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Voice/Fax Over IP Networks
Models MVP400 and MVP800
User Guide
User Guide
S0000008 Revision A
MultiVOIP400 (Model MVP400) and
MultiVOIP 800 (Model MVP 800)
This publication may not be reproduced, in whole or in part, without prior expressed written permission from Multi-
Tech Systems, Inc. All rights reserved.
Copyright © 2000, 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
(6/5/00)
Record of Revisions
Description
Added H.323 protocol support; covers software version 3.45. All pages at revision A.
Patents
This Product is covered by one or more of the following U.S. Patent Numbers: 5.301.274; 5.309.562; 5.355.365;
5.355.653; 5.452.289; 5.453.986. Other Patents Pending.
TRADEMARK
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.
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 and Description
Chapter 2 - Installation
Chapter 3 - Software Loading and Configuration
Chapter 4 - MultiVOIP 400/800 Software
iii
Chapter 5 - Remote Configuration and Management
Chapter 6 - Warranty, Service and Tech Support
Appendixes
Glossary
Index
iv
Chapter 1 - Introduction and Description
6
MultiVOIP 400/800 User Guide
Introduction
Welcome to Multi-Tech's new voice/fax gateway, the MultiVOIP, models MVP400 and MVP800
(Hereafter referred together as MultiVOIP 400/800). The MultiVOIP 400/800 allows analog voice and fax communication over a traditional data communications/data networking digital Internet. Multi-
Tech’s new voice/fax gateway technology allows voice and fax communication to be transmitted, with no additional expense, over your existing communications Internet, which has traditionally been dataonly. To access this free voice and fax communication, all you have to do is connect the MultiVOIP
400/800 to a phone or to your existing in-house phone switch and then to your existing Internet connection. Once configured, the MultiVOIP 400/800 allows voice and fax to travel down the same path as your traditional data communications.
The MultiVOIP 400/800 supports the H.323 standards-based protocol enabling your MultiVOIP 400/
800 to participate in real-time conferencing with other third-party VOIP Gateways or endpoints that support the H.323 protocol (e.g., 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 or gateway. 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.
The MVP400 is designed with four voice/fax channels (which offer three voice/fax interfaces per channel), a 10 Mbps Ethernet LAN interface, and a command port for configuration. The MVP800 is designed with eight voice/fax channels, 10 Mbps Ethernet LAN interface, and command port.
System management is provided through the command port using bundled Windows® software which provides easy-to-use configuration menus and a comprehensive on-line help system.
Figure 1-1. MultiVOIP 400/800
Chapter 1 - Introduction and Description
Preview of this Guide
This guide describes the MultiVOIP and tells you how to install and configure the unit. The information contained in each chapter is as follows:
Chapter 1 - Introduction and Description
Chapter 1 describes the MultiVOIP and provides a typical application. Front panel indicator and back panel connector descriptions are provided. In addition, a list of relevant specifications is provided at the end of the chapter.
Chapter 2 - Installation
provides information on unpacking and cabling your MultiVOIP. The installation procedure describes each cable connection.
Chapter 3 - Software Loading and Configuration
provides instructions for software loading and initial configuration. The MultiVOIP software diskettes are Windows® based. Later chapters, as well as your on-line help program will describe the
MultiVOIP software in more detail.
Chapter 4 - MultiVOIP Software
describes the MultiVOIP software package designed for the Windows ® environment. This chapter describes the software from an applications standpoint, and in so doing, not every screen is shown, nor is each field within a screen defined. For explanations and parameters of each field within a dialog box please refer to the on-line help system provided within the software.
Chapter 5 - Remote Configuration and Management
provides procedures for changing the configuration of a remote MultiVOIP. Remote configuration allows you to change the configuration of a unit by simply connecting two modems between the two MultiVOIPs and remotely controlling the unit. Chapter 5 also describes typical client applications (i.e., Telnet and Web-based management) used for remote configuration of the
MultiVOIP.
Chapter 6 - Warranty, Service and Tech Support
provides instructions on getting service for your MultiVOIP at the factory, a statement of the limited warranty, information about our Internet presence and user bulletin board service, and space for recording information about your MultiVOIP prior to calling Multi-Tech’s Technical Support.
7
MultiVOIP 400/800 User Guide
8
Typical Application
Before Voice Over IP (VOIP), i.e., voice over the Internet, a corporate office had a data connection to the Internet and a voice connection to the public switched telephone network (PSTN). With VOIP, the two networks can be tied together. To accomplish this, a MultiVOIP is connected between the public switched telephone network and the data network as shown in the typical VOIP application in Figure
1-2. A MultiVOIP 400/800 at the corporate office is tied into the public switched telephone network thru the corporate PBX and to the data network. The corporate MultiVOIP is an MVP400/800 with four voice/fax channels connected to the PBX allowing the sales, marketing and regional offices to simultaneously call in or fax on any one of the corporate telephone numbers. The sales office is using a MultiVOIP MVP 200 that is connected to the sales office Key Telephone System (KTS) in which a sales attendent answers in-coming calls and directs them to their proper destination. The marketing office and regional offices also use MVP200s. The marketing office is set up with the
MultiVOIP connected to the E&M trunks on the PBX. The regional office uses an FXS interface to connect two anlaog telephones directly to the local telephone network.
In this configuration, the corporate MultiVOIP is connected to analog trunks on the corporate PBX and the Ethernet connection is plugged into the hub on the data network. The data network is connected via a router to the Internet. In our typical application, any user at the corporate office can pick up a telephone and call the sales, marketing, or regional office. To accomplish this, the caller picks up the phone and dials a trunk extension, for example extension 4. This is the same principle as dialing 9 for an outside line in a corporate environment. A second dial tone is heard, the caller then dials extension 201 at the sales office. This rings the KTS at the sales office. The sales receptionist answers the call and then directs the call (for example, you wanted to talk to the person at extension 5125) and a voice conservation takes place.
Corporate Office
Optional
H.323 Gatekeeper
IP Address 201.22.122.110
Port Number 1719 Workstation
Web Server
Workstation
LAN
MultiVOIP
IP Address
201.22.122.118
Mask 255.255.255.128
HUB
Analog Connections
Channel 1: E&M Channel 2: E&M
Channel 3: E&M Channel 4: E&M
104
103
102
101
4 5 6 7
Router (with Diffserv)
IP Address 201.22.122.1
Mask 255.255.255.128
Internet/Intranet
IP Network
Sales Office
Workstation
Workstation
MultiVOIP
IP Address 205.24.123.119
Mask 255.255.255.240
#201 (5127)
#202
LAN
FXS Interface
HUB
Router (with Diffserv)
IP Address 206.25.124.110
Mask 255.255.255.240
KTS
522-5128
522-5125 522-5126
522-5124
Router Static IP
Address 209.96.211.90
512-4120
512-4121
512-4122
512-4123
P
B
X
ISP
ISP
ISP
PSTN Connection
(T1/E1, PRI, etc.)
PSTN
PSTN
PSTN
Marketing Office
Workstation
Workstation
Router Static IP
Address 207.26.125.121
LAN
Router (with Diffserv)
IP Address 208.25.124.110
Mask 255.255.255.240
HUB
9
301
10
302
E&M Interface
Router Static IP
Address 209.96.211.90
Workstation
Workstation
LAN
FXS Interface
532-6128
532-6129
Router (with Diffserv)
IP Address 210.33.4.110
Mask 255.255.255.240
Regional Office
#401
#402
Figure 1-2. Example of a MultiVOIP application
The sales office is set up with a two voice/fax channel MultiVOIP MVP200 connected to the LAN with one voice/fax channel connected directly to the local Public Switch Telephone Network (PSTN) and the second voice/fax channel connected to the trunk side of the KTS. This allows an attendent to direct incoming calls to any one of the telephones or fax machines connected to the Key Telephone
System at the sales office. This allows the corporate caller to call either the sales office or instead of dialing extension 201, dials extension 202 and is now connected to the local telephone network at the
Chapter 1 - Introduction and Description sales office. The corporate caller could then dial any telephone number from the sales office PSTN and this would be a local call.
The regional office is set up very similar to the corporate office with the exception that a MVP200 is connected between the network and the PBX. This allows two voice/fax channels to be used at the regional office. A person in the regional office can pick up a telephone and dial a trunk extension (for example, trunk extension 10) on the local PBX. This is the same as dialing a 9 for an outside line. A second dial tone is heard, the caller would then dial the corporate MultiVIOP at one of the following extensions (101 thru 104). When the third dial tone is heard, the caller could then dial any telephone at the corporate office by dialing its extension, for example, extension 4123.
To configure a MultiVOIP, the COM port of a PC is connected to the Command port on the MultiVOIP.
Configuration software is loaded onto your PC and your unique LAN parameters must be established.
The configuration software is based on a standard Windows Graphical User Interface (GUI) which simplifies your selection process to a single parameter group within a dialog box. For example, your
LAN IP parameters are contained in a single dialog box (See below). You can configure your network
IP address and mask for the MultiVOIP 400/800 and the gateway address for the corporate router on the same dialog box.
For your corporate MultiVOIP, the Ethernet Frame Type is Type II, the IP Address is 201.22.122.118, the Subnet Mask Address is 255.255.255.128, and the router Gateway Address is 201.22.122.1.
Once the LAN parameters are established, you can set up the voice channel parameters.
The channel setup parameters define the voice side of the MultiVOIP, that is, the voice channel interface; FXS (Ground and Loop Start) are for connecting to a standard analog telephone set, FXO
(Foreign Exchange Office) interface connects to the station side of a PBX, and E&M (Ear and Mouth) connects to the trunk side of the PBX. Along with each interface there are additional parameters that need to be considered, such as for FXO, the dialing options for DTMF (Touchtone) or Pulse, the method of disconnecting (Current Loss or Tone Detection), and for E&M, signaling, mode, and the wink timer settings in milliseconds.
In our typical application, you would configure the corporate office channel parameters for an E&M interface. With this interface, you would have to consult with your in-house telephone people to determine how your PBX is configured. They would have to determine E&M signaling options (dial
9
MultiVOIP 400/800 User Guide tone or wink), 2-wire or 4-wire connection, and if the wink option is the signaling option, the wink timer delay in milliseconds. The default is 250 milliseconds.
Additional channel setup parameters cover the voice coder, DTMF gain, voice gain, and faxing in the
Voice/Fax tab of the Channel Setup dialog box. The most important parameter is this group is to ensure that the voice coder is the same for all MultiVOIPs in the network. The Billing/Misc tab handles the billing options, automatic disconnect options, and the dynamic jitter buffer options. The jitter options in this tab handle voice break up which can be particularly disruptive to voice communications. For the most part, these parameters can remain in their default values. The
Regional tab defines the country or region the MultiVOIP is being used in.
Once you have completed channel setup, 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. 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.
10
The decision on building the phone directory database is contained in 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
Chapter 1 - Introduction and Description option. You can not mix the Proprietary PhoneBook with the Gatekeeper. If you choose the
Gatekeeper option, interoperability is less of a concern and you can communicate with other third party endpoints that support H.323 (e.g., Microsoft Netmeeting). (2) If you choose the Proprietary
PhoneBook, you establish a master-slave relationship in that 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.
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 register with the gatekeeper. The gatekeeper functions are 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/TRCP for sequencing audio packets. The H.323 Gatekeeper also provides call-authorization for both accepting and placing calls in its zone, handles hunting (call-rollover), and certain monitoring features (i.e., call permissioning and address resolution).
So, if you choose the Gatekeeper option, initially you need to communicate with the administrator of the Gatekeeper to preregister 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 is a name, e-mail like address, etc, 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 you use the same port number.
The Gatekeeper administrator will then enter your information into the Gatekeeper data base. This concludes the preregisteration.
Now, 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
Voice Channel 1. For instance, in our typical application channel 1 of the Corporate MultiVOIP uses extension 101. The Description is optional, but can be helpful if it is assigned to a particular individual or department, or in this case it defines the channel interface. The H323 ID that was assigned to this phone number is an e-mail like address which identifies the individual that is using this extension and the domain name of the Gatekeeper (in this example corporatehd). The IP Address of the Corporate
MultiVOIP is 201.022.122.118 and the default port number 1720 is used.
So now when you “come alive”, the Gatekeeper will register you with the above alias address. No other H323 endpoint can use this alias. This is like your own telephone number.
Now, if you choose the Proprietary PhoneBook option in the Phone Directory Database dialog box instead of the Gatekeeper option, the Database Type group would become active and the RAS
Parameters group is inactive (greyed out).
11
MultiVOIP 400/800 User Guide
Now, lets change the typical application to not have the Gatekeeper control the call session. When you elect to use the Proprietary PhoneBook, 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 data base allows you to see all the participants in your network and provides you with there phone numbers.
Lets again use the corporate MultiVOIP and we will set up the database so that the corporate
MultiVOIP can call the sales, marketing, or regional offices. To do this, the Phone Directory Database will have four entries for the corporate office, and two entries for the sales office, two for the marketing office and two for the regional office. Extension 101 at the corporate office is tied to voice channel 1, 102 is tied to channel 2, etc. The Description again ties to the type of interface used on the corporate MultiVOIP (E&M). The Hunt Group in this situation is set for No Hunt. But if you wanted to activate a Hunt Group, i.e., 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. So that, say extension
101 is busy, the corporate MultiVOIP would roll over to extension 102.
Again, the IP Address of the corporate MultiVOIP needs to be added and the port number is 1720.
This adds phone number 101 of the corporate MultiVOIP to the proprietary data base. Now, to add extension 102 to the proprietary data base, all you have to do is change the Phone Number and
Description to support channel 2 of the corporate MultiVOIP. After you have added channel 2, you need to include the two entries for each office.
The proprietary data base would then appear as in the following dialog box and when the office
MultiVOIP is turned on, the current data base would be down loaded.
12
Chapter 1 - Introduction and Description
Front Panel Description
The front panel contains three groups of LEDs that provide the status of the Ethernet connection,
Voice/Fax channels, and general status of the MultiVOIP. The front panel is shown in Figure 1-3, and a description of each LED follows.
Figure 1-3. Front Panel
ETHERNET
RCV
LNK
XMT
COL
Receive Data indicator blinks when packets are being received from the local area network.
Link indicator lights when the Ethernet link senses voltage from a concentrator or external device.
Transmit Data indicator blinks when packets are being transmitted to the local area network.
Collision indicator lights when a collision is detected on the Ethernet link.
VOICE/FAX CHANNEL
FXS
FXO
E&M
FAX
XMT
RCV
XSG
RSG
Foreign Exchange Station indicator lights when the voice/fax channel is configured for
FXS operation.
Foreign Exchange Office indicator lights when the voice/fax channel is configured for
FXO operation.
Ear and Mouth indicator lights when the voice/fax channel is configured for E&M operation.
Fax indicator lights when there is fax traffic on the voice/fax channel.
Transmit indicator blinks when voice packets are being transmitted to the local area network.
Receive indicator blinks when voice packets are being received from the local area network.
Transmit Signal indicator lights when the FXS-configured channel is off-hook, the FXOconfigured channel is receiving a ring from the Telco, or the M lead is active on the E&M configured channel (i.e., the MultiVOIP is receiving a ring from the PBX).
Receive Signal indicator lights when the FXS-configured channel is ringing, the FXOconfigured channel has taken the line off-hook, or the E lead is active on the E&Mconfigured channel.
BOOT (BTG)
The BTG indicator lights when the MultiVOIP is booting or downloading setup.
POWER (PWR)
The Power indicator lights when power is applied to the MultiVOIP.
13
MultiVOIP 400/800 User Guide
Back Panel Description
The cable connections for the MultiVOIP are made at the back panel. Connectors include Power,
Command Port (RS232), Ethernet (10BASE-T), and Voice/Fax Channels (E&M, FXO and FXS). The cable connectors are shown in Figure 1-4 and defined in the following groups.
VOICE/
FAX
CHANNEL
8
VOICE/
FAX
CHANNEL
4
E&M FXO FXS
VOICE/
FAX
CHANNEL
7
VOICE/
FAX
CHANNEL
3
E&M FXO FXS
VOICE/
FAX
CHANNEL
6
VOICE/
FAX
CHANNEL
2
E&M FXO FXS
VOICE/
FAX
CHANNEL
5
VOICE/
FAX
CHANNEL
1
E&M FXO FXS
CHANNEL 10
CHANNEL 9
CHANNEL 8
CHANNEL 7
CHANNEL 6
CHANNEL 5
CHANNEL 4
CHANNEL 3
CHANNEL 2 (RS232/V.35)
CHANNEL 1 (RS232/V.35)
Figure 1-4. Back Panel
10BASET COMMAND PORT
ETHERNET
EXT. COMPOSITE LINK (RS232/V.35)
INTERNAL
COMPOSITE
LINK MONITOR
XMT RCV
T1 DSU
GND
I
POWER
O
Power Connector
The Power connector is used to connect the external power supply to the MultiVOIP. The Power connector is a 7-pin circular DIN connector. A separate power cord is connected to the power supply and the live AC grounded outlet.
Command Port Connector
The Command Port connector is used to configure the MultiVOIP using a PC with a serial port and running Windows® software. The Command Port connector is a DB-25 female connector.
10Base-T (Ethernet) Connector
The Ethernet 10Base-T connector is used to connect the MultiVOIP to a LAN using unshielded twisted cable. This connector is a keyed RJ-45 jack.
Voice/Fax Channel
The Voice/Fax channel connectors include three options per channel: E&M, FXO and FXS.
E&M - This connector is used if you are connecting VOICE/FAX CHANNEL _ to the E&M trunk on a
PBX. This connector is an RJ-45 jack.
FXO - This connector is used if you are connecting VOICE/FAX CHANNEL _ to the station side of a
PBX. This connector is an RJ-11 jack.
FXS - This connector is used if you are connecting VOICE/FAX CHANNEL _ to a station device, etc., an analog telephone, KTS telephone system, or fax machine. This connector is an RJ-11 jack.
14
Chapter 1 - Introduction and Description
Specifications
• One 1 Meg by 32 byte at 70 nanosecond SIMM is 4 Mb DRAM
Caution: SIMM speed and size cannot be mixed
• Two Meg 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 with a DB-25 female connector
Voice/Fax Channel
• Two RJ-11 jacks (FXO and FXS)
• One RJ-45 jack (E&M)
Electrical/Physical
• Voltage - 115 VAC (Standard), 240 Volts AC (Optional)
• Frequency - 47 to 63 Hz
• Power Consumption - 18 Watts
• Dimensions - 3.75" high x 17.4" wide x 8" deep
8.9cm high x 44.2cm wide x 20.3cm deep
• Weight - 7.4 pounds (3.4 kg)
15
MultiVOIP 400/800 User Guide
16
Chapter 2 - Installation
MultiVOIP 400/800 User Guide
Installing Your MultiVOIP
The basic steps of installing your MultiVOIP 400/800 network involve unpacking the units, connecting the cables, and configuring the units using management software (MultiVOIP Configuration). This process results in a fully functional Voice Over IP network. A general description is provided below and detailed instructions are provided in Chapter 3, Software Loading and Configuration .
Installing and Configuring Your MultiVOIP
The VOIP administrator must first install the MultiVOIP 400/800 software and then configure each
MultiVOIP 400/800 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 pre-register or remotely 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; as well as the selection of appropriate channel interface type for each of the Voice/Fax channels. Once all connections have been made, the VOIP administrator configures the unit and builds the Phone Directory Database that will reside with the Master unit.
Once configuration of the “Master” MultiVOIP has been completed, the administrator moves on to configure the MultiVOIP(s) designated as “Slave” units. Again, unique LAN IP addresses, subnet masks, and Gateway IP addresses are assigned, and each Voice/Fax channel is configured for the appropriate channel interface type. When this is done, the Phone Directory Database option is set to
Slave, and the IP address of the Master MultiVOIP is entered. Once all Slave units are configured, the process moves on to 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 remotely or pre-registered with the Gatekeeper (i.e., 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 (i.e., the Proprietary phonebook database or the Gatekeeper phonebook database).
Safety Warning Telecom
1.
Never install telephone wiring during a lighting storm.
2.
Never install telephone jacks in wet locations unless the jack is specifically designed for wet locations.
3.
This product is to be used with UL and cUL listed computers.
4.
Never touch uninsulated telephone wires or terminals unless the telephone line has been disconnected at the network interface.
5.
Use caution when installing or modifying telephone lines.
6.
Avoid using a telephone (other than a cordless type) during an electrical storm. There may be a remote rist of electrical shock from lighting.
7.
Do not use the telephone to report a gas leak in the vicinity of the leak.
8.
To reduct the risk of fire, use only No. 26 AWG or larger telecommunications line cord.
18
Unpacking Your MultiVOIP
Remove all items from the box.
Voice/Fax over IP Networks
Chapter 2 - Installation
Figure 2-1. Unpacking
Safety Warnings
Caution: Danger of explosion if battery is incorrectly replaced.
A lithium battery on the voice/fax channel 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 Sytems for battery replacement.
The E&M, FXS, and Ethernet ports are not designed to be connected to a Public Telecommunication
Network.
Valid VOIP Network Connections
The following VOIP network interface connections (calls) can be made.
• FXS to FXS
• FXS to E&M
• FXS to FXO
• FXO to FXO
• FXO to FXS
• FXO to E&M
• E&M to E&M
• E&M to FXS
• E&M to FXO
19
MultiVOIP 400/800 User Guide
Cabling Procedure
Cabling involves connecting the master MultiVOIP to your LAN and telephone equipment.
1.
If you are connecting any Voice/Fax Channel to an E&M trunk other than type 2, perform the
E&M Jumper Block Positioning procedure before connecting power to the unit.
2.
Connect one end of the power supply to a live AC outlet and connect the other end to the
MultiVOIP as shown in Figure 2-2. The power connector is a 7-pin circular DIN connector.
VOICE/
8
E&M FXO FXS
VOICE/
7
E&M FXO FXS
VOICE/
6
E&M FXO FXS
VOICE/
5
E&M FXO FXS
CHANNEL 10
CHANNEL 9
CHANNEL 8
CHANNEL 7
CHANNEL 6
CHANNEL 5
CHANNEL 4
CHANNEL 3
CHANNEL 2 (RS232/V.35)
CHANNEL 1 (RS232/V.35)
10BASET
ETHERNET
COMMAND PORT
EXT. COMPOSITE LINK (RS232/V.35)
T1 DSU
GND
I
POWER
O
Voice/Fax Channel 1 - 8
Connections
Power Connection
E&M FXO FXS
E&M
FXO
FXS
Command Port Connection
PSTN
Ethernet Connection
Figure 2-2. Cable Connections
3.
Connect the MultiVOIP to a PC by using a DB-25 (male) to DB-9 (female) cable. Plug the DB-25 end of the cable into the Command port of the MultiVOIP and the other end into the PC serial port. See Figure 2-2.
4.
Connect a network cable to the ETHERNET 10BASET connector on the back of the MultiVOIP.
Connect the other end of the cable to your network.
5.
If you are connecting a station device (e.g., analog telephone, fax machine, or Key Telephone
System (KTS) to your MultiVOIP, connect one end of an RJ11 phone cord to the Voice/Fax
Channel 1 FXS connector on the back of the MultiVOIP and the other end to the station device.
If you are connecting the station side of a telephone switch (PBX) to your MultiVOIP, connect one end of an RJ11 phone cord to the Voice/Fax Channel 1 FXO connector on the back of the
MultiVOIP and the other end to the phone jack.
If you are connecting an E&M trunk from a telephone switch to your MultiVOIP, connect one end of an RJ45 phone cord to the Voice/Fax Channel 1 E&M connector on the back of the MultiVOIP and the other end to the trunk. Refer to Appendix B for E&M cabling pinout.
If you are connecting to an E&M trunk, you need ensure that the E&M trunk jumper is in the correct position for the E&M type trunk. The default E&M jumper position is E&M type 2. To change the E&M jumper position, perform the E&M jumper block positioning procedure.
6.
Repeat the above step to connect the remaining telephone equipment to each Voice/Fax
Channel on your MultiVOIP.
7.
Turn on power to the MultiVOIP by placing the ON/OFF switch on the back panel to the ON position. Wait for the BTG LED on the MultiVOIP to go OFF before proceeding. This may take a couple of minutes to go OFF.
Proceed to the next section to load the MultiVOIP software.
20
Chapter 2 - Installation
E&M Jumper Block Positioning Procedure
A jumper block exists for each voice/fax channel. The jumper block is to the right of each set of channel jacks. The jumper block contains 8-pairs of pins. The jumper plug fits over three pairs of pins on the jumper block. The E&M type number is labeled on the pc board. The jumper plug needs to be centered on the E&M type number. Perform the following procedure to change E&M jumper position.
1.
Ensure that power is removed from the MultiVOIP
2.
Remove the front panel by loosening the two Phillips quarter turn screws.
3.
Slide the top cover back off the chassis to expose the rear panel.
4.
To change a jumper position, lift the jumper plug up off the jumper block and move to the new position, ensuring that the center jumper is centered on the E&M type number.
2 (Default)
1,3
4
5
Figure 2-3. E&M Jumper Block
5.
Change the jumper position for each voice/fax channel that is connecting to an E&M trunk that is not a type 2. If you have two voice/fax channel boards in your unit and you need to change the jumpers on the second board, remove the six screws from the top board and disconnect the ribbon cable from the top board.
6.
Slide the top cover back on to the chassis.
7.
Replace the front panel and secure it by tightening the two Phillips quarter turn screws.
8.
Return to the Cabling Procedure section.
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MultiVOIP 400/800 User Guide
22
Chapter 3 - Software Loading and Configuration
MultiVOIP 400/800 User Guide
Installing Your MultiVOIP Software
The following installation procedures do not provide every screen or option in the process of installing the MultiVOIP 400/800 software. It is assumed that a technical person with a thorough knowledge of
Windows and the software loading process is doing the installation. Once you have installed the software, you will be instructed on how to configure your MultiVOIP 400/800, and finally, on how to deploy your MultiVOIP 400/800. Additional information on the MultiVOIP 400/800 software is provided in the on-line 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
MultiVOIP 400/800” will explain these differences.
The MultiVOIP 400/800 software and User Guide are contained on the MultiVOIP 400/800 CD. The
CD is auto-detectable, so when you insert it into your CD ROM drive it will start up automatically.
When you have finished configuring your MultiVOIP 400/800, you can view and print the User Guide by clicking on the Install Manuals icon.
CAUTION: If you are installing a MultiVOIP 400/800 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 MultiVOIP 400/800 has been properly cabled and that it is powered on.
2.
Before you insert the MultiVOIP 400/800 CD into your CD ROM, you need to determine how you will configure your MultiVOIP 400/800, via the LAN or connected directly to the Command port. If you are configuring your MultiVOIP 400/800 via your network, you need to have your PC configured for network communications (i.e., TCP/IP stack loaded). Your PC and the MultiVOIP
400/800 must be on the same physical LAN segment.
Insert the MultiVOIP 400/800 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 screen to appear.
24
If the Multi-Tech Installation CD Screen does not appear automatically, click My Computer, then right click the CD ROM drive icon, click Open, and then click the Autorun icon.
3.
When the Multi-Tech Installation CD Screen is displayed, click the Install Software icon.
Chapter 3 - Software Loading and Configuration
4.
The MultiVOIP 800 Setup welcome screen is displayed.
Press Enter or click Next > to continue.
5.
The Choose Destination Location dialog box is displayed. Follow the on-screen instructions.
You can either choose the Destination Location of your MultiVOIP 400/800 software or select the default destination by clicking Next >. If you click Browse, you can select a different destination folder for the MultiVOIP 400/800 software.
6.
The Select Program Folder dialog box enables you to choose where you want the program file to be located.
Verify the path and click Next > to continue.
25
MultiVOIP 400/800 User Guide
7.
The Copying program files ... screen is displayed followed by the MultiVOIP 800 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 MultiVOIP 400/800. From the Select Port drop-down list, choose the COM port of your PC.
Click OK to continue.
8.
The Setup Complete dialog is displayed.
Click Finish to continue.
9.
The following message is displayed:
Click Yes to continue.
10. The following message is displayed.
26
Click Yes to continue.
Chapter 3 - Software Loading and Configuration
Configuring Your MultiVOIP 400/800
The following steps provide instructions for configuring your MultiVOIP 400/800. The configuration sequence includes IP Protocol default setup, Channel setup, and Phone Directory Database setup.
The Phone Directory Database setup is configured differently depending on whether or not the
Gatekeeper function is available and enabled on the Phone Directory Database dialog box (See
Step 26).
11. The IP Protocol Default Setup dialog box is displayed.
The default Frame Type is TYPE_II. If this does not match your IP network, change the Frame
Type by clicking the drop-down arrow and selecting SNAP. The available Frame Type choices are
TYPE_II and SNAP.
12. In the Ethernet group, enter the IP Address, Subnet Mask, and Gateway Address unique to your IP LAN in the corresponding fields.
The IP address is the unique LAN IP address that is assigned to the MultiVOIP 400/800, and the
Gateway address is the IP address of the device connecting your MultiVOIP 400/800 to the
Internet.
Click OK when you are finished.
13. The Channel Setup dialog box is displayed. The four tabs in this dialog box define the channel interface, voice/fax parameters, billing/miscellaneous parameters, and regional telephone parameters for each channel.
Configure each channel for the type of interface you are connecting to. The Interface tab defaults to Channel 1 in the Select Channel field. To change the channel number, click the drop-down arrow and the list of channels is displayed. Highlight the channel you want to configure.
Note: Feature options are enabled or disabled (grayed out) according to the interface type that you select. The one option available for all interface types is the Inter Digit Time option. This 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 setting is 2 seconds.
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MultiVOIP 400/800 User Guide
14. The Interface group defaults to FXS (Loop Start). Select the interface option that corresponds to the interface type being connected to the Voice/Fax Channel 1 jack on the back panel of the
MultiVOIP 400/800.
FXS (Loop Start): If a station device; e.g., an analog telephone, fax machine, or KTS (Key
Telephone System) is connected to the Voice/Fax connector on the back of the unit, FXS (Loop
Start) will likely be the correct Interface.
FXS (Ground Start): If the station device uses ground start, then choose the FXS (Ground Start) option. Refer to the device’s user documentation.
For both FXS Loop Start and FXS Ground Start , the Ring Count FXS window allows you to set the maximum number of rings output on the FXS interface before hanging up and releasing the line to another call. The default setting is 8 rings.
Note: Zero (0) means no rings - caller hears a busy tone.
FXO: If you are using an analog extension from your PBX, then choose the FXO option. Check with your in-house phone personnel to verify the connection type.
If FXO is selected, the Dialing Options Regeneration, Flash Hook Timer, and Ring Count groups are enabled. Check with your local in-house phone personnel to verify whether your local
PBX dial signaling is Pulse or tone (DTMF). Then, set the Regeneration option accordingly. The
Flash Hook Timer allows you to enter the time, in milliseconds, for the duration of the flash hook signals output on the FXO interface. The default setting is 600 milliseconds. The Ring Count
FXO window allows you to set the number of rings received on the FXO interface before the
MultiVOIP 400/800 answers the incoming call. The default setting is 2 rings.
Note: Zero (0) means that the MultiVOIP 400/800 never answers.
For FXO-to-FXO communications, you can enable a specific type of FXO Disconnect; Current
Loss, Tone Detection, or Silence Detection. (Check with your in-house phone personnel to verify the preferred type of disconnect to use.) Enabling Tone Detection activates the
Disconnect Tone Sequence options. For Disconnect Tone Sequence, you can select from dropdown lists either one or two tones that will cause the line to be disconnected; the person hanging up a call must then hit the key(s) that will produce those tones. For Silence Detection, select
One Way or Two Way, then set the timer for the number of seconds of silence before disconnect.
Note that the default value of 15 seconds may be shorter than desired for your application.
E&M: If you are connecting to an analog E&M trunk on your PBX, then choose the E&M interface option to enable the E&M Options group. Check with your local in-house phone personnel to determine if the signaling is Dial Tone or Wink and if the connection is 2-wire or 4-wire. If Wink signaling is used, then the Wink Timer is enabled with a default of 250 milliseconds. The range of the Wink Timer is from 100 to 350 milliseconds. Consult with your local in-house phone personnel for this timer setting.
Note: After configuring a given channel, you can copy that channel’s configuration by clicking the
Copy button and everything on the Interface tab will be copied to the designated channel.
15. Repeat the above step to configure the interface type for each voice/fax channel.
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Chapter 3 - Software Loading and Configuration
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 in to the voice channel before it is sent over the Internet to the remote MultiVOIP 400/800; and, 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 drop-down 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 drop-down lists in the
DTMF Gain group.
Note: Only change the DTMF gain under the direction of Multi-Tech Technical Support supervision.
18. To change the voice coder, first select the channel by clicking the Select Channel down arrow
(highlighting the channel number) then click Manual in the Coder group. To select the appropriate coder, click the Selected Coder down arrow and highlight your new voice coder entry.
If you changed the voice coder, ensure that the same voice coder is used on the voice/fax channel you are calling; otherwise, you will always get a busy signal.
Note: If you allow the Coder to be selected automatically, then you need to select the Max
Bandwidth from the drop-down list. Check with your VOIP 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 and the fax volume in the two drop-down lists and change the jitter value in milliseconds.
When receiving fax packets from a remote MultiVOIP 400/800, 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 MultiVOIP 400/800 uses a Jitter Buffer. The Jitter Value field allows the MultiVOIP
400/800 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 50 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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MultiVOIP 400/800 User Guide
20. You can enable the voice/fax advanced features by clicking (checking) the silence compression, echo cancellation, or forward error correction options.
The Silence Compression option defines whether silence compression is enabled (checked) for this voice channel. If silence compression is enabled, the MultiVOIP 400/800 will not transmit voice packets when silence is detected, thereby reducing the amount of network bandwidth that is being used by the voice channel.
The Echo Cancellation option defines whether echo cancellation is enabled (checked) for this voice channel. If echo cancellation is enabled, the MultiVOIP 400/800 will remove echo which improves the quality of sound.
The Forward Error Correction (FEC) option 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.
Note: After configuring a given channel, you can copy that channel’s configuration by clicking the
Copy button and everything on the Voice/Fax tab will be copied to the designated channel.
21. The Billing/Misc tab displays the parameters for auto call, automatic disconnection, billing options, and dynamic jitter buffer.
30
If you want to dedicate a local voice/fax channel to a remote voice/fax channel (so you will not have to dial the remote channel), click the Auto Call Enable option in the Auto Call group. Then enter the phone number of the remote MultiVOIP 400/800 in the Phone Number field.
22. 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.
Chapter 3 - Software Loading and Configuration
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 MultiVOIP 400/800, 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 MultiVOIP 400/800 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the
MultiVOIP 400/800 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 MultiVOIP
400/800 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.
Note: After configuring a given channel, you can copy that channel’s configuration to the other channel by clicking the Copy button. Everything on the Billing/Misc tab will be copied to the designated channel.
If your country/region is not the default USA, click the Regional tab and proceed to step 25; otherwise, proceed to step 26 to begin building your phone directory database.
25. To change the Tone Pairs on the Regional tab, click the Country/Region down arrow and highlight your specific country or region.
Note: If your country or region is not listed, click the Custom button to define it.
The Tone Pairs group enables you to select/modify the parameters according to choice. Click
OK when finished. Proceed to step 26 to begin building your phone directory database.
26. The Phone Directory Database dialog box is displayed. This dialog box enables you to add, delete, edit phone entries, or print all entries currently in the Phone Directory Database, and view the telephone numbers defined in each Hunt group. You can build a proprietary phonebook (Go to step 27); or, if an H.323 Gatekeeper is available and enabled, you can register your phone numbers with the Gatekeeper (Go to the following section “Registering with a Gatekeeper Phone
Directory”).
31
MultiVOIP 400/800 User Guide
Registering with a Gatekeeper Phone Directory
This section describes what a Gatekeeper is and 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 provides call control services to registered endpoints. The Gatekeeper performs 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.
The Phone Directory Database dialog box is dispayed.
Enable the Gatekeeper option, enter the IP Address of the Gatekeeper, accept the Port default, and then click the Add (+) button.
The Add/Edit Phone Entry dialog box is displayed.
32
The information provided in the Add/Edit Phone Directory dialog box depends on how the
Gatekeeper will be set up with the Gatekeeper administrator. Phonebook information can be registered with the H.323 Gatekeeper in two ways: a.
If the Gatekeeper is set up to allow remote registration, enter the Phone Number, Description,
Channel, H323 ID, the IP Address (optional), and accept the Port default.
Note: To avoid possible phone number and H323 ID duplication, you will need to contact the
Gatekeeper administrator and verify that these entries are unique. An H323 ID is a unique identifier (most often implemented in the form of an e-mail address) that represents the name of the gateway with the name of the Gatekeeper appended to the end (e.g., [email protected]).
Chapter 3 - Software Loading and Configuration b.
If the Gatekeeper does not allow remote registration, you must contact the Gatekeeper administrator (by phone or e-mail) and provide the IP Address (mandatory) to be registered
(entered manually) by the administrator at the Gatekeeper site. This process is called preregistering.
Note: The Gatekeeper administrator assigns the H323 ID. Together (you and the Gatekeeper administrator) determine the Phone Number and H323ID entries.
For both 1) remote registration and 2) pre-registration, enter the unique phone number of the local device in the Phone Number field (e.g. 101) and indicate in the Voice Channel field that the device is connected to Channel 1.
The Description field 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 identifies the phone number you entered in the previous step. For this example, you could enter a description such as
“New York Office 1”.
CAUTION: The default setting for the Port Number Gatekeeper is 1720. 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.
Repeat this process for all H.323 endpoints in your network. Once all H.323 endpoints have been entered, click OK on the Phone Directory Database dialog box and proceed to step 54.
33
MultiVOIP 400/800 User Guide
Building a Proprietary Phonebook Directory
27. To build your proprietary MultiVOIP 400/800 Phone Directory (i.e., in an H.323 environment without the Gatekeeper option enabled), you will first need to enable (check) the Proprietary
Phonebook option and then configure the “Master” MultiVOIP 400/800 and then add the “Slave”
MultiVOIP 400/800s (or other H.323 endpoints). Configuring the “Slave” MultiVOIP 400/800 is discussed later.
The first MultiVOIP 400/800 to be configured is designated the “Master” and contains the proprietary phonebook database. All subsequent MultiVOIP 400/800s 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 400/800 as it comes on-line.
28. To configure the “Master” MultiVOIP 400/800, make certain that the Proprietary Phonebook and
Master options are enabled. The Master IP Address, Send Status Report to Master, and RAS
Parameters group will be disabled (grayed out).
Note: 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
MultiVOIP 400/800s” for details).
Click Add (+) to begin building your phone directory database. The Add/Edit Phone Entry dialog box is displayed.
34
29. Enter the unique phone number of the local device in the Phone Number field (e.g., 101) and indicate that the local device is connected to Channel 1 in the Voice Channel field.
Chapter 3 - Software Loading and Configuration
30. The Description field 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 (e.g., normally the “Master” MultiVOIP 400/800 resides at the entity’s main office; therefore, for this example you could enter a description such as “New York Office 1”).
31. The Station Identification group includes a Hunt Group drop-down 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.
Note: 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 MultiVOIP 400/800 in the IP Address field (e.g., 204.022.122.118).
Note: The Port field 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
(101), destination details (204.022.122.118 Channel 1), and description (New York Office 1).
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MultiVOIP 400/800 User Guide
33. To configure Channel 2 on the Master MultiVOIP 400/800, click Add (+) and the Add/Edit Phone
Entry dialog box is displayed again.
34. Enter the phone number for the MultiVOIP 400/800 in the Station Information group Phone
Number field (e.g., 102).
35. Click inside the Description field and enter a description for the remote MultiVOIP 400/800 phone number for Channel 2. For example, “New York Office 2.”
36. In the Station Identification group, select HUNT GROUP #1 from the Hunt Group drop-down list, enter the New York Office 2’s IP Address (204.022.122.118), 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.
36
You have completed configuration of the “Master” MultiVOIP 400/800. Both voice channels belong to Hunt Group # 1. If a call from an H.323 endpoint (a MultiVOIP 400/800 or a standalone
H.323 endpoint) to Phone Number 101 is unable to be connected, it will automatically connect to the next available phone number in Hunt Group #1, i.e., Phone Number 102.
Chapter 3 - Software Loading and Configuration
38. At this point it is time to add all other phone numbers (slave units and stand-alone units) to the
Phone Directory database. To add Channel 1 of the Slave MultiVOIP 400/800, click Add (+) and the Add/Edit Phone Entry dialog box is displayed again.
39. Enter the phone number for the “Slave” MultiVOIP 400/800 in the Phone Number field (e.g.,
201).
40. Click inside the Description field and enter a description for the remote MultiVOIP 400/800 phone number for Channel 1; for example, “London Office 1.”
41. In the Station Identification group, select HUNT GROUP #2 from the Hunt Group drop-down list, enter the London Office 1’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.
42. Click OK and you are returned to the Phone Directory Database dialog box which now includes the remote phone number and related information in the Phone Number list.
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MultiVOIP 400/800 User Guide
43. To add Channel 2 of the Slave MultiVOIP 400/800, click Add (+) and the Add/Edit Phone Entry dialog box is displayed again.
44. Enter the phone number for the remote (Slave) MultiVOIP 400/800 in the Station Information group Phone Number field (e.g., 202).
45. Click inside the Description field and enter a description for the remote MultiVOIP 400/800 phone number for Channel 2. For example, “London Office 2.”
46. In the Station Identification group, select HUNT GROUP #2 from the Hunt Group drop-down list, enter the London Office 2’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.
Note: Depending on your requirements, you may want calls that cannot make a connection to
London Office 1 (Hunt Group #2) to roll over to the New York office instead. In this case, you would configure that phone entry to be listed as a member of HUNT GROUP #1.
47. Click OK and you are returned to the Phone Directory Database dialog box which now includes the remote phone number and related information in the Phone Number list.
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Chapter 3 - Software Loading and Configuration
48. To configure a stand-alone endpoint (e.g., a PC with Netmeeting software), click Add (+) and the
Add/Edit Phone Entry dialog box is displayed again.
49. Enter the phone number for the stand-alone endpoint in the Station Information group Phone
Number field (e.g., 301).
50. Click inside the Description field and enter a description for the remote MultiVOIP 400/800 phone number. For example, “Human Resources Desk.”
Note: Because the H.323 endpoint is not a MultiVOIP 400/800, the Phone Directory database ignores the Voice Channel entry, i.e., it does not matter what value is entered.
51. In the Station Identification group, select NO HUNT from the Hunt Group drop-down list, enter the Human Resource Desk’s IP Address (e.g., 202.198.100.04), 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.
52. 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.
53. When you have finished, click OK to download the setup configuration to the MultiVOIP 400/800.
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MultiVOIP 400/800 User Guide
54. The Checking MultiVOIP 800 dialog box is displayed.
55. Once the setup program receives a response from the MultiVOIP 400/800, the Writing Setup dialog box is displayed indicating that the setup configuration is being written to the MultiVOIP
400/800.
56. After the setup has been written to the MultiVOIP 400/800, the unit is rebooted.
57. Check to ensure that the BOOT LED on the MultiVOIP 400/800 is Off after the download is complete. This may take several minutes as the MultiVOIP 400/800 reboots.
58. You are returned to the Multi-Tech Installation CD screen from which you can load the Acrobat
Reader to your PC. This allows you to view and/or print the User Guide by clicking on the Install
Manuals icon.
40
At this time your master MultiVOIP 400/800 is configured. Proceed to the “Configuring Your Slave
MultiVOIP 400/800s” section.
Chapter 3 - Software Loading and Configuration
Configuring Your Slave MultiVOIP 400/800s
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
MultiVOIP 400/800 at the company’s subsidiary office in London would need to be configured as a
“Slave.”.
CAUTION: If you are installing a MultiVOIP 400/800 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 MultiVOIP 400/800 and connect it to the Command port on the Slave MultiVOIP 400/800.
2.
Win 3.1 users - from the Program Manager, click the MultiVOIP 800 Configuration icon in the
MultiVOIP 400/800 Program Group. The Main menu is displayed.
Win2000/NT/98/95 users - from your desktop, click Programs I MultiVOIP 800 I MultiVOIP 800
Configuration. The Main menu is displayed.
3.
Click IP to display the IP Setup dialog box.
Click (check) the Diffserv box to enable differentiated services on routers that support this service. This feature provides priority to voice packets so that 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, change the Frame
Type by clicking on the drop-down arrow. The Frame Type choices are TYPE_II and SNAP.
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MultiVOIP 400/800 User Guide
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 MultiVOIP 400/800, and the
Gateway Address is the IP address of the device (e.g., network router) connected to the Internet/
Intranet.
Click OK when you are finished. The Main menu is displayed.
5.
From the Main menu, click Voice Channels to display the Channel Setup dialog box. The
Channel Setup dialog box is displayed. The four tabs in this dialog box define the channel interface, voice/fax parameters, Billing/Misc parameters, and regional telephone parameters for each channel.
42
Configure each channel for the type of interface you are connecting to. The Interface tab defaults to Channel 1 in the Select Channel field. To change the channel number, click the drop-down arrow and the list of channels is displayed. Highlight the channel you want to configure.
Feature options are enabled or disabled (grayed out) according to the interface type that you select. The one option available for all interface types is the Inter Digit Time option. This 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.
6.
The Interface group defaults to FXS (Loop Start). Select the interface option that corresponds to the interface type being connected to the Voice/Fax Channel 1 jack on the back panel of the
MultiVOIP 400/800.
FXS (Loop Start): If a station device; e.g., an analog telephone, fax machine, or KTS (Key
Telephone System) is connected to the Voice/Fax connector on the back of the unit, FXS (Loop
Start) will likely be the correct Interface option.
FXS (Ground Start): If the station device uses ground start, then choose the FXS (Ground Start) option. Refer to the device’s user documentation.
For both FXS Loop Start and FXS Ground Start , the Ring Count FXS window allows you to set the maximum number of rings output on the FXS interface before hanging up and releasing the line to another call. The default setting is 8 counts.
Note: Zero (0) means no rings - caller hears a busy tone.
FXO: If you are using an analog extension from your PBX, then choose the FXO option. Check with your in-house phone personnel to verify the connection type.
Chapter 3 - Software Loading and Configuration
If FXO is selected, the Dialing Options Regeneration, Flash Hook Timer, and Ring Count groups are enabled. Check with your local in-house phone personnel to verify whether your local
PBX dial signaling is Pulse or tone (DTMF). Then, set the Regeneration option accordingly. The
Flash Hook Timer allows you to eter the time, in milliseconds, for the duration of the flash hook signals output on the FXO interface. The default setting is 600 milliseconds. The Ring Count
FXO window allows you to set the number of rings received on the FXO interface before the
MultiVOIP 400/800 answers the incoming call. The default setting is 2 counts.
Note: Zero (0) means that the MultiVOIP 400/800 never answers.
For FXO-to-FXO communications, you can enable a specific type of FXO Disconnect; Current
Loss, Tone Detection, or Silence Detection. (Check with your in-house phone personnel to verify the preferred type of disconnect to use.) Enabling Tone Detection activates the
Disconnect Tone Sequence options. For Disconnect Tone Sequence, you can select from dropdown lists either one or two tones that will cause the line to be disconnected; the person hanging up a call must then hit the key(s) that will produce those tones. For Silence Detection, select
One Way or Two Way, then set the timer for the number of seconds of silence before disconnect.
Note that the default value of 15 seconds may be shorter than desired for your application.
E&M: If you are connecting to an analog E&M trunk on your PBX, then choose the E&M interface option to enable the E&M Options group. Check with your local in-house phone personnel to determine if the signaling is Dial Tone or Wink and if the connection is 2-wire or 4-wire. If Wink signaling is used, then the Wink Timer is enabled with a default of 250 milliseconds. The range of the Wink Timer is from 100 to 350 milliseconds. Consult with your local in-house phone personnel for this timer setting.
Note: After configuring a given channel, you can copy that channel’s configuration by clicking the
Copy button and everything on the Interface tab will be copied to the designated channel.
7.
Repeat the above step to configure the interface type for voice/fax channels 2 - 4 or 2 - 8.
8.
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.
9.
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 in to the voice channel before it is sent over the Internet to the remote MultiVOIP 400/800; and, 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 drop-down lists in the Voice Gain group. The valid range is +31dB to –31dB with a recommended/default value of 0.
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MultiVOIP 400/800 User Guide
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 drop-down lists in the
DTMF Gain group.
Note: Only change the DTMF gain under the direction of Multi-Tech Technical Support supervision.
10. To change the voice coder, first select the channel by clicking the Select Channel down arrow
(highlighting the channel number) then click Manual in the Coder group. To select the appropriate coder, click the Selected Coder down arrow and highlight your new voice coder entry.
If you changed the voice coder, ensure that the same voice coder is used on the voice/fax channel you are calling; otherwise, you will always get a busy signal.
Note: If you allow the Coder to be selected automatically, then you need to select the Max
Bandwidth from the drop-down list. Check with your VOIP administrator to determine how much bandwidth is available.
11. The Fax group enables you to send/receive faxes on the selected voice/fax channel. You can set the maximum baud rate for faxes and the fax volume in the two drop-down lists and change the jitter value in milliseconds.
When receiving fax packets from a remote MultiVOIP 400/800, 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 MultiVOIP 400/800 uses a Jitter Buffer. The Jitter Value field allows the MultiVOIP
400/800 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 50 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.
12. You can enable the voice/fax advanced features by clicking (checking) the silence compression, echo cancellation, or forward error correction options.
The Silence Compression option defines whether silence compression is enabled (checked) for this voice channel. If silence compression is enabled, the MultiVOIP 400/800 will not transmit voice packets when silence is detected, thereby reducing the amount of network bandwidth that is being used by the voice channel.
The Echo Cancellation option defines whether echo cancellation is enabled (checked) for this voice channel. If echo cancellation is enabled, the MultiVOIP 400/800 will remove echo-delay which improves the quality of sound.
The Forward Error Correction (FEC) option 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.
Note: After configuring a given channel, you can copy that channel’s configuration by clicking the
Copy button and everything on the Voice/Fax tab will be copied to the designated channel.
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Chapter 3 - Software Loading and Configuration
The Billing/Misc tab displays the parameters for auto call, automatic disconnection, billing options and dynamic jitter buffer.
13. If you want to dedicate a local voice/fax channel to a remote voice/fax channel (so you will not have to dial the remote channel), click the Auto Call Enable option in the Auto Call group. Then enter the phone number of the remote MultiVOIP 400/800 in the Phone Number field.
14. 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.
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.
15. 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.
16. A minimum and maximum set of values can be set for Dynamic Jitter Buffer. When receiving voice packets from a remote MultiVOIP 400/800, 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 MultiVOIP 400/800 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the
MultiVOIP 400/800 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 MultiVOIP
400/800 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.
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MultiVOIP 400/800 User Guide
Note: After configuring a given channel, you can copy that channel’s configuration to the other channel by clicking the Copy button. Everything on the Billing/Misc tab will be copied to the designated channel.
If your country/region is not the default USA, click the Regional tab and proceed to step 17; otherwise, proceed to step 18 to begin building your phone directory database.
17. To change the Tone Pairs on the Regional tab, click the Country/Region down arrow and highlight your specific country or region.
Note: If your country or region is not listed, click the Custom button to define it.
The Tone Pairs group enables you to select/modify the parameters according to choice. Click
OK when finished and proceed to step 6 to begin building your phone directory database.
18. 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.
46
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.
Chapter 3 - Software Loading and Configuration
19. Enter the IP address (204.022.122.118) of the New York Office MultiVOIP 400/800 in the Master
IP Address field and enable the Send Status Report to Master so that status reports are sent to the Master MultiVOIP 400/800.
Note: In a Dial-On-Demand (DOD) network, you should leave Send Status Report to Master disabled (not checked). 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.
20. Click OK to return to the Main menu.
21. Click Download Setup to write the new configuration to the slave unit. The Save Setup dialog box is displayed.
22. Select (check) the Save Current Setup as User Default Configuration and click OK. The Writing
Setup dialog box is displayed as the setup configuration is written to the MultiVOIP 400/800.
After the setup is written to the MultiVOIP 400/800, it reboots.
23. Check that the Boot LED on the MultiVOIP 400/800 is off after the download is complete. This may take several minutes as the MultiVOIP 400/800 reboots.
24. You are returned to the Main menu.
Your MultiVOIP 400/800 is operational at this time.
Repeat the process for each of the slave units. When all slaves have been configured, go to
“Deploying the Network.”
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MultiVOIP 400/800 User Guide
Deploying the VOIP Network
With the Proprietary Protocol option enabled on the Phone Directory Database dialog box, the
VOIP Administrator must develop the VOIP Dialing Directory and deploy the pre-configured slave
MultiVOIP 400/800s to their remote sites. The remote site administrators need only connect power to the pre-configured MultiVOIP 400/800, connect the MultiVOIP 400/800 to their 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 MultiVOIP
400/800s are configured as “Master” and cannot be downloaded. In this case, each MultiVOIP 400/
800 Phone Book will be programmed with phone numbers for its own channels. These phone numbers are remotely registered or pre-registered with the H.323 Gatekeeper (See the “Registering with a Gatekeeper Phone Directory” section discussed earlier.
VOIP Administrator
The VOIP administrator is responsible for the following two steps.
1.
Establish your VOIP Dialing Directory based on your Phone Directory Database for the numbers to connect the MultiVOIP 400/800s to your VOIP network and the telephone extension number you need to connect to the Voice/Fax channels. A sample VOIP Dialing Directory is provided below for your consideration and use.
Note: The # character in the dialing sequence in the table below acts as a delimiter between the
VOIP phone number and the PBX extension.
VOIP Dialing Directory
Call Process
To call from Dialing
Sequence
Headquarters to
Sales Office
Pick up telephone and dial a trunk extension number (e.g., 1, 2, 3, or4).
Second dial tone is generated, dial the Sales
Office MultiVOIP (201).
Attendent connects you to Sales office extension of called party
1#
201#
5123
Headquarters to
Marketing Office
Pick up telephone and dial an extension number (e.g., 1,2,3, or4).
Second dial tone is generated, dial the
Marketing Office MultiVOIP (401).
Third dial tone is generated, dial the line access code (9).
Then dial the extension number of your calling party (6128).
2#
401#
9#
6128
3#
301
Headquarters to
Regional Office
Pick up telephone and dial an extension number (e.g., 1,2,3, or 4).
Second dial tone is generated, dial the
Regional Office MultiVOIP (301).
Marketing Office
to
Headquarters
Pick up telephone and dial a trunk extension number (e.g., 9 or 10).
Second dial tone is generated, dial the
Headquarters MultiVOIP (101)
Third dial tone is generated, dial extension number of your calling party (4124).
Marketing Office
to
Regional Office
Pick up telephone and dial a trunk extension number (e.g., 9 or 10).
Second dial tone is generated, dial the regional MultiVOIP (301) and telephone rings.
9#
101#
4124
10#
301
2.
Send the slave MultiVOIP 400/800s to their remote sites.
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Chapter 3 - Software Loading and Configuration
Remote Site Administrator
The following steps are for MultiVOIP 400/800 H.323 endpoints. For non-MultiVOIP 400/800 H.323
endpoints, refer to the appropriate installation documentation.
3.
Unpack your MultiVOIP 400/800.
4.
Connect one end of the power supply to a live AC outlet and connect the other end to the Power connection on your MultiVOIP 400/800 (See Figure 3-1).
Voice/Fax Channel
Connections
10BASET
E&M FXO FXS
ETHERNET POWER
E&M
FXO
FXS
Power Connection
PSTN
Ethernet Connection
Figure 3-1. Remote Site Cable Connection
5.
Connect a network cable to the ETHERNET 10Base-T (RJ-45) connector on the back of your
MultiVOIP 400/800.
6.
If you are connecting a station device (e.g., analog telephone, fax machine, or Key Telephone
System (KTS) to your MultiVOIP 400/800, connect the smaller end of a special adapter cable
(supplied) to the Voice/Fax Channel 1 FXS connector on the back of the MultiVOIP 400/800 and the other end to the station device.
If you are connecting a PBX extension to your MultiVOIP 400/800, connect the smaller end of a special adapter cable (supplied) to the Voice/Fax Channel 1 FXO connector on the back of the
MultiVOIP 400/800 and the other end to the PBX extension.
If you are connecting an E&M trunk from a telephone switch to your MultiVOIP 400/800, connect one end of an RJ-45 phone cord to the Voice/Fax Channel 1 E&M connector on the back of the unit and the other end to the trunk phone jack.
If you are connecting to an E&M trunk, you need to ensure that the E&M trunk jumper is in the correct position for the E&M type trunk. The default E&M jumper position is E&M type 2. To change the E&M jumper position, perform the E&M jumper block positioning procedure.
7.
Repeat the above step to connect the remaining telephone equipment to each Voice/Fax
Channel on your MultiVOIP.
8.
Turn on power to the MultiVOIP 400/800 by placing the ON/OFF switch on the back panel to the
ON position. Wait for the BOOT LED on the MultiVOIP 400/800 to go OFF before proceeding.
This may take a couple of minutes.
9.
At this time your VOIP network should be fully operational. Dial one of the sites in your network using the dialing directory supplied by your network administrator.
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MultiVOIP 400/800 User Guide
If your H.323 endpoint is not a MultiVOIP 400/800 (e.g., a PC with Netmeeting software), the
Systems Administrator for your VOIP network should do the following.
1.
Make certain that the H.323 endpoint (e.g., PC with Netmeeting, router with voice, etc.) have been properly connected to the network. Refer to the appropriate end user documentation.
2.
Acquire the relevant IP Addresses and/or H323 IDs (H323 IDs are required if Gatekeeper is enabled) from the other H.323 endpoints that will participate in H.323 calls.
3.
Enter these IP Addresses and H323 IDs (H323 IDs are required if Gatekeeper is enabled) in your phonebook directory.
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Chapter 4 - MultiVOIP 400/800 Software
MultiVOIP 400/800 User Guide
Introduction
This chapter describes various features of the MultiVOIP 400/800 software that enable you to change
(update) the configuration of your MultiVOIP 400/800. The basic configuration parameters were established during the loading of the software ( Chapter 3 ). The MultiVOIP 400/800 software and configuration utilities described in this chapter enable you to change that initial configuration as necessary.
The primary interface to the MultiVOIP 400/800 software is the Main menu (MultiVOIP 800 Setup is on the title bar) which includes individual buttons that enable you to quickly and easily select a desired function. These features are discussed in detail in the MultiVOIP 400/800 Configuration section later in this chapter.
The MultiVOIP 800 Configuration (Main menu) utility along with eight other configuration utilities provide full software functionality for your MultiVOIP 400/800. The Configuration Port Setup utility enables you to change the method by which you access the MultiVOIP 400/800 (i.e., through a direct connection of a PC to the Command Port on the MultiVOIP 400/800, or via your Internet or LAN connection to the LAN port on the MultiVOIP 400/800). The Date and Time Setup utility enables you to easily set the date and time used for data logging in the MultiVOIP 400/800. The Download
Factory Defaults utility enables you to return the configuration to the original factory settings. The
Download Firmware utility enables you to download new versions of firmware as enhancements become available. The Download User Defaults utility enables you to repeat the download user defaults process (part of software installation) and update the MultiVOIP 400/800 configuration with any necessary changes. The Download Voice Coders utility enables you to download voice coders to the MultiVOIP 400/800 after repair or upgrade. The Download H.323 Stack utility enables you to download the H.323 protocols to the MultiVOIP 400/800 after repair or upgrade. The Uninstall
MultiVOIP 800 Configuration utility removes most of the MultiVOIP 400/800 software from your PC.
The MultiVOIP 400/800 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 MultiVOIP 400/800 software operates in a Microsoft Windows environment. The MultiVOIP 400/
800 program group contains icons for all the utilities described above. In Windows 2000/NT/98/95, you can access the individual utility programs either by clicking Start | Programs | MultiVOIP 800 |
(utility), or by double-clicking the utility icon in the MultiVOIP 400/800 program group shown here:
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Chapter 4 - MultiVOIP Software
MultiVOIP 400/800 Configuration
The MultiVOIP 800 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 MultiVOIP 400/800 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
MultiVOIP 400/800 or is communicating with the Ethernet port. The last field on the status bar is the
Rights field which displays either Read/Write or Read only rights. The first user to communicate with the MultiVOIP 400/800 has Read/Write rights that enable the user to view and/or change the configuration of the MultiVOIP 400/800. Any additional users have Read Only rights and can only display the configuration of the MultiVOIP 400/800 but are prohibited from changing the configuration.
53
MultiVOIP 400/800 User Guide
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 MultiVOIP 400/800 is operating. The Channel Setup dialog box, accessed by clicking the Voice Channels button on the
Setup menu, has four tabs that display the following categories of channel information -- Interface,
Voice/Fax, Billing/Misc, and Regional.
54
Interface
The Interface tab defines the parameters related to the physical interface of the voice/fax channel.
Depending on the interface type selected (FXS, FXO, or E&M), other options on the interface tab will be grayed out (become inactive) indicating that they do not apply to the selected interface. The Inter
Digit Time feature applies to all interface types.
The Inter Digit Time (in seconds) option in the Dialing Options group defines the amount of time the
MultiVOIP 400/800 waits between digits as they are entered by the user. If this timer expires, the
MultiVOIP 400/800 will immediately attempt to match the digits entered to an entry in the Phone
Directory Database. The range for this option is 2 to 100 with a default of 2.
If the interface type is FXO, the Regeneration group in the Dialing Options group defines how the
MultiVOIP 400/800 recreates telephone numbers that were detected at the remote end. You can select Pulse (for rotary dial telephones) or DTMF Tone dialing (touch-tone), depending upon the dialing type that is supported by the PBX or exchange. When FXO is the Interface, the Flash Hook
Timer field is enabled (activated) Enter the time, in milliseconds, for the duration of the flash hook signals output on the FXO interface. The default is 600 milliseconds.
FXS Interface
The FXS Interface is used to connect telephones, fax machines, key telephone systems, etc., to the
MultiVOIP 400/800. In addition, you need to select either Loop Start or Ground Start. Most of the equipment mentioned will use Loop Start which is the default.
FXO Interface
The FXO Interface is used to connect PBX extensions or central office telephone lines. You also, need to select DTMF or Pulse dialing in the Regeneration field of the Dialing Options group. If you are unsure of the correct selection, contact the personnel in charge of your PBX or your local telephone company to determine whether pulse or DTMF should be used.
Chapter 4 - MultiVOIP Software
E&M Interface
The E&M Interface is used to connect PBX E&M trunks. You will need to select between Dial Tone or
Wink signaling and also between 2-wire and 4-wire mode. If wink signaling is selected, the wink timer field becomes active with a range from 100 to 350 milliseconds. Contact the personnel in charge of your PBX to determine the proper configuration of these settings.
FXO Disconnect On
The FXO Disconnect On option applies when two MultiVOIP 400/800s are used in an FXO-to-FXO configuration. When you have an FXO-to-FXO configuration, you need to determine the method of terminating the call. Three methods of terminating the call are provided: Current Loss, Tone
Detection, or Silence Detection. Current Loss is the preferred method.
Current Loss has to be supported by your PBX or local telephone company. Current Loss terminates the call when the PBX or local telephone company switch detects a person hanging up the phone and opens the local circuit for a minimum of 600 milliseconds.
Tone Detection terminates the call when the party who wishes to disconnect enters a one or two digit sequence on the telephone keypad. Valid digits are zero to nine, *, #, and A thru D.
Note: A through D are extended DTMF tones supported by some PBX or central office equipment and are not the same as letters a - d on the standard telephone key pad.
Silence Detection can be silence in one direction or silence in both directions for a specified amount of time. The amount of time is defined by the entry in the Silence Timer. The range of the Timer is from one to 65535 seconds (roughly 18 hours). The default is 15 seconds.
Ring Count
This field enables you to enter the maximum number of rings output on the FXS interface (default is
8) before hanging up and releasing the line to another call or the number of rings (default is 2) that must be received before the FXO port answers an incoming call.
A setting of 0 (zero) on the FXS interface disables the generation of rings. The caller will receive a
“Busy” tone.
A setting of 0 (zero) on the FXO interface causes the FXO port to ignore rings from the attached PBX or exchange, disabling access to the MultiVOIP 400/800.
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MultiVOIP 400/800 User Guide
Voice/Fax
The Voice/Fax tab controls voice and DTMF gain, voice coder, fax settings, and advanced options.
56
The Voice Gain group enables you to select the Input and Output voice gain. Gain is the increased signaling power that occurs as the signal is boosted by the MultiVOIP 400/800. The Input Gain dropdown list defines the input gain for this voice channel. Before your MultiVOIP 400/800 digitizes voice, the volume can be increased or decreased. Input gain modifies the level of the audio coming in to the voice channel before it is sent over the Internet to the remote MultiVOIP 400/800. The valid range for this option is +31dB to –31dB. The recommended and default value is 0. The Output Gain drop-down list defines the voice output gain for this voice channel. Before your MultiVOIP 400/800 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 fields control the gain in dB (decibels) of the
High and Low tones in the tone pairs; the default gain values are -4 dB and -7 dB, respectively.
DTMF Gain should not be changed except under supervision of MultiTech’s Technical Support.
The MultiVOIP 400/800 supports many state-of-the art ITU (International Telecommunications Union) voice coders. The Voice Coder drop-down menu enables you to select from a range of coders with specific bandwidths. The higher the bps rate, the more bandwidth is used. The channel that you are calling has to have the same voice coder selected; otherwise, you will always get a Busy signal.
The Fax group enables a fax machine to transmit and receive faxes through the MultiVOIP 400/800. If a fax machine is connected to one of the voice/fax channels, the Max Baud Rate should be set to match the baud rate of the fax machine (refer to user documentation). The Fax 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 (see Chapter 6 - Warranty, Service and Tech Support ).
The Jitter Value setting defines the inter-arrival 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 MultiVOIP 400/
800 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 a particular 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.
Chapter 4 - MultiVOIP Software
Billing/Misc
This tab controls the parameters for auto call, automatic disconnection, billing options, and dynamic jitter buffer.
The Auto Call option allows the local MultiVOIP 400/800 to call a remote MultiVOIP 400/800 without the user having to dial a Phone Directory Database number. As soon as you access the local
MultiVOIP 400/800 voice/fax channel, the MultiVOIP 400/800 immediately connects to the remote
MultiVOIP 400/800 that you identified in the Remote MultiVOIP 400/800 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.
Billing Options can be used to track the cost of Inbound and/or Outbound calls on any of the three interfaces (FXO, FXS, or E&M). The amount to be charged in cents is entered in the Charge ( )
Cents field together with the associated time duration in the Per ( ) Seconds field. While a given call is active, the accumulated charges can then be viewed on the Call Progress dialog box. When the call ends, the charges are transferred to a Log File that can be viewed by highlighting the call event in the Log Entries dialog box and selecting Details.
Dynamic Jitter Buffer defines a minimum and a maximum jitter value for voice communications.
When receiving voice packets from a remote MultiVOIP 400/800, 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 MultiVOIP 400/800 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the
MultiVOIP 400/800 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 MultiVOIP 400/800 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.
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MultiVOIP 400/800 User Guide
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
The Regional tab controls the voice communications for the country or region in which the MultiVOIP
400/800 is being used.
From the Country/Region drop-down list you can select the country or region for which you are configuring the MultiVOIP 400/800. The Tone Pairs group always displays the tones used in the country or region currently selected. In addition to Australia, Central America, Chile, Europe, France,
Japan, UK, and USA, there is a Custom selection (with defaults identical to USA) that will make the
Custom button active. Clicking the Custom button enables you to edit the Tone Pairs and establish custom sets of tone pairs for Dial Tone, Ring, and Busy on a Custom Tone Pair Settings dialog box.
The Pulse Generation Ratio group contains two ratios: the 60/40 is for the USA, and the 67/33 ratio is for international applications.
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Chapter 4 - MultiVOIP Software
Changing the Phone Directory Database
The Phone Directory Database dialog box displays all the phone numbers in your MultiVOIP 400/
800 network. The database displays the phone numbers in numerical order with destination details
(i.e., IP Address (or H323 ID), channel assignment, and a brief description of the entry (e.g., New
York Office 1). The method for changing the phone directory database is dependent on whether the
Gatekeeper option or the Proprietary Phonebook option is enabled.
Access this database by clicking the Phone Book button on the Main MultiVOIP 400/800 menu.
Proprietary Phonebook Enabled
With the Proprietary Phonebook option enabled, you can add, delete, or edit any entry in the
MultiVOIP 400/800 phone directory database. You can change the master - slave relationship of the entry, view/print entries, and view the details of Hunt Groups.
To add an entry to the Phone Directory database, perform the steps as described in Chapter 3. To edit an entry in the Phone Directory Database, click on the relevant entry in the Phone Direcory database dialog box and then click the Edit button. The Add/Edit Phone Entry dialog box will be displayed with relevant information.
In the Station Information group, enter/change the unique phone number for the device in the
Phone Number field (e.g., 101); enter/change the Channel number (1 - 4 or 1 - 8) in the Voice
Channel field; and provide/change the optional Description. In the Station Identification group, select the Hunt group from the Hunt Group drop-down list and select the appropriate Hunt Group you want the phone number to be associated with; enter the appropriate IP Address of the entry in the IP Address field (e.g., 204.22.122.118); and accept the Port default.
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MultiVOIP 400/800 User Guide
In the Station Information group, the phone number does not have to be a conventional phone number; for example, it does not have to be 717-5565. It can be a single digit or several digits. For example, you could enter a phone number of 101 with a description of New York Office 1.
The Station Identification group enables you to assign the entry to a Hunt Group, provide the IP
Address, 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.
To establish the phone directory database for a remote MultiVOIP 400/800, you do not have to enter phone numbers, but you have to check the Slave option in the Database Type group. When you click the Slave option, the Update Database From group becomes active. You need to enter the IP address of the MultiVOIP 400/800 that you established as the Master.
Clicking the Slave Status button on the Phone Directory Database dialog box allows you 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 is displayed with its IP Address, current line status, and the description of the phone number.
You can view the details of the current Hunt Group (e.g., HUNT GROUP #1) configuration by clicking on the Phone Directory Database’s Hunt button. The current Phone numbers for HUNT GROUP #1 are displayed.
60
Highlight 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.
Chapter 4 - MultiVOIP Software
With Gatekeeper Enabled
Enabling the Gatekeeper option on the Phone Directory Database dialog box means that 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 conferences on the LAN.
Adding and/or editing entries in the H.323 Gatekeeper database must be done from the Add/Edit
Phone Entry dialog box. Deleting entries is done by highlighting the entry to be deleted in the Phone
Directory Database dialog box and clicking the Delete button.
To change a parameter of an existing entry already registered with the Gatekeeper database, highlight the entry currently listed in the Phone Directory Database dialog box and click the Edit button. The Add/Edit Phone Entry dialog box is displayed with the current information.
If you simply want to change the Description field (e.g., John no longer works at the company and you want to change the name to “Bob’s Desk”), enter the new description and click OK. The update will be made automatically to the Gatekeeper database. However, if you need to change any of the other parameters in the Add/Edit Phone Entry dialog box, you will need to consult with the
Gatekeeper administrator. Contact the Gatekeeper administrator (e.g., by phone or e-mail) and determine together that any changes to the phone entry’s parameters will not create duplication.
When consulting with the Gatekeeper administrator, note that the Station Information group phone number does not have to be a conventional phone number. For example, it does not have to be 717-
5565. It can be a single digit or several digits. You could enter a phone number of 101 with a description of New York Office 1 and assign it to channel 1.
Note: The phone number must not conflict with other endpoints.
When consulting with the Gatekeeper administrator, note that the Station Identification group H323
ID is a unique identifier that represents the name of the gateway with the name of the Gatekeeper appended to the end (e.g., [email protected]). 1720 is the H.323 industry standard Port number of all H.323 endpoints.The IP Address field is for the IP Address of the New York 1 endpoint and is an optional (redundant) entry.
Note: The Port field is enabled (activated) only when the IP Address of the H.323 endpoint is entered.
When you enter this information and click OK, the information is updated automatically at the
Gatekeeper database.
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MultiVOIP 400/800 User Guide
Changing IP Parameters
The IP Setup dialog box establishes the IP addressing for the local Ethernet LAN and defines the
Internet gateway address. The IP Setup dialog box is accessed by clicking the IP button on the
MultiVOIP 800 Setup menu.
With IP Setup dialog box displayed you can change the status of differential services, 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.
Clicking (checking) the Enable Diffserv box enables Differentiated Services (default is disabled).
Differentiated Services provides priority to voice packets so that they are not delayed whenever large data files need to be downloaded. You will need to check with your systems administrator to determine if any routers in the VOIP network support this feature. If the answer is yes, then you would want to enable this function.
The Frame Type drop-down 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, change the Frame Type by clicking the drop-down arrow. 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.
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Chapter 4 - MultiVOIP Software
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 MultiVOIP 800 Setup menu by clicking the Call Progress button.
The ratio of Packets Lost versus Packets Received provides a general indication of the integrity of the Internet connection. To reduce the frequency of lost packets, select a low-bit-rate coder, such as,
G.723 or Netcoder. In addition, enabling the Forward Error Correction option on the Voice/Fax tab on the Channel Setup dialog box will enable the MultiVOIP 400/800 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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MultiVOIP 400/800 User Guide
Applications Setup
Clicking the Others button 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 MultiVOIP 400/800 for Internet security. These remote configuration applications enable remote viewing and changing of the MultiVOIP 400/800 configuration, or updating firmware, from anywhere on the connected internetwork.
Verify that the desired applications are enabled (checked). All applications are checked (enabled) as the default condition. To disable a given application, click to uncheck the check box and disable support.
SNMP related operations can be performed only when the SNMP Agent is enabled (checked) on this dialog box. The IP address of the system (i.e., SNMP Manager) that will receive the Traps from the
MultiVOIP 400/800 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 MultiVOIP 400/800 currently supports a maximum of two community users at a time, and they can be assigned either Read/Write or Read Only rights.
The Password group enables you to enter a password, up to 13 alphanumeric characters, to be used for Internet Security. Once the password is entered (in the MultiVOIP Password field) and confirmed (in the Confirm Password field), remote users will be queried to enter the password before gaining access to the MultiVOIP 400/800 for configuration purposes.
Note: If you forget your password, contact Multi-Tech Technical Support for instructions.
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For more information on using these applications, click the on-line Help button or refer to Chapter 5,
Remote Configuration and Management.
Chapter 4 - MultiVOIP Software
Viewing Statistics
The Statistics dialog box enables you to view statistics for major events of the MultiVOIP 400/800 operation. This dialog box is accessed by clicking the Statistics button on the MultiVOIP 800 Setup menu.
Statistics can be a helpful troubleshooting tool. For example, viewing the Voice Channel statistics you can see the attempted and completed calls, call duration, average call length, bytes/packets sent and received, etc.
IP Statistics
IP is a connection-less network protocol residing in the network layer of a conventional OSI layered
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.
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MultiVOIP 400/800 User Guide
For the most part, these statistics are informational, and their use as a troubleshooting tool will be contingent on the applications running in the upper layers. For example, if you were having problems connecting to the MultiVOIP 400/800’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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Chapter 4 - MultiVOIP Software
Viewing Logs
The Log Entries dialog box displays a chronological history of all calls into and out of this unit. This dialog box is accessed from the Logs button on the Statistics dialog box.
The Log Entries dialog box displays each call as a sequentially numbered Event with the date, time, duration of the call, the status of the call (Successful or Unsuccessful), Mode (Voice or Fax), and the from and to numbers.
Viewing Log Entry Details
The Log Entry Details dialog box displays the status of a completed call. This dialog box displays the same details as the Call Progress dialog box after a call is completed.
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MultiVOIP 400/800 User Guide
Viewing Channel Totals
The Channel Totals dialog box displays Outgoing and Incoming calls with their Attempted and
Completed numbers for each channel on this MultiVOIP 400/800. 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 the Generate button. This function provides a hard copy of the Log Entries dialog box.
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Chapter 4 - MultiVOIP Software
Upgrade Procedures
Whenever you upgrade your version of the MultiVOIP 400/800 software, you must first install the new software on your PC. Then, download the Firmware, the Factory Defaults, the Voice Coders, and/or the H.323 stack to upgrade the MultiVOIP 400/800 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.
Three utility programs included in the MultiVOIP 400/800 software are to be used only after the unit has been repaired or upgraded. They are Download Firmware, Download Voice Coders, and
Download H.323 Stack.
Download Firmware
If you have obtained a new firmware version from the Multi-Tech Web site, the Multi-Tech FTP site, or another source, do the following:
Win3.1 users - In the Program Manager, double-click the Download Firmware icon in the
MultiVOIP 400/800 program group. The Open dialog box appears displaying the file list containing a single file, mtvoip.bin.
Win2000/NT/98/95 users - Click Programs | MultiVOIP 800 | Download Firmware. The Open dialog box appears displaying the file list containing a single file, mtvoip.bin.
After you select the mtvoip.bin file and click OK, the Downloading Code dialog box is displayed with a progress bar so you can monitor the download process.
After the new firmware file is downloaded to the MultiVOIP 400/800, the unit reboots and is then ready for use with upgraded firmware.
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MultiVOIP 400/800 User Guide
Download Coders
If you have obtained a new coders file, do the following:
Win3.1 users - In the Program Manager, double-click the Download Coders icon in the MultiVOIP
400/800 program group. The Open dialog box appears displaying the file list containing a single file, coders.hst.
Win2000/NT/98/95 users - Click Programs | MultiVOIP 800 | Download Coders. The Open dialog box appears displaying the file list containing a single file, coders.hst.
After you select the coders.hst file and click OK, the Downloading Voice Coder dialog box is displayed with a progress bar so you can monitor the download process.
After the file is downloaded to the MultiVOIP 400/800, the unit reboots and is then ready for use with upgraded voice coders.
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Chapter 4 - MultiVOIP Software
Download H.323 Stack
If you have obtained a new H.323 stack, do the following:
Win3.1 users - In the Program Manager, double-click the Download H.323 Stack icon in the
MultiVOIP 400/800 program group. The Open dialog box appears displaying the file list containing a single file, coders.hst.
Win2000/NT/98/95 users - Click Programs | MultiVOIP 800 | Download H.323 Stack. The Open dialog box appears displaying the file list containing a single file, h323.pdl.
After you select the h323.pdl file and click OK, the Downloading H.323 Stack dialog box is displayed with a progress bar so you can monitor the download process.
After the file is downloaded to the MultiVOIP 400/800, the unit reboots and is then ready for use with the upgraded H.323 stack.
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MultiVOIP 400/800 User Guide
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Chapter 5 - Remote Configuration and Management
MultiVOIP 400/800 User Guide
Introduction
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
Remote configuration requires the MultiVOIP 400/800 software to be loaded on the local PC. The local PC then controls the remote MultiVOIP 400/800 either via the modem connection or the LAN.
Modem-Based
To remotely configure a MultiVOIP 400/800, a local PC needs to be connected to a dial-up line and the MultiVOIP software configured to call the remote MultiVOIP. The remote MultiVOIP needs to have a modem connected to a dial-up line and the Command Port. Once the connection to the remote unit is made, you can change the configuration as required. Once the configuration is changed, you can download the new configuration to the remote MultiVOIP. Refer to the Modem-Based Remote
Configuration Procedure in this chapter to remotely configure a MultiVOIP 400/800.
1.
At the remote site, remove the serial cable from the PC to the Command Port connector on the back panel of the MultiVOIP 400/800.
2.
At the remote site, connect a special cable (Remote Configuration Cable) from the Command
Port connector on the back panel of the MultiVOIP 400/800 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 MultiVOIP 400/800 software on the local PC. When installed, click Start | Programs |
MultiVOIP 800 | Configuration Port Setup, or double click on the Configuration Port icon in the MultiVOIP 400/800 program group.
5.
The MultiVOIP 800 Setup dialog box is displayed.
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Chapter 5 - Remote Configuration and Management
Verify that the Communication Type is set for COM Port and the Select Port field is set for the
COM port of your local PC.
In the Dial String field, enter the AT command for dialing (ATDT) plus the phone number of the remote MultiVOIP 400/800.
If your Modem Initialization String, Initialization Response, or Connect Response values are different than the defaults in the dialog box, refer to your modem user documentation and change the default values to match your modem.
Click OK when you are satisfied with your selections.
6.
Run the MultiVOIP 400/800 Configuration program. Click Start | Programs | MultiVOIP 800 |
MultiVOIP Configuration, or double click on the MultiVOIP Configuration icon in the MultiVOIP
400/800 program group.
7.
The Dialing dialog box is displayed while software is dialing the remote MultiVOIP 400/800.
8.
The Reading Setup dialog box is displayed.
9.
The MultiVOIP 800 Setup menu is displayed. This is the dialog box of the remote MultiVOIP
400/800. Refer to the online Help provided with your software for a description of each dialog box and field within a dialog box.
10. After you have changed the configuration of the remote MultiVOIP 400/800, click Download
Setup to update the configuration. The remote MultiVOIP 400/800 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 is displayed.
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 may want to remove it from the Port Setup dialog box.
14. At the remote site, reconnect the MultiVOIP 400/800 to the serial port of the PC and from the
MultiVOIP 400/800 program group double click on the MultiVOIP Configuration icon to verify that the MultiVOIP 400/800 is running.
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MultiVOIP 400/800 User Guide
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.
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 MultiVOIP 400/800 software on the local PC. When installed, click Start | Programs |
MultiVOIP 800 | Configuration Port Setup, or double-click the Configuration Port Setup icon in the MultiVOIP 400/800 program group.
3.
The MultiVOIP Port Setup dialog box is displayed.
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Verify that IP is set for the Communication Type.
In the MultiVOIP 400/800 IP Address field, enter the IP Address of the remote MultiVOIP 400/
800.
4.
Click OK when you are satisfied with your selections.
5.
Run the MultiVOIP 400/800 Configuration program. Click Start | Programs | MultiVOIP 800 |
MultiVOIP Configuration, or double click the MultiVOIP Configuration icon in the MultiVOIP
400/800 program group.
The Reading Setup dialog box is displayed.
Chapter 5 - Remote Configuration and Management
6.
The MultiVOIP 800 Setup dialog box is displayed. This is the dialog box of the remote MultiVOIP
400/800. Refer to the online Help provided with your MultiVOIP 400/800 for the definition of each dialog box and field within a dialog box.
7.
After you have changed the configuration of the remote MultiVOIP 400/800, click Download
Setup to update the configuration. The remote MultiVOIP 400/800 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 on the MultiVOIP Configuration icon in the MultiVOIP 400/800 program group to verify that the MultiVOIP 400/800 is running.
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MultiVOIP 400/800 User Guide
Remote Management
This section describes typical client applications that can be used to configure the MultiVOIP 400/800 remotely. It is important to note that although any subsequent changes to configuration can be made using these applications, the initial setup and configuration of the MultiVOIP 400/800 must be done on the local PC using the MultiVOIP 400/800 software provided with your unit.
Although establishing access to the MultiVOIP 400/800 varies between applications, the configuration functions mirror those of the MultiVOIP 400/800 software. For more information on MultiVOIP 400/
800 software, refer to Chapter 4 - MultiVOIP 400/800 Software .
Telnet
A typical Telnet client application is described next. The MultiVOIP 400/800 has a built-in Telnet
Server that enables Telnet client PCs to access the MultiVOIP 400/800. A typical Telnet client is allowed to configure the MultiVOIP 400/800. In addition, the MultiVOIP 400/800 can be remotely accessed and configured from anywhere on the Internet through its Web interface.
The TCP/IP stack has to be loaded before the Telnet client (a Windows application) will run. The
Telnet Server option has to be selected from the Applications Setup dialog box using the MultiVOIP
400/800 Configuration icon. Double click the Telnet icon (or shortcut) and a blank Telnet screen is displayed. Click Connect | Remote System and the Connect dialog box is displayed. Select from the drop-down list (or enter) a Host Name (the IP address of the MultiVOIP 400/800). In this example, the Host Name is 192.168.2.8. Then select telnet from the Port drop-down list and vt100 from the Term Type drop-down list.
Once you have entered a valid Host Name (IP address), Port, and Term Type, click the Connect button to be connected to the target MultiVOIP. The MultiVOIP Telnet Server screen is displayed.
MultiVOIP Telnet Server Menu
The MultiVOIP Telnet Server menu provides three basic options: Voice over IP Configuration, Phone
Directory Database, and Phone Directory Configuration. A further option enables you to close the
Telnet session.
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Chapter 5 - Remote Configuration and Management
Voice over IP Configuration
Selecting Option 1 displays the Main menu, which allows further configuration options. These options include Protocol Stacks (option 1), Applications (option 2), System Information (option 3), and
Voice Channels (option 4). For further descriptions of these options, refer to Chapter 4 - MultiVOIP
400/800 Software .
Phone Directory Database
Selecting Option 2 allows you to add entries to the Phone Directory Database. Refer to Chapter 4 -
MultiVOIP 400/800 Software , for more details on the database.
Phone Directory Configuration
Selecting Option 3 allows you to configure and manage the Phone Directory. The various options are described in detail in Chapter 4 - MultiVOIP 400/800 Software .
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MultiVOIP 400/800 User Guide
WEB Management
The MultiVOIP 400/800 can be accessed, via a standard Web browser, from anywhere on the connected Internet. In order to provide this support, the WEB Server option has to be enabled from the Others button in the Main menu which displays the Applications Setup dialog box (see Chapter
4 - MultiVOIP 400/800 Software ).
Once enabled, users can access the MultiVOIP 400/800 by entering its IP address in the Address field of their web browser. The following screen is displayed.
If a Password was entered in the Applications Setup dialog box, then enter the password and click on the Enter button.
80
From this screen you can access all the configuration options. Refer to Chapter 4 - MultiVOIP 400/
800 Software , for a description of the various options.
Chapter 6 - Warranty, Service and Tech Support
MultiVOIP 400/800 User Guide
Introduction
This chapter starts out with statements about your MultiVOIP 400/800 2-year warranty. The next section, Tech Support, should be read carefully if you have questions or problems with your
MultiVOIP 400/800. It includes the technical support phone numbers, space for recording your product information, and an explanation of how to send in your MultiVOIP 400/800 should you require service. The final section explains how to receive support from the Internet.
Limited 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 years from date of purchase, or if proof of purchase is not provided, two years 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.
Addendum for North American Products
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), a telephone number, and if the product is out of warranty, a check or purchase order for repair charges.
Extended two-year overnight replacement agreements are available for selected products. Please refer to our Overnight Replacement Agreement for details on rates and coverages. Please direct your questions regarding technical matters, product configuration, verification that the product is defective, etc., to our Technical Support department at 1-800-972-2439.
Please direct your questions regarding repair expediting, receiving, shipping, billing, etc., to our
Repair Accounting department at (800) 328-9717 or (763) 785-3500.
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.
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Chapter 6 - Warranty, Service and Tech Support
Addendum for International Products
Distributors should contact Amex, Inc., for information about the repairs for your Multi-Tech product.
Amex, Inc.
2724 Summer Street NE Minneapolis, MN 55413
U.S.A. Tel: +(763) 331-3251
Fax: +(763) 331-3180
Please direct your questions regarding technical matters, product configuration, verification that the product is defective, etc., to our Technical Support department nearest you. When calling the U.S., please direct your questions regarding repair expediting, receiving, shipping, billing, etc., to our
Repair Accounting department at +(763) 785-3500 in the U.S.A., or a nearby Multi-Tech office which is listed on the “Multi-Tech Corporate Offices” sheet in this International Distributor Resource Kit.
Repairs for damages caused by lightning storms, water, power surges, incorrect installation, physical abuse, or uOut of Warranty Repair Costs
Refer to Multi-Tech System's web site at http://www.multitech.com for information about out of warranty repair costs.ser-caused damages are billed on a time-plus-materials basis.
Service
If your tech support specialist decides that service is required, your MultiVOIP 400/800 may be sent
(freight prepaid) to our factory. Return shipping charges will be paid by Multi-Tech Systems.
Include the following with your MultiVOIP 400/800:
• a description of the problem
• return billing and return shipping addresses
• contact name and phone number
• check or purchase order number for payment if the MultiVOIP 400/800 is out of warranty
(Check with your technical support specialist for the standard repair charge for your
MultiVOIP 400/800)
• if possible, note the name of the technical support specialist with whom you spoke
If you need to inquire about the status of the returned product, be prepared to provide the serial
number of the product sent.
Send your MultiVOIP 400/800 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 MultiVOIP 400/800 on the availability of local service and/or loaner units in your part of the country.
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MultiVOIP 400/800 User Guide
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: [email protected]
Internet: http://www.thesupplynet.com
Tech Support
Multi-Tech has an excellent staff of technical support personnel available to help you get the most out of your Multi-Tech product. If you have any questions about the operation of this unit, call 1-800-972-
2439. Please fill out the MultiVOIP 400/800 information (below), and have it available when you call.
If your MultiVOIP 400/800 requires service, the tech support specialist will guide you on how to send in your MultiVOIP 400/800 (refer to the next section).
Recording MultiVOIP 400/800 Information
Please fill in the following information on your Multi-Tech MultiVOIP. This will help tech support in answering your questions. (The same information is requested on the Warranty Registration Card.)
Model No.: _________________________
Serial No.: _________________________
Software Version: ____________________
The model and serial numbers are on the bottom of your MultiVOIP 400/800.
Please note status of your MultiVOIP 400/800 including LED indicators, screen messages, diagnostic test results, problems with a specific application, etc. Use the space below to note the MultiVOIP 400/
800 status:
________________________________________________________________________________________________________
________________________________________________________________________________________________________
________________________________________________________________________________________________________
________________________________________________________________________________________________________
______________________________________________________________________________________________________________
______________________________________________________________________________________________________
About the Internet
Multi-Tech’s presence includes a Web site at: http://www.multitech.com
and an ftp site at: ftp://ftp.multitech.com
84
Appendixes
MultiVOIP 400/800 User Guide
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 nonhierarchical; MAC addresses are never the same as IP addresses.
When a host needs to send a datagram to another host on the same network, the sending application must know both the IP and MAC addresses of the intended receiver. Unfortunately, the IP process may not know the MAC address of the receiver. The Address Resolution Protocol (ARP), described in RFC 826 (located at ftp://ds.internic.net/rfc/rfc826.txt) provides a mechanism for a host to determine a receiver’s MAC address from the IP address. In the process, the host sends an ARP packet in a frame containing the MAC broadcast address; and then the ARP request advertises the destination IP address and asks for the associated MAC address. The station on the LAN that recognizes its own IP address will send an ARP response with its own MAC address. An ARP message is carried directly in an IP datagram.
Other address resolution procedures have also been defined, including those which allow a diskless processor to determine its IP address from its MAC address (Reverse ARP, or RARP), provides a mapping between an IP address and a frame relay virtual circuit identifier (Inverse ARP, or InARP), and provides a mapping between an IP address and ATM virtual path/channel identifiers (ATMARP).
The TCP/IP protocol suite comprises two protocols that correspond roughly to the OSI Transport and
Session Layers; these protocols are called the Transmission Control Protocol (TCP) and the User
Datagram Protocol (UDP). Individual applications are referred to by a port identifier in TCP/UDP messages. The port identifier and IP address together form a “socket”. Well-known port numbers on the server side of a connection include 20 (FTP data transfer), 21 (FTP control), 23 (Telnet), 25
(SMTP), 43 (whois), 70 (Gopher), 79 (finger), and 80 (HTTP).
TCP, described in RFC 793 ( ftp://ds.internic.net/rfc/rfc793.txt) provides a virtual circuit (connectionoriented) communication service across the network. TCP includes rules for formatting messages, establishing and terminating virtual circuits, sequencing, flow control, and error correction. Most of the applications in the TCP/IP suite operate over the “reliable” transport service provided by TCP.
UDP, described in RFC 768 (ftp://ds.internic.net/rfc/rfc768.txt) provides an end-to-end datagram
(connectionless) service. Some applications, such as those that involve a simple query and response, are better suited to the datagram service of UDP because there is no time lost to virtual
86
Appendix A - TCP/IP Description 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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MultiVOIP 400/800 User Guide
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.
88
Appendix B - Cabling Diagrams
Appendix B - Cabling Diagrams
Command Port Cable
13 12 11 10 9 8 7 6 5 4 3 2 1
25 24 23 22 21 20 19 18 17 16 15 14
To COMMAND PORT
Connector
PIN NO.
7
8
5
6
3
4
1
2
20
22
25
PIN NO.
7
8
5
6
3
4
1
2
20
22
25
TRANSMIT DATA (BA)
RECEIVE DATA (BB)
SIGNAL GROUND (AB)
LAN Cable
To
DTE
Device
(Terminal
Device i.e. ASCII
Terminal)
1 2 3 4 5 6 7 8
Pin
3
6
1
2
Circuit Signal Name
TD+ Data Transmit Positive
TD- Data Transmit Negative
RD+ Data Receive Positive
RD- Data Receive Negative
Voice/Fax Channel Connectors
1 2 3 4 5 6 7 8 2 3 4 5 2 3 4 5
Pin Connections
6
7
4
5
8
2
3
E&M Desc.
Function
1 M Input
E
T1
Output
4-Wire Output
R
T
R1
SG
SB
4-Wire Input, 2-Wire
4-Wire Input, 2-Wire
4-Wire Output
(Signal Ground) Output
(Signal Battery) Output
FXO Description FXS Description
4
5
2
3
N/C
Ring
Tip
N/C
2
3
4
5
N/C
Tip
Ring
N/C
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MultiVOIP 400/800 User Guide
Remote Configuration Cable
To
COMMAND PORT
Connector
PIN NO.
7
8
5
6
3
4
1
2
20
22
25
Male
PIN NO.
7
8
5
6
3
4
1
2
20
22
25
Male
FRAME GROUND
TRANSMIT DATA (TX)
RECEIVE DATA (RX)
REQUEST TO SEND (RTS)
CLEAR TO SEND (CTS)
SIGNAL GROUND
To
DCE
Device
(Communication
Device i.e. Modem)
90
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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MultiVOIP 400/800 User Guide
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
MVP400, MVP800
AU7USA-26050-DV-N
RJ-11C or RJ-11W
Multi-Tech Systems Inc.
2205 Woodale Drive
Mounds View, MN 55112
(763) 785-3500 Fax (763) 785-9874
92
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.
93
MultiVOIP 400/800 User Guide
94
Glossary
MultiVOIP 400/800 User Guide
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 56 Kbps leased (private) line service. Similar to services of MCI and Sprint.
ASDS is available in nx56/64 Kbps, 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 64 Kbps "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.
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 64 Kbps and one D-channel at 64 Kbps
(2B+D).
96
Glossary
Bell Operating Companies (BOC): The family of corporations created during the divestiture of AT&T. BOCs are independent companies which service a specific region of the US. Also called Regional Bell Operating Companies (RBOCs).
Bell Pub 41450: The Bell publication defining requirements for data format conversion, line conditioning, and termination for direct DDS connection.
Bell Pub 62310: The Bell publication defining requirements for data format conversion, line conditioning, and termination for direct DDS connection.
Binary Synchronous Communication (BSC): A form of telecommunication line control that uses a standard set of transmission control characters and control character sequences, for binary synchronous transmission of binary-coded data between stations.
Bit (Binary digIT): A bit is the basis of the binary number system. It can take the value of 1 or 0. Bits are generally recognized as the electrical charge generated or stored by a computer that represent some portion of usable information.
Bit Error Rate Test (BERT): A device or routine that measures the quality of data transmission. A known bit pattern is transmitted, and the errors received are counted and a BER (bit error rate) is calculated. The BER is the ratio of received bits in error relative to the total number of bits received, expressed in a power of 10.
Bit robbing: The use of the least significant bit per channel in every sixth frame for signaling. The line signal bits "robbed" from the speech pat conveys sufficient pre-ISDN telephony signaling information with the remaining line signal bits providing sufficient line signaling bits for recreating the original sound. See "robbed bit signaling".
Blue Alarm: An error indication signal consisting of all 1s indicating disconnection or attached device failure. Contrast "Red Alarm" and
"Yellow Alarm".
Bps (bits per second): A unit to measure the speed at which data bits can be transmitted or received. Bps differs from baud when more than one bit is represented by a single cycle of the carrier.
Bridges: 1. A functional unit that interconnects two local area networks that use the same logical link protocol but may use different medium access control protocols. 2. A functional unit that interconnects multiple LANs (locally or remotely) that use the same logical link control protocol but that can use different medium access control protocols. A bridge forwards a frame to another bridge based on the medium access control (MAC) address. 3. In the connection of local loops, channels, or rings, the equipment and techniques used to match circuits and to facilitate accurate data transmission.
Buffer: A temporary storage register or Random Access Memory (RAM) used in all aspects of data communications which prevents data from being lost due to differences in transmission speed. Keyboards, serial ports, muxes and printers are a few examples of the devices that contain buffers.
Bus: A common channel between hardware devices either internally between components in a computer, or externally between stations in a communications network.
Byte: The unit of information a computer can handle at one time. The most common understanding is that a byte consists of 8 binary digits
(bits), because that's what computers can handle. A byte holds the equivalent of a single character (such as the letter A).
C
Call Setup Time: The time to establish a circuit-switched call between two points. Includes dialing, wait time, and CO/long distance service movement time.
Carrier Group Alarm (CGA): A T1 service alarm generated by a channel bank when an OOF condition occurs for a predefined length of time (usually 300mS to 2.5 seconds). The CGA causes the calls using a trunk to be dropped and for trunk conditioning to be applied.
Carrier signal: An analog signal with known frequency, amplitude and phase characteristics used as a transport facility for useful 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.
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.
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MultiVOIP 400/800 User Guide
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 64 Kbps digital circuit usually has 8 Kbps used for signaling. ISDN has two 64 Kbps circuits, and a 16 Kbps 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 super-frame. A
D4 framing bit is used to identify the channel and the signaling frame. Signalling for voice channels is carried in-band for every channel, along with the encoded voice. See "robbed-bit signaling".
Data Communications Equipment (DCE): Any device which serves as the portal of entry from the user equipment to a telecommunications facility. A modem is a DCE for the phone network (PSTN) that is commonly on site at the user’s premises. Packet Switched Networks have another level of DCE which is most often located at a central office.
Data Link Connection Identifier (DLCI): One of the six components of a frame relay frame. Its purpose is to distinguish separate virtual circuits across each access connection. Data coming into a frame relay node is thus allowed to be sent across the interface to the specified
"address". The DLCI is confirmed and relayed to its destination, or if the specification is in error, the frame is discarded.
Dataphone Digital Service (DDS): A private line digital service that offers 2400, 4800, 9600 and 56 Kbps data rates on an inter-LATA basis by AT&T and on an intra-LATA basis by the BOCs.
Data Service Unit (DSU): A device that provides a digital data service interface directly to the data terminal equipment. The DSU provides loop equalization, remote and local testing capabilities, and a standard EIA/CCITT interface.
Dedicated Line: A communication line that is not switched. The term leased line is more common.
Default: This is a preset value or option in software packages, or in hardware configuration, that is used unless you specify otherwise.
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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 (64 Kbps) for digital voice conversation using PCM (pulse coded modulation).
Digital Service, level 1 (DS1): The 1.544 Mbps voice standard (derived from an older Bell System standard) for digitized voice transmission in North America. The 1.544 Mbps consists of 24 digitally-encoded 64 Kbps voice channels (north America) and 2.048 Mbps (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 64 Kbps (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
E&M: A telephony trunking system used for either switch-to-switch, or switch-to-network, or computer/telephone system-to-switch connection.
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.
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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 56 Kbps (DS0 rate) and 1.544 Mbps (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 64 Kbps multiples, usually up to 768 Kbps.
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.
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 1024 Kbps) 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.
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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).
I
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.
Hexadecimal: A base 16 numbering system used to represent binary values. Hex uses the numbers 0-9 and the letters A-F: usually notated by an "h" (e.g., "4CF h", read "four charley fox, hex"). The result is that one hex digit represents a 4-bit value.
Implicit congestion management: A method of informing the terminal that the network is busy. This method relies on the end-system protocol to detect and fix the congestion problem. (TCP/IP is an example of a protocol using only implicit congestion management.) See also
"explicit congestion management".
In-band: Refers to the type of signalling over the conversion path on an ISDN call. Contrast "out-of-band".
Insufficient Ones: A T1 error condition that is logged when less than one 1 in 16 0s or less than 12.5 % average 1s density is received.
Inter Exchange Carrier (IEC): The long distance company (LE) who's central office provides the point of reference for T1 access. Any common carrier authorized by the FCC to carry customer transmissions between LATAs.
Internet: Refers to the computer network of many millions of university, government and private users around the world. Each user has a unique Internet Address.
Internet Address (IP Address): A unique 32-bit address for a specific TCP/IP host on a network. Normally printed in dotted decimal format
(e.g., 129.128.44.227).
Internet Protocol (IP): A protocol used to route data from its source to its destination in an Internet environment. The Internet Protocol was designed to connect to local area networks. Although there are many protocols that do this, IP refers to the global system of interconnecting computers. It is a highly distributed protocol (each machine only worries about sending data to the next step in the route).
Internetwork Packet Exchange (IPX): A NetWare communications protocol used to route messages from one node to another. IPX packets include network addresses and can be routed from one network to another. An IPX packet can occasionally get lost when crossing networks, thus IPX does not guarantee delivery of a complete message. Either the application has to provide that control, or NetWare's SPX protocol must be used.
Interoperable: Devices from different vendors that can exchange information using a standard's base protocol.
I/O Addresses: Locations within the I/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.
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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 64 Kbps and higher speeds.
Contrast Public Switched
Telephone Network (PSTN).
ITU-TSS (formerly CCITT): International Telecommunications Union-Telecommunications Sector; the United Nations organization that prepares standards ("Recommendations") for resolving communications issues and problems.
K
Key Telephone System (KTS): Phone devices with multiple buttons that let you select incoming or outgoing CO phone lines directly. Similar in operation to a PBX, except a KTS you don't have to dial a "9" for a call outside the building.
Key Service Unit (KSU): A small device containing the switching electronics for a business key telephone system (KTS).
Key Set: A phone set with several buttons for call holding, line pickup, intercom, autodialing, etc. Also called a touchtone phone (Ericsson) and a KTS (Key Telephone Set).
L
LAPB: Link Access Procedure Balanced; based on the X.25 Layer 2 specification. A full-duplex point-to-point bit-synchronous protocol commonly used as a data link control protocol to interface X.25 DTEs. LAPB is the link initialization procedure that establishes and maintains communications between the DTE and the DCE.
LAPD: Link Access Protocol for the D-Channel; based on the ISDN Q.921 specification. A full-duplex point-to-point bit-synchronous linklevel protocol for ISDN connections; different from LAPB in its framing sequence. Transmission is in units called "frames", and a frame may contain one or more X.25 packets.
Line Coding: The representation of 1s and 0s on a T1 line. The two methods of line coding commonly used, B8ZS and AMI, differ in the restrictions placed on user data. T1 line coding ensures that sufficient timing information is sent with the digital signal to ensure recovery of all the bits at the far end. Timing information on the T1 line is included in the form of 1s in the data stream; a long string of 0s in the data stream could cause problems recovering the data.
Line Termination (LT): The electronics at the ISDN network side of the user/network interface that complements the NT1 at the user side.
The LT and the NT1 together provide the high-speed digital line signals required for BRI access.
Listed Directory Number (LDN): The main number assigned by the telco; the number listed in the phone directory and also provided by
Directory Assistance. Some devices can have more than one LDN, such as ISDN devices that have one LDN for voice and another LDN for data.
Local Area Network (LAN): 1. A computer network located on a user's premises within a limited geographical area. Communication within a local area network is not subject to external regulations; however, communication across the LAN boundary may be subject to some form of regulation. 2. A LAN does not use store and forward techniques. 3. A network in which a set of devices are connected to one another for a communication and that can be connected to a larger network.
Local Access and Transport Area (LATA): A post-divestiture geographical area generally equivalent to a Standard Metropolitan Statistical
Area. At divestiture, the territory served by the Bell system was divided into approximately 161 LATAs. The Bell Operating Companies (BOCs) provide Intra-LATA services.
Local Exchange Carrier (LEC): The local phone company which provides local (i.e., not long distance) transmission services. AKA "telco".
LECs provide T1 or FT1 access to LDCs (unless the T1 circuit is completely intra-LATA). Inter-LATA T1 circuits are made up of a combination of Access and Long Haul facilities.
Local Management Interface (LMI): A specification for frame relay equipment that defines status information exchange.
Local Loop: A transmission path, typically twisted-pair wire, between an individual subscriber and the nearest public telecommunications network switching center. The wires provide ISDN service, but require an NT1 at the user end and an LT at the network end. (AKA, "loop" or
"subscriber loop".)
Logical Link Control (LLC2): In a local area network, the protocol that governs the exchange of transmission frames between data stations independently of how the transmission medium is shared. The LLC2 protocol was developed by the IEEE 802 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).
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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.
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.
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Outage: The measure of the time during which a circuit is not available for use due to service interrupt. Outage is the complement of circuit
"availability" (100% minus % available = % outage).
Out-of-band: Signaling that is separated from the channel carrying the information (i.e., the voice/data/video signal is separate from the carrier signal). Dialing and various other "supervisory" signals are included in the signaling element. Contrast "In-band" signaling.
Out of Frame (OOF): A T1 alarm condition that is logged on the loss of 2, 3 or 4 of 5 consecutive FT framing bits.
P
Packet: 1. In data communication, a sequence of binary digits, including data and control signals, that is transmitted and switched as a composite whole. The data, control signals and, possibly, error control information are arranged in a specific format. 2. Synonymous with data frame. 3. In TCP/IP, the unit of data passed across the interface between the Internet layer and the link layer. A packet includes an IP header and data. A packet can be a complete IP datagram or a fragment of an IP diagram. 4. In X.25, a data transmission information unit. A group of data and control characters, transferred as a unit, determined by the process of transmission. Commonly used data field lengths in packets are 128 or 256 bytes. 5. The field structure and format defined in the CCITT X.25 recommendation.
Packet Assembler/Dissembler (PAD): Used by devices to communicate over X.25 networks by building or stripping X.25 information on or from a packet.
Packet Data: The information format ("packetized") used for packet-mode calls.
Packet Mode: Refers to the switching of chunks of information for different users using statistical multiplexing to send them over the same transmission facility.
Parity bit: An extra bit attached to each byte of synchronous data used to detect errors in transmission.
Permanent Virtual Circuit (PVC): A connection between two endpoints dedicated to a single user. In ISDN, PVCs are established by network administration and are held for as long as the user subscribes to the service.
Physical Unit (PU): The component that manages and monitors the resources (such as attached links and adjacent link stations) associated with a node, as requested by an SSCP via an SSCP-PU session. An SSCP activates a session with the physical unit in order to indirectly manage, through the PU, resources of the node such as attached links. This term applies to type 2.0, type 4, and type 5 nodes only.
Point-of-Presence (POP): The central office's end points of the long distance carriers.
Point-to-Point Protocol (PPP): A protocol that lets a PC user access TCP/IP (Internet member) using an ISDN terminal adapter or a highspeed modem over a standard phone line.
Port: A location for input or output data exchange. Computers, muxes, etc. have ports for various purposes.
Primary Rate Interface (PRI): Used on ISDN. In North America, and Japan, PRI is one 64Kbps D channel and 23 B channels. Elsewhere, it is one D channel and 30 B channels.
Primitive: An abstract representation of interaction across the access points indicating that information is being passed between the service user and the service provider. The OSI Reference Model defines four types of primitives: Request, Indication, Response and Confirm.
Private Branch Exchange (PBX): A phone exchange located on the customer's premises. The PBX provides a circuit switching facility for phone extension lines within the building, and access to the public phone network. See also "Exchange".
PROM (Programmable Read Only Memory - pronounced "prom"): A permanent memory chip that can be programmed or filled by the customer after by the manufacturer has set initial values. Contrast with ROM.
Protocol: 1. A set of semantic and syntactic rules that determines the behavior of functional units in achieving communication. 2. In Open
Systems Interconnection architecture, a set of semantic and syntactic rules that determine the behavior of entities in the same layer in performing communication functions. 3. In SNA, the meanings of and the sequencing rules for requests and responses used for managing the network, transferring data, and synchronizing the states of network components. 4. Synonymous with line control discipline.
PSTN (Public Switched Telephone Network): A worldwide public voice phone network that is used as a telecommunications medium for the transmission of voice, data and other information.
Public Data Network (PDN): A packet-switched network that is available to the public for individual ("subscriber") use. Typically, controlled by a government or a national monopoly.
Public Switched Telephone Network (PSTN): The group of circuit-switching voice carriers, which are commonly used as analog data communications services.
Pulse Code Modulation (PCM): 1. In data communication, variation of a digital signal to represent information; for example, by means of pulse amplitude modulation (PAM), pulse duration modulation (PDM), or pulse position modulation (PPM). 2. Transmissions of analog information in digital form through sampling and encoding the samples with a fixed number of bits.
Pulse dialing: One of two methods of dialing a phone, usually associated with rotary-dial phones. Compare with "tone dialing".
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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 generally-accepted ideas, proposals, research results, etc. are published.
Ring Down Box: A device that emulates a CO by generating POTS calls for testing and product demos.
Ring Down Circuit: A tie line connecting phones where picking up one phone automatically rings another phone. A feature used for emergencies to alert the person at the other phone of the incoming call.
RJ-11: An industry standard interface used for connecting a phone to a modular wall outlet; comes in 4-and 6-wire packages.
RJ-45: An 8-wire modular connector for voice and data circuits.
Robbed Bit Signaling: The popular T1 signaling mechanism where the A and B bits are sent by each side of the T1 termination and are
"buried" in the voice data of each voice channel in the T1 circuit. Since the bits are "robbed" infrequently, voice quality is remains relatively uncompromised. See "bit robbing".
The robbed-bit signaling technique is used in D4 channel banks to convey signaling information. The eighth (least significant) bit of each of the 24 8-bit time slots is "robbed" every sixth frame to convey voice-related signaling information such as on-hook, off-hook, etc., for each channel.
Router: A device that connects two networks using the same networking protocol. It operates at the Network Layer (Layer 3) of the OSI model for forwarding decisions.
Routing Information Protocol (RIP): A distance vector-based protocol that provides a measure of distance, or hops, from a transmitting workstation to a receiving workstation.
RS-232-C: An EIA standard for a serial interface between computers and peripheral devices (modem, mouse, etc.). It uses a 25-pin DB-25, or a 9-pin DB-9 connector. The RS-232 standard defines the purposes, electrical characteristics and timing of the signals for each of the 25 lines.
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 .
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MultiVOIP 400/800 User Guide
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 56 Kbps. 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.
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 Mbps 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 1.544 Mbps. T1 uses two pairs of normal UTP, and can handle 24 voice conversations, each digitized at 64 Kbps. T1 is a standard for digital transmission in the U.S., Canada, Japan and Hong Kong. T1 is the access method for highspeed services such as ATM, frame relay, and SMDS. See also T Carrier, T1 line and FT1.
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 64 Kbps 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 56/64 Kbps 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.
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Glossary
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 64 Kbps maximum bandwidth. "Time slot" implies the time division multiplexing organization of the T1 signal.
Toll Call: A call to a location outside of your local service area (i.e., a long distance call).
Tone dialing: One of two methods of dialing a phone, usually associated with Touch-Tone® (push button) phones. Compare with pulse dialing.
Topology: Physical layout of network components (cables, stations, gateways, and hubs). Three basic interconnection topologies are star, ring, and bus networks.
Transmission Control Protocol (TCP): A communications protocol used in Internet and in any network that follows the US Department of
Defense standards for internetwork protocol. TCP provides a reliable host-to-host protocol between hosts in packet-switched communications networks and in interconnected systems of such networks. It assumes that the Internet protocol is the underlying protocol.
Transport Layer: Layer 4 of the Open Systems Interconnection (OSI) model; provides reliable, end-to-end delivery of data, and detects transmission sequential errors.
Transport Protocol Data Unit (TPDU): A transport header, which is added to every message, contains destination and source addressing information that allows the end-to-end routing of messages in multi-layer NAC networks of high complexity. They are automatically added to messages as they enter the network and can be stripped off before being passed to the host or another device that does not support TPDU's.
Trunk: Transmission links that interconnect switching offices.
TSR (terminate and stay resident): A software program that remains active and in memory after its user interface is closed. Similar to a daemon in UNIX environments.
Tunneling: Encapsulation data in an IP packet for transport across the Internet.
Twisted pair wiring: A type of cabling with one or more pairs of insulated wires wrapped around each other. An inexpensive wiring method used for LAN and telephone applications, also called UTP wiring.
U
UART (Universal Asynchronous Receiver/Transmitter) (pronounced "you art"): A chip that transmits and receives data on the serial port. It converts bytes into serial bits for transmission, and vice versa, and generates and strips the start and stop bits appended to each character.
UNIX: An operating system developed by Bell Laboratories that features multiprogramming in a multi-user environment.
Unshielded Twisted Pair (UTP): Telephone-type wiring. Transmission media for 10Base-T.
V
V.25bis: An ITU-T standard for synchronous communications between a mainframe or host and a modem using HDLC or other characteroriented 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 A.323 endpoints and MCUs managed by a single Gatekeeper. A zone must include may include LAN segments connected by using routers.
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MultiVOIP 400/800 User Guide
108
Index
Index
A
About the Internet ........................................... 84
Accessories, ordering ..................................... 84
Adding phone entries
Gatekeeper ................................................. 32
Proprietary .................................................. 34
Application, typical ........................................... 8
Applications setup, changing .......................... 64
Archie ............................................................. 87
Auto Call ......................................................... 57
Automatic disconnection, billing/security ........ 57
B
Back Panel ..................................................... 14
Billing options ................................................. 57
Billing/Misc default setup, Master ............. 31, 45
Billing/Misc parameters, changing .................. 57
Boot LED ........................................................ 13
Building a proprietary phonebook ................... 34
C
Cabling the MultiVOIP
Local site .................................................... 20
Remote site ................................................ 49
Call progress, viewing .............................. 63, 67
Canadian limitations notice ............................ 93
Channel interface default setup, Master ... 27, 42
Channel interface default setup, Slave ........... 42
Channel interface parameters, changing ....... 54
Channel totals, viewing .................................. 68
Coder, voice/fax ............................................. 56
Collision .......................................................... 13
Command port cable, diagram ....................... 89
Command port connector ............................... 14
Configuration port setup ................................. 52
Configuration utilities, overview ...................... 52
Configuring the MultiVOIP .............................. 27
Configuring the MultiVOIP, overview .............. 18
Configuring your Slave MultiVOIPs ................ 41
Connectors
Command ................................................... 14
Power ......................................................... 14
Current loss .................................................... 55
Custom tone pair settings ............................... 58
D
Deploying the VOIP network
Remote site administrator ........................... 49
VOIP administrator ..................................... 48
Deploying the VOIP network, overview .......... 18
Differentiated services .................................... 62
DNS ................................................................ 87
Download firmware ......................................... 52
Downloading coders ....................................... 70
Downloading firmware .................................... 69
Downloading H.323 stack ............................... 71
DTMF gain, voice/fax ..................................... 56
Dynamic jitter buffer ....................................... 57
E
E&M connector ............................................... 14
E&M interface ................................................. 55
E&M jumper block positioning ........................ 21
EMC, safety and terminal directive compliance93
Endpoint, H.323 .............................................. 61
Ethernet connector ......................................... 14
Ethernet LEDs ................................................ 13
F
Fax branding statement .................................. 91
FCC Part 15 statement .................................. 91
FCC Part 68 Telecom statement .................... 92
Finger ............................................................. 87
Firewall, installing with MultiVOIPs ................. 24
Flash Hook Timer ........................................... 54
Front Panel, description ................................. 13
FTP (File Transfer Protocol) ........................... 87
FXO connector ............................................... 14
FXO disconnect on ......................................... 55
FXO interface ................................................. 54
FXS connector ............................................... 14
FXS interface ................................................. 54
G
Gatekeeper
Overview ....................................................... 6
Registering with .......................................... 32
Gopher ........................................................... 87
H
I
H.323 endpoint ............................................... 61
H.323 protocol
Overview ....................................................... 6
HTML ............................................................. 87
HTTP .............................................................. 87
Industry Canada statement ............................ 91
Installing MultiVOIP software ......................... 24
Installing software, overview .......................... 18
Inter digit time ................................................. 54
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MultiVOIP 400/800 User Guide
Internet
Multi-Tech's Web site .................................. 84
Internet protocol (IP) ...................................... 88
IP default setup, Master ................................. 27
IP default setup, Slave ................................... 41
IP parameters, changing ................................ 62
IP statistics ..................................................... 65
J
Jitter buffer, dynamic ...................................... 57
Jitter value ...................................................... 57
L
LAN cable, diagram ........................................ 89
LAN-based remote configuration .................... 76
LEDs, front panel ........................................... 13
Limited warranty ............................................. 82
Log entries, reports ........................................ 68
Log entry details, viewing ............................... 67
M
Max Baud rate, voice/fax ................................ 56
Modem-based remote configuration ............... 74
Multi-Tech Installation CD screen ................... 24
MultiVOIP 400/800
Accessories, ordering ................................. 84
Back panel, description .............................. 14
Cabling diagrams ........................................ 89
Cabling procedure ...................................... 20
Configuring ................................................. 27
Configuring the Slave MultiVOIP ................ 41
Deploying the VOIP network ....................... 48
E&M positioning procedure ......................... 21
Firewall, installing with MVP400/800 .......... 24
Front panel, description .............................. 13
Installing the MultiVOIP software ................ 24
Introduction ................................................... 6
Service ........................................................ 83
Technical specifications .............................. 15
Technical support ........................................ 84
Typical application ........................................ 8
Unpacking ................................................... 19
Upgrade procedure ..................................... 69
Valid VOIP network connections ................. 19
MultiVOIP Main menu, introduction ................ 52
MultiVOIP Main menu, screen ........................ 53
MultiVOIP, Telnet server menu ....................... 78
O
Optimization factor ......................................... 58
Ordering accessories ..................................... 84
P
Packets lost/received ..................................... 63
Password, SNMP ........................................... 64
Phone directory
Building a proprietary phonebook ............... 34
Registering with a Gatekeeper ................... 32
Phone directory database, changing .............. 59
Ping ................................................................ 87
POP ................................................................ 87
Power
Connector ................................................... 14
Power LED ..................................................... 13
Proprietary phonebook directory .................... 34
Pulse generation ratio .................................... 58
R
Receive Data .................................................. 13
Regeneration .................................................. 54
Regional default setup, Master ................. 31, 46
Regional parameters, changing ..................... 58
Registering with a Gatekeeper ....................... 32
Regulatory information ................................... 91
Remote configuration
LAN-based procedure ................................ 76
Modem-based procedure ............................ 74
Remote management ..................................... 78
Telnet .......................................................... 78
WEB management ...................................... 80
Remote site administrator, deployment .......... 49
Remote site cable connection ........................ 49
Reports, log entries ........................................ 68
Ring count, FXO and FXS .............................. 55
Ringer equivalence number ........................... 93
S
Safety Warning
Battery and port interfaces ......................... 19
Telecom ...................................................... 18
Service ........................................................... 83
Silence detection ............................................ 55
Slave MultiVOIP, configuring .......................... 41
Slave MultiVOIPs, configuring ........................ 41
Slave status .................................................... 60
SMTP ............................................................. 87
SNMP ............................................................. 87
SNMP agent, enabling ................................... 64
SNMP statistics .............................................. 66
Software ......................................................... 52
Description .................................................. 52
Software installation procedure ...................... 24
Specifications
Command Port ........................................... 15
Electrical/Physical ....................................... 15
Ethernet Port .............................................. 15
Station identification ................................. 60, 61
110
Statistics, viewing
IP ................................................................ 65
SNMP ......................................................... 66
T
TCP packets ................................................... 65
TCP/IP ...................................................... 78, 86
Technical support ........................................... 84
Telnet ........................................................ 78, 87
Proxy Server Configuration ........................ 80
Telnet server menu ......................................... 78
Tone detection ................................................ 55
Tone pairs, customizing .................................. 58
Traceroute ...................................................... 87
Transmit Data ................................................. 13
Typical application ............................................ 8
U
UDP packets .................................................. 65
Uninstall ProxyServer configuration ............... 52
Unpacking your MultiVOIP ............................. 19
Upgrading software version ............................ 69
V
Valid VOIP network connections .................... 19
Viewing statistics ............................................ 65
Voice coder, voice/fax .................................... 56
Voice over IP configuration ............................. 79
Voice/fax channel connectors ........................ 14
Voice/Fax channel connectors, diagram ........ 89
Voice/Fax channel LEDs ................................ 13
Voice/Fax default setup, Master ............... 29, 43
Voice/Fax parameters, changing .................... 56
VOIP administrator, deployment ..................... 48
W
Warranty ......................................................... 82
WEB Server ................................................... 80
Index
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