable
connectors are shown and defined belos.
POWER
Power
Connector
ETHERNET COMMAND PHONE
Ethernet
Command
(10BaseT) Port (RJ-45)
Connector Connector
Phone
(RJ-11)
Connector
Back Panel
Power Connector
The Power connector is used to connect the external power supply to the MVP120.
Ethernet Connector
The Ethernet connector is used to connect the MVP120 to a LAN using unshielded twisted
cable. This connector is an RJ-45 jack.
Command Connector
The Command connector is used to configure the MVP120 using a PC with an available
serial port and running Windows software. The Command connector is an RJ-45 jack (an
adapter cable is provided to convert to a standard serial port DB9 female connector).
Phone Connector
Connects the FXO Voice/Fax Channel to an FXO/PBX port or PTSN. This connector is an
RJ-11 jack.
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Chapter 1 - Introduction to the MultiVOIP
Specifications
•
One 4 MB DRAM (1 Meg by 32-bit, 70 nanosecond SIMM)
Caution: SIMM speed and size cannot be mixed
•
Two Megabytes of flash memory
Ethernet Port
•
Single Ethernet Interface - 10Base-T (twisted pair) keyed RJ-45 connector.
Command Port
•
Single 19.2 Kbps asynchronous Command Port using an RJ-45 to DB9 cable with a DB9
female connector
Voice/Fax Channel
•
One RJ-11 jack (FXO)
Power Supply
•
Voltage - 115 VAC (Standard), 240 Volts AC (Optional)
•
Frequency - 47 to 63 Hz
•
Power Consumption - 4.3 Watts
Product
•
Dimensions - 1.0" high x 4.3" wide x 5.6" deep
(2.5 cm x 10.8 cm x 14.2 cm)
•
Weight - 8 oz. (224 g)
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Voice / Fax over IP Networks
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Chapter 2
Installation
Chapter 2 - Installation
Chapter 2 - Installation
Overview of the Installation Process
The basic steps of installing your MVP120 network involve unpacking the units, connecting
the cables, and configuring (Configuring is discussed in detail in Chapter 3) the units using
the included management software (MultiVOIP Configuration). The recommended installation
process includes four phases that, when completed, result in a fully functional Voice Over IP
network.
Unpacking and Cabling the MultiVOIP
The MultiVOIP package includes the MultiVOIP 100, a printed Quick Start Guide, a system
CD, a power supply, and a RJ-45 cable. If any one of these items is missing, please contact
you sales representative. Detailed cabling directions are included in this guide.
Installing the Software and Configuring Your MVP120
The VOIP administrator must first install the MVP120 software and then configure each
MVP120 for its specific function. During the configuration process, it’s important to note that
the Phone Directory Database is configured differently depending on whether or not you have
Gatekeeper support on your VOIP network.
If your VOIP network supports Gatekeeper software, you must register all H.323 endpoints
with the Gatekeeper. The procedure for doing this is explained in the section “Registering
with a Gatekeeper Phone Directory”.
If your VOIP network does not have Gatekeeper software or the Gatekeeper software is not
enabled, then you must build a proprietary phonebook with a “Master” MultiVOIP and “Slave”
MultiVOIPs. The “Master” unit includes the assignment of a unique LAN IP address, subnet
mask, and Gateway IP address.
Once configuration of the “Master” MultiVOIP has been completed, the administrator moves
on to configure the MVP120(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 the “Deploying the
VOIP Network” section.
Deploying the VOIP Network
The final phase of the installation is deployment of the network. When the remote 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 Proprietary Phonebook will be loaded into their units within minutes of being
connected and powered up. For remote VOIPs that were configured with the Gatekeeper
option enabled, each MultiVOIP will be registered with the Gatekeeper (the Gatekeeper
phonebook directory is not downloaded to the remote units). The final task of the VOIP
administrator or the Gatekeeper administrator is to develop the VOIP Dialing Directory based
on the appropriate phone directory database (the Proprietary phonebook database or the
Gatekeeper phonebook database).
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Chapter 2 - Installation
Unpacking Your MVP120
www.multitech.com
Remove all items from the box.
Voice/Fax over IP Networks
CL
MADE
IN U.S
.A
U.S.A
MADE IN
Unpacking
Telecom Safety Warning
1.
2.
3.
4.
5.
6.
7.
8.
Never install phone wiring during a lightning storm.
Never install phone jacks in wet locations unless the jacks are designed for wet
locations.
This product is to be used with UL and cUL listed computers.
Never touch uninsulated phone wires or terminals unless the phone line has been
disconnected at the network interface.
Use caution when installing or modifying phone lines.
Avoid using a phone during an electrical storm. There may be a remote risk of electrical
shock from lightning.
Do not use the phone to report a gas leak in the vicinity of the leak.
To reduce the risk of fire, use only No. 26 AWG or larger Telecommunication line cord.
Other Safety Warnings
1.
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.
Caution: Danger of explosion if battery is incorrectly replaced.
2.
The Command port and the Ethernet port are not designed to be connected to a Public
Telecommunication Network.
3.
Use only the power source supplied with your product or an equivalent power source
supplying the minimum power requirements
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Chapter 2 - Installation
Cabling Your MVP120
Cabling your MVP120 involves making the proper Power, Command Port, and Network
connections. The graphic below shows the back panel connectors and the associated cable
connections. The Cabling Procedure section provides step-by-step instructions for cabling
your MVP120.
POWER
ETHERNET COMMAND PHONE
Cable Connections
Cabling Procedure
1.
Using the supplied cable, connect the power supply to a live AC outlet, then plug the
power supply into the MVP120 as shown above.
2.
Connect the MVP120 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 MVP120
and connect the other end to the PC’s serial port.
3.
Connect a network cable to the Ethernet 10Base-T connector on the back of the
MVP120. Connect the other end of the cable to your network.
4.
If you are connecting a PSTN or FXO/PBX port to your MVP120, connect one end of
an RJ-11 cable to the Phone connector on the back of the MVP120 and the other
end to the PSTN or FXO port on the PBX.
5.
Turn on power to the MVP120 by setting the power switch on the right-side panel to
the On position. Wait for the Boot LED on the MVP120 to go off before proceeding.
This may take a couple of minutes.
Note: Since the MVP120 doesn’t have a power LED, no LEDs will be on after
booting unless the Ethernet link is active.
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Voice / Fax over IP Networks
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Chapter 3
Installing the MVP120 Software
Chapter 3 - Installing the MVP120 Software
Chapter 3 - Installing the MVP120 Software
The section covers the software installation. Note that not every screen or option in the
process is included in this chapter. Additional information on the MVP120 software is
provided in Chapter 4 and Chapter 5 and in the online Help.
Note: The phonebook directory configuration process is different depending on whether or
not you have an enabled H.323 Gatekeeper resident in your network. The section on
“Configuring Your MVP120” will explain these differences.
The MVP120 software and User Guide are contained on the MVP120 CD. The CD is autodetectable, so when you insert it into your CD ROM drive it will start up automatically. When
you have finished configuring your MVP120, you can view and print the User Guide by
clicking on the Install Manuals icon.
CAUTION: If you are installing a MVP120 behind a Firewall, the Firewall must support H.323.
Refer to your Firewall user documentation to enable H.323 support.
1.
Make certain that your MVP120 has been properly cabled and that it is powered on.
2.
Insert the MVP120 CD into a CD-ROM drive. The CD is auto-detectable, so it starts
automatically. It may take 10 to 20 seconds for the Multi-Tech Installation CD window
to appear.
If the Multi-Tech Installation CD window does not appear automatically, click My
Computer, then right-click the CD-ROM drive icon, click Open, then click the
Autorun icon.
3.
When the Multi-Tech Installation CD window displays, click the Install Software icon.
You will be prompted to select your software:
H.323 or Proprietary. Select Proprietary.
4.
The Welcome/Setup dialog box displays.
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Chapter 3 - Installing the MVP120 Software
Press Enter or click Next to continue.
5. The Choose Destination Location dialog box displays. Follow the on-screen
instructions.
You can either choose the Destination Location of your MVP120 software or select the
default destination by clicking Next. If you click Browse, you can select a different
destination folder for the MVP120 software.
6.
In the Select Program Folder dialog box, select where you want the program file to be
located.
Verify the path and click Next to continue.
7.
The Copying program files window displays, followed by the MVP120 Setup dialog
box. This dialog box enables you to select the COM port of your PC that is connected to
the Command port of the MVP120. From the Select Port list, choose the COM port of
your PC.
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Chapter 3 - Installing the MVP120 Software
Click OK to continue.
8.
The Setup Complete dialog displays.
Click Finish to continue.
9.
The following message displays:
Click Yes to continue.
10.
The following message displays.
Click Yes to continue.
11.
The IP Protocol Default Setup dialog box displays as you continue now with the
configuring of your MultiVOIP, which is continued in Chapter 4.
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Voice / Fax over IP Networks
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Chapter 4
Configuring the Master MultiVOIP
Chapter 4 - Configuring the Master MultiVOIP
Chapter 4 - Configuring the Master MultiVOIP
The following steps provide instructions for configuring your MVP120. The configuration
sequence includes IP Protocol default setup, Channel setup, and Phone Directory Database
setup. The numbering of steps continues from the previous section.
12.
The IP Protocol Default Setup dialog box displays.
The default Frame Type is TYPE_II. If this does not match your IP network, select SNAP
from the Frame Type list. The available Frame Type choices are TYPE_II and SNAP.
13.
In the Ethernet group, enter the IP Address, Subnet Mask, and Gateway Address
unique to your IP LAN.
The IP address is the unique LAN IP address that is assigned to the MVP120, and the
Gateway address is the IP address of the device connecting your MVP120 to the
Internet.
Click OK when you are finished.
14.
The Channel Setup dialog box displays. The four tabs in this dialog box define the
Channel Interface, Voice/Fax parameters, Billing/Miscellaneous parameters, and
Regional telephone parameters.
Check with your in-house telephone personnel to verify whether your local PBX dial
signaling is Pulse or DTMF (tone). Select the regeneration option accordingly.
In the Inter Digit Time box, enter the maximum amount of time between dialed digits in
milliseconds that the unit will wait before mapping the dialed digits to an entry in the
Phone Directory Database. If too much time elapses between digits, the wrong numbers
will be mapped and you will hear a rapid busy signal. If this happens, hang up and dial
again. The default setting is 2 seconds.
In the Flash Hook Timer box, enter the amount of time the duration of flash hook signals
output on the interface. The default setting is 600 milliseconds.
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15.
The Ring Count FXO box enables you to set the number of rings received on the
FXO interface before hanging up and releasing the line to another call. The default
setting is 2 rings.
Note: Zero (0) means no incoming calls will be answered.
For FXO-to-FXO communications, you can enable a specific kind of FXO
Disconnect Options: Current Loss, Tone Detection, or Silence Detection. Check
with your in-house phone personnel to verify the type of disconnect to use. If current
loss is checked, the VOIP will hang up when it detects a loss of current on the FXO
(phone) port. For tone detection, select from the lists one or two tones that will
cause the line to disconnect. The person hanging up the call must then press the key
or keys that produce those tones to hang up a call. For silence detection, select
One Way or Two Way, then set the timer for the number of seconds of silence before
disconnect. The default value of 15 seconds may be shorter than desired for your
application.
The Message Waiting Light check box must be selected on the originating and
answering voice channel. This enables the number dialed to connect you to the
appropriate voice channel, then output that number on the voice channel.
16.
The Voice/Fax tab displays the parameters for the voice gain, DTMF (Dual Tone
Multi-Frequency) gain, voice coder, faxing, and advanced features such as Silence
Compression, Echo Cancellation, and Forward Error Correction.
17.
You can set up the input and output voice gain so that the volume can be increased
or decreased. Input gain modifies the level of the audio coming into the voice channel
before it is sent over the Internet to the remote MVP120. Output gain modifies the
level of the audio being output to the device attached to the voice channel. Make your
selections from the Input and Output lists in the Voice Gain group. The valid range is
+31dB to –31dB with a recommended/default value of 0.
You can also 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 lists
in the DTMF Gain group. When the DTMF Out of Band check box is selected, the
tones are passed via RTP packets, and the receiving VOIP reproduces the DTMF
tones instead of passing them through like voice.
Note: Only change the DTMF gain under the direction of Multi-Tech Technical
Support.
18.
To change the voice coder, select the Manual option in the Coder group. Select the
appropriate coder from the Selected Coder list.
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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.
Note: If Automatic Coder is selected, then you need to select the Max Bandwidth from
the list. Check with your Network Administrator to determine how much bandwidth is
available.
19.
The Fax group enables you to send/receive faxes on the selected voice/fax channel. You
can set the maximum baud rate for faxes, the fax volume, and change the jitter value in
milliseconds.
When receiving fax packets from a remote MVP120, it is possible for individual packets
to be delayed or received out of order due to traffic conditions on the network. To
compensate for this effect, the MVP120 uses a Jitter Buffer. The Jitter Value box allows
the MVP120 to wait a user-definable period of time, in milliseconds, for delayed or out of
order fax packets. The range of allowable Jitter Values is 0 to 400 with a default of 400
milliseconds.
If you do not plan to send or receive faxes on a given voice/fax channel, you can disable
faxes in the Fax group.
20.
You can enable the voice/fax advanced features by selecting the Silence Compression,
Echo Cancellation, or Forward Error Correction options.
The Silence Compression option defines whether silence compression is enabled for
the voice channel. If Silence Compression is enabled, the MVP120 will not transmit
voice packets when silence is detected, thereby reducing the amount of network
bandwidth that is being used by the voice channel.
If Echo Cancellation is enabled, the MVP120 will remove echo which improves the
quality of sound.
The Forward Error Correction (FEC) feature allows some of the voice packets that
were corrupted or lost to be recovered. FEC adds an additional 50% overhead to the
total network bandwidth consumed by the voice channel.
21.
The Billing/Misc tab displays the parameters for auto call, automatic disconnection,
billing options, and dynamic jitter buffer.
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 number), click the Auto Call Enable option in the
Auto Call group. Then enter the phone number of the remote channel in the Phone
Number box.
22.
The Automatic Disconnection group provides three options to be used singly or in
combination.
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The Jitter Value defines the average inter-arrival packet deviation (in milliseconds)
before the call is automatically disconnected. Jitter is the inter-arrival packet deviation
(phase shift of digital pulses) over the transmission medium that causes voice breakup
which can be particularly disruptive to voice communications. The default is 20
milliseconds. A higher value means that the voice transmission will be more accepting of
jitter. A lower value will be less tolerant of jitter.
Consecutive Packets Lost defines the number of consecutive packets that are lost after
which the call is automatically disconnected. The default setting is 30.
Call Duration defines the maximum length of time (in seconds) that a call remains
connected before the call is automatically disconnected. The default setting is 180
seconds. A call limit of three minutes may be too short for most configurations.
Therefore, you may want to increase this default value.
23.
You can set billing options for inbound and/or outbound calls by checking them in the
Billing Options group and then entering the charge in cents per number of seconds.
24.
A minimum and maximum set of values can be set for Dynamic Jitter Buffer. When
receiving voice packets from a remote MVP120, it is possible to experience varying
delays between packets due to traffic conditions on the network. This is called Jitter. To
compensate for this effect, the MVP120 uses a Dynamic Jitter Buffer. The Jitter Buffer
allows the MVP120 to wait for delayed voice packets by automatically adjusting the
length of the Jitter Buffer between configurable minimum and maximum values. An
Optimization Factor adjustment controls how quickly the length of the Jitter Buffer is
increased when jitter increases on the network. The length of the jitter buffer directly
effects the voice delay between MVP120 gateways.
The Minimum Jitter Value default setting is 150 milliseconds, the Maximum Jitter Value
default setting is 300 milliseconds, and the Optimization Factor default setting is 7.
If your country/region is not the default USA, click the Regional tab and proceed to step
25; otherwise, click OK and proceed to step 26 to begin building your phone directory
database.
25.
To change the Tone Pairs on the Regional tab, select your country or region from the
Country/Region list. If your country or region is not listed, click Custom to define it.
The Tone Pairs group enables you to select and modify the parameters. Click OK when
finished. Proceed to Step 26 to begin building your phone directory database.
26.
The Phone Directory Database dialog box displays with the Proprietary PhoneBook
option enabled and no phone numbers entries displayed in the database. This dialog box
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enables you to select either the GateKeeper or Proprietary PhoneBook. Once you have
choosen the type of Phone Book database, you can proceed to registering with a
Gatekeeper in the following section (entitled, Registering with a Gatekeeper Phone
Directory).
If you are building a proprietary phone book, proceed to page 31. On page 31 the
Building a Proprietary Phonebook Directory section starts with Step 27. It is a
continuation of Step 26 on this page.
Registering with a Gatekeeper Phone Directory
This section describes how to register H.323 endpoints with the Gatekeeper. The H.323
Gatekeeper function resides at a PC acting as the central point for all calls within its zone and
providing call control services to registered endpoints. The Gatekeeper performs two important call
control functions: address translation from LAN aliases to IP addresses, and bandwidth
management where the network manager has specified a threshold for the number of
simultaneous conferences on the LAN.
In a GateKeeper environment, you will be enabling the GateKeeper option, entering an IP address
for the GateKeeper, accepting the default port number, and if the GateKeeper network is servicing
Fast Start, accept the defaults in the Q.931 Parameters group. However, if this network zone is
primarily non-fast start supported, disable Use Fast Start.
27.
Enable the Gatekeeper option
28.
If the GateKeeper network employs Fast Start, then accept the Use Fast Start option
(default). You may have to verify this with the GateKeeper administrator.
29.
Enter the Gatekeeper IP Address in the IP Address box.
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30.
Accept the default Port Number 1719.
CAUTION: The default setting for the Gatekeeper Port Number is 1719. This can be
changed to a different value by the Gatekeeper administrator. If you decide to change the
default Port Number, you must use the same number on the Gatekeeper and all other
H.323 endpoints.
31.
When you are finished with this dialog box, click Add to begin building your phone directory
database. The Add/Edit Phone Entry dialog box displays.
32.
Enter the unique phone number of the local device in the Phone Number box.
33.
Leave the Description box blank.
34.
Enter the Voice Channel number corresponding to the phone number entered. MVP120
will always be Channel 1.
35.
Fill in the H.323 ID / alias box with a description to identify the phone number. For this
example, you could enter “New York Office”.
36.
Enter the IP Address of the MultiVOIP you are currently configuring in the IP Address box.
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37.
Click OK when you are finished and the Phone Directory Database dialog box displays
with the first entry in the window.
38.
Click OK.
39.
The following dialog box displays.
Click OK to download default setup.
40.
After the setup has been written to the MVP120, the unit is rebooted.
41.
Verify that the BOOT LED on the MVP120 is off after the download completes. This may
take several minutes as the MVP120 reboots.
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Building a Proprietary Phonebook Directory
Note: This section starts at Step 27 as it is a continuation from Step 26 on pages 27-28.
27.
To build your proprietary MVP120 Phone Directory (in an H.323 environment without the
Gatekeeper option enabled), you will first need to select the Proprietary Phonebook
option and then configure the “Master” MVP120 and then the “Slave” MVP120s (or other
H.323 endpoints). Configuring the “Slave” MVP120 is discussed later in this chapter.
The MultiVOIP configured as Master contain the proprietary phonebook database. All
ohter MultiVOIPs added to the proprietary phonebook database are designated Slaves.
The master database contains the phone numbers of all H.323 endpoints available for
communication on an IP network. This database is downloaded to each Slave MultiVOIP
as it comes online.
28.
To configure the “Master” MVP120, make certain that the Proprietary Phonebook and
Master options are enabled. The Slave option, Master IP Address, and Send Status
Report to Master will be disabled. The Slave Status button displays the Slave VOIP
Status dialog box used for viewing phone number, IP address, status, and description of
slave units (See “Configuring Your Slave MVP120s” for details). Note: In the Q.931
Parameters group, Use Fast Start is checked for compatibility with other H.323 devices
that support Fast Start Capability.
Click Add to begin building your phone directory database. The Add/Edit Phone Entry
dialog box displays.
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29.
Enter a unique phone number for the local VOIP in the Phone Number box and enter 1 for
the Voice/Fax channel in the Voice Channel box. This number must be 1 for each MVP120.
Note: MVP120 and MVP110 will only have one entry and will be Channel 1.
30.
The Description box is optional, but can be useful in associating the channel to the
extension. 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. Normally the “Master”
MVP120 resides at the entity’s main office; therefore, for this example you could enter a
description such as “New York Office”).
31.
The Station Identification group includes a Hunt Group list. This list enables you to
indicate which Hunt Group you want the phone number to be associated with; or, you can
select NO HUNT if you don’t want this entry to participate in hunting. Hunting is a series of
telephone lines organized in such a way that if the first line is busy the next line is hunted and
so on until a free line is found. For this example, assign the phone entry to HUNT GROUP
#1.
Once you have assigned this entry to a Hunt Group (or NO HUNT), you must enter the IP
Address of the Master MVP120 in the IP Address box.
Note: The Port box becomes active as you begin to enter the IP Address. The entry is the
H.323 industry standard Port value (1720) used to communicate with other H.323 endpoints.
32.
Click OK to return to the Phone Directory Database dialog box. It now includes phone
number, destination details, and description.
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33.
Add all other phone numbers (slave units and standalone units) to the Phone Directory
database. To add a channel of a slave MultiVOIP, click Add and the Add/Edit Phone Entry
dialog box displays.
34.
Enter the phone number for the MultiVOIP in the Station Information group Phone Number
box.
35.
Click inside the Description box and enter a description for the remote MultiVOIP phone
number for the Voice/Fax Channel.
36.
In the Station Identification group, select HUNT GROUP #2 from the Hunt Group list, enter
the London Office’s IP Address (202.056.039.100), and accept the H.323 industry standard
Port value (1720) used to communicate with other H.323 endpoints.
37.
Click 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.
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38.
To configure a stand-alone endpoint (a PC with NetMeeting software), click Add and the
Add/Edit Phone Entry dialog box displays again.
39.
Enter the phone number for the stand-alone endpoint in the Station Information group
Phone Number box. For example, 301.
40.
Click inside the Description box and enter a description for the remote stand-alone phone
number. For example, “Human Resources Desk”.
Note: Because the H.323 endpoint is not a MVP120, the Phone Directory database ignores
the Voice Channel entry. Set the Channel to 1.
41.
In the Station Identification group, select NO HUNT from the Hunt Group list, enter the
Human Resource Desk’s IP Address and accept the H.323 industry standard Port value
(1720) used to communicate with other H.323 endpoints.
Note: This stand-alone was not configured as part of a Hunt Group. However, depending on
your requirements, you could configure a stand-alone to be part of a Hunt Group.
42.
Click OK and you are returned to the Phone Directory Database dialog box which now
includes the stand-alone phone number and related information in the Phone Number list.
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43.
When you have finished, click OK to download the setup configuration to the
MVP120.
44.
Click OK. After the setup is written to the MVP120, the unit reboots.
45.
Verify that the BOOT LED on the MVP120 is off after the download is complete. This
may take several minutes as the MVP120 reboots.
At this time your master MVP120 is configured. Proceed to the “Configuring Your
Slave MVP120s” section.
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Chapter 5
Configuring the Slave MultiVOIPs
Chapter 5 - Configuring the Slave MultiVOIPs
Chapter 5 - Configuring the Slave MultiVOIPs
If the Proprietary Phonebook option on the Phone Directory Database dialog box was
enabled, then you will need to configure all remote H.323 endpoints as “Slave” units. For
example, the MVP120 at the company’s subsidiary office in London would need to be
configured as a “Slave.”
CAUTION: If you are installing a MVP120 behind a Firewall, the Firewall must support H.323.
Refer to your Firewall user documentation to enable H.323 support.
1.
Disconnect the PC from the Command port of the Master MVP120 and connect it to
the Command port on the Slave MVP120.
2.
From your desktop, click Programs I MultiVOIP 100 I MultiVOIP Configuration. The
Main menu displays.
3.
Click IP to display the IP Setup dialog box.
Select the Enable Diffserv box if you have routers that support Diffserv (sometimes
called IP Precedence). This feature gives priority to voice packets so they are not
delayed because of large data files being downloaded.
The default Frame Type is TYPE_II. If this does not match your IP network, select the
Frame Type from the Frame Type list. The Frame Type choices are TYPE_II and
SNAP.
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4.
In 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 the unique IP address that you assign to the MVP120, and the Gateway
Address is the IP address of the device (such as a network router) connected to the Internet/
Intranet.
Click OK when you are finished. The Main menu displays.
5.
From the Main menu, click Voice Channels to display the Channel Setup dialog box. The
Channel Setup dialog box displays. The four tabs in this dialog box define the channel
interface, voice/fax parameters, Billing/Misc parameters, and regional telephone parameters
for the channel.
Check with your in-house telephone personnel to verify whether your local PBX dial signaling
is Pulse or DTMF (tone). Select the Regeneration option accordingly.
The Inter Digit Time option defines the maximum amount of time that the unit will wait before
mapping the dialed digits to an entry in the Phone Directory Database. If too much time
elapses between digits, and the wrong numbers are mapped, you will hear a rapid busy
signal. If this happens, it will be necessary to hang up and dial again. The default is 2
seconds. In the Flash Timer box, enter the time, in milliseconds, for the duration of flash
hook signals output on the interface.
The Message Waiting Light check box must be selected on the originating and answering
voice channel. This enables the number dialed to connect you to the appropriate voice
channel, then output that number on the voice channel. This feature does not work FXS to
FXS.
6.
The Ring Count FXO box enables you to set the maximum number of rings output on the
FXO interface before hanging up and releasing the line to another call. The default setting is
2 rings.
Note: Zero (0) means no incoming calls will be answered.
For FXO-to-FXO communications, you can enable a specific kind of FXO Disconnect
Options: Current Loss, Tone Detection, or Silence Detection. Check with your in-house
phone personnel to verify the type of disconnect to use. If current loss is checked, the VOIP
will hang up when it detects a loss of current on the FXO (phone) port. For tone detection,
select from the lists one or two tones that will cause the line to disconnect. The person
hanging up the call must then press the key or keys that produce those tones to hang up a
call. For silence detection, select One Way or Two Way, then set the timer for the number
of seconds of silence before disconnect. The default value of 15 seconds may be shorter
than desired for your application.
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7.
The Voice/Fax tab displays the parameters for the voice gain, DTMF (Dual Tone MultiFrequency) gain, voice coder, faxing, and advanced features such as Silence Compression,
Echo Cancellation, and Forward Error Correction.
8.
Input gain modifies the level of the audio coming in to the voice channel before it is sent over
the Internet to the remote MVP120. Output gain modifies the level of the audio being output
to the device attached to the voice channel. Make your selections from the Input and Output
lists. The valid range is +31dB to –31dB with a recommended/default value of 0.
You can also 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 lists in the DTMF Gain
group.
The DTMF Out of Band check box is selected so the MVP120 will reproduce the DTMF
tones rather than passing them through with voice. When this option is checked the DTMF is
sent via RTP packets.
Note: Only change the DTMF Gain under direction os Multi-Tech Technical Support.
9.
To change the voice coder, select the Manual in the Coder group. Select the appropriate
coder from the Selected Coder list.
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.
10.
The Fax group enables you to send/receive faxes. You can set the maximum baud rate for
faxes and the fax volume in the two lists and change the jitter value in milliseconds.
When receiving fax packets from a remote MVP120, it is possible for individual packets to be
delayed or received out of order due to traffic conditions on the network. To compensate for
this effect, the MVP120 uses a Jitter Buffer. The Jitter Value box allows the MVP120 to wait
a user-definable period of time, in milliseconds, for delayed or out of order fax packets. The
range of allowable Jitter Values is 0 to 400 with a default of 400 milliseconds.
If you do not plan to send or receive faxes on a given voice/fax channel, you can disable
faxes in the Fax group.
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11.
You can enable the voice/fax advanced features by clicking (checking) the silence
compression, echo cancellation, or forward error correction options.
Silence Compression - Defines whether silence compression is enabled for the voice
channel. If silence compression is enabled, the MVP120 will not transmit voice packets when
silence is detected, thereby reducing the amount of network bandwidth that is being used by
the voice channel.
Echo Cancellation - Defines whether echo cancellation is enabled for this voice channel. If
echo cancellation is enabled, the MVP120 will remove echo-delay which improves the quality
of sound.
Forward Error Correction (FEC) - Defines whether forward error correction is enabled
(checked) for this voice channel. The FEC feature allows some of the voice packets that
were corrupted (or lost) to be recovered. FEC adds an additional 50% overhead to the total
network bandwidth consumed by the voice channel.
The Billing/Misc tab displays the parameters for auto call, automatic disconnection, billing
options and dynamic jitter buffer.
12.
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 MVP120 in the Phone Number box.
13.
The Automatic Disconnection group provides three options to be used singly or in
combination.
The Jitter Value defines the average inter-arrival packet deviation (in milliseconds) before
the call is automatically disconnected. Jitter is the inter-arrival packet deviation (phase shift of
digital pulses) over the transmission medium that causes voice breakup which can be
particularly disruptive to voice communications. The default setting is 20 milliseconds. A
higher value means that the voice transmission will be more accepting of jitter. A lower value
will be less tolerant of jitter.
Consecutive Packets Lost defines the number of consecutive packets that are lost after
which the call is automatically disconnected. The default setting is 30 packets.
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Call Duration defines the maximum length of time (in seconds) that a call remains
connected before the call is automatically disconnected. The default setting is 180 seconds.
A call limit of three minutes may be too short for most configurations. Therefore, you may
want to increase this default value.
14.
You can set billing options for inbound and/or outbound calls by selecting them in the Billing
Options group and then entering the charge in cents per number of seconds.
15.
A minimum and maximum set of values can be set for Dynamic Jitter Buffer. When
receiving voice packets from a remote MVP120, it is possible to experience varying delays
between packets due to traffic conditions on the network. This is called Jitter. To compensate
for this effect, the MVP120 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the MVP120
to wait for delayed voice packets by automatically adjusting the length of the Jitter Buffer
between configurable minimum and maximum values. An Optimization Factor adjustment
controls how quickly the length of the Jitter Buffer is increased when jitter increases on the
network. The length of the jitter buffer directly effects the voice delay between MVP120
gateways.
The Minimum Jitter Value default setting is 150 milliseconds, the Maximum Jitter Value
default setting is 300 milliseconds, and the Optimization Factor default setting is 7.
16.
To change the Tone Pairs on the Regional tab, select your country or region from the
Country/Region list.
Note: If your country or region is not listed, click Custom to define it.
The Tone Pairs group enables you to select/modify the parameters according to choice.
Click OK when finished.
17.
From the Main menu, click Phone Book to display the Phone Directory Database dialog
box. Make certain the Proprietary Phonebook option is enabled and in the Database Type
group, click the Slave option. The Master IP Address field becomes active.
Note: Refer to pages 28-30 if you had chosen Gatekeeper as the type of phone book
database during the master phone book setup.
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Note: After you have enabled the Slave option, the Slave Status button is replaced by the
Update button. Once your Phone Directory database has been established, you can click
this button to refresh the entries in the Phone Directory Database window.
18.
Enter the IP address (204.022.122.118) of the New York Office MVP120 in the Master IP
Address box and enable the Send Status Report to Master so that status reports are sent
to the Master MVP120.
Note: In a Dial-On-Demand (DOD) network, you should leave Send Status Report to
Master disabled. This allows the router to disconnect whenever there is no voice activity.
Note that Slaves with Send Status Report to Master disabled will show up as “Unknown”
when viewing Slave status on the Master.
19.
Click OK to return to the Main menu.
20.
Click Download Setup to write the new configuration to the slave unit. The Save Setup
dialog box displays.
21.
Select the Save Current Setup as User Default Configuration check box and click OK.
The Writing Setup dialog box displays as the setup configuration is written to the MVP120.
22.
Check that the Boot LED on the MVP120 is off after the download is complete. This may
take several minutes as the MVP120 reboots.
23.
You are returned to the Main menu.
Your MVP120 is operational at this time. Repeat the process for each of the slave units. When all
slaves have been configured, go to “Deploying the VOIP Network”.
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Chapter 6
Deploying the VOIP Network
Chapter 6 - Deploying the VOIP Network
Chapter 6 - Deploying the VOIP Network
For a Proprietary Phone Directory database, the VOIP administrator can deploy the preconfigured slave MVP120s to their remote sites. The remote site administrators need only
connect power to the pre-configured MVP120, connect the MVP120 to the Ethernet LAN and
predefined telephone equipment, and then wait for the phone directory database to be
downloaded.
With the Gatekeeper option enabled on the Phone Directory Database dialog box, all MVP120s
are configured as Gatekeeper and cannot be downloaded. In this case, each MVP120 Phone
Book will be programmed with phone numbers for its own channel. The phone number is
registered with the H.323 Gatekeeper. See the “Registering with a Gatekeeper Phone Directory”
section discussed earlier.
Remote Site Administrator
The following steps are for MVP120 H.323 endpoints. For non-MVP120 H.323 endpoints, refer to
the appropriate installation documentation.
1.
Connect one end of the power supply to a live AC outlet and connect the other end to the
Power connection on your MVP120 (See Figure 5).
Vo i c e / F a x C h a n n e l
Connection
P H O N E
ETHERNET POWER
FX X
O
FO
P S T N
Power Connection
Ethernet Connection
Remote Site Cable Connection
3.
Connect a network cable to the ETHERNET (RJ-45) connector on the back of your
MVP120.
4.
If you are connecting an FXO / PBX port or PSTN to your MVP120, connect one end of
an RJ-11 cable to the Phone connector on the back of the MVP120 and the other end to
the FXO or PSTN.
5.
Turn on power to the MVP120 by placing the ON/OFF switch on the right side panel in
the ON position. Wait for the BOOT LED on the MVP120 to go off before proceeding.
This may take a couple of minutes.
6.
At this time your VOIP network should be fully operational. Dial one of the sites in your
network.
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Chapter 7
Using the MultiVOIP Software
Chapter 7 - Using the MultiVOIP Software
Chapter 7 - Using the MultiVOIP Software
This chapter describes various features of the MVP120 software that enable you to change
the configuration of your MVP120. The default configuration parameters were entered during
the loading of the software. The MVP120 software and configuration utilities described in this
chapter enable you to change that initial configuration as necessary.
The primary interface to the MVP120 software is the Main menu (MultiVOIP 100 Setup is on
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 MVP120 Configuration section later in
this chapter.
The MVP120 Configuration (Main menu) utility along with nine other configuration utilities
provide full software functionality for your MVP120.
Configuration Port Setup enables you to change the method by which you access the
MVP120, whether through a direct connection from a PC to the Command Port on the
MVP120 or via your Internet or LAN connection to the LAN port on the MVP120.
Date and Time Setup enables you to easily set the date and time used for data logging in
the MVP120.
Download Factory Defaults enables you to return the configuration to the original factory
settings.
Download Firmware enables you to download new versions of firmware as
enhancements become available.
Download User Defaults enables you to repeat the download user defaults process (part
of software installation) and update the MVP120 configuration with any necessary
changes.
Download Voice Coders enables you to download voice coders to the MVP120 after
repair or upgrade.
Download H.323 Stack enables you to download the H.323 protocols to the MVP120
after repair or upgrade.
Uninstall MVP120 Configuration removes most of the MVP120 software from your PC.
Upgrade Software downloads boot code, factory defaults, firmware, voice coders, and
H.323 stack; and then reboots the MVP120.
The MVP120 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 Help menu bar) displays an Index tab with a
list of entries. Click an entry, then click the Display button to display the text associated with
that topic.
Before You Begin
The MVP120 software operates in a Microsoft Windows environment. The MVP120 program
group contains icons for all the utilities described above. You can open the individual utility
programs by clicking Start | Programs | MultiVOIP 100 | (utility).
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MVP120 Configuration
The MVP120 Setup menu consists of 10 buttons, 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 MVP120 password.
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 MVP120 or is communicating with the Ethernet port. The Rights
box which displays whether you have Read/Write or Read-Only rights. The first user to
communicate with the MVP120 command port has Read/Write rights that enable the user to
view and/or change the configuration of the MVP120. Any additional simultaneous users
have Read-Only rights and can only display the configuration of the MVP120 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,
billing and security, and voice communications for the country and region in which the
MVP120 is operating. The Channel Setup dialog box, accessed by clicking Voice Channels
on the Setup menu, has four tabs that display the following categories of channel information
-- Interface, Voice/Fax, Billing/Misc, and Regional.
Interface Tab
Check with your in-house telephone personnel to verify whether your local PBX dial signaling
is Pulse or DTMF (tone). Select the Regeneration option accordingly.
In the Inter Digit Time box, enter the maximum amount of time between dialed digits in
milliseconds that the unit will wait before mapping the dialed digits to an entry in the Phone
Directory Database. If too much time elapses between digits, the wrong numbers will be
mapped and you will hear a rapid busy signal. If this happens, hang up and dial again. The
default setting is 2 seconds.
In the Flash Hook Timer box, enter the amount of time the duration of flash hook signals
output on the interface. The default setting is 600 milliseconds.
The Ring Count FXO box enables you to set the maximum number of rings output on the
FXO interface before hanging up and releasing the line to another call. The default setting is
2 rings.
Note: Zero (0) means no incoming calls will be answered.
The Message Waiting Light check box must be selected on the originating and answering
voice channel. This enables the number dialed to connect you to the appropriate voice
channel, then output that number on the voice channel.
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For FXO-to-FXO communications, you can enable a specific kind of FXO Disconnect Options:
Current Loss, Tone Detection, or Silence Detection. Check with your in-house phone
personnel to verify the type of disconnect to use. If current lossis checked, the VOIP will hang up
when it detects a loss of current on the FXO (phone) port. For tone detection, select from the
lists one or two tones that will cause the line to disconnect. The person hanging up the call must
then press the key or keys that produce those tones to hang up a call. For silence detection,
select One Way or Two Way, then set the timer for the number of seconds of silence before
disconnect. The default value of 15 seconds may be shorter than desired for your application.
Voice/Fax Tab
The Voice/Fax tab controls voice and DTMF gain, voice coder, fax settings, and advanced
options.
In the Voice Gain group you can select the Input and Output voice gain. Gain is the increased
signaling power that occurs as the signal is boosted by the MVP120. The Input Gain list defines
the input gain for this voice channel. Before your MVP120 digitizes voice, the volume can be
increased or decreased. Input gain modifies the level of the audio coming into the voice channel
before it is sent over the Internet to the remote MVP120. The valid range for this option is +31dB
to –31dB. The recommended and default value is 0. The Output Gain list defines the voice
output gain for this voice channel. Before your MVP120 converts digital voice back to analog, the
volume can be increased or decreased. The output gain modifies the level of the audio being
output to the device attached to the voice channel. The valid range for this option is +31dB to –
31dB. The recommended and default value is 0.
The DTMF Gain (Dual Tone Multi-Frequency) group controls the volume level of the digital tones
sent out for Touchtone dialing. The Gain High and Gain Low lists 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.
The DTMF Out of Band check box is selected so the MVP120 will reproduce the DTMF tones
rather than passing them through with voice. When this option is checked the DTMF is sent via
RTP packets.
The MVP120 supports many state-of-the art ITU (International Telecommunications Union) voice
coders. The Voice Coder list 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 has to have the same voice coder selected. Otherwise, you will always get a Busy signal.
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The Fax group enables a fax machine to transmit and receive faxes through the MVP120. If a
fax machine is connected to the via PBX port or PTSN, the Max Baud Rate should be set to
match the baud rate of the fax machine (refer to user documentation). The Fax Volume setting
controls the output level of the fax tones, and this setting should be changed only under the
direction of Multi-Tech’s Technical Support personnel. The Jitter Value setting defines the interarrival packet deviation (in milliseconds) for the fax transmission. A higher value will increase
the amount of delay (allowing for a higher percentage of packets to be reassembled) and a
lower value would decrease the amount of delay (a lower percentage of packets would be
reassembled).
The Advanced Features group allows you to enable Silence Compression so that a MVP120
will not transmit voice packets when silence is detected, thereby reducing the amount of
network bandwidth that is being used by the voice channel; Echo Cancellation for the voice
channel will remove echo and improve the quality of sound; and, Forward Error Correction
allowing some of the voice packets that were corrupted (or lost) to be recovered. FEC adds an
additional 50% overhead to the total network bandwidth consumed by the voice channel.
Billing/Misc Tab
This tab controls the parameters for auto call, automatic disconnection, billing options, and
dynamic jitter buffer.
The Auto Call option allows the local MVP120 to call a remote MVP120 without the user
having to dial a Phone Directory Database number. As soon as you access the local MVP120
voice/fax channel, the MVP120 immediately connects to the remote MVP120 that you identified
in the Remote MVP120 Phone Number field of this option.
The Automatic Disconnection group provides three options which can be used singly or in
any combination. The Jitter Value defines the average inter-arrival packet deviation (in
milliseconds) before the call is automatically disconnected.The default is 20 milliseconds. A
higher value means voice transmission will be more accepting of jitter. A lower value is less
tolerant of jitter.
Consecutive Packets Lost defines the number of consecutive packets that are lost after
which the call is automatically disconnected. The default is 30 packets.
Call Duration defines the maximum length of time (in seconds) that a call remains connected
before the call is automatically disconnected. The default is 180 seconds. A call limit of three
minutes may be too short for most configurations. Therefore, you may want to increase this
default value.
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Billing Options can be used to track the cost of Inbound and/or Outbound calls on the FXO
interface. The amount to be charged in cents is entered in the Charge ( ) Cents box together
with the associated time duration in the Per ( ) Seconds box. 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 selecting the call event in the Log
Entries dialog box and selecting Details.
Dynamic Jitter Buffer defines a minimum and a maximum jitter value for voice communications.
When receiving voice packets from a remote MVP120, it is possible to experience varying delays
between packets due to traffic conditions on the network. This is called Jitter. To compensate for
this effect, the MVP120 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the MVP120 to wait
for delayed voice packets by automatically adjusting the length of the Jitter Buffer between
configurable minimum and maximum values. An Optimization Factor adjustment controls how
quickly the length of the Jitter Buffer is increased when jitter increases on the network. The length
of the jitter buffer directly effects the voice delay between MVP120 gateways.
The default minimum dynamic jitter buffer of 150 milliseconds is the minimum delay that would be
acceptable over a low jitter network. The default maximum dynamic jitter buffer of 300
milliseconds is the maximum delay tolerable over a high jitter network.
The Optimization Factor determines how quickly the length of the Dynamic Jitter Buffer is
changed based on actual jitter encountered on the network. Selecting the minimum value of 0
means low voice delay is desired, but increases the possibility of jitter induced voice quality
problems. Selecting the maximum value of 12 means highest voice quality under jitter conditions
is desired at the cost of increased voice delay.
The Optimization Factor can be configured in the range of 0 to 12 with a default setting of 7.
Regional Tab
The Regional tab controls the voice communications for the country or region in which the
MVP120 is being used.
From the Country/Region list, select the country or region for which you are configuring the
MVP120. 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) . 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 MultiVOIP provides two phone directory database architectures; the propreitary database
and a database using an H.323 protocol gatekeeper that provides a centralized call control
center. The gatekeeper centralized call control center contains the phone directory database
when all VOIP gateways and endpoints support the H.323 protocol. The proprietary database is
based on a master and slave relationship in which the master VOIP maintains the phone
directory database and distributes it to its slave VOIPs. The centralized call center handles call
control, call routing, address translation from LAN aliases to IP addresses, and bandwidth
management. The H.323 protocol allows other third party gateways and end points that support
the H.323 protocol standards to participate in the VOIP network (e.g., Microsoft NetMetting®).
The database displays the phone numbers in numerical order with destination details (IP Address
or H323 ID \ alias), channel assignment, and a brief description of the entry. The method for
changing the phone directory database is dependent on whether the Gatekeeper option or the
Proprietary Phonebook option is enabled.
If the GateKeeper option is enabled, the RAS Parameters group is enabled and the IP address
of the GateKeeper needs to be enterred in the IP Address window. The Port Number is the port
on the GateKeeper which it communicates with its endpoints. The Q.931 Parameters group is
enabled in both the GateKeeper and Proprietary Phone Book Database architectures. The Use
Fast Start option is used when the VOIP network supports Fast Start capability.
In the GateKeeper phone directory database, the phone directory database is developed through
the Add/Edit Phone Entry dialog box. The Add/Edit Phone Entry dialog box defines the Station
Information, phone number and voice channel of the unit, and station identification, H323 ID /
alias which defines the LAN alias and the IP address of the local unit. In the GateKeeper phone
directory database, only the phone entry of the local unit displays.
If the Proprietary PhoneBook option is enabled, the Database Type group is active which
defines the Master and Slave relationship. If the database type is master, then the Add, Delete,
Edit, Hunt, and Print buttons at the top of the database dialog box are active. This allows the
master database to build the phone directory. You can click Slave Status to view the status of
the slave units.
The Add/Edit Phone Entry dialog box for the Proprietary PhoneBook is used to add, delete, and
edit entries. This information displays in the Phone Directory Database dialog box.
If the database type is set to slave, then the IP address of the master MultiVOIP needs to be
entered in the Master IP Address window, the Send Status Report to Master option can be
enabled, and all the buttons at the top of the directory database dialog box become inactive,
except for the Print button.
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Proprietary Phone Directory Database
In the Proprietary Phone Directory Database, you can add, delete, or edit any entry in the
phone directory database and you can set up Hunt groups that locate another phone number
if the called number is busy. You can print the phone directory database so that you have a
hardcopy of the phone directory.
To add an entry to the Phone Directory database, click Add and the Add/Edit Phone Entry
dialog box displays.
The Add/Edit Phone Entry dialog box contains two groups of information: the Station
Information which contains the phone number, an optional description window, and the
voice channel number. The Station Identification group contains the Hunt Group listing
and the IP Address window for the IP Address of the MultiVOIP assigned the phone number.
The Port number is not used in the proprietary phone book.
The Station Information identifies the calling unit by the phone number and voice channel of
the unit doing the calling. The Phone Number does not have to be a conventional telephone
number. It can be, for example, a three digit number like 101.
The Description is similar to a local telephone book listing. It identifies the calling party. The
voice channel window defines the voice channel associated with the phone number.
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The Station Identification group enables you to assign the entry to a Hunt Group, provide
the IP Address of the MultiVOIP being assigned the phone number, and accept the H.323
industry standard Port number. A Hunt Group is a series of telephone lines organized in such
a way that if the first line is busy the next line is hunted and so on until a free line is found. It
is a set of links which provides a common resource and which is assigned a single hunt
group designation. A user requesting that designation may then be connected to any member
of the Hunt Group.
You can view the details of the current Hunt Group configuration by clicking the Phone
Directory Database’s Hunt button. The current Phone numbers for HUNT GROUP #1 are
displayed.
Select the Hunt Group you wish to view. The Phone no’s window displays the telephone
numbers associated with that Hunt Group and the No. of Entries field displays the running
total of entries.
Note: You can change the name of the Hunt Group by clicking on the entry that you want to
change, editing the change in the Hunt Group name window, and then clicking the Set button.
Click Slave Status on the Phone Directory Database dialog box to view the status of all the
slave units in your VOIP network (Send Status Report to Master must be enabled on the
Slave). The Phone Number of each Slave displays with its IP Address, current line status,
and the description of the phone number.
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Gatekeeper Phone Directory Database
With the Gatekeeper Phone Directory Database, the Gatekeeper acts as the central point
for all calls within its zone and provides call control services to registered endpoints. The
Gatekeeper performs address translation from LAN aliases to IP addresses and provides
bandwidth management where the network manager has specified a threshold for the number
of simultaneous calls on the LAN. The H.323 ID is an alias. The Gatekeeper may use other
information as an alias such as a URL or an e-mail address.
When the Gatekeeper Phone Directory Database option is enabled, the RAS Parameters
group is enabled with the IP address of the Gatekeeper displayed in the IP Address box. The
Port Number is the port of the endpoint communicating with the Gatekeeper. If this number is
changed, it should only be changed with consultation with Gatekeeper administrator. The port
numbers have to match with the Gatekeeper.
If the H.323 Gatekeeper network supports Q.931 Fast Start servicing, then the Use Fast Start
option on all endpoints should be enabled. The Call Signalling Port of 1720 is the port on the
MultiVOIP unit supporting the Q.931 parameters.
The Phone Directory Database in a H.323 Gatekeeper network only displays the station
information and station identification of the local unit. The station information and identification
must be entered in conjunction with the Gatekeeper administrator so that the identification of
the endpoint is the same. To add an entry to the Database, click Add. The Add/Edit Phone
Entry dialog box displays.
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The Add/Edit Phone Entry dialog box contains two groups of information: 1. Station
Information which contains the phone number, description window which can be left blank,
and the voice channel number. Station Identification group contains the H.323 ID and the IP
Address window for the IP Address of the MultiVOIP assigned the phone number, and the
Port number of the MultiVOIP unit communicating with the Gatekeeper. This port number has
to match the port number pair used by the Gatekeeper. If the port number on either end is
changed, communication between endpoint and the Gatekeeper is lost.
The Station Information identifies the calling unit by the phone number, and voice channel
of the unit doing the calling. The Phone Number does not have to be a conventional
telephone number. It can be a three digit number such as 101. The Description box and the
H323 ID/alias box may contain the same information. It is recommended that the
Description box be left blank and the identifying information be enterred in the H323 ID/alias
window. The voice channel box defines the voice channel associated with the phone number.
The Station Identification group allows you to define the MultiVOIP unit by entering the
H323 ID/alias which can be the same, as for example, your name in your local telephone
book listing. The Gatekeeper associates the H323 ID/alias with the address of the local unit
in the IP Address window. The Port number 1720 is the port of the MultiVOIP communicating
with the Gatekeeper.
The Station Information and Identification of the MultiVOIP unit have to be identical to the
information used by the Gatekeeper in order for the MultiVOIP unit to be registered with the
Gatekeeper. The Gatekeeper can allow an open registration or a secure registration in which
the endpoints are pre-defined by the Gatekeeper. The registration method is determined by
the Gatekeeper administrator and will require communication with each endpoint in order to
develop the H.323 compatible network.
The Phone Directory Database for the local unit holds the local phone numbers, destination
details of the IP address, port number, and channel number of the local unit. The description
is the same as the entry in the H323 ID entry in the Add/Edit Phone Entry dialog box.
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Changing IP Parameters
The IP Setup dialog box establishes the IP address for the local Ethernet LAN and defines the
Internet gateway address. The IP Setup dialog box is accessed by clicking IP on the MVP120
Main menu.
With IP Setup dialog box displayed you can change the status of Diffserv, the Ethernet Frame
Type, the IP address and IP Mask of your H.323 endpoint, and the Gateway Address of the IP
address of the device connected to the Internet.
Selecting the Enable Diffserv check box enables Diffserv. Diffserv provides priority to voice
packets so that they are not delayed whenever large data files need to be downloaded. Check
with your systems administrator to determine if any routers in the VOIP network support this
feature. If so, enable this function.
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The Frame Type list enables you to change the Ethernet Frame Type so that it matches your
IP network. If the current entry does not match your IP network, select the Frame Type from
the Frame Type list. The Frame Type choices are TYPE_II and SNAP.
The Port Address group enables you to change the unique IP Address and IP Mask of the
local LAN.
The Gateway Address group enables you to change the gateway IP Address of the device
connected to the Internet/Intranet.
Viewing Call Progress
The Call Progress dialog box is a read-only screen that displays the status of a call in
progress. This dialog box is accessed from the MVP120 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-bitrate 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 MVP120 to
recover many of the lost packets.
The Jitter value (measured in milliseconds) 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 Others on the Setup menu displays the Applications Setup dialog box. This dialog
box allows you to enable SNMP Agent (the default is disabled) and set up all the necessary
parameters; enable or disable various remote configuration methods such as TFTP (Trivial
File Transfer Protocol) Server, Web Server, Dumb Terminal Management, and Telnet Server;
and assign a Password to the MVP120 for security. These applications enable remote
viewing and changing of the MVP120 configuration, or updating firmware, from anywhere on
the connected network.
Verify that the desired applications are selected. All applications are selected by default. To
disable a given application, clear its check box.
SNMP related operations can be performed only when the SNMP Agent check box is
selected on this dialog box. The IP address of the system (i.e., SNMP Manager) that will
receive the Traps from the MVP120 should be entered in the IP Address field in the Trap
Manager group. The Community Name 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 MVP120 currently supports a maximum of two
community users at a time, and they can be assigned either Read/Write or Read Only rights.
Note: if you have SNMP client software and enable SNMP Agent, you can (in the lower right
corner of the dialog box) choose to read logs through the SNMP Manager instead of the
COM port.
The Password group enables you to enter a password, up to 13 alphanumeric characters, to
be used for security. Once the password is entered and confirmed, remote users will be
queried to enter the password before gaining access to the MVP120 for configuration
purposes.
Note: If you forget your password, contact Technical Support for instructions.
For more information on using these applications, 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 MVP120
operation. This dialog box is accessed by clicking Statistics on the MVP120 Main menu.
The Voice Channel statistics shows the attempted and completed calls, call duration, average
call length, bytes/packets sent and received. These statistics can be a helpful troubleshooting
tool.
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), 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.
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 MVP120’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: (1) the devices that are to
be managed, (2) agents, and (3) 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 a chronological history of all calls into and out of this unit.
This dialog box is opened by clicking Logs 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 MVP120. 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.
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 Generate.
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 MVP120 software, you must first install the new
software on your PC. Then use the Upgrade Software function, the Factory Defaults, or
download the Firmware, the Voice Coders, and the H.323 stack to upgrade the MVP120 itself.
Before starting the upgrade process, view the current configuration and write down important
data such as your IP address, and voice channel configurations; these settings must be put back
in place after the software has been upgraded. Multi-Tech also recommends that you use the
Print button on the Phone Directory Database dialog box to have a copy of the phone directory
contents.
Upgrade Software
Note: The software upgrade can be done from the command port or via IP utilizing tftp.
If you have obtained a new software version from the Multi-Tech Web site or another source, do
the following:
1. Install the software by clicking Programs | MultiVOIP 100 | Upgrade Software.
2. Click Yes. The download process begins.
3.
Enter your LAN IP address, mask, and gateway address in the IP Protocol Default Setup
dialog box. Click OK when finished.
4.
Enter your current configuration in the Channel Setup dialog box.
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5.
Click OK when finished. The Checking MultiVOIP dialog box displays. When you click OK in
this box, theThe MultiVOIP firmware, voice coders, and H.323 stack are downloaded, and then
the MVP120 reboots.
Upgraded versions of the Boot Code (firmware, coders, and H.323 stack) can be downloaded
individually using the following manual procedures.
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Manual Upgrade Procedure
Follow this procedure only if you want to manually upgrade your MultiVOIP.
Note: Steps 4-8 may be performed via the command port or over IP.
1.
Run MultiVOIP Configuration from your old version of the MultiVOIP software. Note the
current settings. Your MultiVOIP will be reset to factory defaults during this upgrade.
2.
Unistall your old version of MultiVOIP software by selecting the Unistall MultiVOIP
Configuration option from the program group.
3.
Install the H.323 compatible software from the MultiVOIP CD.
4.
a. From the program group of the MultiVOIP software, select Download Firmware.
b. Enter “*.upg” in the File name box and press ENTER.
c. Select the file named boot_100.upg, if you are using the command port.
OR
Select tftp.upg if you are using IP.
5.
From the program group of the MultiVOIP software, select Download Factory Defaults.
6.
From the program group of the MultiVOIP software, select Download Firmware. Select
the file named mtvoip.bin.
7.
From the program group of the MultiVOIP software, select Download Voice Coders.
Select the default file.
8.
From the program group of the MultiVOIP software, select Download H323 Stack.
Select the default file.
9.
Select MultiVOIP Configuration from the program group and reconfigure your MultiVOIP.
The upgrade is complete.
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Chapter 8
Remote Configuration and Management
Chapter 8 - Remote Configuration and Management
Chapter 8 - Remote Configuration and Management
This chapter provides procedures for viewing or changing the configuration of a remote unit. Two
methods are provided to access a remote unit; the first method is modem based and the second
method uses IP. Within the IP method, three applications can be used: 1) LAN-Based using
TFTP (Trivial File Transfer Protocol), 2) Telnet as a client application, or 3) a standard web
browser on the Internet.
Remote configuration requires the MVP120 software to be loaded on the local PC. The local PC
then controls the remote MVP120 either via the modem connection or the LAN.
Modem-Based
To remotely configure a MVP120, a local PC with a modem needs to be connected to a dial-up
line and the MVP120 software configured to call the remote MVP120. The remote MVP120 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 required. Once the configuration is
changed, you can download the new configuration to the remote MVP120. Refer to the ModemBased Remote Configuration Procedure in this chapter to remotely configure a MVP120.
1.
At the remote site, remove the serial cable from the PC to the Command Port connector
on the back panel of the MVP120.
2.
At the remote site, connect a special cable (Remote Configuration Cable) from the
Command Port connector on the back panel of the MVP120 to the RS-232 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.
3.
At the main site, connect your local PC to a modem that is connected to a dial-up line.
4.
Install the MVP120 software on the local PC. When installed, click Start | Programs |
MultiVOIP 100 | Configuration Port Setup, or double-click the Configuration Port icon
in the MultiVOIP 100 program group.
5.
The MultiVOIP 100 Setup dialog box displays.
Verify that the Communication Type is set for COM Port and the COM port of your local
PC is selected in the Select Port list.
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Chapter 8 - Remote Configuration and Management
In the Dial String box, enter the AT command for dialing (ATDT) plus the phone number of
the remote MVP120.
If your Modem Initialization String, Initialization Response, or Connect Response values are
different from 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.
6.
Run the MVP120 Configuration program. Click Start | Programs | MultiVOIP 100 |
MultiVOIP Configuration, or double-click the MultiVOIP Configuration icon in the
MultiVOIP 100 program group.
7.
The Dialing dialog box displays while software is dialing the remote MultiVOIP.
8.
The Reading Setup dialog box displays.
9.
The MultiVOIP 100 Setup menu displays. This is the dialog box of the remote MultiVOIP.
Refer to the online Help for a description of the user interface.
10.
After 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.
11.
Click Exit when downloading is complete.
12.
The Hangup connection? dialog box displays.
Click Yes to disconnect the phone connection to the remote site.
13.
If the same telephone number is not going to be used again in the immediate future, you can
remove it from the Port Setup dialog box.
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Chapter 8 - Remote Configuration and Management
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.
1.
You 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. It is recommended that you ping each MultiVOIP to verify
the connection and its IP address.
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.
2.
Install the MVP120 software on the local PC. When installed, click Start | Programs |
MultiVOIP 100 | Configuration Port Setup, or double-click the Configuration Port Setup
icon in the MultiVOIP 100 program group.
3.
The MultiVOIP Port Setup dialog box displays.
Verify that IP is set for the Communication Type.
In the MultiVOIP IP Address box, enter the IP Address of the remote MultiVOIP.
4.
Click OK when you are satisfied with your selections.
5.
Run the MultiVOIP Configuration program. Click Start | Programs | MultiVOIP 100 |
MultiVOIP Configuration, or double-click the MultiVOIP Configuration icon in the
MultiVOIP 100 program group.
6.
The MultiVOIP 100 Setup dialog box displays. 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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Chapter 8 - Remote Configuration and Management
7.
After 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.
8.
Click Exit when downloading is complete.
9.
Double-click the MultiVOIP Configuration icon in the MultiVOIP 100 program group to
verify that the MultiVOIP is running.
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Chapter 8 - Remote Configuration and Management
Telnet
A typical Telnet client application is described next. The MVP120 has a built-in Telnet Server that
enables Telnet client PCs to access the MVP120. A typical Telnet client is allowed to configure the
MVP120. In addition, the MVP120 can be remotely accessed and configured from anywhere on the
Internet through its Web interface.
The Telnet Server option has to be selected from the Applications Setup dialog box using the
MultiVOIP 100 Configuration icon. Click Start I Run and enter telnet in the Run window and a blank
Telnet screen displays. Click Connect | Remote System and the Connect dialog box displays.
Select from the drop-down list (or enter) a Host Name (the IP address of the MultiVOIP). Then select
Telnet from the Port list and vt100 from the Term Type list.
Once you have entered a valid Host Name (IP address), Port, and Term Type, click Connect to be
connected to the target MultiVOIP. The MultiVOIP Telnet Server menu displays.
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.
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).
Phone Directory Database
Selecting Option 2 enables you to add entries to the Phone Directory Database.
Phone Directory Configuration
Selecting Option 3 enables you to configure and manage the Phone Directory.
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Chapter 8 - Remote Configuration and Management
Web Management
The MVP120 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
Applications Setup dialog box (see Chapter 4 - MVP120 Software).
Once enabled, you can access the MVP120 by entering its IP address in your web browser. The
following page displays.
If a Password was entered in the Applications Setup dialog box, then enter the password and the
Enter.
From this page you can access all the configuration options. Refer to Chapter 4 - MVP120 Software,
for a description of the configuration options.
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Chapter 9
Warranty, Service, and Technical Support
Chapter 9 - Warranty, Service, and Technical Support
Chapter 9 - Warranty, Service, and Technical Support
This chapter starts out with statements about your MVP120 2-year warranty. The next section, Tech
Support, should be read carefully if you have questions or problems with your MVP120. It includes
the technical support phone numbers, space for recording your product information, and an
explanation of how to send in your MVP120 should you require service. The final section explains
how to receive support from the Internet.
Multi-Tech Systems, Inc. Warranty & Repairs Policies
Warranty
Multi-Tech Systems, Inc., (hereafter “MTS”) warrants that its products will be free from defects in
material or workmanship for a period of two, five, or ten years (depending on model) from date of
purchase, or if proof of purchase is not provided, two, five, or ten years (depending on model) from
date of shipment.
MTS MAKES NO OTHER WARRANTY, EXPRESS 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 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 PRICE FOR DEFECTIVE PRODUCTS.
Repair Procedures for U.S. and Canadian Customers
In the event that service is required, products may be shipped, freight prepaid, to our Mounds View,
Minnesota factory:
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, MN 55112
Attn: Repairs, Serial # ____________
A Returned Materials Authorization (RMA) is not required. Return shipping charges (surface) will be
paid by MTS.
Please include, inside the shipping box, a description of the problem, a return shipping address (must
have street address, not P.O. Box), your telephone number, and if the product is out of warranty, a
check or purchase order for repair charges.
For out of warranty repair charges, go to www.multitech.com/documents/warranties
Extended two-year overnight replacement service agreements are available for selected products.
Please call MTS at (888) 288-5470, extension 5308 or visit our web site at
http://www.multitech.com/programs/orc/ for details on rates and coverage’s.
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Chapter 9 - Warranty, Service, and Technical Support
Please direct your questions regarding technical matters, product configuration, verification
that the product is defective, etc., to our Technical Support department at (800) 972-2439 or
email tsupport@multitech.com. Please direct your questions regarding repair expediting,
receiving, shipping, billing, etc., to our Repair Accounting department at (800) 328-9717 or
(763) 717-5631, or email mtsrepair@multitech.com.
Repairs for damages caused by lightning storms, water, power surges, incorrect installation,
physical abuse, or user-caused damages are billed on a time-plus-materials basis.
Repair Procedures for International Customers (Outside U.S.A. and
Canada)
Your original point of purchase Reseller may offer the quickest and most economical repair
option for your Multi-Tech product. You may also contact any Multi-Tech sales office for
information about the nearest distributor or other repair service for your Multi-Tech product.
http://www.multitech.com/COMPANY/offices/DEFAULT.ASP
In the event that factory service is required, products may be shipped, freight prepaid to our
Mounds View, Minnesota factory. Recommended international shipment methods are via
Federal Express, UPS or DHL courier services, or by airmail parcel post; shipments made by
any other method will be refused. A Returned Materials Authorization (RMA) is required for
products shipped from outside the U.S.A. and Canada. Please contact us for return
authorization and shipping instructions on any International shipments to the U.S.A. Please
include, inside the shipping box, a description of the problem, a return shipping address
(must have street address, not P.O. Box), your telephone number, and if the product is out of
warranty, a check drawn on a U.S. bank or your company’s purchase order for repair
charges. Repaired units shall be shipped freight collect, unless other arrangements are
made in advance.
Please direct your questions regarding technical matters, product configuration, verification
that the product is defective, etc., to our Technical Support department nearest you or email
tsupport@multitech.com. When calling the U.S., please direct your questions regarding
repair expediting, receiving, shipping, billing, etc., to our Repair Accounting department at
+(763) 717-5631 in the U.S.A., or email mtsrepair@multitech.com.
Repairs for damages caused by lightning storms, water, power surges, incorrect installation,
physical abuse, or user-caused damages are billed on a time-plus-materials basis.
Repair Procedures for International Distributors
Procedures for International Distributors of Multi-Tech products are on the distributor web
site.
http://www.multitech.com/PARTNERS/login/
Copyright ã Multi-Tech Systems, Inc. 2001
10-Sep-01
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Chapter 9 - Warranty, Service, and Technical Support
Service
If your tech support specialist decides that service is required, your MVP120 may be sent (freight
prepaid) to our factory. Return shipping charges will be paid by Multi-Tech Systems.
Include the following with your MVP120:
• 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 MVP120 is out of warranty (Check
with your technical support specialist for the standard repair charge for your MVP120)
• 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 MVP120 to this address:
MULTI-TECH SYSTEMS, INC.
2205 WOODALE DRIVE
MOUNDS VIEW, MINNESOTA 55112
ATTN: SERVICE OR REPAIRS
You should also check with the supplier of your MVP120 on the availability of local service and/or
loaner units in your part of the country.
Ordering Accessories
SupplyNet, Inc. supplies replacement transformers, cables and connectors for select Multi-Tech
products. You can place an order with SupplyNet via mail, phone, fax or the Internet at:
Mail:
SupplyNet, Inc.
614 Corporate Way
Valley Cottage, NY 10989
Phone: 800 826-0279
Fax:
914 267-2420
Email: info@thesupplynet.com
Internet:
http://www.thesupplynet.com
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Chapter 9 - Warranty, Service, and Technical Support
Technical 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 1800-972-2439. Please fill out the MVP120 information (below), and have it available when you
call. If your MVP120 requires service, the tech support specialist will guide you on how to send it
in (refer to the next section).
Recording MVP120 Information
Please fill in the following information on your Multi-Tech MVP120. 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 MVP120.
Please note status of your MVP120 including LED indicators, screen messages, diagnostic test
results, problems with a specific application, etc. Use the space below to note the MVP120
status:
________________________________________________________________________________________________________
________________________________________________________________________________________________________
________________________________________________________________________________________________________
______________________________________________________________________________________________________________
______________________________________________________________________________________________________
About the Internet
Multi-Tech’s presence includes a Web site at:
http://www.multitech.com
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Appendixes
Appendix A - TCP/IP Description
Appendix A - TCP/IP Description
TCP/IP (Transmission Control Protocol/Internet Protocol) 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 official 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
non-hierarchical; 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
(connection-oriented) 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.
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Appendix A - TCP/IP Description
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
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
1157 ( ftp://ds.internic.net/rfc/rfc1157.txt) is widely deployed in local and wide area network.
SNMP Version 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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Appendix A - TCP/IP Description
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
Appendix B - Cabling Diagrams
Command Port Cable
1 2 3 4 5 6 7 8
RJ-45
DB9F
1
4
2
7
3
8
CLEAR TO SEND
4
3
TRANSMIT DATA
5
2
RECEIVE DATA
6
6
7
1
8
5
PIN NO.
To Command
Port Connector
PIN NO.
To DTE
Device
(PC)
SIGNAL GROUND
LAN Cable
1 2 3 4 5 6 7 8
Pin
Circuit Signal Name
1
2
3
6
TD+ Data Transmit Positive
TD- Data Transmit Negative
RD+ Data Receive Positive
RD- Data Receive Negative
Phone Connector
2 3 4 5
Pin Connections
FXO
Description
2
3
4
5
N/C
Tip
Ring
N/C
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Appendix B - Cabling Diagrams
Remote Configuration Cable
9-Pin Male
To Command port
adapter cable
25-Pin Male
To DCE Device
(e.g., modem)
Receive Data (RX)
2
2
Receive Data (RX)
Transmit Data (TX)
3
3
Transmit Data (TX)
Signal Ground
5
7
Signal Ground
Clear to Send (CTS)
8
4
Request to Send (RTS)
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Appendix C - Regulatory Information
Appendix C - Regulatory Information
Class A Statement
FCC Part 15
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 fax
branding.)
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.
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Appendix C - Regulatory Information
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 and ringer equivalence number (REN). If
requested, 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. An FCC-compliant telephone cord and
modular plug is provided with this equipment. This equipment is designed to be
connected to the telephone network or premises wiring using a compatible modular jack
which is Part 68 compliant. See installation instructions for details.
3.
The ringer equivalence number (REN) is used to determine the number of devices which
may be connected to the telephone line. Excessive REN’s on the telephone line may
result in the devices not ringing in response to an incoming call. In most, but not all
areas, the sum of the REN’s should not exceed five (5.0). To be certain of the number of
devices that may be connected to the line, as determined by the total REN’s, contact the
telephone company to determine the maximum REN for the calling area.
4.
If this equipment causes harm to the telephone network, the telephone company will
notify you in advance that temporary discontinuance of service may be required. But if
advance notice isn’t practical, the telephone 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 telephone company may make changes in its facilities, equipment, operations, or
procedures that could affect the operation of the equipment. If this happens, the
telephone 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 telephone network, the
telephone 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.
This equipment cannot be used on public coin service provided by the telephone
company. Connection to Party Line Service is subject to state tariffs. (Contact the state
public utility commission, public service commission or corporation commission for
information.)
9.
If so required, this equipment is hearing-aid compatible.
Manufacturer:
Trade name:
Model Numbers:
FCC Registration Number:
Modular Jack (USOC):
Service Center in U.S.A.:
Multi-Tech Systems, Inc.
MultiVOIP
MVP120
AU7USA-25715-DF-N
RJ-11C or RJ-11W
Multi-Tech Systems Inc.
2205 Woodale Drive
Mounds View, MN 55112
(763) 785-3500 Fax (763) 785-9874
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Appendix C - Regulatory Information
Canadian Limitations Notice
Ringer Equivalence Number
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.
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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Voice / Fax over IP Networks
100
Glossary
Glossary
Glossary
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
compressed 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 distribution 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.
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Glossary
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).
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
information. 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.
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Glossary
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.
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.
CODEC (COmpression/DEcompression): The term is used to describe the conversion of voice signals from their analog form to
digital signals acceptable to modern digital PBXs and digital transmission systems. It then converts those digital signals back to
analog so that you can hear and understand what the other person is saying. In some phone systems, the CODEC is in the PBX
and shared by many analog phone extensions. In other phone systems, the CODEC is actually in the phone. Thus the phone itself
sends out a digital signal and can, as a result, be more easily designed to accept a digital RS-232-C signal.
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 user data 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 superframe. 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".
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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 interLATA 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").
Digital Loopback: A technique used for testing the circuitry of a communications device. Can be initiated locally, or remotely (via a
telecommunications 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
modulation).
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
intermediate 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
E1: The European equivalent of the North American 1.544M bps T-1, except that E-1 carries 2.048M bps. It is characterized by
thirty 64 Kbps digital channels for voice or data calls, plus a 64 Kbps channel for signaling and a 64 Kbps channel for framing
(synchronization) and maintenance.
E&M: A telephony trunking system used for either switch-to-switch, or switch-to-network, or computer/telephone system-to-switch
connection.
Glossary
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.
Endpoint: The clients in an H.323 network. They are typically video conferencing, audio conferencing, or other multimedia systems
implemented by end users to communicate in real time. The H.323 standard requires that every endpoint support certain functions
and codecs (Coder/Decoder) that have previously been defined by the ITU.
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 congestion".
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.
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, an FXO connection must be configured.
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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/dissassembler that
can support multiple protocols and provide basic "routing" functions.
G
Gatekeeper: An H.323 entity that provides address translation, control access, and sometimes bandwidth management to the LAN
for H.323 endpoints.
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. 3. An H.323 entity that provides real-time, two-way communications between H.323 terminals on the
LAN and other ITU terminals on a WAN, or to another H.323 Gateway.
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
H.323: An umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia
communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS).
H.323 Endpoint: The clients in an H.323 network. They are typically video conferencing, audio conferencing, or other multimedia
systems implemented by end users to communicate in real time. The H.323 standard requires that every endpoint support the
G.711 ITU standard for speech compression, H.245 protocol for controlling media between H.323 endpoints, Q.931 signal protocol
for establishing and terminating calls, RAS (Registration/Admissions/Status) channel data stream used to communicate with a
gatekeeper (optional), and RTP/RTCP (Real-Time Protocol/Real-Time Control Protocol) for carrying packetized real-time media on
IP networks.
H.323 Entity: Any H.323 component, including terminals, Gateways, Gatekeepers, Multipoint Controllers (MCs), Multipoint
Processors (MPs), and Multipoint Control Units (MCUs).
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.
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 endsystem 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.
Glossary
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/O address 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
link-level 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.
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.
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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
committee 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.
Multicast: A process of transmitting from one source to many destinations. The actual mechanism may be different for different
LAN technologies.
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.
Multipoint Conference: A conference between three or more terminals, which may be on the LAN or on the Circuit Switched
Network.
Multipoint Control Unit (MCU): An H.323 endpoint on the LAN which enables three or more terminals and Gateways to participate
in a multipoint conference. The MCU includes a mandatory Multipoint Controller and optional Multipoint Processors.
Multipoint Controller (MC): An H.323 entity which provides for the control of three or more terminals in a multipoint conference.
Multipoint Processor (MP): An H.323 entity which provides for the processing of audio, video, and/or data streams in a multipoint
conference. The MP provides for the mixing, switching, or other processing of media streams under the control of the MC.
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.
Glossary
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.
PBX (Private Branch Exchange): 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".
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.
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PRI (Primary Rate Interface): 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. PRI is the ISDN equivalent of a T-1 circuit.
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.
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
Quality of Service (QoS): Guarantees network bandwidth and availability for applications.
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.
RAS Channel: An unreliable channel used to convey the Registration, Admissions and Status messages and bandwidth changes
between two H.323 entities.
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 generallyaccepted 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, offhook, etc., for each channel.
Glossary
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.
RS-232-C: An EIA standard for a serial interface between computers and peripheral devices (modem, mouse, etc.). It uses a 25pin 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.
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 networks.
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Glossary
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.
TCP/IP was developed by the US Department of Defense to link dissimilar computers across dissimilar and unreliable LANs.
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 high-speed services such as ATM, frame relay, and SMDS. See also T Carrier, T1 line and FT1.
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-by-frame 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.
Terminal: 1) 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. 2) An H.323 endpoint that provides for real-time, two-way
communications with another terminal, gateway, or H.323 MCU. An H.323 terminal must provide audio and may also provide video
and/or data.
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 packetswitched 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.
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Glossary
Trunk: A communication line between two switching systems or offices. Central office trunks connect a PBX to the switching
system at the central office. A tie trunk connects two PBXs.
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.
V
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).
Z
Zone: A collection of all H.323 endpoints and MCUs managed by a single Gatekeeper. A zone must include may include LAN
segments connected by using routers.
Multi-Tech Systems, Inc. MultiVOIP User Guide
101
Index
E
Index
A
About the Internet ...........................................
Accessories, ordering .....................................
Add/Edit Phone Entry .....................................
Applications setup, changing ..........................
Archie .............................................................
Auto Call ................................................... 40,
Auto Call Enable ....................................... 26,
Automatic Disconnection .......................... 26,
Automatic disconnection, billing/security ........
78
77
31
59
81
50
40
40
50
B
Back panel, description ..................................
Billing options .................................................
Billing/Misc default setup, Master ............. 27,
Billing/Misc parameters, changing ..................
Billing/Misc tab ...............................................
Boot LED ........................................................
Building a proprietary phonebook ...................
13
51
41
50
40
12
31
F
Fax branding statement ..................................
FCC Part 15 statement ..................................
FCC Part 68 Telecom statement ....................
Finger .............................................................
Firewall ...........................................................
Flash Timer box ........................................ 24,
Front panel, description ..................................
FTP (File Transfer Protocol) ...........................
85
85
86
81
20
48
12
81
G
Gatekeeper ....................................................
Registering phone numbers ........................
Glossary .........................................................
Gopher ...........................................................
28
28
89
81
H
C
Cable Connection
Remote Site ................................................
Cabling ...........................................................
Local site ....................................................
Cabling the MVP120
Remote site ................................................
Call Duration ............................................ 27,
Call progress, viewing .............................. 58,
Canadian limitations notice ............................
Channel interface default setup, Master .........
Channel interface default setup, Slave ...........
Channel totals, viewing ..................................
Coder, voice/fax .............................................
Collision ..........................................................
Command connector ......................................
Command port cable, diagram .......................
Configuration port setup .................................
Configuration utilities ......................................
Configuration utilities, overview ......................
Configuring Your Client MultiVOIPs ................
Connectors
Command ...................................................
Power .........................................................
Consecutive Packets Lost ........................ 27,
Custom tone pair settings ...............................
EMC, safety and terminal directive compliance87
Ethernet (10Base-T) connector ...................... 13
Ethernet, front panel ....................................... 12
44
18
18
44
41
62
87
38
38
63
49
12
13
83
46
46
46
36
13
13
40
51
H.323 protocol
Overview ....................................................... 6
HTML ............................................................. 81
HTTP .............................................................. 81
Hunt Group ..................................................... 32
I
Industry Canada statement ............................
Installing MultiVOIP software .........................
Inter Digit Time ......................................... 24,
Internet
Multi-Tech's Web site ..................................
Internet protocol (IP) ......................................
IP default setup, Master .................................
IP default setup, Slave ...................................
IP parameters, changing ................................
IP statistics .....................................................
85
20
48
78
82
24
37
57
60
J
Jitter buffer, dynamic ...................................... 51
Jitter Value ............................................... 26, 40
Jitter value ...................................................... 50
D
DNS ................................................................
DTMF gain, voice/fax .....................................
Dynamic Jitter Buffer ......................................
Dynamic jitter buffer .......................................
81
49
27
51
102
Index
L
LAN cable, diagram ........................................
LAN-based remote configuration ....................
LEDs
56K DSU
RD ...........................................................
Ethernet
CL ...........................................................
LNK .........................................................
RD ...........................................................
TD ...........................................................
Log entries, reports ........................................
Log entry details, viewing ...............................
R
83
70
12
12
12
12
12
63
62
M
Manual Upgrade Procedure ........................... 66
Max Baud rate, voice/fax ................................ 50
Message Waiting Light ................................... 48
Modem-based remote configuration ............... 68
MultiVOIP Main menu, introduction ................ 46
MultiVOIP Main menu, screen ........................ 47
MultiVOIP, Telnet server menu ....................... 72
MVP120
Back panel .................................................. 13
Cabling diagrams ........................................ 83
Configuring the Slave MultiVOIP ................ 37
Deploying the VOIP network ....................... 44
Front panel ................................................. 12
Installing the MultiVOIP software ................ 20
Introduction ................................................... 6
Service ........................................................ 77
Specifications ............................................. 14
Technical support ........................................ 78
Typical application ........................................ 8
Unpacking ................................................... 17
Upgrade procedure ..................................... 64
O
Optimization Factor ........................................ 27
Optimization factor ......................................... 51
Ordering accessories ..................................... 77
P
Packets lost/received .....................................
Password, SNMP ...........................................
Phone directory database, changing ..............
Ping ................................................................
POP ................................................................
Power connector ............................................
Proprietary phonebook
Building .......................................................
Pulse generation ratio ....................................
58
59
52
81
81
13
31
51
Receive data ..................................................
Regional default setup, Master ................. 27,
Regional parameters, changing .....................
Registering with the Gatekeeper ....................
Regulatory information ...................................
Remote configuration
LAN-based procedure ................................
Modem-based procedure ............................
Remote management
Telnet ..........................................................
WEB management ......................................
Remote site administrator, deployment ..........
Remote site cable connection ........................
Reports, log entries ........................................
Ring Count FXO .............................................
Ringer equivalence number ...........................
12
41
51
28
85
70
68
71
73
44
44
63
48
87
S
Safety warning
General .......................................................
Telecom ......................................................
Send Status Report to Master ........................
Service ...........................................................
Slave status ....................................................
SMTP .............................................................
SNMP .............................................................
SNMP agent, enabling ...................................
SNMP statistics ..............................................
Software .........................................................
Description ..................................................
Software installation procedure ......................
Specifications
Command Port ...........................................
Ethernet Port ..............................................
Station Identification .......................................
Station identification .......................................
Statistics, viewing
IP ................................................................
SNMP .........................................................
17
17
31
77
54
81
81
59
61
46
46
20
14
14
32
54
60
61
T
TCP packets ................................................... 60
TCP/IP ............................................................ 80
Technical specifications .................................. 14
Technical support ........................................... 78
Telnet ........................................................ 71, 81
Proxy Server Configuration ........................ 73
Telnet server menu ......................................... 72
Tone Pairs ................................................ 27, 41
Tone pairs, customizing .................................. 51
Traceroute ...................................................... 81
Transmit data ................................................. 12
Typical application ............................................ 8
103
Index
U
UDP packets ..................................................
Uninstall ProxyServer configuration ...............
Unpacking your MVP120 ...............................
Upgrading software version ............................
60
46
17
64
V
Viewing statistics ............................................
Voice coder, voice/fax ....................................
Voice over IP configuration .............................
Voice/Fax 1 and 2, front panel ........................
Voice/Fax channel connectors, diagram ........
Voice/Fax default setup, Master ............... 25,
Voice/Fax parameters, changing ....................
60
49
72
12
83
39
49
W
WEB Server ................................................... 73
104
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