MVP 130-210-410-810AV User Guide

Voice / Fax over IP Networks

User Guide for Voice/IP Gateways

Digital Models (T1, E1, ISDN-PRI):MVP-2410/3010

Analog/BRI Models: MVP-130/210/410/810 MVP-410ST/810ST

2

User GuideS000249IAnalog MultiVOIP Units (Models MVP130, MVP210, MVP410, MVP810)ISDN-BRI MultiVOIP Units (Models MVP410ST, and MVP810ST)Digital MultiVOIP Units (Models MVP2410, & MVP3010)Upgrade Units (MVP24-48 and MVP30-60)This publication may not be reproduced, in whole or in part, without prior expressedwritten permission from Multi-Tech Systems, Inc. All rights reserved.Copyright © 2004, by Multi-Tech Systems, Inc.

Multi-Tech Systems, Inc. makes no representations or warranties with respect to thecontents hereof and specifically disclaims any implied warranties of merchantability orfitness for any particular purpose. Furthermore, Multi-Tech Systems, Inc. reserves theright to revise this publication and to make changes from time to time in the contenthereof without obligation of Multi-Tech Systems, Inc. to notify any person ororganization of such revisions or changes.

Record of RevisionsRevision Description A Initial Release. (05/10/02)

B Index added. (05/24/02) C Updated for 4.03/6.03 software. (10/11/02) D Updated for 4.04/6.04/8.04/9.04 software. (03/20/03) Add

embedded gatekeeper models, ISDN-BRI models,MultiVantage Apx., SPP protocol, & Call State Apx.

E Remove MultiVantage. (04/18/03) F Update ISDN-BRI info in SW version 5.02c. (06/04/03) G Add MVP130 information. (06/30/03)

H Revisions to ISDN-BRI & MVP130 content. (08/15/03) I Updated for x.06 software release. (04/29/04)

PatentsThis Product is covered by one or more of the following U.S. Patent Numbers:6151333, 5757801, 5682386, 5.301.274; 5.309.562; 5.355.365; 5.355.653; 5.452.289;5.453.986. Other Patents Pending.

TrademarkTrademark of Multi-Tech Systems, Inc. is the Multi-Tech logo. Windows andNetMeeting are registered trademarks of Microsoft.

Multi-Tech Systems, Inc.2205 Woodale DriveMounds View, Minnesota 55112(763) 785-3500 or (800) 328-9717; U.S. Fax: 763-785-9874Technical Support: (800) 972-2439http://www.multitech.com

http://www.multitech.com/

3

CONTENTSCHAPTER 1: OVERVIEW.......................................................................................7

ABOUT THIS MANUAL...............................................................................................8INTRODUCTION TO TI MULTIVOIPS (MVP2410 & MVP24-48).............................11

T1 Front Panel LEDs..........................................................................................16INTRODUCTION TO EI MULTIVOIPS (MVP3010 & MVP30-60)............................18

E1 Front Panel LEDs .........................................................................................23E1 LED Descriptions ..........................................................................................24

INTRODUCTION TO ANALOG MULTIVOIPS (MVP130, MVP-210/410/810 &MVP428) ................................................................................................................25

Analog MultiVOIP Front Panel LEDs................................................................30INTRODUCTION TO ISDN-BRI MULTIVOIPS (MVP410ST & MVP810ST) ..........34

ISDN BRI MultiVOIP Front Panel LEDs ...........................................................38ISDN-BRI MultiVOIP LED Descriptions ...........................................................39

COMPUTER REQUIREMENTS ....................................................................................40SPECIFICATIONS ......................................................................................................41

Specs for Digital T1 MultiVOIP Units................................................................41Specs for Digital E1 MultiVOIP Units................................................................42Specs for Analog/BRI MultiVOIP Units..............................................................43

INSTALLATION AT A GLANCE ..................................................................................44RELATED DOCUMENTATION....................................................................................44

CHAPTER 2: QUICK START INSTRUCTIONS ................................................45

INTRODUCTION........................................................................................................46MULTIVOIP STARTUP TASKS .................................................................................46

Phone/IP Details *Absolutely Needed* Before Starting the Installation............47Gather IP Information...................................................................................................47Gather Telephone Information (T1) .............................................................................47Gather Telephone Information (E1) .............................................................................48Gather Telephone Information (Analog) ......................................................................48Gather Telephone Information (ISDN BRI) .................................................................49Obtain Email Address for VOIP (for email call log reporting).....................................50Identify Remote VOIP Site to Call ...............................................................................50Identify VOIP Protocol to be Used...............................................................................50

Placement ...........................................................................................................51The Command/Control Computer (Specs & Settings) ........................................51Quick Hookups....................................................................................................52Load MultiVOIP Control Software onto PC.......................................................57Phone/IP Starter Configuration..........................................................................58Phonebook Starter Configuration (with remote voip).........................................62

Outbound Phonebook ...................................................................................................62Inbound Phonebook......................................................................................................66

Phonebook Tips ..................................................................................................69Phonebook Example ...........................................................................................72Connectivity Test ................................................................................................77Troubleshooting..................................................................................................81

Contents MultiVOIP User Guide

4

CHAPTER 3: MECHANICAL INSTALLATION AND CABLING.................110

INTRODUCTION......................................................................................................111SAFETY WARNINGS ...............................................................................................111

Lithium Battery Caution ...................................................................................111Safety Warnings Telecom..................................................................................111

UNPACKING YOUR MULTIVOIP............................................................................112Unpacking the MVP2410/3010.........................................................................112Unpacking the MVP-410x/810x........................................................................113Unpacking the MVP210....................................................................................114Unpacking the MVP130....................................................................................115

RACK MOUNTING INSTRUCTIONS FOR MVP-2410/3010 & MVP-410X/810X......116Safety Recommendations for Rack Installations ...............................................11719-Inch Rack Enclosure Mounting Procedure..................................................118

CABLING ...............................................................................................................119Cabling Procedure for MVP2410/3010............................................................119Cabling Procedure for MVP-410/810...............................................................121Cabling Procedure for MVP-410ST/810ST ......................................................126Cabling Procedure for MVP210.......................................................................130Cabling Procedure for MVP130.......................................................................134

CHAPTER 4: SOFTWARE INSTALLATION ...................................................110

INTRODUCTION......................................................................................................111LOADING MULTIVOIP SOFTWARE ONTO THE PC..................................................111UN-INSTALLING THE MULTIVOIP CONFIGURATION SOFTWARE...........................118

CHAPTER 5: TECHNICAL CONFIGURATION FOR DIGITAL T1/E1MULTIVOIPS (MVP2410, MVP3010) .................................................................121

CONFIGURING THE DIGITAL T1/E1 MULTIVOIP...................................................122LOCAL CONFIGURATION........................................................................................124

Pre-Requisites...................................................................................................124IP Parameters..............................................................................................................124T1 Telephony Parameters (for MVP2410) .................................................................125E1 Telephony Parameters (for MVP3010) .................................................................126SMTP Parameters (for email call log reporting).........................................................127

Local Configuration Procedure (Summary) .....................................................128Local Configuration Procedure (Detailed).......................................................129Modem Relay ....................................................................................................150

CHAPTER 6: TECHNICAL CONFIGURATION FOR ANALOG/BRIMULTIVOIPS (MVP130, MVP-210, MVP-410, MVP-810 & MVP-410ST/810ST) ..........................................................................................................205

CONFIGURING THE ANALOG/BRI MULTIVOIP .....................................................304LOCAL CONFIGURATION........................................................................................307

Pre-Requisites...................................................................................................307IP Parameters..............................................................................................................307Analog Telephony Interface Parameters (for MVP130/210/410/810) .......................308ISDN-BRI Telephony Parameters (for MVP-410ST/810ST) .....................................309

MultiVOIP User Guide Contents MultiMVP3

5

SMTP Parameters (for email call log reporting).........................................................310Local Configuration Procedure (Summary) .....................................................311Local Configuration Procedure (Detailed).......................................................312Modem Relay ....................................................................................................335

CHAPTER 7: T1 PHONEBOOK CONFIGURATION ......................................304

CONFIGURING THE MVP2410 MULTIVOIP PHONEBOOKS...................................305T1 PHONEBOOK EXAMPLES...................................................................................330

3 Sites, All-T1 Example.....................................................................................330Configuring Mixed Digital/Analog VOIP Systems ...........................................336Call Completion Summaries .............................................................................345Variations in PBX Characteristics....................................................................348

CHAPTER 8: E1 PHONEBOOK CONFIGURATION ......................................349

MVP3010 INBOUND AND OUTBOUND MULTIVOIP PHONEBOOKS .......................350Free Calls: One VOIP Site to Another.............................................................351Local Rate Calls: Within Local Calling Area of Remote VOIP.......................352National Rate Calls: Within Nation of Remote VOIP Site ...............................354Inbound versus Outbound Phonebooks.............................................................355

PHONEBOOK CONFIGURATION PROCEDURE...........................................................359E1 PHONEBOOK EXAMPLES...................................................................................379

3 Sites, All-E1 Example ....................................................................................379Configuring Digital & Analog VOIPs in Same System.....................................386

Call Completion Summaries.......................................................................................395Variations in PBX Characteristics....................................................................398International Telephony Numbering Plan Resources .......................................399

CHAPTER 9: ANALOG/BRI PHONEBOOK CONFIGURATION .................403

CHAPTER 10: OPERATION AND MAINTENANCE ......................................403

OPERATION AND MAINTENANCE ...........................................................................404System Information screen................................................................................404Statistics Screens ..............................................................................................406About Call Progress..........................................................................................406About Logs ........................................................................................................412About IP Statistics.............................................................................................416About Link Management...................................................................................420About Packetization Time .................................................................................423About T1/E1 and BRI Statistics.........................................................................426About Registered Gateway Details ...................................................................438

MULTIVOIP PROGRAM MENU ITEMS .....................................................................440Date and Time Setup.........................................................................................442Obtaining Updated Firmware...........................................................................442Implementing a Software Upgrade ...................................................................446

Identifying Current Firmware Version .......................................................................446Downloading Firmware..............................................................................................447Downloading CAS Protocols......................................................................................450Downloading Factory Defaults...................................................................................452

Setting and Downloading User Defaults ..........................................................454

Contents MultiVOIP User Guide

6

Downloading IFM Firmware............................................................................456Setting a Password (Windows GUI) .................................................................458Setting a Password (Web Browser GUI) ..........................................................461Un-Installing the MultiVOIP Software .............................................................462Upgrading Software..........................................................................................464

FTP SERVER FILE TRANSFERS (“DOWNLOADS”) ..................................................465WEB BROWSER INTERFACE ...................................................................................475SYSLOG SERVER FUNCTIONS ................................................................................480

CHAPTER 11 WARRANTY, SERVICE, AND TECH SUPPORT...................483

LIMITED WARRANTY.............................................................................................484REPAIR PROCEDURES FOR U.S. AND CANADIAN CUSTOMERS ...............................484TECHNICAL SUPPORT ............................................................................................486

Contacting Technical Support ..........................................................................486

CHAPTER 12: REGULATORY INFORMATION ............................................487

EMC, Safety, and R&TTE Directive Compliance.............................................488FCC DECLARATION ..............................................................................................488

Industry Canada ...............................................................................................489FCC Part 68 Telecom.......................................................................................489Canadian Limitations Notice ............................................................................490

APPENDIX A: EXPANSION CARD INSTALLATION (MVP24-48 & MVP30-60).............................................................................................................................491

INSTALLATION.......................................................................................................492OPERATION............................................................................................................494

APPENDIX B: CABLE PINOUTS ......................................................................495

APPENDIX B: CABLE PINOUTS..............................................................................496Command Cable ...............................................................................................496Ethernet Connector...........................................................................................496T1/E1 Connector...............................................................................................497Voice/Fax Channel Connectors ........................................................................497ISDN BRI RJ-45 Pinout Information ................................................................499ISDN Interfaces: “ST” and “U” .....................................................................500

APPENDIX C: TCP/UDP PORT ASSIGNMENTS ...........................................501

WELL KNOWN PORT NUMBERS .............................................................................502PORT NUMBER ASSIGNMENT LIST.........................................................................502

APPENDIX D: INSTALLATION INSTRUCTIONS FOR MVP428 UPGRADECARD.......................................................................................................................503

INSTALLATION INSTRUCTIONS FOR MVP428 UPGRADE CARD..............................504

INDEX .....................................................................................................................509

7

Chapter 1: Overview

Overview MultiVOIP User Guide

8

About This ManualThis manual is about Voice-over-IP products made by Multi-Tech Systems,Inc. It describes four product groups.

1. T1 Digital MultiVOIP units, models MVP2410, and the capacity-doubling add-on expansion card, model MVP24-48 (which fits theMVP2410 only).

2. E1 Digital MultiVOIP units, models, MVP3010 and the capacity-doubling add-on expansion card, model MVP30-60.

3. Analog MultiVOIP units,models MVP810, MVP410, MVP210, & MVP130.

4. ISDN-BRI MultiVOIP units, models MVP410ST & MVP810ST.These MultiVOIP units can inter-operate with the earlier generation ofMultiVOIP products (the MVP200, MVP400, MVP800, MVP120, etc.)

In this manual, an “x” suffix to a model number means the assertion applies toall suffix types of that model number. E.g., “MVP410x” refers collectively toMVP410 and MVP410ST.

The table below describes the vital characteristics of the various modelsdescribed in this manual.

MultiVOIP User Guide Overview

9

MultiVOIP Product Family

ModelDescription MVP-

2410MVP24-48

MVP3010

MVP30-60

Function T1digitalVOIPunit

T1digitalVOIPadd-oncard

E1digitalVOIPunit

E1digitalVOIPadd-oncard

Capacity 24channels

24addedchannels

30channels

30addedchannels

Chassis/Mounting

19” 1Urackmount

circuitcardonly

19” 1Urackmount

circuitcardonly

ModelDescription MVP

810MVP428

MVP410

MVP210

MVP130

Function analogvoip

add-oncard

analogvoip

analogvoip

analogvoip

Capacity 8channels

4 addedchannels

4channels

2channels

1

channel

Chassis/Mounting

19” 1Urackmount

circuitcardonly

19” 1Urackmount

Tabletop

tabletop

ModelDescription MVP810ST MVP410ST

Function ISDN-BRI voip ISDN-BRI voipCapacity 4 ISDN lines

(8 B-channels)2 ISDN lines(4 B-channels)

Chassis/Mounting

19” 1U rack mount 19” 1U rack mount

1. “BRI” means Basic Rate Interface.


10

How to Use This Manual. In short, use the index and the examples.When our readers crack open this large manual, they generally need one of twothings: information on a very specific software setting or technical parameter(about telephony or IP) or they need help when setting up phonebooks for theirvoip systems. The index gives quick access to voip settings and parameters.It’s detailed. Use it. The best way to learn about phonebooks is to wadethrough examples like those in our chapters on T1 (North American standard)Phonebooks and E1 (Euro standard) Phonebooks. Also, the quick setup info ofthe printed Quick Start Guide is replicated in this manual for your convenience.Finally, this manual is meant to be comprehensive. If you notice thatsomething important is lacking, please let us know.

Additional Resources. The MultiTech web site (www.multitech.com) offersboth a list of Frequently Asked Questions (the MultiVOIP FAQ) and acollection of resolutions of issues that MultiVOIP users have encountered(these are Troubleshooting Resolutions in the searchable Knowledge Base).

Variable Model/Version Icon and Typography. The MultiVOIP productfamily is a coordinated set of products that can operate with each other in aseamless fashion. For example, both the digital and analog MultiVOIP unitsuse the same graphic user interface (GUI) in the MultiVOIP configurationsoftware and both operate under a single GUI in the MultiVoipManager remotemanagement software. Because this is the case, the various model numbers andversion numbers of MultiVOIP family products will each appear in variousdialog boxes and commands. But instead of showing these dialog boxes oncefor each model in this manual, we substitute the following icon.

Figure 1-1: Variable Model/Version Icon

It indicates that, whatever MultiVOIP model you are using, all details exceptthe very model and version numbers themselves will be the same regardless ofthe MultiVOIP model used. Also, in some cases, we will use othertypographic devices, like blank underlining(“MultiVOIP ____”) to denote information that applies to anyand all of the products in this product family.


11

Introduction to TI MultiVOIPs (MVP2410 &MVP24-48)

We proudly present MultiTech’s T1 Digital Multi-VOIP products. The MVP2410 is a rack-mount model; and the MVP24-48 is an add-onexpansion card that doubles the capacity of the MVP2410 without addinganother chassis. These voice-over-IP products have fax capabilities. Thesemodels adhere to the North American standard of T1 trunk telephony usingdigital 24-channel time-division multiplexing, which allows 24 phoneconversations to occur on the T1 line simultaneously. They can alsoaccommodate T1 lines of the ISDN Primary Rate Interface type (ISDN-PRI).

Figure 1-2. MultiVOIP MVP2410 LEDs

Scale-ability. The MVP2410 is tailored to companies needing more than afew voice-over-IP lines, but not needing carrier-class equipment. Whenexpansion is needed, the MVP2410 can be field-upgraded into a dual T1 unitby installing the MVP24-48 kit, which is essentially a second MultiVOIPmotherboard that fits in an open expansion-card slot in the MVP2410. Theupgraded dual unit then accommodates two T1 lines.

T1 VOIP Traffic. The MVP2410 accepts its outbound traffic from a T1 trunkthat’s connected to either a PBX or to a telco/carrier. The MVP2410transforms the telephony signals into IP packets for transmission on LANs,WANs, or the Internet. Inbound IP data traffic is converted to telephony dataand signaling.

When connected to PBX. When connected to a PBX, the MVP2410 creates anetwork node served by 10/100-Base T connections. Local PBX phoneextensions gain toll-free access to all phone stations directly connected to theVOIP network. Phone extensions at any VOIP location also gain toll-freeaccess to the entire local public-switched telephone network (PSTN) at everyother VOIP location in the system.

When connected to PSTN. When the T1 line(s) connected to the MVP2410are connected directly to the PSTN, the unit becomes a Point-of-Presenceserver dedicated to local calls off-net.


12

H.323, SIP & SPP. Being H.323 compatible, the MVP2410 can place calls totelephone equipment at remote IP network locations that also contain H.323compatible voice-over-IP gateways. It will interface with H.323 software andH.323 gatekeeper units. H.323 specifications also bring to voip telephonymany special features common to conventional telephony. H.323 features ofthis kind that have been implemented into the MultiVOIP include Call Hold,Call Waiting, Call Name Identification, Call Forwarding (from the H.450standard), and Call Transfer (H.450.2 from H.323 Version 2). The fourthversion of the H.323 standard improves system resource usage (esp. logicalport or socket usage) by handling call signaling more compactly and allowinguse of the low-overhead UDP protocol instead of the error-correcting TCPprotocol where possible.

The MultiVOIP is also SIP-compatible. (“SIP” means Session InitiationProtocol.) However, H.450 Supplementary Services features can be used underH.323 only and not under SIP.

SPP (Single-Port Protocol) is a non-standard protocol developed by Multi-Tech. SPP is not compatible with the “Proprietary” protocol used in Multi-Tech’s earlier generation of voip gateways. SPP offers advantages in certainsituations, especially when firewalls are used and when dynamic IP addressassignment is needed. However, when SPP is used, certain features of SIP andH.323 will not be available and SPP will not inter-operate with voip systemsusing H.323 or SIP.

Data Compression & Quality of Service. The MultiVOIP MVP2410 comesequipped with a variety of data compression capabilities, including G.723,G.729, and G.711 and features DiffServ quality-of-service (QoS) capabilities.

VOIP Functions. The MultiVOIP MVP2410 gateway performs four basicfunctions: (a) it converts a dialed number into an IP address, (b) it sends voiceover the data network, (c) it establishes a connection with another VOIPgateway at a remote site, and (d) it receives voice over the data network. Voiceis handled as IP packets with a variety of compression options. Each T1connection to the MultiVOIP provides 24 time-slot channels to connect to thetelco or to serve phone or fax stations connected to a PBX.

Ports. The MVP2410 has one 10/100 Mbps Ethernet LAN interface and oneCommand port for configuration. An MVP2410 upgraded with the MVP24-48kit will have two Ethernet LAN interfaces and two Command ports.

PSTN Failover Feature. The MultiVOIP can be programmed to divert callsto the PSTN temporarily in case the IP network fails.

Gatekeeper. T1 voip systems can have gatekeeper functionality by adding, asan endpoint, a Multi-Tech standalone gatekeeper (special software residing inseparate hardware). Gatekeepers are optional but useful within voip systems.The gatekeeper acts as the ‘clearinghouse’ for all calls within its zone.MultiTech’s stand-alone gatekeeper software performs all of the standardgatekeepers functions (address translation, admission control, and bandwidth


13

control) and also supports many valuable optional functions (call controlsignaling, call authorization, bandwidth management, and call management).

Management. Configuration and system management can be done locallywith the MultiVOIP configuration software. After an IP address has beenassigned locally, other configuration can be done remotely using theMultiVOIP web browser GUI. Remote system management can be done withthe MultiVoipManager SNMP software or via the MultiVOIP web browserGUI. All of these control software packages are included on the Product CD.


14

While the web GUI’s appearance differs slightly, its content and organizationare essentially the same as that of the Windows GUI (except for logging).

The primary advantage of the web GUI is remote access for control andconfiguration. The controller PC and the MultiVOIP unit itself must both beconnected to the same IP network and their IP addresses must be known.

Once you’ve begun using the web browser GUI, you can go back to theMultiVOIP Windows GUI at any time. However, you must log out of the webbrowser GUI before using the MultiVOIP Windows GUI.


15

Logging of System Events. MultiTech has built SysLog Server functionalityinto the software of the MultiVOIP units. SysLog is a de facto standard forlogging events in network communication systems.

The SysLog Server resides in the MultiVOIP unit itself. To implement thisfunctionality, you will need a SysLog client program (sometimes referred to asa “daemon”). SysLog client programs, both paid and freeware, can be obtainedfrom Kiwi Enterprises, among other firms. See www.kiwisyslog.com. SysLogclient programs essentially give you a means of structuring console messagesfor convenience and ease of use.

MultiTech Systems does not endorse any particular SysLog client program.SysLog client programs by any qualified provider should suffice for use withMultiVOIP units. Kiwi’s brief description of their SysLog program indicatesthe typical scope of such programs. “Kiwi Syslog Daemon is a freewareSyslog Daemon for the Windows platform. It receives, logs, displays andforwards Syslog messages from hosts such as routers, switches, Unix hosts andany other syslog enabled device. There are many customizable optionsavailable.”

http://www.kiwisyslog.com/


16

Supplementary Telephony Services. The H.450 standard (an addition toH.323) brings to voip telephony more of the premium features found in PSTNand PBX telephony. MultiVOIP units offer five of these H.450 features: CallTransfer, Call Hold, Call Waiting, Call Name Identification (not the same asCaller ID), and Call Forwarding. (The first four features are found in the“Supplementary Services” window; the fifth, Call Forwarding, appears in theAdd/Edit Inbound phonebook screen.) Note that the first three features areclosely related. All of these H.450 features are supported for H.323 operationonly; they are not supported for SIP or SPP.

T1 Front Panel LEDsThe MVP2410 and MVP24-48 both use a common main circuit board ormotherboard. Consequently the LED indicators are the same for both.

Active LEDs. The MVP2410 front panel has two sets of identical LEDs. Inthe MVP2410 as shipped (that is, without an expansion card), the left-hand setof LEDs is functional whereas the right-hand set is not.

When the MVP2410 has been upgraded with an MVP24-48 kit, the right-handset of LEDs will also become active.

Figure 1-3: MVP2410 LEDs

T1 LED Descriptions. The descriptions below apply to the digital T1MultiVOIP units. The MVP2410 has four sets of LEDs plus a lone LED at itsfar right end. As viewed from the front of the MVP2410, it is the two leftgroups that are active and present feedback about the operation of the unit. Ifan MVP24-48 expansion card is added to the MVP2410, the two LED groupson the right become operational with respect to the second T1 connection.


17

MVP2410 Front Panel LED Definitions

LED NAME DESCRIPTION

Power Indicates presence of power.

Boot After power up, the Boot LED will be on for about 10seconds while the MVP2410 is booting.

FDX Full-Duplex & Collision LED. This LED indicates whetherthe Ethernet connection is half-duplex or full-duplex (FDX)and, in half-duplex mode, indicates occurrence of datacollisions. LED is on constantly for full-duplex mode; LEDis off constantly for half-duplex mode. When operating inhalf-duplex mode, the LED will flash during data collisions.

LNK Link/Activity LED. This LED is lit if Ethernet connectionhas been made. It is off when the link is down (i.e., when noEthernet connection exists). While link is up, this LED willflash off to indicate data activity.

T1 When lit, indicates presence of T1 connection.

E1 E1. Not supported.

PRI PRI. On if T1 line is of ISDN-Primary-Rate type.

ONL Online. This LED is on when frame synchroni-zationhas been established on the T1/E1 link.

IC IC LED is on when Internal Clocking is selected inT1/E1 configuration.

LC Indicates Loss of Carrier.

LS Indicates Loss of Signal.

Test For testing purposes only.


18

Introduction to EI MultiVOIPs(MVP3010 & MVP30-60)

We proudly present MultiTech’s E1 Digital Multi-VOIP products. TheMVP3010 is a rack-mount model and the MVP30-60 is an add-on expansioncard that doubles the capacity of the MVP3010 without adding another chassis.All of these voice-over-IP products have fax capabilities. All adhere to theEuropean standard of E1 trunk telephony using digital 30-channel time-division multiplexing, which allows 30 phone conversations to occur on the E1line simultaneously. All can also accommodate E1 lines of the ISDN PrimaryRate Interface type (ISDN-PRI).

Figure 1-4. MultiVOIP MVP3010 Chassis

Scale-ability. The MVP3010 is tailored to companies needing more than afew voice-over-IP lines, but not needing carrier-class equipment. Whenexpansion is needed, the MVP3010 can be field-upgraded into a dual E1 unitby installing the MVP30-60 kit, which is essentially a second MultiVOIPmotherboard that fits into an open expansion-card slot in the MVP3010. Theupgraded dual unit then accommodates two E1 lines.

E1 VOIP Traffic. The MVP3010 accepts its outbound traffic from an E1trunk that’s connected to either a PBX or to a telco/carrier. The MVP3010transforms the telephony signals into IP packets for transmission on LANs,WANs, or the Internet. Inbound IP data traffic is converted to telephony dataand signaling.

When connected to PBX. When connected to a PBX, the MVP3010 creates anetwork node served by 10/100-Base T connections. Local PBX phoneextensions gain toll-free access to all phone stations directly connected to theVOIP network. Phone extensions at any VOIP location also gain local-rateaccess to the entire local public-switched telephone network (PSTN) at everyother VOIP location in the system.

When connected to PSTN. When the E1 line(s) connected to the MVP3010are connected directly to the PSTN, the unit becomes a Point-of-Presenceserver dedicated to local calls off-net.


19

H. 323, SIP, & SPP. Being H.323 compatible, the MVP3010 can place callsto telephone equipment at remote IP network locations that also contain H.323compatible voice-over-IP gateways. It will interface with H.323 software andH.323 gatekeeper units. H.323 specifications also bring to voip telephonymany special features common to conventional telephony. H.323 features ofthis kind that have been implemented into the MultiVOIP include Call Hold,Call Waiting, Call Identification, Call Forwarding (from the H.450 standard),and Call Transfer (H.450.2 from H.323 Version 2). The fourth version of theH.323 standard improves system resource usage (esp. logical port or socketusage) by handling call signaling more compactly and allowing use of the low-overhead UDP protocol instead of the error-correcting TCP protocol wherepossible.

The MultiVOIP is also SIP-compatible. (“SIP” means Session InitiationProtocol.) However, H.450 Supplementary Services features can be usedunder H.323 only and not under SIP.


Data Compression & Quality of Service. The MultiVOIP3010 comesequipped with a variety of data compression capabilities, including G.723,G.729, and G.711 and features DiffServ quality-of-service (QoS) capabilities.

VOIP Functions. The MultiVOIP MVP3010 gateway performs four basicfunctions: (a) it converts a dialed number into an IP address, (b) it sends voiceover the data network, (c) it establishes a connection with another VOIPgateway at a remote site, and (d) it receives voice over the data network. Voiceis handled as IP packets with a variety of compression options. Each E1connection to the MultiVOIP provides 30 time-slot channels to connect to thetelco or to serve phone or fax stations connected to a PBX.

Ports. The MVP3010 also has a 10/100 Mbps Ethernet LAN interface, and aCommand port for configuration. An MVP3010 upgraded with the MVP30-60kit will have two Ethernet LAN interfaces and two Command ports.



20

Gatekeeper. E1 voip systems can have gatekeeper functionality by adding, asan endpoint, a Multi-Tech standalone gatekeeper (special software residing inseparate hardware). Gatekeepers are optional but useful within voip systems.The gatekeeper acts as the ‘clearinghouse’ for all calls within its zone.MultiTech’s stand-alone gatekeeper software performs all of the standardgatekeepers functions (address translation, admission control, bandwidthcontrol, and zone management) and also supports many valuable optionalfunctions (call control signaling, call authorization, and bandwidthmanagement).



21





22






23


E1 Front Panel LEDsBecause the MVP3010 and MVP30-60 both use a common main circuit card ormotherboard, the LED indicators are the same for both.

Figure 1-5: MVP3010 LEDs

Active LEDs. The MVP3010 front panel has two sets of identical LEDs. Inthe MVP3010 as shipped (that is, without an expansion card), the left-hand setof LEDs is functional whereas the right-hand set is not.

When the MVP3010 has been upgraded with an MVP30-60 kit, the right-handset of LEDs will also become active.


24

E1 LED Descriptions

MVP3010 Front Panel LED Definitions

LED NAME DESCRIPTION


Boot After power up, the Boot LED will be on for about 10seconds while the MVP3010 is booting.

FDX Full-Duplex & Collision LED. This LED indicates whetherthe Ethernet connection is half-duplex or full-duplex (FDX)and, in half-duplex mode, indicates occurrence of datacollisions. LED is on constantly for full-duplex mode; LEDis off constantly for half-duplex mode. When operating inhalf-duplex mode, the LED will flash during data collisions.

LNK Link/Activity LED. This LED is lit if Ethernet connectionhas been made. It is off when the link is down (i.e., whenno Ethernet connection exists). While link is up, this LEDwill flash off to indicate data activity.

T1 T1. Not supported.

E1 E1. When lit, indicates presence of E1 connection.

PRI PRI. On if E1 line is of ISDN-Primary-Rate type.

ONL Online. This LED is on when frame synchronizationhas been established on the T1/E1 link.

IC IC LED is on when Internal Clocking is selected inT1/E1 configuration.

LC Indicates Loss of Carrier.

LS Indicates Loss of Signal.

Test For testing purposes only. For testing purposes only.


25

Introduction to Analog MultiVOIPs(MVP130, MVP-210/410/810 & MVP428)

VOIP: The Free Ride. We proudly present Multi-Tech's MVP130 and MVP-210/410/810 generation of MultiVOIP Voice-over-IP Gateways. All of thesemodels allow voice/fax communication to be transmitted at no additionalexpense over your existing IP network, which has ordinarily been data only.To access this free voice and fax communication, you simply connect theMultiVOIP to your telephone equipment and your existing Internet connection.These analog MultiVOIPs inter-operate readily with T1 or E1 MultiVOIPunits.

R C V X M T C OL LN K X MT RC V X S G R SG

X M T R CV XS G RS G X MT RC V X S G R SG XM T R C V X SG RS G

XM T R C V X SG RS G

X MT RC V X S G R SG

X MT RC V X S G R SG

Voi ce/Fax 5 Voice/ Fax 6 Voice/Fax 7 Voi ce/ Fax 8

Voice/Fax 1 Voice/ Fax 2 Voice/Fax 3 Voi ce/ Fax 4Ether netBootPowerX M T R CV XS G RS G

Figure 1-6: MVP-410/810 Chassis

Figure 1-7: MVP-210 Chassis


26

Figure 1-8: MultiVOIP MVP130 Chassis

Capacity. MultiVOIP model MVP810 is an eight-channel unit, the modelMVP410 is a four-channel, the model MVP210 is a two-channel units, and theMVP130 is a single-channel unit. All of these MultiVOIP units have a10/100Mbps Ethernet interface and a command port for configuration. TheMVP428 is an expansion circuit card for the four-channel MVP410 that turns itinto an eight-channel voip.

Mounting. Mechanically, the MVP410 and MVP810 MultiVOIPs aredesigned for a one-high industry-standard EIA 19-inch rack enclosure. Bycontrast, MVP130 and the MVP210 are tabletop units. The product must beinstalled by qualified service personnel in a restricted-access area, inaccordance with Articles 110-16, 10-17, and 110-18 of the National ElectricalCode, ANSI/NFPA 70.

Phone System Transparency. These MultiVOIPs inter-operate with atelephone switch or PBX, acting as a switching device that directs voice andfax calls over an IP network. The MultiVOIPs have “phonebooks,” directoriesthat determine to who calls may be made and the sequences that must be usedto complete calls through the MultiVOIP. The phonebooks allow the phoneuser to interact with the VOIP system just as they would with an ordinary PBXor telco switch. When the phonebooks are set, special dialing sequences areminimized or eliminated altogether. Once the call destination is determined,the phonebook settings determine whether the destination VOIP unit must stripoff or add dialing digits to make the call appear at its destination to be a localcall.

H. 323, SIP, & SPP. Being H.323 compatible, the analog MultiVOIP unit canplace calls to telephone equipment at remote IP network locations that alsocontain H.323 compatible voice-over-IP gateways. It will interface with H.323


27

software and H.323 gatekeeper units. H.323 specifications also bring to voiptelephony many special features common to conventional telephony. H.323features of this kind that have been implemented into the MultiVOIP includeCall Hold, Call Waiting, Call Identification, Call Forwarding (from the H.450standard), and Call Transfer (H.450.2 from H.323 Version 2). The fourthversion of the H.323 standard improves system resource usage (esp. logicalport or socket usage) by handling call signaling more compactly and allowinguse of the low-overhead UDP protocol instead of the error-correcting TCPprotocol where possible.



Data Compression & Quality of Service. The analog MultiVOIP unit comesequipped with a variety of data compression capabilities, including G.723,G.729, and G.711 and features DiffServ quality-of-service (QoS) capabilities.


Gatekeepers. For voip systems built with MultiTech’s analog gateway units,users can have a stand-alone gatekeeper (gatekeeper software residing inseparate hardware). Gatekeepers are optional but useful within voip systems.The gatekeeper acts as the ‘clearinghouse’ for all calls within its zone.MultiTech’s stand-alone gatekeeper software performs all of the standardgatekeepers functions (address translation, admission control, and bandwidthcontrol) and also supports many valuable optional functions (call controlsignaling, call authorization, bandwidth management, and call management).


28





29







30


Analog MultiVOIP Front Panel LEDsLED Types. The MultiVOIPs have two types of LEDs on their front panels:

(1) general operation LED indicators (for power, booting, andethernet functions), and

(2) channel operation LED indicators that describe the data traffic andperformance in each VOIP data channel.

Active LEDs. On both the MVP410 and MVP810, there are eight sets ofchannel-operation LEDs. However, on the MVP410, only the lower four setsof channel-operation LEDs are functional. On the MVP810, all eight sets arefunctional.

Figure 1-9. MVP410/810 LEDs


31

Similarly, the MVP210 has the general-operation indicator LEDs and two setsof channel-operation LEDs, one for each channel.

Figure 1-10. MVP210 LEDs

Finally, the MVP130 has the general-operation indicator LEDs and a set ofchannel-operation LEDs for its single voip channel.

Figure 1-11. MVP130 LEDs


32

Analog MultiVOIP LED Descriptions

MVP210/410/810 Front Panel LED DefinitionsLED NAME DESCRIPTION

General Operation LEDs (one set on each MultiVOIP model)


Boot After power up, the Boot LED will be on briefly while theMultiVOIP is booting. It lights whenever the MultiVOIP isbooting or downloading a setup configuration data set.

Ethernet FDX. LED indicates whether Ethernet connection is half-duplex or full-duplex (FDX) and, in half-duplex mode,indicates occurrence of data collisions. LED is onconstantly for full-duplex mode; LED is off constantly forhalf-duplex mode. When operating in half-duplex mode,the LED will flash during data collisions.LNK. Link/Activity LED. This LED is lit if Ethernetconnection has been made. It is off when the link is down(i.e., when no Ethernet connection exists). While link is up,this LED will flash off to indicate data activity.

Channel-Operation LEDs (one set for each channel)

XMT Transmit. This indicator blinks when voice packets arebeing transmitted to the local area network.

RCV Receive. This indicator blinks when voice packets arebeing received from the local area network.

XSG Transmit Signal. This indicator lights when the FXS-configured channel is off-hook, the FXO-configuredchannel is receiving a ring from the Telco, or the M lead isactive on the E&M configured channel. That is, it lightswhen the MultiVOIP is receiving a ring from the PBX.

RSG Receive Signal. This indicator lights when the FXS-configured channel is ringing, the FXO-configured channelhas taken the line off-hook, or the E lead is active on theE&M-configured channel.


33

MVP130 Front Panel LED DefinitionsLED NAME DESCRIPTION

General Operation LEDs




Channel-Operation LEDs

TX Transmit. This indicator blinks when voice packets arebeing transmitted to the local area network.

RX Receive. This indicator blinks when voice packets arebeing received from the local area network.

XS Transmit Signal. This indicator lights when theFXS-configured channel is off-hook or the FXO-configured channel is receiving a ring from the Telcoor PBX.

RS Receive Signal. This indicator lights when the FXS-configured channel is ringing or the FXO-configuredchannel has taken the line off-hook.


34

Introduction to ISDN-BRI MultiVOIPs(MVP410ST & MVP810ST)

VOIP: The Free Ride. We proudly present Multi-Tech's MVP-410ST/810STgeneration of MultiVOIP Voice-over-IP Gateways. All of these models allowvoice/fax communication to be transmitted at no additional expense over yourexisting IP network, which has ordinarily been data only. To access this freevoice and fax communication, you simply connect the MultiVOIP to yourtelephone equipment and your existing Internet connection. These ISDN BasicRate Interface (ISDN-BRI) MultiVOIPs inter-operate readily with T1 or E1MultiVOIP units (T1 and E1 MultiVOIP units can operate in ISDN PrimaryRate Mode, ISDN-PRI, as well).

RCV XMT COL LNK

EthernetBootPower ISDN 1 ISDN 2 ISDN 3 ISDN 4

D XMT RCV XMT RCV

Ch 1 Ch 2

D XMT RCV XMT RCV

Ch 3 Ch 4

D XMT RCV XMT RCV

Ch 5 Ch 6

D XMT RCV XMT RCV

Ch 7 Ch 8

Figure 1-12: MVP-410ST/810ST ChassisCapacity. MultiVOIP model MVP810ST accommodates four ISDN-BRI lines(eight B-channels) and model MVP410ST accommodates two ISDN-BRIchannels (four B-channels). Both of these MultiVOIP units have a 10/100MbpsEthernet interface and a command port for configuration.

Mounting. Mechanically, the MVP410ST and MVP810ST MultiVOIPs aredesigned for a one-high industry-standard EIA 19-inch rack enclosure. Theproduct must be installed by qualified service personnel in a restricted-accessarea, in accordance with Articles 110-16, 10-17, and 110-18 of the NationalElectrical Code, ANSI/NFPA 70.

Phone System Transparency. These MultiVOIPs inter-operate with atelephone switch or PBX, acting as a switching device that directs voice andfax calls over an IP network. The MultiVOIPs have “phonebooks,” directoriesthat determine to who calls may be made and the sequences that must be usedto complete calls through the MultiVOIP. The phonebooks allow the phoneuser to interact with the VOIP system just as they would with an ordinary PBXor telco switch. When the phonebooks are set, special dialing sequences areminimized or eliminated altogether. Once the call destination is determined,the phonebook settings determine whether the destination VOIP unit must stripoff or add dialing digits to make the call appear at its destination to be a localcall.


35

H. 323, SIP, & SPP. Being H.323 compatible, the BRI MultiVOIP unit canplace calls to telephone equipment at remote IP network locations that alsocontain H.323 compatible voice-over-IP gateways. It will interface with H.323software and H.323 gatekeeper units. H.323 specifications also bring to voiptelephony many special features common to conventional telephony. H.323features of this kind that have been implemented into the MultiVOIP includeCall Hold, Call Waiting, Call Identification, Call Forwarding (from the H.450standard), and Call Transfer (H.450.2 from H.323 Version 2). The fourthversion of the H.323 standard improves system resource usage (esp. logicalport or socket usage) by handling call signaling more compactly and allowinguse of the low-overhead UDP protocol instead of the error-correcting TCPprotocol where possible.



Data Compression & Quality of Service. The BRI MultiVOIP unit comesequipped with a variety of data compression capabilities, including G.723,G.729, and G.711 and features DiffServ quality-of-service (QoS) capabilities.


Gatekeeper. At this writing, ISDN-BRI MultiVOIP systems can havegatekeeper functionality only by adding, as an endpoint, a standalonegatekeeper (special software residing in separate hardware). Gatekeepers areoptional but useful within voip systems. The gatekeeper acts as the‘clearinghouse’ for all calls within its zone. MultiTech’s embedded and stand-alone gatekeeper software packages both perform all of the standardgatekeepers functions (address translation, admission control, bandwidthcontrol, and zone management) and also support many valuable optionalfunctions (call control signaling, call authorization, bandwidth management,and call management). The stand-alone gatekeeper is, however, slightly morefeature-rich than the embedded gatekeeper. For more details, see the“Embedded Gatekeeper” chapter of this manual and the manual onMultiTech’s stand-alone gatekeeper.


36

Management. Configuration and system management can be done locallywith the MultiVOIP configuration software. After an IP address has beenassigned locally, other configuration can be done remotely using theMultiVOIP web browser GUI. Remote system management can be done withthe MultiVOIP web browser GUI. Neither of these is available yet. The webGUI will be in release 5.04, however. All of these control software packagesare included on the Product CD.




37







38

Supplementary Telephony Services. This is available in 5.04 but not 5.02c.The H.450 standard (an addition to H.323) brings to voip telephony more ofthe premium features found in PSTN and PBX telephony. MultiVOIP unitsoffer five of these H.450 features: Call Transfer, Call Hold, Call Waiting, CallName Identification (not the same as Caller ID), and Call Forwarding. (Thefirst four features are found in the “Supplementary Services” window; the fifth,Call Forwarding, appears in the Add/Edit Inbound phonebook screen.) Notethat the first three features are closely related. All of these H.450 features aresupported for H.323 operation only; they are not supported for SIP or SPP.

ISDN BRI MultiVOIP Front Panel LEDsLED Types. The MultiVOIPs have two types of LEDs on their front panels:

(1) general operation LED indicators (for power, booting, andethernet functions), and

(2) channel operation LED indicators that describe the data traffic andperformance in each VOIP data channel.

Active LEDs. On the MVP810ST, there are four sets of ISDN-operationLEDs. On the MVP410ST, there are two sets of ISDN-operation LEDs. Eachset contains one “D” LED and two sets of channel operation LEDs (XMT andRCV).

Figure 1-13. MVP-410ST/810ST LEDs


39

ISDN-BRI MultiVOIP LED Descriptions

MVP-410ST/810ST Front Panel LED DefinitionsLED NAME DESCRIPTION

General Operation LEDs (one set on each MultiVOIP model)




D-Channel Operation LEDs (one for each ISDN line)

D ISDN D-channel & physical layer indicator. One “D” LEDfor each ISDN-BRI connection. The “D” LED is off whenthe BRI physical layer is de-activated.* It flashes when aconnection is being established on the physical layer. It ison when the physical layer has been activated. It flickers toindicate D-channel traffic.*If the voip is running in terminal mode and its BRI line isunplugged, the D LED goes off. However, if the voip isrunning in network mode and its BRI line is unplugged, itsLED will flash at regular interval.

B-Channel Operation LEDs (one for each B-channel)

XMT Transmit. This indicator blinks when voice packets arebeing transmitted onto the B-channel.

RCV Receive. This indicator blinks when voice packets arebeing received on the B-channel.


40

Computer RequirementsThe computer on which the MultiVOIP’s configuration program is installedmust meet these requirements:

• must be IBM-compatible PC with MS Windows operatingsystem;

• must have an available COM port for connection to theMultiVOIP.

However, this PC does not need to be connected to the MultiVOIPpermanently. It only needs to be connected when local configuration andmonitoring are done. Nearly all configuration and monitoring functions can bedone remotely via the IP network.


41

Specifications

Specs for Digital T1 MultiVOIP Units

Digital T1 MultiVOIP SpecificationsParameter……/Model MVP-2410

MVP-2410w/ MVP24-48ExpansionCard

OperatingVoltage/Current

100-240 VAC1.2 - 0.6 A

100-240 VAC1.2 - 0.6 A

MainsFrequencies

50/60 Hz 50/60 Hz

PowerConsumption

17 watts 27 watts

MechanicalDimensions

1.75”H x17.4”W x8.75”D

4.5cm H x44.2 cm W x22.2 cm D

1.75”H x17.4”W x8.75”D

4.5cm H x44.2 cm W x22.2 cm D

Weight 7.1 lbs.(3.2 kg)

7.5 lbs.(3.4 kg)


42

Specs for Digital E1 MultiVOIP Units

Digital E1 MultiVOIP SpecificationsParameter……/Model

MVP-3010 MVP-3010w/ MVP30-60ExpansionCard


100-240 VAC1.2 - 0.6 A

100-240 VAC1.2 - 0.6 A

MainsFrequencies

50/60 Hz 50/60 Hz

PowerConsumption

17 watts 27 watts


1.75”H x17.4”W x8.75”D

4.5cm H x44.2 cm W x22.2 cm D

1.75”H x17.4”W x8.75”D

4.5cm H x44.2 cm W x22.2 cm D


7.5 lbs.(3.4 kg)


43

Specs for Analog/BRI MultiVOIP Units

Parameter/Model

MVP210 MVP410 MVP810 orMVP410 + 428


Externaltransformer:3A @5V

100-240 VAC1.2 - 0.6 A

100-240 VAC1.2 - 0.6 A

MainsFrequencies

50/60 Hz 50/60 Hz 50/60 Hz

PowerConsumption

19 watts 29 watts 46 watts


6.2” W x9” D x1.4” H

15.8cm W x22.9cm D x3.6cm H

1.75” H x17.4” W x8.5” D

4.5cm H x44.2 cm W x21.6 cm D

1.75” H x17.4” W x8.5” D

4.5cm H x44.2 cm W x21.6 cm D

Weight 1.8lbs (.82kg)2.6lbs (1.17kg)with transformer

7.1 lbs.(3.2 kg)

7.7 lbs.(3.5 kg)

Parameter……/Model MVP410ST MVP810ST MVP130OperatingVoltage/Current

100-240VAC1.2-0.6 A

100-240VAC1.2-0.6 A

100-240VAC1.0 A

MainsFrequencies

50/60 Hz 50/60 Hz 50/60 Hz

PowerConsumption

12 watts 18 watts 9.7 watts (withphone off hook)


Same as MVP410 Same as MVP810 4.3" W x 5.6" D1.0" H

10.8 cm W X14.2 cm D X2.95 cm H


6.75 lbs.(3.06 kg)

8 oz.(23 g)


44

Installation at a GlanceThe basic steps of installing your MultiVOIP network involve unpacking theunits, connecting the cables, and configuring the units using managementsoftware (MultiVOIP Configuration software) and confirming connectivitywith another voip site. This process results in a fully functional Voice-Over-IPnetwork.

Related DocumentationThe MultiVOIP User Guide (the document you are now reading) comes inelectronic form and is included on your system CD. It presents in-depthinformation on the features and functionality of Multi-Tech’s MultiVOIPProduct Family.

The CD media is produced using Adobe AcrobatTM for viewing and printingthe user guide. To view or print your copy of a user guide, load AcrobatReaderTM on your system. The Acrobat Reader is included on the MultiVOIPCD and is also a free download from Adobe’s Web Site:

www.adobe.com/prodindex/acrobat/readstep.html

This MultiVOIP User Guide is also available on Multi-Tech’s Web site at:

http://www.multitech.com

Viewing and printing a user guide from the Web also requires that you havethe Acrobat Reader loaded on your system. To select the MultiVOIP User Guide fromthe Multi-Tech Systems home page, click Documents and then click MultiVOIP Family inthe product list drop-down window. All documents for this MultiVOIP Product Family will bedisplayed. You can then choose User Guide (MultiVOIP Product Family) to view or downloadthe .pdf file.

Entries (organized by model number) in the “knowledge base” and‘troubleshooting resolutions’ sections of the MultiTech web site (found under“Support”) constitute another source of help for problems encountered in thefield.

http://www.adobe.com/prodindex/acrobat/readstep.html


45

Chapter 2: Quick Start Instructions

Quick Start Instructions MultiVOIP User Guide

46

IntroductionThis chapter gets the MultiVOIP up and running quickly. The details we’veskipped to make this brief can be found elsewhere in the manual (see Table ofContents and Index).

MultiVOIP Startup TasksTask Summary

● Collecting Phone/IPDetails (vital!)

The MultiVOIP must be configured tointerface with your particular phone systemand IP network. To do so, certain detailsmust be known about those phone and IPsystems.

● Placement Decide where you’ll mount the voip.

● Command/ControlComputer Setup:Specs & Settings

Some modest minimum specifications mustbe met. A COM port must be set up.

● Hookup Connect power, phone, and data cables perdiagram.

● Software Installation This is the configuration program.It’s a standard Windows softwareinstallation.

● Phone/IP StarterConfiguration

You will enter phone numbers and IPaddresses. You’ll use default parametervalues where possible to get the systemrunning quickly.

● Phonebook StarterConfiguration

The phonebook is where you specify howcalls will be routed. To get the systemrunning quickly, you’ll make phonebooksfor just two voip sites.

● Connectivity Test You’ll find out if your voip system cancarry phone calls between two sites. Thatmeans you’re up and running!

● Troubleshooting Detect and remedy any problems that mighthave prevented connectivity.

MultiVOIP User Guide Quick Start Instructions

47

Phone/IP Details *Absolutely Needed*Before Starting the Installation

Gather IP Information

➼ Ask your computer networkadministrator.

Info needed to operate:all MultiVOIP models.

IP Network Parameters:Record for each VOIP Sitein System

• IP Address

• IP Mask

• Gateway• Domain Name Server (DNS) Info (not implemented; for future use)

Gather Telephone Information (T1)

➼ T1 Phone Parameters

Ask phone company orPBX maintainer.

Info needed to operate:MVP2410

T1 Telephony Parameters:Record for this VOIP Site

• Which frame format is used? ESF___ or D4___• Which CAS or PRI protocol is used? ______________• Clocking: Does the PBX or telco switch use

internal or external clocking? _________________Note that the setting used in the voip unit will be theopposite of the setting used by the telco/PBX.

• Which line coding is used? AMI___ or B8ZS___• Pulse shape level?: (most commonly 0 to 40 meters)


48

Phone/IP Details *Absolutely Needed* (cont’d)Gather Telephone Information (E1)

➼ E1 Phone Parameters



E1 Telephony Parameters:Record for this VOIP Site

• Which frame format is used? Double Frame_____MultiFrame w/ CRC4_____

MultiFrame w/ CRC4 modified_____• Which CAS or PRI protocol is used? ______________• Clocking: Does the PBX or telco switch use


• Which line coding is used? AMI___ or HDB3___

• Pulse shape level?: (most commonly 0 to 40 meters)

Gather Telephone Information (Analog)➼ Analog Phone Parameters

Ask phone company ortelecom manager.

Needed for:MVP810MVP410MVP210MVP130

Analog Telephony Interface Parameters:Record for this VOIP Site

• Which interface type (or “signaling”) is used? E&M_____ FXS/FXO_____

• If FXS, determine whether the line will be used for aphone, fax, or KTS (key telephone system)

• If FXO, determine if line will be an analog PBXextension or an analog line from a telco central office

• If E&M, determine these aspects of the E&M trunkline from the PBX:• What is its Type (1, 2, 3, 4, or 5)?• Is it 2-wire or 4-wire?• Is it Dial-Tone or Wink?


49

Gather Telephone Information (ISDN BRI)

➼ ISDN-BRI Phone Parameters


Needed for:MVP810STMVP410ST

ISDN-BRI Telephony Interface Parameters:Record them for this VOIP Site

• In which country is this voip installed?

• Which operator (switch type) is used?

• What type of line coding use required,A-law or u-law?

• Determine which BRI ports will be network side andwhich BRI ports will be terminal side.

• If you are connecting the MultiVOIP to networkequipment with a “U” interface, an NT1 device mustbe connected between them.


50

Phone/IP Details Often Needed/Wanted

Obtain Email Address for VOIP (for email call log reporting)

required if log reports ofVOIP call traffic

are to be sent by emailOptional

SMTP ParametersPreparation Task:

Ask Mail Serveradministrator to set upemail account (withpassword) for theMultiVOIP unit itself. Besure to give a uniqueidentifier to eachindividual MultiVOIP unit.

Get the IP address of themail server computer, aswell.

T o : I .T . D e p a r t m e n t

r e : e m a i l a c c o u n t f o r V O I P

[email protected]

Identify Remote VOIP Site to CallWhen you’re done installing the MultiVOIP, you’ll want to confirm that it isconfigured and operating properly. To do so, it’s good to have another voipthat you can call for testing purposes. You’ll want to confirm end-to-endconnectivity. You’ll need IP and telephone information about that remote site.

If this is the very first voip in the system, you’ll want to coordinate theinstallation of this MultiVOIP with an installation of another unit at a remotesite.

Identify VOIP Protocol to be UsedWill you use H.323, SIP, or SPP? Each has advantages and disadvantages.Although it is possible to mix protocols in a single VOIP system, it is highlydesirable to use the same VOIP protocol for all VOIP units in the system. SPPis a non-standard protocol developed by Multi-Tech. SPP is not compatiblewith the “Proprietary” protocol used in Multi-Tech’s earlier generation of voipgateways.


51

PlacementMount your MultiVOIP in a safe and convenient location where cables foryour network and phone system are accessible. Rack-mounting instructionsare in Chapter 3: Mechanical Installation & Cabling.

The Command/Control Computer (Specs & Settings)The computer used for command and control of the MultiVOIP

(a) must be an IBM-compatible PC,(b) must use a Microsoft operating system,(c) must be connected to your local network (Ethernet) system, and(d) must have an available serial COM port.

The configuration tasks and control tasks the PC will have to do with theMultiVOIP are not especially demanding. Still, we recommend using areasonably new computer. The computer that you use to configure yourMultiVOIP need not be dedicated to the MultiVOIP after installation iscomplete.

COM port on controller PC. You’ll need an available COM port on thecontroller PC. You’ll need to know which COM port is available for use withthe MultiVOIP (COM1, COM2, etc.).


52

Quick HookupsHookup for MVP2410 & MVP3010

Digital Voice

Trunk O

l

10 /100 RS-232

Ethernet Command

Cabling to your IP network.RJ-45 connector.T1/E1/PRI cabling to your PBX,

and/or to the PSTN.RJ-45 connector.

On/Off Switch Power CableReceptacle

Cabling to computer runningMultiVOIP software.RJ-45 to serial connector (DB9).

GroundingScrew

T1/E1 MultiVOIP Hookup(MVP-2410/3010)

Cabling to POTS line(RJ-11 connector) for remoteconfiguration via the built-inmodem.

Command

Modem


53

Hookup for MVP410 & MVP810

E&M FXS/FXO E&M FXS/FXOE&M FXS/FXOE&M FXS/FXO

E&M FXS/FXOE&M FXS/FXO E&M FXS/FXOE&M FXS/FXO

GroundingScrew:

Connect toEarth Ground

On/OffSwitch

Power CableReceptacle

Cabling to your IP network.RJ-45 connector.

Cabling to computer runningMultiVOIP software.

Connector at MultiVOIP: DB-25.Connector at computer: DB-9.

Cabling to phone equipment.

MVP810 has 8 connector pairs.MVP410 has 4 connector pairs.Only 1 connector of any pair isused at a time.

Analog MultiVOIP HookupMVP-410/810

E&M (RJ-45 connector):connects to E&M trunk linefrom PBX or telco office.

FXS (RJ-11 connector):connects to phone, fax,or key phone system.

FXO (RJ-11 connector):connects to analog phone lineor analog PBX extension.

Cabling to POTS line,for remote configuration.RJ-11 connector.

Command

Modem

DID-DPO (RJ-11 connector):connects to DID linefrom telco office. DPO functionalitynot yet supported.

EthernetCommand


54

Hookup for MVP410ST & MVP810ST

GroundingScrew:

Connect toEarth Ground

On/OffSwitch

Power CableReceptacleCabling to computer running

MultiVOIP software.Connector at MultiVOIP: DB-25.Connector at computer: DB-9.

Cabling to phone equipment.

MVP810ST has 4 ISDN connectors.MVP410ST has 2 ISDN connectors.

ISDN MultiVOIP HookupMVP-410ST/810ST

ISDNn (RJ-45 connector):connects to ISDN BRI linefrom PBX or telco office.Or connects to ISDN phoneor terminal adapter.

NT1 Device required betweenvoip interface (ports ISDN1 - ISDN4)and network equipment with“U” interface. Not needed forconnection to network equipmentwith “S/T” interface.

CommandISDN1 ISDN2 ISDN3 ISDN4


55

Hookup for MVP210

��

��

��

��

��

E&MFXS/FXO FXS/FXO RS232

COMMAND10/100 POWER

ETHERNETE&M

��

��

��

��

��

��

��

� �


56

Hookup for MVP130

Power Connection

Network Connection

Ethernet CommandPower FXS/FXO

Command Port Connection

PBX

PSTNTelephony Connection

FXS FXO

Hub


57

Load MultiVOIP Control Software onto PCFor more details, see Chapter 4: Software Installation.

1. MultiVOIP must be properly cabled. Power must be turned on.

2. Insert MultiVOIP CD into drive. Allow 10-20 seconds for Autorun to start.If Autorun fails, go toMy Computer | CD ROM drive | Open. Click Autorun icon.

3. At first dialog box, click Install Software.

4. At ‘welcome’ screen, click Next.

5. Follow on-screen instructions. Accept default program folder location andclick Next.

6. Accept default icon folder location. Click Next. Files will be copied.

7. Select available COM port on command/control computer.

8. At completion screen, click Finish.

9. At the prompt “Do you want to run MultiVOIP Configuration?,” click No.Software installation is complete.


58

Phone/IP Starter ConfigurationFull details here:

MVP2410MVP3010

Chapter 5: Technical Configuration forDigital T1/E1 MultiVOIPsin User Guide.

MVP130MVP210MVP410MVP810

Chapter 6: Technical Configuration forAnalog/BRI MultiVOIPsin User Guide

1. Open MultiVOIP program: Start | MultiVOIP xxx | Configuration.

2. Go to Configuration | IP. Enter the IP parameters for your voip site.

3. Do you want to configure and operate the MultiVOIP unit using the webbrowser GUI? (It has the same functionality as the local Windows GUI, butoffers remote access.)If NO, skip to step 5.If YES, continue with step 4.

4. Enable Web Browser GUI (Optional). To do configuration and operationprocedures using the web browser GUI, you must first enable it. To do so,follow these steps. (The browser used must be Internet Explorer 6.0 orabove; or Netscape 6.0 or above.)

A. Be sure an IP address has beenassigned to the MultiVOIP unit(this must be done in theMultiVOIP Windows GUI).

E. Open web browser.(Note: The PC being used must beconnected to and have an IPaddress on the same IP networkthat the voip is on.)

B. Save Setup in Windows GUI. F. Browse to IP address ofMultiVOIP unit.

C. Close the MultiVOIP WindowsGUI.

G. If username and password havebeen established, enter themwhen prompted by voip.

D. Install Java program fromMultiVOIP product CD.(Must be Java Runtime Environment1.4.2_01 or above.)NOTE: Required on first use ofWeb Browser GUI only.

H. Use web browser GUI toconfigure or operate voip.

Need moreinfo?

See “Web Browser Interface” in Operation &Maintenance chapter of User Guide (on CD).


59

Once you’ve begun using the web browser GUI, you can go back to theMultiVOIP Windows GUI at any time. However, you must log out of theweb browser GUI before using the MultiVOIP Windows GUI.

5. Go to Configuration | Voice/Fax. Select Coder | “Automatic.” At theright-hand side of the dialog box, click Default. If you know any specificparameter values that will apply to your system, enter them. Click CopyChannel. Select Copy to All. Click Copy. At main Voice/Fax Parametersscreen, click OK to exit from the dialog box.

6. Enter telephone system information.

Analog MultiVOIPsMVP130,

MVP-210/410/810

Digital MultiVOIPsMVP-2410/3010

Go toConfiguration | Interface.Enter parameters obtainedfrom phone company or PBXadministrator.

Go toConfiguration | T1/E1/ISDN.Enter parameters obtainedfrom phone company or PBXadministrator.

ISDN-BRI MultiVOIPsMVP-410ST/810ST

Go to Configuration | ISDN BRI.Enter parameters obtained from phone company orPBX administrator.If the voip is connected to BRI extensions of a PBX ora phone company, then select "Terminal"in the ISDN BRI Parameters screen.If the voip is connected to ISDN terminal adaptersand/or ISDN phones, then select "Network"in the ISDN BRI Parameters screen.

7. Go to Configuration | Regional Parameters. Select the Country/Regionthat fits your situation. Click Default and confirm.

8. Go to Configuration | Regional Parameters. In the Country Selectionfor Built-In Modem field (drop-down list), select the country that best fitsyour situation. (This may not be the same as your selection for theCountry/Region field. The selections in the Country Selection for Built-In Modem field entail more detailed groupings of telephony parameters thando the Country/Region values.)

Click OK to exit from the Regional Parameters dialog box.


60

9. Do you want the phone-call logs produced by the MultiVOIP to be sent outby email (to your Voip Administrator or someone else)?If NO, skip to step 11.If YES, continue with step 10.

10. Go to Configuration | SMTP.SMTP lets you send phone-call log records to the Voip Administrator byemail. Select Enable SMTP.You should have already obtained an email address for the MultiVOIP itself(this serves as the origination email account for email logs that theMultiVOIP can email out automatically).

Enter this email address in the “Login Name” field.Type the password for this email account.

Enter the IP address of the email server where the MultiVOIP’s emailaccount is located in the “Mail Server IP Address” field.

Typically the email log reports are sent to the Voip Administrator but theycan be sent to any email address. Decide where you want the email logs sentand enter that email address in the “Recipient Address” field.

Whenever email log messages are sent out, they must have a standardSubject line. Something like “Phone Logs for Voip N” is useful. If youhave more than one MultiVoip unit in the building, you’ll need a uniqueidentifier for each one (select a useful name or number for “N”). In this“Subject” field, enter a useful subject title for the log messages.

In the “Reply-To Address” field, enter the email address of your VoipAdministrator.

11. Go to Configuration | Logs.Select “Enable Console Messages.” (Not applicable if using Web GUI.)

To allow log reports by email (if desired), click SMTP. Click OK.

To do logging with a SysLog client program, click on “SysLog Server –Enable” in the Logs screen. To implement this function, you must install aSysLog client program. For more info, see the “SysLog Server Functions”section of the Operation & Maintenance chapter of the User Guide.


61

Phone/IP Starter Configuration (continued)12. Enable premium (H.450) telephony features. (Not supported in BRI 502c

software.)

Go to Supplementary Services. Select any features to be used.For Call Hold, Call Transfer, & Call Waiting, specify the key sequence thatthe phone user will press to invoke the feature. For Call NameIdentification, specify the allowed name types to be used and a caller-iddescriptor.

If Call Forwarding is to be used, enable this feature in theAdd/Edit Inbound Phone Book screen.

After making changes, click on OK in the current configuration screenbefore moving on to the next configuration screen.

13. Go to Save Setup | Save and Reboot. Click OK. This will save theparameter values that you have just entered.

The MultiVOIP’s “BOOT” LED will light up while the configuration file isbeing saved and loaded into the MultiVOIP. Don’t do anything to theMultiVOIP until the “BOOT “LED is off (a loss of power at this point couldcause the MultiVOIP unit to lose the configuration settings you have made).

END OF PROCEDURE.


62

Phonebook Starter Configuration (with remote voip)If the topic of voip phone books is new to you, it may be helpful to read thePhoneBook Tips section (page 69) before starting this procedure.

To do this part of the quick setup, you need to know of another voip that youcan call to conduct a test. It should be at a remote location, typicallysomewhere outside of your building. You must know the phone number andIP address for that site. We are assuming here that the MultiVOIP will operatein conjunction with a PBX.

You must configure both the Outbound Phonebook and the InboundPhonebook. A starter configuration only means that two voip locations will beset up to begin the system and establish voip communication.

Outbound Phonebook1. Open the MultiVOIP program

(Start | MultiVOIP xxx | Configuration

2. Go to Phone Book | PhoneBook Modify | Outbound Phonebook| Add Entry.

3. On a sheet of paper, write down the calling code of the remote voip (areacode, country code, city code, etc.) that you’ll be calling.

Follow the example that best fits your situation.

North America,Long-Distance Example

Euro, National CallExample

Technician in Seattle (area206) must set up one voipthere, another in Chicago(area 312, downtown).

Answer: Write down 312.

Technician in central London(area 0207) to set up voipthere, another in Birmingham(area 0121).

Answer: write down 0121.

Euro, International Call ExampleTechnician in Rotterdam (country 31; city 010) to setup one voip there, another in Bordeaux (country 33;area 05).



63

4. Suppose you want to call a phone number outside of your building using aphone station that is an extension from your PBX system (if present). Whatdigits must you dial? Often a “9” or “8” must be dialed to “get an outsideline” through the PBX (i.e., to connect to the PSTN). Generally, “1 “or “11”or “0” must be dialed as a prefix for calls outside of the calling code area(long-distance calls, national calls, or international calls).

On a sheet of paper, write down the digits that you must dial before you candial a remote area code.



Seattle-Chicago system.

Seattle voip works with PBXthat uses “8” for all voipcalls. “1” must immediatelyprecede area code of dialednumber.


London/Birming. system.

London voip works withPBX that uses “9” for all out-of-building calls whether byvoip or by PSTN. “0” mustimmediately precede areacode of dialed number.


Euro, International Call ExampleRotterdam/Bordeaux system.Rotterdam voip works with PBX where “9” is used forall out-of-building calls. “0” must precede allinternational calls.



64

5. In the “Destination Pattern” field of the Add/Edit Outbound Phonebookscreen, enter the digits from step 4 followed by the digits from step 3.



Seattle-Chicago system.Answer: enter 81312 as

Destination Pat-ternin Outbound Phonebook of Seattle voip.

London/Birming. system.Leading zero of Birminghamarea code is dropped whencombined with national-dialing access code. (Suchpractices vary by country.)Answer: enter 90121 as

Destination Pat-ternin OutboundPhonebook ofLondon voip.Not 900121.

Euro, International Call ExampleRotterdam/Bordeaux system.

Answer: enter 903305 as Destination Pattern inOutbound Phonebook of Rotterdam voip.


65

6. Tally up the number of digits that must be dialed to reach the remote voipsite (including prefix digits of all types). Enter this number in the “TotalDigits” field.




To complete Seattle-to-Chicagocall, 81312 must be followed bythe 7-digit local phone numberin Chicago.

Answer: enter 12 as number ofTotal Digits inOutbound Phonebook of Seattle voip.


To complete London-to-Birmingham call, 90121 mustbe followed by the 7-digit localphone number in Birmingham.

Answer: enter 12 as number ofTotal Digits inOutbound Phonebook of London voip.

Euro, International Call ExampleRotterdam/Bordeaux system.To complete Rotterdam-to-Bordeaux call, 903305 must befollowed by 8-digit local phone number in Bordeaux.

Answer: enter 14 as number of Total Digits in OutboundPhonebook of Rotterdam voip.

7. In the “Remove Prefix” field, enter the initial PBX access digit(“8” or “9”).




Answer: enter 8 in “RemovePrefix” field ofSeattle OutboundPhonebook.


Answer: enter 9 in “RemovePrefix” field ofLondon OutboundPhonebook.


Answer: enter 9 in “Remove Prefix” field of OutboundPhonebook for Rotterdam voip.

Some PBXs will not ‘hand off’ the “8” or “9” to the voip. But for those PBX unitsthat do, it’s important to enter the “8” or “9” in the “Remove Prefix” field in theOutbound Phonebook. This precludes the problem of having to make two inbound


66

phonebook entries at remote voips, one to account for situations where “8” is used asthe PBX access digit, and another for when “9” is used.

8. Select the voip protocol that you will use (H.323, SIP, or SPP).

9. Click OK to exit from the Add/Edit Outbound Phonebook screen.

Inbound Phonebook1. Open the MultiVOIP program.

(Start | MultiVOIP xxx | Configuration

2. Go to Phone Book | PhoneBook Modify | Inbound Phonebook| Add Entry.

3. In the “Remove Prefix” field, enter your local calling code (area code,country code, city code, etc.) preceded by any other “access digits” that arerequired to reach your local site from the remote voip location (think of it asthough the call were being made through the PSTN – even though it will notbe).




Seattle is area 206. Chicagoemployees must dial 81 beforedialing any Seattle number onthe voip system.

Answer: 1206 is prefix to beremoved by local(Seattle) voip.


Inner London is 0207 area.Birmingham employees mustdial 9 before dialing any Londonnumber on the voip system.

Answer: 0207 is prefix to beremoved by local(London) voip.

Euro, International Call ExampleRotterdam/Bordeaux system.Rotterdam is country code 31, city code 010. Bordeauxemployees must dial 903110 before dialing any Rotterdamnumber on the voip system.

Answer: 03110 is prefix to be removed by local (Rotterdam)voip.


67

4. In the “Add Prefix” field, enter any digits that must be dialed from yourlocal voip to gain access to the PSTN.




On Seattle PBX, “8” is used toget an outside line.

Answer: 8 is the prefix to beadded by local(Seattle) voip.


On London PBX, “9” is used toget an outside line.

Answer: 9 is the prefix to beadded by local(London) voip.

Euro, International Call ExampleRotterdam/Bordeaux system.On Rotterdam PBX, “9” is used to get an outside line.

Answer: 9 is prefix to be added by local (Rotterdam) voip.

5. In the “Channel Number” field, enter “0.” A zero value means the voip unitwill assign the call to an available channel. If desired, specific channels canbe assigned to specific incoming calls (i.e., to any set of calls received with aparticular incoming dialing pattern).


68

6. In the “Description” field, it is useful to describe the ultimate destination ofthe calls. For example, in a New York City voip system, “incoming calls toManhattan office,” might describe a phonebook entry, as might thedescriptor “incoming calls to NYC local calling area.” The descriptionshould make the routing of calls easy to understand. (40 characters max.)




Possible Description:.Free Seattle access, allemployees


Possible Description:.Local-rate London access,all employees


Possible Description:. Local-rate Rotterdam access, allemployees

7. Repeat steps 2-6 for each inbound phonebook entry. When all entries arecomplete, go to step 8.

8. Click OK to exit the inbound phonebook screen.

9. Click on Save Setup. Highlight Save and Reboot. Click OK.

Your starter inbound phonebook configuration is complete.


69

Phonebook TipsPreparing the phonebook for your voip system is a complex task that, at first,seems quite daunting. These tips may make the task easier.

1. Use Dialing Patterns, Not Complete Phone Numbers. You will notgenerally enter complete phone numbers in the voip phonebook. Instead,you’ll enter “destination patterns” that involve area codes and other digits. Ifthe destination pattern is a whole area code, you’ll be assigning all calls to thatarea code to go to a particular voip that has a unique IP address. If yourdestination pattern includes an area code plus a particular local phone exchangenumber, then the scope of calls sent through your voip system will be narrowed(only calls within that local exchange will be handled by the designated voip,not all calls in that whole area code). In general, when there are fewer digits inyour destination pattern, you are asking the voip to handle calls to moredestinations.

2. The Four Types of Phonebook Digits Used. Important!“Destination patterns” to be entered in your phonebook will generally consistof:

(a) calling area codes,(b) access codes,(c) local exchange numbers, and(d) specialized codes.

Although voip phonebook entries may look confusing at first, it’s useful toremember that all the digits in any phonebook entry must be of one of thesefour types.

(a) calling area codes. There are different names for these around the world:“area codes,” “city codes,” “country codes,” etc. These codes, are used whenmaking non-local calls. They always precede the phone number that would bedialed when making a local call.


70

(b) access codes. There are digits (PSTN access codes) that must be dialed togain access to an operator, to access the publicly switched ‘long-distance’calling system(North America), to access the publicly switched ‘national’calling system (Europe and elsewhere), or to access the publicly switched‘international’ calling system (worldwide).

There are digits (PBX access codes) that must be dialed by phones connectedto PBX systems or key systems. Often a “9” must be dialed on a PBX phoneto gain access to the PSTN (‘to get an outside line’). Sometimes “8” must bedialed on a PBX phone to divert calls onto a leased line or to a voip system.However, sometimes PBX systems are ‘smart’ enough to route calls to a voipsystem without a special access code (so that “9” might still be used for allcalls outside of the building).

There are also digits (special access codes) that must be dialed to gain accessto a particular discount long-distance carrier or to some other closed orproprietary telephone system.

(c) local exchange numbers. Within any calling area there will be manylocal exchange numbers. A single exchange may be used for an entire smalltown. In cities, an exchange may be used for a particular neighborhood(although exchanges in cities do not always cover easily discernible areas).Organizations like businesses, governments, schools, and universities are alsocommonly assigned exchange numbers for their exclusive use. In some cases,these organizational-assigned exchanges can become non-localized because theexchange is assigned to one facility and linked, by the organization’s privatenetwork, to other sometimes distant locations.

(d) specialized codes. Some proprietary voip units assign, to sites and phonestations, numbers that are not compatible with PSTN numbering. This can alsooccur in PBX or key systems. These specialized numbers must be handled ona case-by-case basis.

3. Knowing When to Drop Digits. Example

When calling area codes and accesscodes are used in combination, aleading “1” or “0” must sometimes bedropped.

Phonebook Entry ➠

Area code for Inner London islisted as “0207.” However, ininternational calls the leading“0” is dropped.

InternationalAccess Code

U.K.Country

Code

Leading ZeroDropped fromArea Code


71

4. Using a Comma. Detail, = 1-second pauseCommas are used in telephone dialingstrings to indicate a pause to allow adial tone to appear (common on PBXand key systems). Commas may beused only in the “Add Prefix” field ofthe Inbound Phonebook.

In many PBX systems(not needed in all)

5. Ease of Use. The phonebook setup determines how easy the voip system isto use. Generally, you’ll want to make it so dialing a voip call is very similarto dialing any other number (on the PSTN or through the PBX).

6. Avoid Unintentional Calls to Official/Emergency Numbers. Dialing avoip call will typically be somewhat different than ordinary dialing. Becauseof this, it’s possible to set up situations, quite unwittingly, where phone usersmay be predisposed to call official numbers without intending to do so.Conversely, a voip/PBX system might also make it difficult to place anofficial/emergency call when one intends to do so. Study your phonebooksetup and do some dialing on the system to avoid these pitfalls.

7. Inbound/Outbound Pattern Matching. In general, the Inbound Phonebookentries of the local voip unit will match the Outbound Phonebook entries of theremote voip unit. Similarly, the Outbound Phonebook entries of the local voipunit will match the Inbound Phonebook entries of the remote voip unit. Therewill often be non-matching entries, but it’s nonetheless useful to notice thematching between the phonebooks.

8. Simulating Network in-lab/on-benchtop. One common method ofconfiguring a voip network is to set up a local IP network in a lab, connect voipunits to it, and perhaps have phones connected on channel banks to make testcalls.


72

Phonebook Example

Flagstaff Office

204.16.49.758-Channel

Analog VoIP(MVP810)

PSTN

PBX System.Main Number:777-5600

30 extensions

Area: 520

Santa Fe Office


40 extensions

PSTN

204.16.49.748-Channel

Analog VoIP(MVP810)

Area: 505

PBX System.Main Number:

333-2700

204.16.49.73

24-ChannelDigital VoIP(MVP2410)

PSTN

Boise OfficeArea: 208

90 extensions

IPNetwork

Each Outbound Phonebook contains twopairs of entries, two entries for eachremote site. Whenever an out-of-townemployee dials a 12-digit numberbeginning with the l isted 5-digitdestination pattern (9+1+area code) ofanother company location, the PBXhands the call to the voip system. Thelocal voip strips off the “ 9” and directsthe call to the IP address of the remotevoip. The remote voip receives the calland hands it to its PBX. The PBX thencompletes the call to the PSTN.

The one-digit Outbound destinationpatterns pertain to 3-digit call ingbetween company employees.

Inbound Phonebook

One Common SituationVoip Example. This company has offices in threedifferent cities. The PBX units all operate alike.Notably, they all give access to outside lines using“ 9.” They all are ‘smart’ enough to identify voip callswithout using a special access digit (“ 8” is used insome systems). Finally, the system operates so thatemployees in any office can dial employees in anyother office using only three digits. Here are thephonebooks needed for that system.

Each Inbound Phonebook containstwo entries. The first entry (4 digits)specifies how incoming calls from theother voip sites w ill be handled ifthey go out onto the local PSTN.Essentially, all those calls come to thereceiving voip w ith a patternbeginning with 1+area code. The localvoip removes those four digitsbecause they aren’t needed whendialing locally. The local voipattaches a “ 9” at the beginning of thenumber to get an outside line. ThePBX then completes the call to thePSTN.

The second Inbound Phonebook entry(1 digit) is for receiving calls fromcompany employees in the other twocities. The out-of-town employeesimply dials 3 digits. The first of thethree digits is uniquely used at eachsite and so acts as a destinationpattern (Boise extensions are 7xx,Santa Fe extensions 2xx, Flagstaffextensions 6xx).

The local voip sees the pattern in itsinbound phone book and notes thefirst digit (here either 2, 5, or 6).To make the match, this first digit,

three-digit number.specific extension identified by thePBX. The PBX can then dial thevoip will send all three digits to theadded back once again so that thefield. This first digit must then be2, 5, or 6 is put in the “ Remove Prefix”


73

Voip Sites with Phonebooks

Flagstaff Office

204.16.49.758-Channel

Analog VoIP(MVP810)

PSTN


30 extensions

Area: 520

PBX System.Main Number:

333-2700

204.16.49.73

24-ChannelDigital VoIP(MVP2410)

PSTN

Boise OfficeArea: 208

90 extensions

Santa Fe Office


40 extensions

PSTN

204.16.49.748-Channel

Analog VoIP(MVP810)

Area: 505

IPNetwork

Boise Voip Boise VoipInbound Phonebook Outbound Phonebook

Prefix toRemove

Prefixto Add

DescriptionIncoming Calls

Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr

DescriptionOutgoing Calls

91208 9, Incoming callsto PSTN,Boise Area

91505 12 none none 204.16.49.74

Outgoing callsto Santa Fearea

7 7 i ncoming callsto extensionsof company’sPBX systemin Boise

2 3 none none 204.16.49.74

3-digit calls toSanta Feemployees(extensions200 to 240)

91520 12 none none 204.16.49.75

Outgoing callsto Flagstaffarea

6 3 none none 204.16.49.75

3-digit calls toFlagstaffemployees(extensions600-630)

Santa Fe Voip Santa Fe VoipInbound Phonebook Outbound Phonebook

Prefix toRemove

Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


91505 9, Incoming callsto PSTN,Santa Fe localcalls

91208 12 none none 204.16.49.73

Outgoing callsto Boise area

2 2 Incoming callsto extensionsof company’sPBX systemin Santa Fe

7 3 none none 204.16.49.73

3-digit calls toBoiseemployees(extensions700-790)

91520 12 none none 204.16.49.75


6 3 none none 204.16.49.75


Flagstaff Voip Flagstaff VoipInbound Phonebook Outbound Phonebook

Prefix toRemove

Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


91520 9 Incoming callsto PSTN,Flagstaff localcalls

91505 12 none none 204.16.49.74


6 6 Incoming callsto extensionsof company’sPBX systemin Flagstaff

2 3 none none 204.16.49.74

3-digit calls toSanta Feemployees(extensions200-240)

91208 12 none none 204.16.49.73


7 3 none none 204.16.49.73



74

Sample Phonebooks EnlargedBoise Voip Boise Voip

Inbound Phonebook Outbound PhonebookPrefix toRemove

Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


91208 9, Incoming callsto PSTN,Boise Area

91505 12 none none 204.16.49.74


7 7 i ncoming callsto extensionsof company’sPBX systemin Boise

2 3 none none 204.16.49.74

3-digit calls toSanta Feemployees(extensions200 to 240)

91520 12 none none 204.16.49.75


6 3 none none 204.16.49.75


Santa Fe Voip Santa Fe VoipInbound Phonebook Outbound Phonebook

Prefix toRemove

Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


91505 9, Incoming callsto PSTN,Santa Fe localcalls

91208 12 none none 204.16.49.73


2 2 Incoming callsto extensionsof company’sPBX systemin Santa Fe

7 3 none none 204.16.49.73


91520 12 none none 204.16.49.75


6 3 none none 204.16.49.75


Flagstaff Voip Flagstaff VoipInbound Phonebook Outbound Phonebook

Prefix toRemove

Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


91520 9 Incoming callsto PSTN,Flagstaff localcalls

91505 12 none none 204.16.49.74


6 6 Incoming callsto extensionsof company’sPBX systemin Flagstaff

2 3 none none 204.16.49.74

3-digit calls toSanta Feemployees(extensions200-240)

91208 12 none none 204.16.49.73


7 3 none none 204.16.49.73



75

Phonebook WorksheetVoip Location/ID:____________________________


Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


Other Details:

Voip Location/ID:____________________________


Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


Other Details:

Voip Location/ID:____________________________


Prefixto Add


Destin.Pattern

TotalDigits

Prefix toRemove

Prefixto Add

IPAddr


Other Details:


76

Enlarged Phonebook Worksheet


77

Connectivity TestThe procedures “Phone/IP Starter Configuration” and “Phonebook StarterConfiguration” must be completed before you can do this procedure.

1. These connections must be made:

Connectionsfor digital MultiVOIPs(MVP-2410/3010)

for analog MultiVOIPs(MVP-130/210/410/810)

MultiVOIP to local PBX MultiVOIP to local phonestation –OR--MultiVOIP to extension of keyphone system

MultiVOIP to command PC MultiVOIP to command PC

MultiVOIP to Internet MultiVOIP to Internet

2. Inbound Phonebook and Outbound Phonebook must both be set up with atleast one entry in each. These entries must allow for connection betweentwo voip units.

3. Console messages must be enabled. (If this has not been done already, go,in the MultiVOIP GUI, to Configuration | Logs and select the “ConsoleMessages” checkbox.

4. You now need to free up the COM port connection (currently being used bythe MultiVOIP program) so that the HyperTerminal program can use it. Todo this, you can either (a) click on Connection in the sidebar and select“Disconnect” from the drop-down box, or (b) close down the MultiVOIPprogram altogether.


78

5. Open the HyperTerminal program.

6. Use HyperTerminal to receive and record console messages from theMultiVOIP unit. To do so, set up HyperTerminal as follows (setup shown isfor Windows NT4; details will differ slightly in other MS operatingsystems):

In the upper toolbar of the HyperTerminal screen, click on theProperties button.

In the “Connect To” tab of the Connection Properties dialogbox, click on the Configure button.

In the next dialog box, on the “General” tab, set “MaximumSpeed” to 115200 bps.

On the “Connection” tab, set connection preferences to:

Data bits: 8

Parity: none

Stop bits: 1

Click OK twice to exit settings dialog boxes.

7. Make VOIP call.

for digital MultiVOIPs(MVP-2410/3010)

for analog MultiVOIPs(MVP-130/210/410/810)

Make call from an extension ofthe local PBX.

Make call on a local phone lineaccessing PSTN directly orthrough key system


79

8. Read console messages recorded on HyperTerminal.

Console Messages from Originating VOIP. The voip unit that originatesthe call will send back messages like that shown below.

[00026975] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[1]TimeStamp : 26975

[00027190] CAS[0] : TX : ABCD = 1, 1, 1, 1[00027190] PSTN: cas seizure detected on 0[00027440] CAS[0] : TX : ABCD = 0, 0, 0, 0[00033290] PSTN:call detected on 0 num=17637175662*[00033290] H323IF[0]:destAddr =

TA:200.2.10.5:1720,NAME:MoundsView,TEL:17637175662,17637175662

[00033290] H323IF[0]:srcAddr = NAME:NewYork,TA:200.2.9.20

[00033440] H323IF [0]:cmCallStateProceeding[00033500] H323[0]: Remote Information (Q931): MultiVOIP

- T1[00033565] CAS[0] : TX : ABCD = 1, 1, 1, 1[00033675] H323IF [0]: MasterSlaveStatus=Slave[00033675] H323IF[0]:FastStart Setup Not Used[00033690] CAS[0] : TX : ABCD = 1, 1, 1, 1[00033755] H323IF[0]: Coder used 'g7231'[00033810] PSTN:pstn call connected on 0


80

Console Messages from Terminating VOIP. The voip unit connected to thephone where the call is answered will send back messages like that shownbelow.

[00170860] H323[0]: New incoming call[00170860] PSTNIF : Placing call on channel 0 Outbound

digit 7175662[00170885] CAS[0] : TX : ABCD = 1, 1, 1, 1[00171095] H323IF [0]: MasterSlaveStatus=Master[00171105] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[7]

TimeStamp : 171105[00171105] H323IF[0]: Coder used 'g7231'[00171110] H323IF[0]:FastStart Setup Not Used[00171110] H323IF[0]: Already opened the outgoing logical

channel[00171110] H323IF[0]: Coder used 'g7231'[00171315] CAS[0] : RX : ABCD = 0, 0, 0, 0,Pstn State[9]

TimeStamp : 171315[00172275] PSTN: dialing digit ended on 0[00172285] PSTN: pstn proceeding indication on 0[00172995] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[12]

TimeStamp : 172995[00173660] CAS[0] : TX : ABCD = 1, 1, 1, 1[00173760] PSTN:pstn call connected on 0

9. When you see the following message, end-to-end voip connectivity hasbeen achieved.

“PSTN: pstn call connected on X”

where x is the number of the voip channel carrying the call

10. If the HyperTerminal messages do not confirm connectivity, go to theTroubleshooting procedure below.


81

TroubleshootingIf you cannot establish connectivity between two voips in the system, followthe steps below to determine the problem.

1. Ping both MultiVOIP units to confirm connectivity to the network.

2. Verify the telephone connections.

A. For MVP2410 or MVP3010.Check cabling. Are connections well seated? To correct receptacle?Is the ONL LED on?

(If on, ONL indicates that the MultiVOIP is online on thenetwork.)

Are T1/E1/PRI Parameter settings correct?

B. For MVP130, MVP210, MVP410, or MVP810.Check cabling. Are connections well seated? To correct receptacle?Are telephone Interface Parameter settings correct?

C. For MVP410ST or MVP810ST.Check cabling. Are connections well seated? To correctreceptacle?If terminal equipment is connected to the voip, then "Network"should be selected for that BRI interface in the ISDN BRIParameters screen.Note: Each BRI interface is separately configurable.If network equipment such as an ISDN BRI PBX or anISDN BRI line from a phone company is connected to the voip,then "Terminal" should be selected for that BRI interface in theISDN BRI Parameters screen.Was the proper country and operator chosen?Was the proper type of line coding (A-law or u-law) chosen?


82

3. Verify phonebook configuration.

4. Observe console messages while placing a call. Look for error messagesindicating phonebook problems, network problems, voice-coder mismatches,etc.

Mechanical Installation & Cabling MultiVOIP User Guide

110

Chapter 3: Mechanical Installationand Cabling

MultiVOIP User Guide Mechanical Installation & Cabling

111

IntroductionThe MultiVOIP models MVP130, and MVP210 are tabletop units and can behandled easily by one person. However, the MVP410, MVP810, MVP2410,and MVP3010 are somewhat heavier units. When these units are to beinstalled into a rack, two able-bodied persons should participate.

Please read the safety notices before beginning installation.

Safety Warnings

Lithium Battery CautionA lithium battery on the voice/fax channel board provides backup power forthe timekeeping capability. The battery has an estimated life expectancy of tenyears.

When the battery starts to weaken, the date and time may be incorrect. If thebattery fails, the board must be sent back to Multi-Tech Systems for batteryreplacement.

Warning: There is danger of explosion if the battery is incorrectly replaced.

Safety Warnings Telecom1. Never install telephone wiring during a lightning storm.

2. Never install a telephone jack in wet locations unless the jack is specificallydesigned for wet locations.

3. This product is to be used with UL and UL listed computers.

4. Never touch uninsulated telephone wires or terminals unless the telephoneline 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 electricalstorm. There may be a remote risk of electrical shock from lightning.

7. Do not use a telephone in the vicinity of a gas leak.

8. To reduce the risk of fire, use only a UL-listed 26 AWG or largertelecommunication line cord.


112

Unpacking Your MultiVOIPWhen unpacking your MultiVOIP, check to see that all of the items shown areincluded in the box. For the various MultiVOIP models, the contents of thebox will be different. Study the particular illustration below that is appropriateto the model you have purchased. If any box contents are missing, contactMultiTech Tech Support at 1-800-972-2439.

Unpacking the MVP2410/3010

Figure 3-1: Unpacking the MVP2410/3010


113

Unpacking the MVP-410x/810x

Voice/Fax over IP Networks

RCV XMT COL LNK XMT RCV XSG RSG

XMT RCV XSG RSG XMT RCV XSG RSG XMT RCV XSG RSG

XMT RCV XSG RSG

XMT RCV XSG RSG

XMT RCV XSG RSG

Voice/Fax 5 Voice/Fax 6 Voice/Fax 7 Voice/Fax 8

Voice/Fax 1 Voice/Fax 2 Voice/Fax 3 Voice/Fax 4EthernetBootPowerXMT RCV XSG RSG

Quick StartGuide

Figure 3-2: Unpacking the MVP-410x/810x


114

Unpacking the MVP210

Voice/Fax over IP Networks200

Quick Start Guide

Figure 3-3: Unpacking the MVP210


115

Unpacking the MVP130

Figure 3-4: Unpacking the MVP130


116

Rack Mounting Instructions forMVP-2410/3010 & MVP-410x/810x

The MultiVOIPs can be mounted in an industry-standard EIA 19-inch rackenclosure, as shown in Figure 3-5.

Figure 3-5: Rack-Mounting (MVP2410/3010 or MVP410x/810x)


117

Safety Recommendations for Rack InstallationsEnsure proper installation of the unit in a closed or multi-unit enclosure byfollowing the recommended installation as defined by the enclosuremanufacturer. Do not place the unit directly on top of other equipment or placeother equipment directly on top of the unit. If installing the unit in a closed ormulti-unit enclosure, ensure adequate airflow within the rack so that themaximum recommended ambient temperature is not exceeded. Ensure that theunit is properly connected to earth ground by verifying that it is reliablygrounded when mounted within a rack. If a power strip is used, ensure that thepower strip provides adequate grounding of the attached apparatus.

When mounting the equipment in the rack, make sure mechanical loading iseven to avoid a hazardous condition, such as loading heavy equipment in rackunevenly. The rack used should safely support the combined weight of all theequipment it supports.

Ensure that the mains supply circuit is capable of handling the load of theequipment. See the power label on the equipment for load requirements (fullspecifications for MultiVOIP models are presented in chapter 1 of thismanual).

Maximum ambient temperature for the unit is 60 degrees Celsius (140 degreesFahrenheit) at 20-90% non-condensing relative humidity. This equipmentshould only be installed by properly qualified service personnel. Only connectlike circuits. In other words, connect SELV (Secondary Extra Low Voltage)circuits to SELV circuits and TN (Telecommunications Network) circuits toTN circuits.


118

19-Inch Rack Enclosure Mounting ProcedureAttaching the MultiVOIP to a rack-rail of an EIA 19-inch rack enclosure willcertainly require two persons. Essentially, the technicians must attach thebrackets to the MultiVOIP chassis with the screws provided, as shown inFigure 3-6, and then secure unit to rack rails by the brackets, as shown inFigure 3-7. Because equipment racks vary, screws for rack-rail mounting arenot provided. Follow the instructions of the rack manufacturer and use screwsthat fit.

1. Position the right rack-mounting bracket on the MultiVOIP usingthe two vertical mounting screw holes.

2. Secure the bracket to the MultiVOIP using the two screws provided.3. Position the left rack-mounting bracket on the MultiVOIP using the

two vertical mounting screw holes.4. Secure the bracket to the MultiVOIP using the two screws provided.5. Remove feet (4) from the MultiVOIP unit.6. Mount the MultiVOIP in the rack enclosure per the rack

manufacture’s mounting procedure.

x

x

Figure 3-6: Bracket Attachment for Rack Mounting(MVP-2410/3010 & MVP-410x/810x)

Figure 3-7: Attaching MultiVOIP to Rack Rail(MVP-2410/3010 & MVP-410x/810x)


119

Cabling

Cabling Procedure for MVP2410/3010Cabling your MultiVOIP entails making the proper connections for power,command port, phone system (T1/E1 line connected to PBX or telco office),and Ethernet network. Figure 3-8 shows the back panel connectors and theassociated cable connections. The following procedure details the stepsnecessary for cabling your MultiVOIP.

1. Connect the power cord to a live AC outlet, then connect it to theMultiVOIP’s power receptacle shown at top right in Figure 3-8.

ETHERNET COMMANDDIGITAL VOICE

ETHERNET COMMAND

10 BASET RS232

DIGITAL VOICE

TRUNK

T1

PBX

PSTN

Telephony Connection

Network Connection


COMMAND

MODEM

Hub

Figure 3-8. Cabling for MVP2410/3010

2. Connect the MultiVOIP to the PC (the computer that will hold theMultiVOIP software) using the RJ-45 to DB9 (female) cable provided withyour unit. Plug the RJ-45 end of the cable into the Command port of theMultiVOIP and connect the other end (the DB9 connector) to the PC serialport you are using (typically COM1 or COM2). See Figure 3-8.

3. Connect a network cable to the Ethernet connector on the back of theMultiVOIP. Connect the other end of the cable to your network.

4. If you intend to configure the MultiVOIP remotely using the MultiVOIPWindows GUI, connect an RJ-11 phone cable between the Command


120

Modem connector (at the rear of the MultiVOIP) and a receptacle served bya telco POTS line. See Figure 3-9.

The Command Modem is built into the MultiVOIP unit. To configure theMultiVOIP remotely using its Windows GUI, you must call into theMultiVOIP’s Command Modem. Once a connection is made, theconfiguration process is identical to local configuration with the WindowsGUI.

ETHERNET COMMANDDIGITAL VOICE

ETHERNET COMMAND

10 BASET RS232

DIGITAL VOICE

TRUNK

COMMAND

MODEM

Telco POTS Line

Grounding Screw

Figure 3-9. MVP-2410/3010 Voip Connections for GND & Remote Config Modem

5. Ensure that the unit is properly connected to earth ground by verifying that itis reliably grounded when mounted within a rack.

This can be accomplished by connecting a grounding wire between thechassis grounding screw (see Figure 3-9) and a metallic object that willprovide an electrical ground.

6. Turn on power to the MultiVOIP by setting the power switch on the rightside panel to the ON position. Wait for the Boot LED on the MultiVOIP togo off before proceeding. This may take a couple of minutes.

Proceed to Chapter 4 “Software Installation.”


121

Cabling Procedure for MVP-410/810

Cabling involves connecting the MultiVOIP to your LAN and telephoneequipment.

1. For DID channels only. If all channels of your MultiVOIP will be usingeither FXS, FXO, or E&M telephony interfaces, skip to step 2.

For any channel on which you are using the DID interface type, you mustchange the jumper on the MultiVOIP circuit card.

a. Disconnect power. Unplug the AC power cord from the wall outlet orfrom the receptacle on the MultiVOIP unit.

b. Using a #1 Phillips driver, remove the three screws (at back of unit) thatattach the main circuit card to the chassis of the MultiVOIP.

x

xx

Screws (3) holding circuit card assembly to chassis.

MVP410/810rear panel

Figure 3-10. MVP-410/810 Rear Screw Locations

c. Pull the main circuit card out about 5 inches (the power connection tothe board prevents it from being removed entirely from the chassis).


122

d. Identify the channels on which the DID interface will be used.

For DIDInterface type

For non-DIDInterface type

Ch 1 Ch 2 Ch 3 Ch 4

Jumpers 1-4

}

}

U9U7 U8 U10

Ch 5 Ch 6 Ch 7 Ch 8

Jumpers 5-8

Jumper Configurations(enlarged)

MVP-410/810

MVP810 only

Generality:For channels using the DIDinterface, the jumper mustnot straddle across thecross-hatched area betweenthe jumper posts.For channels using any non-DIDinterface, it is acceptable that thejumper straddles across thecross-hatched area betweenthe jumper posts.

Upper Circuit Card

Main Circuit Card

Figure 3-11. MVP-410/810 Channel Jumper Settings

e. Position the jumper for each DID channel so that it does not connect the twojumper posts. For DID operation of a voip channel, the MultiVOIP willwork properly if you simply remove the jumper altogether, but that isinadviseable because the jumper might be needed later if a differenttelephony interface is used for that voip channel.

f. Slide the main circuit card back into the MultiVOIP chassis and replace thethree screws.


123

2. Connect the power cord supplied with your MultiVOIP to a live AC outletand to the power connector on the back of the MultiVOIP as shown at topright in Figure 3-12.

Ethernet Connection


COMMAND

10 BASETE&M FXS/FXO E&M FXS/FXO E&M FXS/FXO E&M FXS/FXO

E&M

FXO

PSTN

E&M FXS/FXO

FXS

E&M FXS/FXO E&M FXS/FXO E&M FXS/FXO E&M FXS/FXO

Voice/Fax Channel ConnectionsChannels 1-4 Bottom MVP410/810Channels 5-8 Top MVP810 Only

COMMAND

MODEM

ETHERNET

Command Modem connectorfor remote configuration

Figure 3-12: Cabling for MVP-410/810

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 theMultiVOIP and the other end into the PC serial port. See Figure 3-12.

4. Connect a network cable to the ETHERNET 10BASET connector on theback of the MultiVOIP. Connect the other end of the cable to your network.

5. For an FXS or FXO connection.(FXS Examples: analog phone, fax machine, Key Telephone System.)(FXO Examples: PBX extension, POTS line from telco central office.)Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXOconnector on the back of the MultiVOIP. Connect the other end to the device or phone jack.

For an E&M connection.(E&M Example: trunk line from telephone switch.)Connect one end of an RJ-45 phone cord to the Channel 1 E&M connectoron the back of the MultiVOIP.Connect the other end to the trunk line. Verify that the E&M Type in the E&M Options group of the Interface dialog


124

box is the same as the E&M trunk type supported by the telephone switch.See Appendix B for an E&M cabling pinout.

For a DID connection.(DID Example: DID fax system or DID voice phone lines.)Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXOconnector on the back of the MultiVOIP. Connect the other end to the DID jack.

NOTE: DID lines are polarity sensitive. If, during testing, the DID linerings busy consistently, you will need to reverse the polarity ofone end of the connector (swap the connections of the wires tothe two middle pins of one RJ-11 connector).

6. Repeat step 5 to connect the remaining telephone equipment to each channelon your MultiVOIP. Although a MultiVOIP’s channels are often allconfigured identically, each channel is individually configurable. So, forexample, some channels of a MultiVOIP might use the FXO interface andothers the FXS; some might use the DID interface and others E&M, etc.

7. If you intend to configure the MultiVOIP remotely using the MultiVOIPWindows GUI, connect an RJ-11 phone cable between the CommandModem connector (at the rear of the MultiVOIP) and a receptacle served bya telco POTS line. See Figure 3-13.

The Command Modem is built into the MultiVOIP unit. To configure theMultiVOIP remotely using its Windows GUI, you must call into theMultiVOIP’s Command Modem. Once a connection is made, theconfiguration process is identical to local configuration with the WindowsGUI.

COMMAND

10 BASETE&M FXS/FXO E&M FXS/FXO E&M FXS/FXO E&M FXS/FXO

E&M FXS/FXO E&M FXS/FXO E&M FXS/FXO E&M FXS/FXO COMMAND

MODEM

Telco POTS Line

Grounding Screw

ETHERNET

Command Modem connectorfor remote configuration

MVP-410/810Rear Panel

Figure 3-13. MVP-410/810 Voip Connections for GND &Remote Config Modem



125


9. Turn on power to the MultiVOIP by placing the ON/OFF switch on the backpanel to the ON position. Wait for the Boot LED on the MultiVOIP to go offbefore proceeding. This may take a few minutes.

Proceed to Chapter 4 to load the MultiVOIP software.


126

Cabling Procedure for MVP-410ST/810ST

Cabling involves connecting the MultiVOIP to your LAN and telephoneequipment.

1. Connect the power cord supplied with your MultiVOIP to a live AC outletand to the power connector on the back of the MultiVOIP as shown at topright in Figure 3-14.

Ethernet Connection


COMMAND ETHERNET

10 BASET

TERMINALMODE

ISDN-BRI Connections

ISDN1 & ISDN2 : MVP410ST/810STISDN3 & ISDN4: MVP810ST only

IS DN1 ISDN2 IS DN3 ISDN4

? NETWORKMODE

ISDNTA

PBX

PSTN

NT1Device

*

* NT1 Device is neededif PBX has “U” interface.

COMMAND

MODEM

Command Modem/TA connectorfor remote configuration

Figure 3-14: Cabling for MVP-410ST/810ST

2. 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 theMultiVOIP and the other end into the PC serial port. See Figure 3-14.

3. Connect a network cable to the ETHERNET 10BASET connector on theback of the MultiVOIP. Connect the other end of the cable to your network.


127

4. Terminal Mode. When a voip ISDN connector is to be connected to a PBXextension line or to a telco line, select “Terminal” as the“Layer 1 Interface” in the ISDN Parameters screen. When making cableconnections, an NT1 device will be needed between the MultiVOIP and thePSTN or between the MultiVOIP and any PBX with a “U” interface. (Formore information, see Appendix B: Cable Pinouts in this manual.) Connectcables between voip ISDN connectors and network equipment.

NOTE: In order to operate in Terminal mode, the networkequipment to which you will be connecting (e.g.,PBX) must support D-channel signaling in itsISDN-S/T interface.


128

Network Mode. When a voip ISDN connector is to be connected to an ISDNphone station or to an ISDN terminal adapter (TA), select “Network” as the“Layer 1 Interface” in the ISDN Parameters screen of the MultiVOIPsoftware. Connect cables between voip ISDN connectors and phone or TA.

NOTE. Any ISDN phone stations connected to the MVP-410ST/810ST must provide their own operating power.That is, the MVP-410ST/810ST does not supply power forISDN phone stations.


129

5. Repeat the above step to connect the remaining ISDN telephone equipmentto each ISDN connector on your MultiVOIP. Be aware that you can assigneach ISDN line separately and independently to either Network mode orTerminal mode. That is, all ISDN lines do not have to be assigned in to thesame operating mode.

Ethernet Connection


COMMAND ETHERNET

10 BASET

TERMINALMODE

ISDN-BRI Connections

ISDN1 & ISDN2 : MVP410ST/810STISDN3 & ISDN4: MVP810ST only

IS DN1 ISD N2 IS DN 3 ISDN4

? NETWORKMODE

ISDNTA

PBX

PSTN

NT1Device

*

* NT1 Device is neededif PBX has “U” interface.

Grounding Screw

Figure 3-15: ISDN/BRI Voip Connections for GND & RemoteConfiguration Modem



7. Turn on power to the MultiVOIP by placing the ON/OFF switch on the backpanel to the ON position. Wait for the Boot LED on the MultiVOIP to go offbefore proceeding. This may take a few minutes.



130

Cabling Procedure for MVP210Cabling involves connecting the MultiVOIP to your LAN and telephoneequipment.

1. For DID channels only. If both channels of your MVP210 MultiVOIP willbe using either FXS, FXO, or E&M telephony interfaces, skip to step 2.

For any channel on which you are using the DID interface type, you mustchange the jumper on the MultiVOIP circuit card.

a. Disconnect power. Unplug the AC power cord from the wall outlet orfrom the receptacle on the MultiVOIP unit.

b. Using a #1 Phillips driver, remove the screw (at bottom of unit, near theback-cover end) that attaches the main circuit card to the chassis of theMVP210.

c. Pull the main circuit card out about half way.


131

d. Identify the channels on which the DID interface will be used.

J3 J 7J9J5 J 11

F B 3

L E D 2

R7 2

J1 S 1 0

R 113R114

L E D 1 2

J 15

L E D 7L E D 11 L E D 10L E D1 4 LE D 5 LE D 3 L E D1

R58 R2R 57

L ED 6 LE D4LE D9

R56R74

R2 05

L ED1 3

R5 5

LE D8

Ch1 Ch2

Ch 2 JumperBlock

MVP210 Circuit Board

JP1

JP8

as configuredfor DID Interface

as shipped,for non-DID interfaces

Ch 1 JumperBlock

JP7

JP4

as configuredfor DID Interface

P7

Figure 3-16. MVP210 Channel Jumper Settings

e. Position the jumper for each DID channel so that it does not connect the twojumper posts. For DID operation of a voip channel, the MultiVOIP willwork properly if you simply remove the jumper altogether, but that isinadviseable because the jumper might be needed later if a differenttelephony interface is used for that voip channel.

f. Slide the main circuit card back into the MultiVOIP chassis and replace thescrew at the bottom of the unit.

2. Connect the power cord supplied with your MultiVOIP to the powerconnector on the back of the MultiVOIP and to a live AC outlet as shown inFigure 3-17.


132

Figure 3-17: Cabling for MVP210

3. Connect the MultiVOIP to a PC by using a RJ-45 (male) to DB-9 (female)cable. Plug the RJ-45 end of the cable into the Command port of theMultiVOIP and the other end into the PC serial port. See Figure 3-17.

4. Connect a network cable to the ETHERNET 10/100 connector on the backof the MultiVOIP. Connect the other end of the cable to your network.

5. For an FXS or FXO connection.(FXS Examples: analog phone, fax machine, Key Telephone System.)(FXO Examples: PBX extension, POTS line from telco central office.)Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXOconnector on the back of the MultiVOIP. Connect the other end to the device or phone jack.

For an E&M connection.(E&M Example: trunk line from telephone switch.)Connect one end of an RJ-45 phone cord to the Channel 1 E&M connectoron the back of the MultiVOIP.Connect the other end to the trunk line. Verify that the E&M Type in the E&M Options group of the Interface dialogbox is the same as the E&M trunk type supported by the telephone switch.See Appendix B for an E&M cabling pinout.

For a DID connection.(DID Example: DID fax system or DID voice phone lines.)Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO


133

connector on the back of the MultiVOIP. Connect the other end to the DID jack.

NOTE: DID lines are polarity sensitive. If, during testing, the DID linerings busy consistently, you will need to reverse the polarity ofone end of the connector (swap the connections of the wires tothe two middle pins of one RJ-11 connector).

6. Repeat the above step to connect the remaining telephone equipment to thesecond channel on your MultiVOIP.


This can be accomplished by connecting a grounding wire between thechassis and a metallic object that will provide an electrical ground.

8. Turn on power to the MultiVOIP by placing the ON/OFF switch on the backpanel to the ON position. Wait for the BOOT LED on the MultiVOIP to gooff before proceeding. This may take a few minutes.

Proceed to the Software Installation chapter to load the MultiVOIP software.


134

Cabling Procedure for MVP130

Power Connection

Network Connection

Ethernet CommandPower FXS/FXO


PBX

PSTNTelephony Connection

FXS FXO

Hub

Figure 3-18: Cabling for MVP130

Cabling involves connecting the MultiVOIP to your LAN and telephone equipment.

1. Connect the power cord supplied with your MultiVOIP to the powerconnector on the back of the MultiVOIP and to a live AC outlet as shown inFigure 3-18.

2. Connect the MultiVOIP to a PC by using a RJ-45 (male) to DB-9 (female)cable. Plug the RJ-45 end of the cable into the Command port of theMultiVOIP and the other end into the PC serial port. See Figure 3-18.

3. Connect a network cable to the ETHERNET 10/100 connector on the backof the MultiVOIP. Connect the other end of the cable to your network.

4. If you are connecting a station device such as an analog telephone, a faxmachine, or a Key Telephone System (KTS) (FXS interface), or a PBXextension (FXO interface) to your MultiVOIP, connect one end of an RJ-11phone cord to the Channel 1 FXS/FXO connector on the back MultiVOIPand the other end to the device or phone jack. You will define the interfacein the Interface dialog box in the software when you configure the unit.



110

Chapter 4: Software Installation

MultiVOIP User Guide Software Installation

111

IntroductionConfiguring software for your MultiVOIP entails three tasks:(1) loading the software onto the PC (this is “Software Installation and isdiscussed in this chapter),(2) setting values for telephony and IP parameters that will fit your system (thisis “Technical Configuration” and it is discussed in Chapter 5 for T1/E1MultiVOIP units and in Chapter 6 for analog MultiVOIP units), and(3) establishing “phonebooks” that contain the various dialing patterns forVOIP calls made to different locations (this is “Phonebook Configuration” andit is discussed in Chapters 7, 8, and 9 for T1, E1, and analog MultiVOIP unitsrespectively).

Loading MultiVOIP Software onto the PCThe software loading procedure does not present every screen or option in theloading process. It is assumed that someone with a thorough knowledge ofWindows and the software loading process is performing the installation.

The MultiVOIP software and User Guide are contained on the MultiVOIPproduct CD. Because the CD is auto-detectable, it will start up automaticallywhen you insert it into your CD-ROM drive. When you have finished loadingyour MultiVOIP software, you can view and print the User Guide by clickingon the View Manuals icon.

1. Be sure that your MultiVOIP has been properly cabled and that the power isturned on.


112

2. Insert the MultiVOIP CD into your CD-ROM drive. The CD should startautomatically. It may take 10 to 20 seconds for the Multi-Tech CDinstallation window to display.

If the Multi-Tech Installation CD window does not display 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 dialog box appears, click the InstallSoftware icon.


113

4. A ‘welcome’ screen appears.

Press Enter or click Next to continue.


114

5. Follow the on-screen instructions to install your MultiVOIP software. Thefirst screen asks you to choose the folder location of the files of theMultiVOIP software.

Choose a location and click Next.


115

6. At the next screen, you must select a program folder location for theMultiVOIP software program icon.

Click Next. Transient progress screens will appear while files are beingcopied.


116

7. On the next screen you can select the COM port that the command PCwill use when communicating with the MultiVoip unit. After softwareinstallation, the COM port can be re-set in the MultiVOIP Software (fromthe sidebar menu, select Connection | Settings to access the COM PortSetup screen or use the keyboard shortcut Ctrl + G).

NOTE: If the COM port setting madehere conflicts with the actual COM portresources available in the command PC,this error message will appear when theMultiVOIP program is launched. If thisoccurs, you must reset the COM port.


117

8. A completion screen will appear.

Click Finish.

9. When setup of the MultiVOIP software is complete, you will be prompted torun the MultiVOIP software to configure the VOIP.

Software installation is complete at this point. You may proceed withTechnical Configuration now or not, at your convenience.

Technical Configuration instructions are in the next two chapters of thismanual: Chapter 5 for T1/E1 MultiVOIP units and Chapter 6 for AnalogMultiVOIP units.


118

Un-Installing the MultiVOIP ConfigurationSoftware

1. To un-install the MultiVOIP configuration software, go to Start | Programsand locate the entry for the MultiVOIP program. Select Uninstall.


119

2. Two confirmation screens will appear. Click Yes and OK when you arecertain you want to continue with the uninstallation process.

3. A special warning message similar to that shown below may appearconcerning the MultiVOIP software’s “.bin” file. Click Yes.


120


Click Finish.

121

Chapter 5: Technical Configurationfor Digital T1/E1 MultiVOIPs

(MVP2410, MVP3010)

Technical Configuration (Digital Voips) MultiVOIP User Guide

122

Configuring the Digital T1/E1 MultiVOIPThere are two ways in which the MultiVOIP must be configured beforeoperation: technical configuration and phonebook configuration.

Technical Configuration. First, the MultiVOIP must be configured to operatewith technical parameter settings that will match the equipment with which itinterfaces. There are seven types of technical parameters that must be set.

These technical parameters pertain to(1) its operation in an IP network,(2) its operation with T1/E1 telephony equipment,(3) its transmission of voice and fax messages,(4) its interaction with SNMP (Simple Network Management Protocol)network management software (MultiVoipManager),(5) certain telephony attributes that are common to particular nations orregions,(6) its operation with a mail server on the same IP network (per SMTPparameters) such that log reports about VoIP telephone call traffic can be sentto the administrator by email,(7) implementing some common premium telephony features (Call Transfer,Call Hold, Call Waiting, Call ID – “Supplementary Services”), and(8) selecting the method by which log reports will be made accessible.

The process of specifying values for the various parameters in these sevencategories is what we call “technical configuration” and it is described in thischapter.

Phonebook Configuration. The second type of configuration that is requiredfor the MultiVOIP pertains to the phone number dialing sequences that it willreceive and transmit when handling calls. Both the PBX/telephony equipmentand the other VOIP devices that the MultiVOIP unit interacts with will affectdialing patterns. We call this “Phonebook Configuration,” and it is describedin Chapter 7: T1 Phonebook Configuration and Chapter 8: E1 PhonebookConfiguration of this manual. Chapter 2, the Quick Start Instructions, presentsadditional examples relevant to the T1/E1 voips.

Local/Remote Configuration. The MultiVOIP must be configured locally atfirst (to establish an IP address for the MultiVOIP unit). But changes to thisinitial configuration can be done either locally or remotely.

Local configuration is done through a connection between the “Command”port of the MultiVOIP and the COM port of the computer; the MultiVOIPconfiguration program is used.

Remote configuration is done through a connection between the MultiVOIP’sEthernet (network) port and a computer connected to the same network. Thecomputer could be miles or continents away from the MultiVOIP itself. Thereare two ways of doing remote configuration and operation of the MultiVOIP

MultiVOIP User Guide Technical Configuration (Digital Voips)

123

unit: (1) using the MultiVoipManager SNMP program, or (2) using theMultiVOIP web browser interface program.

MultiVoipManager. MultiVoipManager is an SNMP agent program (SimpleNetwork Management Protocol) that extends the capabilities of the MultiVOIPconfiguration program: MultiVoipManager allows the user to manage anynumber of VOIPs on a network, whereas the MultiVOIP configurationprogram can manage only the VOIP to which it is directly/locally connected.The MultiVoipManager can configure multiple VOIPs simultaneously,whereas the MultiVOIP configuration program can configure only one at atime.

MultiVoipManager may (but does not need to) reside on the same PC as theMultiVOIP configuration program. The MultiVoipManager program is on theMultiVOIP Product CD. Updates, when applicable, may be posted at on theMultiTech FTP site. To download, go to ftp://ftp.multitech.com/MultiVoip/.

Web Browser Interface. The MultiVOIP web browser GUI gives access tothe same commands and configuration parameters as are available in theMultiVOIP Windows GUI except for logging functions. When using the webbrowser GUI, logging can be done by email (the SMTP option).

Functional Equivalence of Interfaces. The MultiVOIP configurationprogram is required to do the initial configuration (that is, setting an IP addressfor the MultiVOIP unit) so that the VOIP unit can communicate with theMultiVoipManager program or with the web browser GUI. Management ofthe VOIP after that point can be done from any of these three programs sincethey all offer essentially the same functionality. Functionally, either theMultiVoipManager program or the web browser GUI can replace theMultiVOIP configuration program after the initial configuration is complete(with minor exceptions, as noted).

WARNING: Do not attempt to interface the MultiVOIP unit with twocontrol programs simultaneously (that is, by accessing theMultiVOIP configuration program via the Command Portand either the MultiVoipManager program or the webbrowser interface via the Ethernet Port). The results ofusing two programs to control a single VOIPsimultaneously would be unpredictable.

ftp://ftp.multitech.com/MultiVoip/


124

Local ConfigurationThis manual primarily describes local configuration with the Windows GUI.After IP addresses have been set locally using the Windows GUI, however,most aspects of configuration (logging functions are an exception) can behandled through the web browser GUI, as well (see the Operation andMaintenance chapter of this manual). In most aspects of configuration, theWindows GUI and web-browser GUI differ only graphically, not functionally.For information on SNMP remote configuration and management, see theMultiVoipManager documentation.

Pre-RequisitesTo complete the configuration of theMultiVOIP unit, you must know severalthings about the overall system.

Before configuring your MultiVOIP Gateway unit, you must know the valuesfor several IP and T1/E1 parameters that describe the IP network system andtelephony system (PBX or telco central office equipment) with which thedigital MultiVOIP will interact. If you plan to receive log reports on phonetraffic by email (SMTP), you must arrange to have an email address assignedto the VOIP unit on the email server on your IP network.

IP ParametersThe following parameters must be known about the network (LAN, WAN,Internet, etc.) to which the MultiVOIP will connect:




• IP Address

• IP Mask



125

Write down the values for these IP parameters. You will need to enter thesevalues in the “IP Parameters” screen in the Configuration section of theMultiVOIP software. You must have this IP information about every VOIP inthe system.

T1 Telephony Parameters (for MVP2410)The following parameters must be known about the PBX or telco central officeequipment to which the T1 MultiVOIP will connect:

➼ T1 Phone Parameters



T1 Telephony Parameters:Record for this VOIP Site

• Which frame format is used? ESF___ or D4___• Which CAS or PRI protocol is used? ______________• Clocking: Does the PBX or telco switch use


• Which line coding is used? AMI___ or B8ZS___

Write down the values for these T1 parameters. You will need to enter thesevalues in the “T1/E1 Parameters” screen in the Configuration section of theMultiVOIP software.


126

E1 Telephony Parameters (for MVP3010)The following parameters must be known about the PBX or telco central officeequipment to which the E1 MultiVOIP will connect:

➼ E1 Phone Parameters



E1 Telephony Parameters:Record for this VOIP Site

• Which frame format is used? Double Frame_____MultiFrame w/ CRC4_____

MultiFrame w/ CRC4 modified_____• Which CAS or PRI protocol is used? ______________• Clocking: Does the PBX or telco switch use


• Which line coding is used? AMI___ or HDB3___

• Pulse shape level?: (most commonly 0 to 40 meters)

Write down the values for these E1 parameters. You will need to enter thesevalues in the “T1/E1 Parameters” screen in the Configuration section of theMultiVOIP software.


127

SMTP Parameters (for email call log reporting)




Ask Mail Serveradministrator to set upemail account (withpassword) for theMultiVOIP unit itself. Besure to give a uniqueidentifier to eachindividual MultiVOIP unit..




[email protected]


128

Local Configuration Procedure (Summary)After the MultiVOIP configuration software has been installed in the‘Command’ PC (which is connected to the MultiVOIP unit), several steps mustbe taken to configure the MultiVOIP to function in its specific setting.Although the summary below includes all of these steps, some are optional.

1. Check Power and Cabling.

2. Start MultiVOIP Configuration Program.

3. Confirm Connection.

4. Solve Common Connection Problems.A. Fixing a COM Port Problem.B. Fixing a Cabling Problem.

5. Familiarize yourself with configuration parameter screens and how to accessthem.

6. Set IP Parameters.

7. Enable web browser GUI (optional).

8. Set Voice/Fax Parameters.

9. Set T1/E1 Parameters.

10. Set ISDN Parameters (if applicable).

11. Set SNMP Parameters (applicable if MultiVoipManager remotemanagement software is used).

12. Set Regional Parameters (Phone Signaling Tones & Cadences and setup forbuilt-in Remote Configuration/Command Modem).

13. Set Custom Tones and Cadences (optional).

14. Set SMTP Parameters (applicable if Log Reports are via Email).

15. Set Log Reporting Method (GUI, locally in MultiVOIP Configurationprogram; SNMP, remotely in MultiVoipManager program; or SMTP, viaemail).

16. Set Supplementary Services Parameters. The Supplementary Servicesscreen allows voip deployment of features that are normally found in PBX orPSTN systems (e.g., call transfer and call waiting).

17. Set Baud Rate (of COM port connection to ‘Command’ PC).

18. View System Information and set updating interval (optional).

19. Save the MultiVOIP configuration.

20. Create a User Default Configuration (optional).


129

Local Configuration Procedure (Detailed)You can begin the configuration process as a continuation of the MultiVOIPsoftware installation. You can establish your configuration or modify it at anytime by launching the MultiVOIP program from the Windows Start menu.

1. Check Power and Cabling. Be sure the MultiVOIP is turned on andconnected to the computer via the MultiVOIP’s Command Port (DB9connector at computer’s COM port; RJ45 connector at MultiVOIP).

You must allow the MultiVOIP to finish booting before you launch theMultiVOIP Configuration Program. The RED boot LED turns itself offwhen the booting process is completed.

2. Start MultiVOIP Configuration Program. Launch the MultiVOIPprogram from the Windows Start menu (from the folder location determinedduring installation).


130

3. Confirm Connection. If the MultiVOIP is set for an available COM portand is correctly cabled to the PC, the MultiVOIP main screen will appear.(If the main screen appears grayed out and seems inaccessible, go to step 4.)


131

In the lower left corner of the screen, the connection status of the MultiVOIPwill be displayed. The messages in the lower left corner will change asdetection occurs. The message “MultiVOIP Found” confirms that theMultiVOIP is in contact with the MultiVOIP configuration program. Skipto step 5.


132

4. Solving Common Connection Problems.

A. Fixing a COM Port Problem. If the MultiVOIP main screen appears but isgrayed out and seems inaccessible, the COM port that was specified for itscommunication with the PC is unavailable and must be changed. An errormessage will appear.

To change the COM port setting, use the COM Port Setup dialog box,which is accessible via the keyboard shortcut Ctrl + G or by going to theConnection pull-down menu and choosing “Settings.” In the “Select Port”field, select a COM port that is available on the PC. (If no COM ports arecurrently available, re-allocate COM port resources in the computer’s MSWindows operating system to make one available.)

Ctrl + G


133

4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by thecomputer, two error messages will appear (saying “Multi-VOIP Not Found”and “Phone Database Not Read”).

In this case, the MultiVOIP is simply disconnected from the network. Forinstructions on MultiVOIP cable connections, see the “Cabling” section ofChapter 3.

5. Configuration Parameter Groups: Getting Familiar, Learning AboutAccess. The first part of configuration concerns IP parameters, Voice/FAXparameters, T1/E1 parameters, SNMP parameters, Regional parameters, SMTPparameters, Supplementary Services parameters, Logs, and SystemInformation. In the MultiVOIP software, these seven types of parameters aregrouped together under “Configuration” and each has its own dialog box forentering values.

Generally, you can reach the dialog box for these parameter groups in one offour ways: pulldown menu, toolbar icon, keyboard shortcut, or sidebar..


134

6. Set IP Parameters. This dialog box can be reached by pulldown menu,toolbar icon, keyboard shortcut, or sidebar.

Accessing “IP Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + Alt + I


135

In each field, enter the values that fit your particular network.


136

The IP Parameters fields are described in the table below.

IP Parameter Definitions

Field Name Values Description

DiffServParameterfields

DiffServ PHB (Per Hop Behavior) valuespertain to a differential prioritizing systemfor IP packets as handled by DiffServ-compatible routers. There are 64 values, eachwith an elaborate technical description. Thesedescriptions are found in TCP/IP standardsRFC2474, RFC2597, and, for present purposes, inRFC3246, which describes the value 34 (34decimal; 22 hex) for Assured Forwardingbehavior (default for Call Control PHB) and thevalue 46 (46 decimal; 2E hexadecimal) forExpedited Forwarding behavior (default for VoipMedia PHB). Before using values other thanthese default values of 34 and 46, consult thesestandards documents and/or a qualified IPtelecommunications engineer.

To disable DiffServ, configure both fields to 0decimal.

The next page explains DiffServ in the context ofthe IP datagram.

Call ControlPHB

0 – 63

default = 34

Value is used toprioritize call setup IPpackets.

Voip MediaPHB

0 – 63default = 46

Value is used to prioritizethe RTP/RTCP audio IPpackets.

Frame Type Type II, SNAP Must be set to matchnetwork’s frame type.Default is Type II.


137

The IP Datagram with Header, Its Type-of-Service field, & DiffServ

bits =>0 4 8 16 19 24 31VERS HLEN TYPE OF

SERVICETOTAL LENGTH

IDENTIFICATION FLAGS FRAGMENT OFFSETTIME TO LIVE PROTOCOL HEADER CHECKSUM

SOURCE IP ADDRESSDESTINATION IP ADDRESS

IP OPTIONS (if any) PADDING …end of header

DATA…

The TOS field consists of eight bits, of which only the first six are used. These sixbits are called the “Differentiated Service Codepoint” or DSCP bits.

The Type of Service or “TOS” field

0 1 2 3 4 5 6 7PRECEDENCE D T R unused

The three precedence have eight values, 0-7, ranging from “normal” precedence(value of 0) to “network control” (value of 7). When set, the D bit requests low delay,the T bit requests high throughput, and the R bit requests high reliability.

Routers that support DiffServ can examine the six DSCP bits and prioritize the packetbased on the DSCP value. The DiffServ Parameters fields in the MultiVOIP IPParameters screen allow you to configure the DSCP bits to values supported by therouter. Specifically, the Voip Media PHB field relates to the prioritizing of audiopackets (RTP and RTCP packets) and the Call Control PHB field relates to theprioritzing of non-audio packets (packets concerning call set-up and tear-down,gatekeeper registration, etc.).

The MultiVOIP Call Control PHB parameter defaults to 34 decimal (22 hex; 100010binary – consider vis-à-vis TOS field above) for Assured Forwarding behavior. TheMultiVOIP Voip Media PHB parameter defaults to the value 46 decimal (2E hex;101110 binary – consider vis-à-vis TOS field above). To disable DiffServ, configureboth fields to 0 decimal.


138

IP Parameter Definitions (cont’d)


IP Parameter fieldsEnable DHCP Y/N

disabled by defaultDynamic Host ConfigurationProtocol is a method forassigning IP address and otherIP parameters to computers onthe IP network in a singlemessage with great flexibility.IP addresses can be static ortemporary depending on theneeds of the computer.

IP Address 4-places, 0-255 The unique LAN IP addressassigned to the MultiVOIP.

IP Mask 4-places, 0-255 Subnetwork address thatallows for sharing of IPaddresses within a LAN.

Gateway 4-places, 0-255 The IP address of the devicethat connects your MultiVOIPto the Internet.

DNS Parameter fieldsEnable DNS Y/N Enables Domain Name

Space/System functionwhere computer names areresolved using a worldwidedistributed database.

DNS Server IPAddress

4-places, 0-255. IP address of specific DNSserver to be used to resolveInternet computer names.

FTP Parameter fieldsFTP ServerEnable

Y/NSee “FTP ServerFile Transfers” inOperation &Maintenancechapter.

MultiVOIP unit has an FTPServer function so thatfirmware and otherimportant operatingsoftware files can betransferred to the voip viathe network.


139

7. Enable Web Browser GUI (Optional). After an IP address for theMultiVOIP unit has been established, you can choose to do any furtherconfiguration of the unit (a) by using the MultiVOIP web browser GUI, or (b)by continuing to use the MultiVOIP Windows GUI. If you want to doconfiguration work using the web browser GUI, you must first enable it. To doso, follow the steps below.

A. Set IP address of MultiVOIP unit using the MultiVOIP Configurationprogram (the Windows GUI).

B. Save Setup in Windows GUI.

C. Close Windows GUI.

D. Install Java program from MultiVOIP product CD (required on first useonly).

E. Open web browser.

F. Browse to IP address of MultiVOIP unit.

G. If username and password have been established, enter them when whenprompted.

H. Use web browser GUI to configure or operate MultiVOIP unit. Theconfiguration screens in the web browser GUI will have the same content astheir counterparts in the Windows GUI; only the graphic presentation will bedifferent.

For more details on enabling the MultiVOIP web GUI, see the “Web BrowserInterface” section of the Operation & Maintenance chapter of this manual.


140

8. Set Voice/FAX Parameters. This dialog box can be reached by pulldownmenu, toolbar icon, keyboard shortcut, or sidebar.

Accessing “Voice/FAX Parameters”

Pulldown Icon

Shortcut Sidebar

Ctrl + H


141



142

Note that Voice/FAX parameters are applied on a channel-by-channel basis.However, once you have established a set of Voice/FAX parameters for aparticular channel, you can apply this entire set of Voice/FAX parameters toanother channel by using the Copy Channel button and its dialog box. Tocopy a set of Voice/FAX parameters to all channels, select “Copy to All” andclick Copy.


143

The Voice/FAX Parameters fields are described in the tables below.

Voice/Fax Parameter DefinitionsField Name Values DescriptionDefault -- When this button is clicked, all

Voice/FAX parameters are set to theirdefault values.

SelectChannel

1-24 (T1)1-30 (E1)

Channel to be configured is selected here.

CopyChannel

-- Copies the Voice/FAX attributes of onechannel to another channel. Attributescan be copied to multiple channels or allchannels at once.

Voice Gain -- Signal amplification (or attenuation) indB.

Input Gain +31dB to –31dB

Modifies audio level entering voicechannel before it is sent over the networkto the remote VOIP. The default &recommended value is 0 dB.

Output Gain +31dB to –31dB

Modifies audio level being output to thedevice attached to the voice channel. Thedefault and recommended value is 0 dB.

DTMF ParametersDTMF Gain -- The DTMF Gain (Dual Tone Multi-

Frequency) controls the volume level ofthe digital tones sent out for Touch-Tonedialing.

DTMF Gain,High Tones

+3dB to-31dB &“mute”

Default value: -4 dB. Not to be changedexcept under supervision of MultiTech’sTechnical Support.

DTMF Gain,Low Tones




144

Voice/Fax Parameter Definitions (cont’d)Field Name Values Description

DTMF ParametersDuration(DTMF)

60 – 3000ms

When DTMF: Out of Band is selected,this setting determines how long eachDTMF digit ‘sounds’ or is held.Default = 100 ms.

DTMF In/Outof Band

Out ofBand, orInband

When DTMF Out of Band is selected(checked), the MultiVOIP detects DTMFtones at its input and regenerates them at itsoutput. When DTMF Inband is selected,the DTMF digits are passed through theMultiVOIP unit as they are received.

FAX ParametersFax Enable Y/N Enables or disables fax capability for a

particular channel.Max BaudRate(Fax, bps)

2400, 4800,7200, 9600,12000, 14400

Set to match baud rate of fax machineconnected to channel (see Fax machine’suser manual). Default = 14400 bps.

Fax VolumeDefault =-9.5 dB

-18.5 dBto –3.5 dB

Controls output level of fax tones. To bechanged only under the direction of Multi-Tech’s Technical Support.

Jitter Value(Fax)

Default =400 ms

Defines the inter-arrival packetdeviation (in milliseconds) for the faxtransmission. A higher value willincrease the delay, allowing a higherpercentage of packets to bereassembled. A lower value willdecrease the delay allowing fewerpackets to be reassembled.

Mode (Fax) FRF 11;T.38(T.38 notcurrentlysup-ported)

FRF11 is frame-relay FAX standardusing these coders: G.711, G.728,G.729, and G.723.1.T.38 is an ITU-T standard for storingand forwarding Faxes via email usingX.25 packets. It uses T.30 faxstandards and includes specialprovisions to preclude FAX timeoutsduring IP transmissions.


145

Voice/Fax Parameter Definitions (cont’d)Coder ParametersCoder Manual or

Auto-maticDetermines whether selection of coderis manual or automatic. WhenAutomatic is selected, the local andremote voice channels will negotiatethe voice coder to be used by selectingthe highest bandwidth coder supportedby both sides without exceeding theMax Bandwidth setting. G.723, G.729,or G.711 are negotiated.

SelectedCoder

G.711 a/ulaw 64kbps;G.726, @16/24/32/40 kbps;G.727, @nine bpsrates;G.723.1 @5.3 kbps,6.3 kbps;G.729,8kbps;Net [email protected], 7.2, 8,8.8, 9.6kbps

Select from a range of coders withspecific bandwidths. The higher the bpsrate, the more bandwidth is used. Thechannel that you are calling must havethe same voice coder selected.

Default = G.723.1 @ 6.3 kbps, asrequired for H.323. Here 64K of digitalvoice are compressed to 6.3K, allowingseveral simultaneous conversationsover the same bandwidth that wouldotherwise carry only one.

To make selections from the SelectedCoder drop-down list, the Manualoption must be enabled.

Maxbandwidth(coder)

11 – 128kbps

This drop-down list enables you toselect the maximum bandwidth allowedfor this channel. The Max Bandwidthdrop-down list is enabled only if theCoder is set to Automatic.

If coder selected automatically, thenenter a value for maximum bandwidth,as directed by VOIP administrator.


146


Advanced FeaturesSilenceCompression

Y/N Determines whether silence compression isenabled (checked) for this voice channel.

With Silence Compression enabled, theMultiVOIP will not transmit voice packetswhen silence is detected, thereby reducingthe amount of network bandwidth that isbeing used by the voice channel.Default = off.

EchoCancellation

Y/N Determines whether echo cancellation isenabled (checked) for this voice channel.

Echo Cancellation removes echo andimproves sound quality. Default = on.

ForwardErrorCorrection

Y/N Determines whether forward errorcorrection is enabled (checked) for thisvoice channel.

Forward Error Correction enablessome of the voice packets that werecorrupted or lost to be recovered. FEC addsan additional 50% overhead to the totalnetwork bandwidth consumed by the voicechannel. Default = Off


147


AutoCall/Offhook AlertParameters

Auto Call /OffhookAlert

AutoCall,OffhookAlert

The AutoCall option enables the localMultiVOIP to call a remote MultiVOIPwithout the user having to dial a PhoneDirectory Database number. As soon as youaccess the local MultiVOIP voice/faxchannel, the MultiVOIP immediatelyconnects to the remote MultiVOIPidentified in the Phone Number box of thisoption. This is essentially a hotline functionthat gives an immediate connection betweentwo points.

The Offhook Alert option is not supportedin T1/E1 MultiVOIP units.

AutoCall applies on a channel-by-channelbasis. It would not be appropriate for thisfunction to be applied to a channel thatserves in a pool of available channels forgeneral phone traffic. AutoCall requires anentry in the Outgoing phonebook of thelocal MultiVOIP and a matched setting inthe Inbound Phonebook of the remote voip.Also, especially for the AutoCall function,


148



OffhookAlert Timer

N/A Not supported in T1/E1 MultiVOIP units.

PhoneNumber

N/A Not supported in T1/E1 MultiVOIP units.


149

Voice/Fax Parameter Definitions (cont’d)


Dynamic Jitter BufferParameters

DynamicJitter Buffer

Dynamic Jitter defines a minimum anda maximum jitter value for voicecommunications. When receiving voicepackets from a remote MultiVOIP,varying delays between packets mayoccur due to network traffic problems.This is called Jitter. To compensate, theMultiVOIP uses a Dynamic JitterBuffer. The Jitter Buffer enables theMultiVOIP to wait for delayed voicepackets by automatically adjusting thelength of the Jitter Buffer betweenconfigurable minimum and maximumvalues. An Optimization Factoradjustment controls how quickly thelength of the Jitter Buffer is increasedwhen jitter increases on the network.The length of the jitter buffer directlyeffects the voice delay betweenMultiVOIP gateways.

The default minimum dynamic jitterbuffer of 60 milliseconds is theminimum delay that would beacceptable over a low jitter network.The default maximum dynamic jitterbuffer of 300 milliseconds is themaximum delay tolerable over a highjitter network.

MinimumJitter Value

60 to 400ms

The default minimum dynamic jitterbuffer of 60 milliseconds is theminimum delay that would beacceptable over a low jitter network.Default = 60 msec


150



Dynamic Jitter BufferParameters

MaximumJitter Value

60 to 400ms

The default maximum dynamic jitterbuffer of 300 milliseconds is themaximum delay tolerable over a highjitter network.Default = 300 msec

Optimizat-ionFactor

0 to 12 The Optimization Factor determineshow quickly the length of the DynamicJitter Buffer is changed based on actualjitter encountered on the network.Selecting the minimum value of 0means low voice delay is desired, butincreases the possibility of jitter-induced voice quality problems.Selecting the maximum value of 12means highest voice quality under jitterconditions is desired at the cost ofincreased voice delay.

Default = 7.

Modem RelayTo place modem traffic onto the voip network (an application called “modem relay”),use Coder G.711 mu-law at 64kbps.


151

Voice/Fax Parameter Definitions (cont’d) )


Auto DisconnectParameters

The Automatic Disconnection grouphas four options which can be usedsingly or in any combination.

Jitter Value 1-65535milli-seconds

The Jitter Value defines the averageinter-arrival packet deviation (inmilliseconds) before the call isautomatically disconnected. The defaultis 150 milliseconds. A higher valuemeans voice transmission will be moreaccepting of jitter. A lower value is lesstolerant of jitter.Inactive by default. When active,default = 150 ms. However, value mustequal or exceed Dynamic MinimumJitter Value.

Call Duration 1-65535seconds

Call Duration defines the maximumlength of time (in seconds) that a callremains connected before the call isautomatically disconnected.Inactive by default.When active, default = 180 sec.This may be too short for mostconfigurations requiring upwardadjustment.

ConsecutivePackets Lost

1-65535 Consecutive Packets Lost defines thenumber of consecutive packets that arelost after which the call is automaticallydisconnected.Inactive by default.When active, default = 30

NetworkDiscon-nection

1 to 65535seconds;Default =300 sec.

Specifies how long to wait beforedisconnecting the call when IP networkconnectivity with the remote site hasbeen lost.


152

9. Set T1/E1/ISDN Parameters. This dialog box can be reached by pulldownmenu, toolbar icon, keyboard shortcut, or sidebar.

Accessing “T1/E1/ISDN Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + T


153



154

T1 Parameters. The parameters applicable to T1 and their values are shownin the figure below. These T1 Parameter fields are described in the tables thatfollow.


155

T1 Parameter DefinitionsField Name Values DescriptionT1/E1/ISDN T1 North American standard.Long-HaulMode

Y/N In Long-Haul Mode, theMultiVOIP automaticallyrecovers received signals as low as–36 dB. The maximum reachablelength with 22 AWG cable is 2000meters. When Long-Haul Mode isdisabled, signals as low as –10 dBcan be received.Default: disabled.

CRC Check(CyclicRedundancyCheck)

Y/N When enabled, allows generationand checking of CRC bits. If notenabled, all check bits in thetransmit direction are set. Onlyapplies to ESF frame format.Default: enabled.

Frame Format F4, D4, ESF,SLC96

Frame Format of MultiVOIPshould match that used by PBXor telco. ESF and D4 arecommonly used.


156

T1 Parameter Definitions (cont’d)Field Name Values DescriptionCAS Protocol E&M Immed Strt

E&M Wink StartE&M Wink with

dial toneFXO Ground StrtFXO Loop StartFXS Ground StrtFXS Loop Start

Channel Associated Signaling(CAS) is a method of incorporatingtelephony signaling info into a T1voice/data stream. In CAS, thesignaling bits (the A, B, C, and Dbits) are multiplexed into the signalstream of each T1 channel. (Bycontrast, in Common ChannelSignaling (CCS), one channelhandles signaling for all otherchannels.) Each CAS protocoldefines the states of the signalingbits during the various stages of acall (IDLE, SEIZED, ANSWER,RING-ON, RING-OFF).

The CAS protocol code allows theVOIP to interact properly with thePBX or central-office switch that itserves. The need to downloadCAS protocols arises for only asmall minority of VOIP users, andonly when PBX/switch is found tobe incompatible with standardprotocols.

Match this parameter to the settingof PBX or central-office switch.


157

T1 Parameter Definitions (cont’d)Field Name Values DescriptionDetect FlashHook

Y/N This setting determines whether ornot the MultiVOIP responds tohook-flash signals.

Detection Time 100 – 1500milliseconds

Minimum hook-flash time that willbe interpreted as a valid flash bythe MultiVOIP.

GenerationTime

100 – 1500milliseconds

In some systems, a MultiVOIPmight receive a hook-flash signalfrom an upstream device (a PBX,voip or other device) and mustreplicate it to a downstream device.This parameter determines theduration of the hook-flash signalthat is passed to a downstreamdevice.


158

T1 Parameter Definitions (cont’d)ISDN Parameters

Field Name Values DescriptionEnableISDN-PRI

Y/N If digital connection is ISDN-PRItype, this box should be checked.When ISDN is enabled, the “CASProtocols” field is grayed out(ISDN has its own signalingmethod).

Terminal/Network

either “Terminal”or “Network”

When “Terminal” is selected, itindicates that the MultiVOIPshould emulate the subscriber(terminal) side of the digitalconnection. When “Network” isselected, it indicates that theMultiVOIP should emulate thecentral office (network) side of thedigital connection.Setting used for MultiVOIP mustbe opposite to the setting used inthe PBX. For example, if the PBXis set to “Terminal,” then theMultiVOIP must be set to“Network.”

Country see table, later thischapter

Country in which MultiVOIP isoperating with ISDN.

Operator see table, later thischapter

Indicates phone switchmanufacturer/model or refers totelco so as to specify the switchingsystem in question. ISDN isimplemented somewhat differentlyin different switches.

Note on Country& Operatoroptions.

__ [ISDN implementation options areshown, arranged by country, in atable below – soon after E1Parameter Definitions.]


159

T1 Parameter Definitions (cont’d)Field Name Values DescriptionLine Build Out 0 dB, -7.5 dB,

-15 dB, -22.5 dBTo reduce the crosstalk on receivedsignals, a transmit attenuator canbe placed in the data path.Transmit attenuation is selectable.Default: O dB

Pulse ShapeLevel

0 to 40 Meters40 to 81 m81 to 122 m122 to 162 m162 to 200 m

Refers to length of cablebetween MultiVOIP andPBX/telco in meters. Mostcommon will be 0 to 40m.

Clocking External/Internal Set opposite to telco/PBX setting.Example: if telco clocking internal,set VOIP clocking as external.

Line Coding AMI / B8ZS Match to PBX or telco.PCM Law A-Law/Mu-Law Match to PBX or telco. “

Mu-law” is analog-to-digitalcompression/expansionstandard used in North America.“A-law” is European standard.

Yellow AlarmFormat

Bit 2 / 1111… Depending on the Frame Formatused, there are choices of YellowAlarm format, as follows:D4: -Bit2 = 0 in every speech

channel-FS bit of frame 12 is forced toone.

ESF: -Bit2 = 0 in every speechchannel–1111111100000000 pattern indata link channel.

Check with your PBX/telcoadministrator for the correct settingor use the default value (1111 … ).


160

E1 Parameters. The parameters applicable to E1 and their values are shownin the figure below. These E1 Parameter fields are described in the tables thatfollow.


161

E1 Parameter DefinitionsField Name Values DescriptionT1/E1/ISDN E1 European standard.Long-HaulMode

Y/N In Long-Haul Mode, theMultiVOIP automaticallyrecovers received signals as low as–36 dB. The maximum reachablelength with 22 AWG cable is 2000meters. When Long-Haul Mode isdisabled, signals as low as –10 dBcan be received.Default: disabled.

CRC Check(CyclicRedundancyCheck)

-- Not applicable to E1.

Frame Format Double Frame;MultiFrame (withCRC4);MultiFrame(w/CRC4,modified)

Frame Format of MultiVOIPshould match that used by PBXor telco.


162

E1 Parameter Definitions (cont’d)Field Name Values DescriptionCAS Protocol E&M Immed Strt

E&M Wink StartE&M Wink with

dial toneFXO Ground StrtFXO Loop StartFXS Ground StrtFXS Loop StartMFR2ITUMFR2 ChinaMFR2 ANI

Channel Associated Signaling(CAS) is a method of incorporatingtelephony signaling info into an E1voice/data stream. In CAS, thesignaling bits (the A, B, C, and Dbits) are multiplexed into the signalstream of each E1 channel. (Bycontrast, in Common ChannelSignaling (CCS), one channelhandles signaling for all otherchannels.) Each CAS protocoldefines the states of the signalingbits during the various stages of acall (IDLE, SEIZED, ANSWER,RING-ON, RING-OFF).

The CAS protocol code allows theVOIP to interact properly with thePBX or central-office switch that itserves. The need to downloadCAS protocols arises for only asmall minority of VOIP users, andonly when PBX/switch is found tobe incompatible with standardprotocols.

Match this parameter to the settingof PBX or central-office switch.


163

E1 Parameter Definitions (cont’d)Field Name Values DescriptionDetect FlashHook

Y/N This setting determines whether ornot the MultiVOIP responds tohook-flash signals.

Detection Time 100 – 1500milliseconds

Minimum hook-flash time that willbe interpreted as a valid flash bythe MultiVOIP.

GenerationTime


In some systems, a MultiVOIPmight receive a hook-flash signalfrom an upstream device (a PBX,voip or other device) and mustreplicate it to a downstream device.This parameter determines theduration of the hook-flash signalthat is passed to a downstreamdevice.


164

E1 Parameter Definitions (cont’d)ISDN ParametersField Name Values DescriptionEnableISDN-PRI

Y/N If digital connection is ISDN-PRItype, this box should be checked.When ISDN is enabled, the “CASProtocols” field is grayed out(ISDN has its own signalingmethod).

Terminal/Network


When “Terminal” is selected, itindicates that the MultiVOIPshould emulate the subscriber(terminal) side of the digitalconnection. When “Network” isselected, it indicates that theMultiVOIP should emulate thecentral office (network) side of thedigital connection.Setting used for MultiVOIP mustbe opposite to the setting used inthe PBX. For example, if the PBXis set to “Terminal,” then theMultiVOIP must be set to“Network.”

Country see table, later thischapter

Country in which MultiVOIP isoperating with ISDN.

Operator see table, later thischapter

Indicates phone switchmanufacturer/model or refers totelco so as to specify the switchingsystem in question. ISDN isimplemented somewhat differentlyin different switches.

Note on Country& Operatoroptions.

__ [ISDN implementation options areshown, arranged by country, in atable below – soon after E1Parameter Definitions.]


165

E1 Parameter Definitions (cont’d)Field Name Values DescriptionLine Build Out 0 dB, -7.5 dB,

-15 dB, -22.5 dBTo reduce the crosstalk on receivedsignals, a transmit attenuator canbe placed in the data path.Transmit attenuation is selectable.Default: O dB

Pulse ShapeLevel

0 to 40 Meters40 to 81 m81 to 122 m122 to 162 m162 to 200 m

Refers to length of cablebetween MultiVOIP andPBX/telco in meters. Mostcommon will be 0 to 40m.

Clocking External/Internal Set opposite to telco/PBX setting.Example: if telco clocking internal,set VOIP clocking as external.

Line Coding AMI / HDB3 Match to PBX or telco.PCM Law A-Law/Mu-Law Match to PBX or telco.

“A-law” is analog-to-digitalcompression/expansionstandard used in Europe.“Mu-law” is North Americanstandard.


166

10. Set ISDN Parameters (if applicable). These parameters are acces-siblein the T1/E1/ISDN Parameters screen. If your T1 or E1 phone line is aPrimary Rate Interface ISDN line, enable ISDN-PRI and set it for theparticular implementation of ISDN that your telco uses. The ISDN typessupported by the digital MultiVOIP units (at press time) are listed below,organized by country.


167

11. Set SNMP Parameters (Remote Voip Management). This dialog box canbe reached by pulldown menu, keyboard shortcut, or sidebar. To make theMultiVOIP controllable by a remote PC running the MultiVoipManagersoftware, check the “Enable SNMP Agent” box on the SNMP Parametersscreen.

Accessing “SNMP Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + M


168

In each field, enter the values that fit your particular system.


169

The SNMP Parameter fields are described in the table below.

SNMP Parameter Definitions


Enable SNMPAgent

Y/N Enables the SNMP code in the firmwareof the MultiVOIP. This must beenabled for the MultiVOIP tocommunicate with and be controllableby theMultiVoipManager software.Default: disabled

Trap Manager Parameters

Address 4 places; n.n.n.nn = 0-255

IP address of MultiVoipManager PC.

CommunityName

-- A “community” is a group of VOIPendpoints that can communicate witheach other. Often “public” is used todesignate a grouping where all end usershave access to entire VOIP network.However, calling permissions can beconfigured to restrict access as needed.

Port Number 162 The default port number of the SNMPmanager receiving the traps is thestandard port 162.

CommunityName 1

Length = 19characters (max.)Case sensitive.

First community grouping.

Permissions Read-Only,

Read/Write

If this community needs to changeMultiVOIP settings, select Read/Write.Otherwise, select Read-Only to viewsettings.

CommunityName 2


Second community grouping


Read/Write



170

12. Set Regional Parameters (Phone Signaling Tones & Cadences and setupfor built-in Remote Configuration/Command Modem).This dialog box can be reached by pulldown menu, keyboard shortcut, orsidebar.

Accessing “Regional Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + R


171

The Regional Parameters screen will appear. For the country selected, thestandard set of frequency pairs will be listed for dial tone, busy tone,‘unobtainable’ tone (fast busy or trunk busy), and ring tone.

Remote Configuration/Command Modem. In the Country Selection forBuilt-In Modem field (drop-down list), select the country that best fits yoursituation. This may not be the same as your selection for theCountry/Region field. The selections in the Country Selection for Built-In Modem field entail more detailed groupings of telephony parameters thando the Country/Region values.



172

The Regional Parameters fields are described in the table below.

“Regional Parameter” Definitions


Country/Region

USA, Japan, UK,Custom

Name of a country or region that uses acertain set of tone pairs for dial tone,ring tone, busy tone, unobtainabletone (fast busy tone), survivabilitytone (tone heard briefly, 2 seconds,after going offhook denotingsurvivable mode of VOIP unit), andre-order tone (a tone patternindicating the need for the user to hangup the phone).In some cases, the tone-pair schemedenoted by a country name may alsobe used outside of that country. The“Custom” option (button) assures thatany tone-pairing scheme worldwidecan be accommodated.

Type column dial tone,ring tone,busy tone,unobtainable tone(fast busy), re-order tone.

Type of telephony tone-pair for whichfrequency, gain, and cadence are beingpresented.


173

“Regional Parameter” Definitions


Frequency 1 frequency in Hertz Lower frequency of pair.Frequency 2 frequency in Hertz Higher frequency of pair.

Gain 1 gain in dB+3dB to –31dBand “mute” setting

Amplification factor of lowerfrequency of pair.This figure describes amplification thatthe MultiVOIPapplies to outbound tones entering theMultiVOIP atthe T1 port. Default: -16dB


Amplification factor of higherfrequency of pair.This figure describes amplification thatthe MultiVOIPapplies to outbound tones entering theMultiVOIP atthe T1 port. Default: -16dB


174

“Regional Parameter” Definitions (cont’d)


Cadence(msec) On/Off

n/n/n/nfour integer timevalues inmilli-seconds; zerovalue for dial-toneindicatescontinuous tone

On/off pattern of tone durations usedto denote phone ringing, phone busy,connection unobtainable (fast busy),and dial tone (continuous anddescribed as “0“). Default valuesdiffer for different countries/regions.Although most cadences have only twoparts (an “on” duration and an “off”duration), some telephony cadenceshave four parts. Most cadences, then,are expressed as two iterations of atwo-part sequence. Although this isredundant, it is necessary to allow forexpression of 4-part cadences.

Custom (button) -- Click on the “Custom” button to bringup the Custom Tone Pair Settingsscreen. This screen allows the user tospecify tone pair attributes that are notfound in any of the standardnational/regional telephony toningschemes.

CountrySelection forBuilt-In Modem(not applicableto MVP130)

country name MultiVOIP units operating with theX.06 software release (and above)include a built-in modem. Theadministrator can dial into this modemto configure the MultiVOIP unitremotely. The country name values inthis field set telephony parameters thatallow the modem to work in the listedcountry. This value may be differentthan the Country/Region value. Forexample, a user may need to choose“Europe” as the Country/Region valuebut “Denmark” as the Country-Selection-for-Built-In-Modem value.


175

13. Set Custom Tones and Cadences (optional) . The Regional Parametersdialog box has a secondary dialog box that allows you to customize DTMFtone pairs to create unique ring-tones, dial tones, busy-tones “unobtainable”tones (fast busy signal) or “re-order” tones (telling the user that they musthang up an off-hook phone) for your system. This screen allows the user tospecify tone-pair attributes that are not found in any of the standardnational/regional telephony toning schemes. To access this customizationfeature, click on the Custom button on the Regional Parameters screen.


176

The Custom Tone-Pair Settings fields are described in the table below.

Custom Tone-Pair Settings DefinitionsField Name Values DescriptionTone Pair dial tone

busy tonering tone,‘unobtainable’ &re-order tones

Identifies the type of telephonysignaling tone for which frequenciesare being specified.

TONE PAIR VALUES About Defaults: US telephonyvalues are used as defaults on thisscreen. However, since this dialogbox is provided to allow customtone-pair settings, default values areessentially irrelevant.

Frequency 1 frequency in Hertz Frequency of lower tone of pair.This outbound tone pair enters theMultiVOIP at the T1/E1 port.

Frequency 2 frequency in Hertz Frequency of higher tone of pair.This outbound tone pair enters theMultiVOIP at the T1/E1 port.


Amplification factor of lowerfrequency of pair. This figuredescribes amplification that theMultiVOIP applies to outboundtones entering the MultiVOIP atthe T1 port. Default = -16dB


Amplification factor of higherfrequency of pair. This figuredescribes amplification that theMultiVOIP applies to outboundtones entering the MultiVOIP atthe T1 port. Default = -16dB


177

Custom Tone-Pair Settings Definitions


Cadence 1 integer time valueinmilli-seconds; zerovalue for dial-toneindicatescontinuous tone

On/off pattern of tone durationsused to denote phone ringing, phonebusy, connection unobtainable tone(fast busy), dial tone (which iscontinuous and described as “0“) &re-order tone. Cadence 1 is durationof first period of tone being “on” inthe cadence of the telephony signal(which could be ring-tone, busy-tone, unobtainable tone, dial tone, orre-order tone).

Cadence 2 duration inmilliseconds

Cadence 2 is duration of first “off”period in signaling cadence.


Cadence 3 is duration of second“on” period in signaling cadence.


Cadence 4 is duration of second“off” period in the signalingcadence, after which the 4-partcadence pattern of the telephonysignal repeats.


178

14. Set SMTP Parameters (Log Reports by Email). The SMTP Parametersscreen is applicable when the VOIP administrator has chosen to receive logreports by email (this is done by selecting the “SMTP” checkbox in theOthers screen and selecting “Enable SMTP” in the SMTP Parametersscreen.). The SMTP Parameters screen can be reached by pulldown menu,keyboard shortcut, or sidebar.

Accessing “SMTP Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + Alt + S

MultiVOIP as Email Sender. When SMTP is used, the MultiVOIP willactually be given its own email account (with Login Name and Password) onsome mail server connected to the IP network. Using this account, theMultiVOIP will then send out email messages containing log reportinformation. The “Recipient” of the log report email is ordinarily the VoIPadministrator. Because the MultiVOIP cannot receive email, a “Reply-To”address must also be set up. Ordinarily, the “Reply-To” address is that of atechnician who has access to the mail server or MultiVOIP or both, and theVoIP administrator might also be designated as the “Reply-To” party. Themain function of the Reply-To address is to receive error or failure messagesregarding the emailed reports.


179

The SMTP Parameters screen is shown below.

“SMTP Parameters” DefinitionsField Name Values DescriptionEnable SMTP Y/N In order to send log reports by email,

this box must be checked. However, toenable SMTP functionality, you mustalso select “SMTP” in the Logs screen.

Login Name alpha-numeric,per emaildomain

This is the User Name for theMultiVOIP unit’s email account.

Password alpha-numeric Login password for MultiVOIP unit’semail account.

Mail Server IPAddress

n.n.n.nfor n= 0 to 255

This mail server must be accessible onthe IP network to which the MultiVOIPis connected.

Port Number 25 25 is a standard port number for SMTP.


180

......“SMTP Parameters” Definitions (cont’d)


Mail Type text or html Mail type in which log reports will besent.

Subject text User specified. Subject line that willappear for all emailed log reports forthis MultiVOIP unit.

Reply-ToAddress

email address User specified. This email addressfunctions as a source email identifier forthe MultiVOIP, which, of course,cannot usefully receive email messages.The Reply-To address provides adestination for returned messagesindicating the status of messages sent bythe MultiVOIP (esp. to indicate whenlog report email was undeliverable orwhen an error has occurred).

RecipientAddress

email address User specified. Email address at whichVOIP administrator will receive logreports.

Mail Criteria Criteria for sending log summary byemail.The log summary email will be sent outeither when the user-specified numberof log messages has accumulated, oronce every day or multiple days, whichever comes first.

Number ofRecords

integer This is the number of log records thatmust accumulate to trigger the sendingof a log-summary email.

Number of Days integer This is the number of days that mustpass before triggering the sending of alog-summary email.


181

The SMTP Parameters dialog box has a secondary dialog box, CustomFields, that allows you to customize email log messages for the MultiVOIP.The MultiVOIP software logs data about many aspects of the call traffic goingthrough the MultiVOIP. The Custom Fields screen lets you pick which aspectswill be included in the email log reports.

“Custom Fields” DefinitionsField Description Field DescriptionSelect All Log report to

include all fieldsshown.

ChannelNumber

Data channelcarrying call.

Start Date,Time

Date and time the phonecall began.

Duration Length of call. Call Mode Voice or fax.PacketsSent

Total packets sent incall.

PacketsReceived

Total packets receivedin call.

BytesSent

Total bytes sent incall.

BytesReceived

Total bytes received incall.

PacketsLost

Packets lost in call. Coder Voice Coder/Compression Rate usedfor call will be listed inlog.


182

“Custom Fields” Definitions (cont’d)

Field Description Field DescriptionOutboundDigits

Digits put out byMultiVOIP onto theT1 or E1 line.

PrefixMatched

When selected, thephonebook prefixmatched in processingcall will be listed in log.

CallStatus

Successful orunsuccessful.

From Details To DetailsGatewayNumber

Originating gateway Gatew N. Completing orterminating gateway

IP Addr IP address wherecall originated.

IP Addr IP address where callwas completed orterminated.

Descript Identifier of sitewhere calloriginated.

Descript Identifier of site wherecall was completed orterminated.

Options When selected, logwill not use/non-useof SilenceCompression andForward ErrorCorrection by calloriginator.

Options When selected, log willnot use/non-use ofSilence Compressionand Forward ErrorCorrection by callterminator.


183


184

15. Set Log Reporting Method. The Logs screen lets you choose how theVoIP administrator will receive log reports about the MultiVOIP’sperformance and the phone call traffic that is passing through it. Log reportscan be received in one of three ways:

A. in the MultiVOIP program (GUI),

B. via email (SMTP), or

C. at the MultiVoipManager remote voip system managementprogram (SNMP).

Accessing “Logs” ScreenPulldown Icon

Shortcut Sidebar

Ctrl + Alt + O

If you enable console messages, you can customize the types of messages tobe included/excluded in log reports by clicking on the “Filters” button andusing the Console Messages Filter Settings screen (see subsequent page).If you use the logging function, select the logging option that applies to your


185

VoIP system design. If you intend to use a SysLog Server program forlogging, click in that Enable check box. The common SysLog logical portnumber is 514. If you intend to use the MultiVOIP web browser GUI forconfiguration and control of MultiVOIP units, be aware that the web browserGUI does not support logs directly. However, when the web browser GUI isused, log files can still be sent to the voip administrator via email (whichrequires activating the SMTP logging option in this screen).

“Logs” Screen DefinitionsField Name Values DescriptionEnable ConsoleMessages

Y/N Allows MultiVOIP debugging messages tobe read via a basic tele-communicationsprogram like HyperTerminal ™ or similarapplication. Normally, this should bedisabled because it consumers MultiVOIPpro-cessing resources. Console messages aremeant for use by tech support personnel.


186

“Logs” Screen Definitions (cont’d)Field Name Values DescriptionFilters (button) Click to access secondary screen on where

console messages can be included/excludedby category and on a per-channel basis. (Seethe Console Messages Filter Settings screenon subsequent page.)

Turn Off Logs Y/N Disables log reporting function.Logs Buttons Only one of these three log reporting

methods, GUI, SMTP, or SNMP, may bechosen.

GUI Y/N User must view logs at the MultiVOIPconfiguration program.

SNMP Y/N Log messages will be delivered to theMultiVoipManager application program.

SMTP Y/N Log messages will be sent to user-specifiedemail address.

SysLog ServerEnable

Y/N This box must be checked if logging is to bedone in conjunction with a SysLog Serverprogram. For more on SysLog Server, seeOperation & Maintenance chapter.

IP Address n.n.n.nfor n=0-255

IP address of computer, connected to voipnetwork, on which SysLog Server program isrunning.

Port 514 Logical port for SysLog Server. 514 iscommonly used.

Online StatisticsUpdationInterval

integer Set the interval (in seconds) at which logginginformation will be updated.


187

To customize console messages by category and/or by channel, click on“Filters” and use the Console Messages Filters Settings screen.


188

16. Set Supplementary Services Parameters. This dialog box can bereached by pulldown menu, keyboard shortcut, or sidebar.

Accessing “Supplementary Services Parameters”

Pulldown Icon

Shortcut Sidebar

Ctrl + Alt +H

Supplementary Services features derive from the H.450 standard, which bringsto voip telephony functionality once only available with PSTN or PBXtelephony. Supplementary Services features can be used under H.323 only andnot under SIP.


189


Of the features implemented under Supplementary Services, three are veryclosely related: Call Transfer, Call Hold, and Call Waiting. Call NameIdentification is similar but not identical to the premium PSTN featurecommonly known as Caller ID.

Call Transfer. Call Transfer allows one party to re-connect the party withwhom they have been speaking to a third party. The first party isdisconnected when the third party becomes connected. Feature is invokedby a programmable phone keypad sequence (for example, #7).

Call Hold. Call Hold allows one party to maintain an idle (non-talking)connection with another party while receiving another call (Call Waiting),while initiating another call (Call Transfer), or while performing someother call management function. Invoked by keypad sequence.

Call Waiting. Call Waiting notifies an engaged caller of an incoming calland allows them to receive a call from a third party while the party withwhom they have been speaking is put on hold. Invoked by keypadsequence.

Call Name Identification. When enabled for a given voip unit (the ‘home’voip), this feature gives notice to remote voips involved in calls.Notification goes to the remote voip administrator, not to individual phonestations. When the home voip is the caller, a plain English descriptor willbe sent to the remote (callee) voip identifying


190

the channel over which the call is being originated (for example, “CallingParty - Omaha Sales Office Line 2”). If that voip channel is dedicated to acertain individual, the descriptor could say that, as well (for example“Calling Party - Harold Smith in Omaha”). When the home voip receives acall from any remote voip, the home voip sends a status message back tothat caller. This message confirms that the home voip’s phone channel iseither busy or ringing or that a connection has been made (for example,“Busy Party - Omaha Sales Office Line ”). These messages appear in theStatistics – Call Progress screen of the remote voip.

Note that Supplementary Services parameters are applied on a channel-by-channel basis. However, once you have established a set of supplementaryparameters for a particular channel, you can apply this entire set of parametersto another channel by using the Copy Channel button and its dialog box. Tocopy a set of Supplementary Services parameters to all channels, select “Copyto All” and click Copy.


191

The Supplementary Services fields are described in the tables below.

Supplementary Services Parameter Definitions


SelectChannel

1-2 (210);1-4 (410);1-8 (810)

The channel to be configured is selectedhere.

Call TransferEnable

Y/N Select to enable the Call Transfer functionin the voip unit.This is a “blind” transfer and the sequenceof events is as follows:Callers A and B are having aconversation.Caller A wants to put B into contact withC.Caller A dials call transfer sequence.Caller A hears dial tone and dials numberfor caller C.Caller A gets disconnected while Caller Bgets connected to caller C.A brief musical jingle is played for thecaller on hold.

TransferSequence

any phonekeypadcharacter

The numbers and/or symbols that thecaller must press on the phone keypad toinitiate a call transfer.The call-transfer sequence can be 1 to 4characters in length using anycombination of digits or characters(* or #).The sequences for call transfer, call hold,and call waiting can be from 1 to 4 digitsin length consisting of any combinationof digits 1234567890*#.


192

Supplementary Services Definitions (cont’d)Field Name Values Description

Call HoldEnable

Y/N Select to enable Call Hold function invoip unit.Call Hold allows one party to maintain anidle (non-talking) connection with anotherparty while receiving another call (CallWaiting), while initiating another call(Call Transfer), or while performing someother call management function.

HoldSequence

phonekeypadcharacters

The numbers and/or symbols that thecaller must press on the phone keypad toinitiate a call hold.The call-hold sequence can be 1 to 4characters in length using anycombination of digits or characters(* or #).

Call WaitingEnable

Y/N Select to enable Call Waiting function invoip unit.

RetrieveSequence

phonekeypadcharacters,twocharactersin length

The numbers and/or symbols that thecaller must press on the phone keypad toinitiate retrieval of a waiting call.The call-waiting retrieval sequence can be1 to 4 characters in length using anycombination of digits or characters(* or #).This is the phone keypad sequence that auser must press to retrieve a waiting call.Customize-able. Sequence should bedistinct from sequence that might be usedto retrieve a waiting call via the PBX orPSTN.


193


Call NameIdentificationEnable

Enables CNI function. Call NameIdentification is not the same as Caller ID.When enabled on a given voip unitcurrently being controlled by theMultiVOIP GUI (the ‘home voip’), CallName Identification sends an identifierand status information to theadministrator of the remote voip involvedin the call. The feature operates on achannel-by-channel basis (each channelcan have a separate identifier).If the home voip is originating the call,only the Calling Party field isapplicable. If the home voip is receivingthe call, then the Alerting Party, BusyParty, and Connected Party fields arethe only applicable fields (and any or allof these could be enabled for a given voipchannel). The status information confirmsback to the originator that the callee (thehome voip) is either busy, or ringing, orthat the intended call has been completedand is currently connected.The identifier and status information aremade available to the remote voip unitand appear in the Caller ID field of itsStatistics – Call Progress screen. (This ishow MultiVOIP units handle CNImessages; in other voip brands, H.450may be implemented differently and thenthe message presentation may vary.)


194


Calling Party,AllowedName Type(CNI)

If the ‘home’ voip unit is originating thecall and Calling Party is selected, thenthe identifier (from the Caller Id field)will be sent to the remote voip unit beingcalled. The Caller Id field gives theremote voip administrator a plain-language identifier of the party that isoriginating the call occurring on a specificchannel.

This field is applicable only when the ‘home’voip unit is originating the call.

Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip in thisexample), Call Name Identification hasbeen enabled, Calling Party has beenenabled as an Allowed Name Type, and“Omaha Sales Office Voipchannel 2” hasbeen entered in the Caller Id field.When channel 2 of the Omaha voip isused to make a call to any other voipphone station (for example, the Denveroffice), the message“Calling Party - Omaha Sales OfficeVoipchannel 2” will appear in the “CallerId” field of theStatistics - Call Progress screenof the Denver voip.


195


AlertingParty,AllowedName Type(CNI)

If the ‘home’ voip unit is receiving thecall and Alerting Party is selected, thenthe identifier (from the Caller Id field)will tell the originating remote voip unitthat the call is ringing.

This field is applicable only when the ‘home’voip unit is receiving the call.

Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip unit inthis example), Call Name Identificationhas been enabled, Alerting Party hasbeen enabled as an Allowed Name Type,and “Omaha Sales Office Voipchannel 2”has been entered in the Caller Id field ofthe Supplementary Services screen.

When channel 2 of the Omaha voipreceives a call from any other voip phonestation (for example, the Denver office),the message “Alerting Party - OmahaSales Office Voipchannel 2” will be sentback and will appear in the Caller Id fieldof the Statistics – Call Progress screenof the Denver voip. This confirms to theDenver voip that the phone is ringing inOmaha.


196


Busy Party,AllowedName Type(CNI)

If the ‘home’ voip unit is receiving a calldirected toward an already engagedchannel or phone station and Busy Partyis selected, then the identifier (from theCaller Id field) will tell the originatingremote voip unit that the channel or calledparty is busy.


Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip unit inthis example), Call Name Identificationhas been enabled, Busy Party has beenenabled as an Allowed Name Type, and“Omaha Sales Office Voipchannel 2” hasbeen entered in the Caller Id field of theSupplementary Services screen.

When channel 2 of the Omaha voip isbusy but still receives a call attempt fromany other voip phone station (forexample, the Denver office), the message“Busy Party - Omaha Sales OfficeVoipchannel 2” will be sent back and willappear in the Caller Id field of theStatistics – Call Progress screen of theDenver voip. This confirms to the Denvervoip that the channel or phone station isbusy in Omaha.


197


ConnectedParty,AllowedName Type(CNI)

If the ‘home’ voip unit is receiving a calland Connected Party is selected, then theidentifier (from the Caller Id field) willtell the originating remote voip unit thatthe attempted call has been completed andthe connection is made.


Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip unit inthis example), Call Name Identificationhas been enabled, Connected Party hasbeen enabled as an Allowed Name Type,and “Omaha Sales Office Voipchannel 2”has been entered in the Caller Id field ofthe Supplementary Services screen.

When channel 2 of the Omaha voipcompletes an attempted call from anyother voip phone station (for example, theDenver office), the message “ConnectedParty - Omaha Sales Office Voipchannel2” will be sent back and will appear in theCaller Id field of the Statistics – CallProgress screen of the Denver voip. Thisconfirms to the Denver voip that the callhas been completed to Omaha.


198


Caller ID This is the identifier of a specific channelof the ‘home’ voip unit. The Caller Idfield typically describes a person, office,or location, for example, “Harry Smith,”or “Bursar’s Office,” or “BarnesvilleFactory.”

Default -- When this button is clicked, allSupplementary Service parameters are setto their default values.

CopyChannel

-- Copies the Supplementary Serviceattributes of one channel to anotherchannel. Attributes can be copied tomultiple channels or all channels at once.


199

17. Set Baud Rate. The Connection option in the sidebar menu has a “Settings”item that includes the baud-rate setting for the COM port of the computer running theMultiVOIP software.

First, it is important to note that the default COM port established by theMultiVOIP program is COM1. Do not accept the default value until youhave checked the COM port allocation on your PC. To do this, check forCOM port assignments in the system resource dialog box(es) of yourWindows operating system. If COM1 is not available, you must change theCOM port setting to COM2 or some other COM port that you haveconfirmed as being available on your PC.

The default baud rate is 115,200 bps.


200

18. View System Information screen and set updating interval (optional).

This dialog box can be reached by pulldown menu, keyboard shortcut, orsidebar.

Accessing the “System Information” Screen

Pulldown Icon

Shortcut Sidebar

Ctrl + Alt +Y


201

This screen presents vital system information at a glance. Its primary use is introubleshooting.

System Information Parameter Definitions


Boot Version nn.nn Indicates the version of the code that isused at the startup (booting) of the voip.The boot code version is independent ofthe software version.

Mac Address alpha-numeric

Denotes the number assigned as the voipunit’s unique Ethernet address.

Up Time days:hours:mm:ss

Indicates how long the voip has beenrunning since its last booting.

FirmwareVersion

alpha-numeric

Indicates version of MultiVOIP firmware.


202

The frequency with which the System Information screen is updated isdetermined by a setting in the Logs screen

19. Saving the MultiVOIP Configuration. When values have been set for allof the MultiVOIP’s various operating parameters, click on Save Setup in thesidebar.


203

20. Creating a User Default Configuration. When a “Setup” (completegrouping of parameters) is being saved, you will be prompted aboutdesignating that setup as a “User Default” setup. A User Default setup maybe useful as a baseline of site-specific values to which you can easily revert.Establishing a User Default Setup is optional.

205

Chapter 6: Technical Configurationfor Analog/BRI MultiVOIPs

(MVP130, MVP-210,MVP-410, MVP-810

& MVP-410ST/810ST)

Technical Configuration (Analog/BRI) MultiVOIP User Guide

304

Configuring the Analog/BRI MultiVOIPThere are two ways in which the MultiVOIP must be configured beforeoperation: technical configuration and phonebook configuration.

Technical Configuration. First, the MultiVOIP must be configured to operatewith technical parameter settings that will match the equipment with which itinterfaces. There are eight types of technical parameters that must be set.

These technical parameters pertain to(1) its operation in an IP network,(2) its operation with telephony equipment,(3) its transmission of voice and fax messages,(4) its interaction with SNMP (Simple Network Management Protocol)network management software (MultiVoipManager),(5) certain telephony attributes that are common to particular nations orregions,(6) its operation with a mail server on the same IP network (per SMTPparameters) such that log reports about VoIP telephone call traffic can be sentto the administrator by email,(7) implementing some common premium telephony features (Call Transfer,Call Hold, Call Waiting, Call ID – “Supplementary Services”), and(8) selecting the method by which log reports will be made accessible.

The process of specifying values for the various parameters in these sevencategories is what we call “technical configuration” and it is described in thischapter.

Phonebook Configuration. The second type of configuration that is requiredfor the MultiVOIP pertains to the phone number dialing sequences that it willreceive and transmit when handling calls. Dialing patterns will be affected byboth the PBX/telephony equipment and the other VOIP devices that theMultiVOIP unit interacts with. We call this “Phonebook Configuration,” and,for analog MultiVOIP units, it is described nominally in Chapter 9: AnalogPhonebook Configuration of this manual. But, in fact, nearly all of thedescriptions and examples for analog phonebook configuration are to be foundin Chapter 7 if the analog voip is operating under the North Americantelephony scheme, or in Chapter 8 if the analog voip is operating under aEuropean telephony scheme. Chapter 2, the Quick Start Instructions, presentsadditional examples relevant to the analog voips.

Local/Remote Configuration. The MultiVOIP must be configured locally atfirst (to establish an IP address for the MultiVOIP unit). But changes to thisinitial configuration can be done either locally or remotely.

MultiVOIP User Guide Technical Configuration (Analog/BRI)

305

Local configuration is done through a connection between the “Command”port of the MultiVOIP and the COM port of the computer; the MultiVOIPconfiguration program is used.

Remote configuration is done through a connection between the MultiVOIP’sEthernet (network) port and a computer connected to the same network. Thecomputer could be miles or continents away from the MultiVOIP itself. Thereare two ways of doing remote configuration and operation of the MultiVOIPunit: (1) using the MultiVoipManager SNMP program, or (2) using theMultiVOIP web browser interface program.

MultiVoipManager. MultiVoipManager is an SNMP agent program (SimpleNetwork Management Protocol) that extends the capabilities of the MultiVOIPconfiguration program: MultiVoipManager allows the user to manage anynumber of VOIPs on a network, whereas the MultiVOIP configurationprogram can manage only the VOIP to which it is directly/locally connected.The MultiVoipManager can configure multiple VOIPs simultaneously,whereas the MultiVOIP configuration program can configure only one at atime.

MultiVoipManager may (but does not need to) reside on the same PC as theMultiVOIP configuration program. The MultiVoipManager program is on theMultiVOIP Product CD. Updates, when applicable, may be posted at on theMultiTech FTP site. To download, go to ftp://ftp.multitech.com/MultiVoip/.

Web Browser Interface. The MultiVOIP web browser GUI gives access tothe same commands and configuration parameters as are available in theMultiVOIP Windows GUI except for logging functions. When using the webbrowser GUI, logging can be done by email (the SMTP option).

ftp://ftp.multitech.com/MultiVoip/


306

Functional Equivalence of Interfaces. The MultiVOIP configurationprogram is required to do the initial configuration (that is, setting an IP addressfor the MultiVOIP unit) so that the VOIP unit can communicate with theMultiVoipManager program or with the web browser GUI. Management ofthe VOIP after that point can be done from any of these three programs sincethey all offer essentially the same functionality. Functionally, either theMultiVoipManager program or the web browser GUI can replace theMultiVOIP configuration program after the initial configuration is complete(with minor exceptions, as noted).

WARNING: Do not attempt to interface the MultiVOIP unit with twocontrol programs simultaneously (that is, by accessing theMultiVOIP configuration program via the Command Portand either the MultiVoipManager program or the webbrowser interface via the Ethernet Port). The results ofusing two programs to control a single VOIPsimultaneously would be unpredictable.


307

Local ConfigurationThis manual primarily describes local configuration with the Windows GUI.After IP addresses have been set locally using the Windows GUI, most aspectsof configuration (logging functions are an exception) can be handled throughthe web browser GUI, as well (see the Operation and Maintenance chapter ofthis manual). In most aspects of configuration, the Windows GUI and web-browser GUI differ only graphically, not functionally. For information onSNMP remote configuration and management, see the MultiVoipManagerdocumentation.

Pre-RequisitesTo complete the configuration of theMultiVOIP unit, you must know severalthings about the overall system.

Before configuring your MultiVOIP Gateway unit, you must know the valuesfor several IP and telephone parameters that describe the IP network systemand telephony system (PBX or telco central office equipment) with which thedigital MultiVOIP will interact. If you plan to receive log reports on phonetraffic by email (SMTP), you must arrange to have an email address assignedto the VOIP unit on the email server on your IP network.

IP ParametersThe following parameters must be known about the network (LAN, WAN,Internet, etc.) to which the MultiVOIP will connect:




• IP Address

• IP Mask



308

Write down the values for these IP parameters. You will need to enter thesevalues in the “IP Parameters” screen in the Configuration section of theMultiVOIP software. You must have this IP information about every VOIP inthe system.

Analog Telephony Interface Parameters(for MVP130/210/410/810)

The following parameters must be known about the PBX or telco central officeequipment to which the analog MultiVOIP will connect:

➼ Analog Phone Parameters


Needed for:MVP810MVP410MVP210MVP130

Analog Telephony Interface Parameters:Record for this VOIP Site

• Which interface type (or “signaling”) is used? E&M_____ FXS/FXO_____

• If FXS, determine whether the line will be used for aphone, fax, or KTS (key telephone system)

• If FXO, determine if line will be an analog PBXextension or an analog line from a telco central office

• If E&M, determine these aspects of the E&M trunkline from the PBX:• What is its Type (1, 2, 3, 4, or 5)?• Is it 2-wire or 4-wire?• Is it Dial Tone or Wink?


309

ISDN-BRI Telephony Parameters(for MVP-410ST/810ST)

The following parameters must be known about the PBX or telco central officeequipment to which the analog MultiVOIP will connect:

➼ ISDN-BRI Phone Parameters


Needed for:MVP810STMVP410ST

ISDN-BRI Telephony Interface Parameters:Record them for this VOIP Site

• In which country is this voip installed?

• Which operator (switch type) is used?

• What type of line coding use required,A-law or u-law?

• Determine which BRI ports will be network side andwhich BRI ports will be terminal side.

Write down the values for these telephony parameters (whether analog orISDN-BRI). You will need to enter these values in the “Interface” screen(analog) or “ISDN Parameters” screen (ISDN-BRI) in the Configurationsection of the MultiVOIP software.


310

SMTP Parameters (for email call log reporting)




Ask Mail Serveradministrator to set upemail account (withpassword) for theMultiVOIP unit itself. Besure to give a uniqueidentifier to eachindividual MultiVOIP unit..




[email protected]


311

Local Configuration Procedure (Summary)After the MultiVOIP configuration software has been installed in the‘Command’ PC (which is connected to the MultiVOIP unit), several steps mustbe taken to configure the MultiVOIP to function in its specific setting.Although the summary below includes all of these steps, some are optional.

1. Check Power and Cabling.

2. Start MultiVOIP Configuration Program.

3. Confirm Connection.

4. Solve Common Connection Problems.

A. Fixing a COM Port Problem.

B. Fixing a Cabling Problem.

5. Familiarize yourself with configuration parameter screens and how to accessthem.

6. Set IP Parameters.

7. Enable web browser GUI (optional).

8. Set Voice/Fax Parameters.

9. Set Telephony Interface Parameters (analog) or ISDN Parameters(ISDN/BRI).

10. Set SNMP Parameters (applicable if MultiVoipManager remotemanagement software is used).

11. Set Regional Parameters (Phone Signaling Tones & Cadences and setup forbuilt-in Remote Configuration/Command Modem).

12. Set Custom Tones and Cadences (optional).

13. Set SMTP Parameters (applicable if Log Reports are via Email).

14. Set Log Reporting Method (GUI, locally in MultiVOIP Configurationprogram; SNMP, remotely in MultiVoipManager program; or SMTP, viaemail).

15. Set Supplementary Services Parameters. The Supplementary Servicesscreen allows voip deployment of features that are normally found in PBX orPSTN systems (e.g., call transfer and call waiting).

16. Set Baud Rate (of COM port connection to ‘Command’ PC).

17. View System Info screen and set updating interval (optional).

18. Save the MultiVOIP configuration.


312

19. Create a User Default Configuration (optional).

When technical configuration is complete, you will need to configure theMultiVOIP’s phonebooks (for all models) and its embedded gatekeeperfunctionality, if present (for MVP-210G, -410G, and 810G only). This manualhas separate chapters describing T1 Phonebook Configuration for North-American-influenced telephony settings and E1 Phonebook Configuration forEuro-influenced telephony settings, as well as a separate EmbeddedGatekeeper chapter.

Local Configuration Procedure (Detailed)You can begin the configuration process as a continuation of the MultiVOIPsoftware installation. You can establish your configuration or modify it at anytime by launching the MultiVOIP program from the Windows Start menu.

1. Check Power and Cabling. Be sure the MultiVOIP is turned on andconnected to the computer via the MultiVOIP’s Command Port (DB9connector at computer’s COM port; RJ45 connector at MultiVOIP).

2. Start MultiVOIP Configuration Program. Launch the MultiVOIPprogram from the Windows Start menu (from the folder location determinedduring installation).


313

3. Confirm Connection. If the MultiVOIP is set for an available COM portand is correctly cabled to the PC, the MultiVOIP main screen will appear.(If the main screen appears grayed out and seems inaccessible, go to step 4.)


314

In the lower left corner of the screen, the connection status of the MultiVOIPwill be displayed. The messages in the lower left corner will change asdetection occurs. The message “MultiVOIP Found” confirms that theMultiVOIP is in contact with the MultiVOIP configuration program. Skipto step 5.


315

4. Solving Common Connection Problems. .

A. Fixing a COM Port Problem. If the MultiVOIP main screen appears but isgrayed out and seems inaccessible, the COM port that was specified for itscommunication with the PC is unavailable and must be changed. An errormessage will appear.

To change the COM port setting, use the COM Port Setup dialog box,which is accessible via the keyboard shortcut Ctrl + G or by going to theConnection pull-down menu and choosing “Settings.” In the “Select Port”field, select a COM port that is available on the PC. (If no COM ports arecurrently available, re-allocate COM port resources in the computer’s MSWindows operating system to make one available.)

Ctrl + G


316

4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by thecomputer, two error messages will appear (saying “Multi-VOIP Not Found”and “Phone Database Not Read”).

In this case, the MultiVOIP is simply disconnected from the network. Forinstructions on MultiVOIP cable connections, see the Cabling section ofChapter 3.

5. Configuration Parameter Groups: Getting Familiar, Learning AboutAccess. The first part of configuration concerns IP parameters, Voice/FAXparameters, Telephony Interface parameters, SNMP parameters, Regionalparameters, SMTP parameters, Supplementary Services parameters, Logs, andSystem Information. In the MultiVOIP software, these seven types ofparameters are grouped together under “Configuration” and each has its owndialog box for entering values.

Generally, you can reach the dialog box for these parameter groups in one offour ways: pulldown menu, toolbar icon, keyboard shortcut, or sidebar. ..


317

6. Set IP Parameters. This dialog box can be reached by pulldown menu,toolbar icon, keyboard shortcut, or sidebar.

Accessing “IP Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + Alt + I


318



319

The IP Parameters fields are described in the table below.

IP Parameter DefinitionsField Name Values DescriptionDiffServParameterfields

DiffServ PHB (Per Hop Behavior) valuespertain to a differential prioritizing systemfor IP packets as handled by DiffServ-compatible routers. There are 64 values, eachwith an elaborate technical description. Thesedescriptions are found in TCP/IP standardsRFC2474, RFC2597, and, for present purposes, inRFC3246, which describes the value 34 (34decimal; 22 hex) for Assured Forwardingbehavior (default for Call Control PHB) and thevalue 46 (46 decimal; 2E hexadecimal) forExpedited Forwarding behavior (default for VoipMedia PHB). Before using values other thanthese default values of 34 and 46, consult thesestandards documents and/or a qualified IPtelecommunications engineer.

To disable DiffServ, configure both fields to 0decimal.

The next page explains DiffServ in the context ofthe IP datagram.

Call ControlPHB

0 – 63

default = 34

Value is used toprioritize call setup IPpackets.

Voip MediaPHB

0 – 63default = 46

Value is used to prioritizethe RTP/RTCP audio IPpackets.

Frame Type Type II, SNAP Must be set to matchnetwork’s frame type.Default is Type II.


320

The IP Datagram with Header, Its Type-of-Service field, & DiffServ

bits =>0 4 8 16 19 24 31VERS HLEN TYPE OF

SERVICETOTAL LENGTH

IDENTIFICATION FLAGS FRAGMENT OFFSETTIME TO LIVE PROTOCOL HEADER CHECKSUM

SOURCE IP ADDRESSDESTINATION IP ADDRESS

IP OPTIONS (if any) PADDING …end of header

DATA…

The TOS field consists of eight bits, of which only the first six are used. These sixbits are called the “Differentiated Service Codepoint” or DSCP bits.

The Type of Service or “TOS” field

0 1 2 3 4 5 6 7PRECEDENCE D T R unused

three precedence have eight values, 0-7, ranging from “normal” precedence (value of0) to “network control” (value of 7). When set, the D bit requests low delay, the T bitrequests high throughput, and the R bit requests high reliability.

Routers that support DiffServ can examine the six DSCP bits and prioritize the packetbased on the DSCP value. The DiffServ Parameters fields in the MultiVOIP IPParameters screen allow you to configure the DSCP bits to values supported by therouter. Specifically, the Voip Media PHB field relates to the prioritizing of audiopackets (RTP and RTCP packets) and the Call Control PHB field relates to theprioritzing of non-audio packets (packets concerning call set-up and tear-down,gatekeeper registration, etc.).

The MultiVOIP Call Control PHB parameter defaults to 34 decimal (22 hex; 100010binary – consider vis-à-vis TOS field above) for Assured Forwarding behavior. TheMultiVOIP Voip Media PHB parameter defaults to the value 46 decimal (2E hex;101110 binary – consider vis-à-vis TOS field above). To disable DiffServ, configureboth fields to 0 decimal.


321



IP Parameter fieldsEnable DHCP Y/N

disabled by defaultDynamic Host ConfigurationProtocol is a method forassigning IP address and otherIP parameters to computers onthe IP network in a singlemessage with great flexibility.IP addresses can be static ortemporary depending on theneeds of the computer.

IP Address 4-places, 0-255 The unique LAN IP addressassigned to the MultiVOIP.

IP Mask 4-places, 0-255 Subnetwork address thatallows for sharing of IPaddresses within a LAN.

Gateway 4-places, 0-255. The IP address of thedevice that connects yourMultiVOIP to the Internet.

Enable DNS Y/N.(feature not yetimplemented; forfuture use)

Enables Domain NameSpace/System functionwhere computer names areresolved using a worldwidedistributed database.

DNS Parameter fieldsEnable DNS Y/N

Default = disabledEnables Domain NameSpace/System functionwhere computer names areresolved using a worldwidedistributed database.

DNS Server IPAddress

4-places, 0-255. IP address of specific DNSserver to be used to resolveInternet computer names.


322



FTP Parameter fieldsFTP ServerEnable

Y/NDefault = disabledSee “FTP ServerFile Transfers” inOperation &Maintenancechapter.

MultiVOIP unit has an FTPServer function so thatfirmware and otherimportant operatingsoftware files can betransferred to the voip viathe network.

FTP Parameter fieldsTDM Routing Y/N;

enabled by defaultAllows calls placedbetween ports on thesame MultiVOIP voicechannel board to berouted over internalTime Division Multiplexbus without conversionto IP. TDM routingeffectively eliminates thedelay introduced by IPconversion.If you require all calls tobe IP routed, disable the“use TDM Routing forIntra-Gateway Calls”option. Since this is notnormally required, wegenerally recommendleaving TDM Routingenabled.


323

7. Enable Web Browser GUI (Optional). After an IP address for theMultiVOIP unit has been established, you can choose to do any furtherconfiguration of the unit (a) by using the MultiVOIP web browser GUI, or (b)by continuing to use the MultiVOIP Windows GUI. If you want to doconfiguration work using the web browser GUI, you must first enable it. To doso, follow the steps below.

A. Set IP address of MultiVOIP unit using the MultiVOIP Configurationprogram (the Windows GUI).

B. Save Setup in Windows GUI.

C. Close Windows GUI.

D. Install Java program from MultiVOIP product CD (on first use only).

E. Open web browser.

F. Browse to IP address of MultiVOIP unit.

G. If username and password have been established, enter them when whenprompted.

H. Use web browser GUI to configure or operate MultiVOIP unit. Theconfiguration screens in the web browser GUI will have the same content astheir counterparts in the Windows GUI; only the graphic presentation will bedifferent.

For more details on enabling the MultiVOIP web GUI, see the “Web BrowserInterface” section of the Operation & Maintenance chapter of this manual.


324

8. Set Voice/FAX Parameters. This dialog box can be reached by pulldownmenu, toolbar icon, keyboard shortcut, or sidebar.

Accessing “Voice/FAX Parameters”

Pulldown Icon

Shortcut Sidebar

Ctrl + H


325



326

Note that Voice/FAX parameters are applied on a channel-by-channel basis.However, once you have established a set of Voice/FAX parameters for aparticular channel, you can apply this entire set of Voice/FAX parameters toanother channel by using the Copy Channel button and its dialog box. Tocopy a set of Voice/FAX parameters to all channels, select “Copy to All” andclick Copy.


327

The Voice/FAX Parameters fields are described in the tables below.

Voice/Fax Parameter DefinitionsField Name Values DescriptionDefault -- When this button is clicked, all

Voice/FAX parameters are set to theirdefault values.

SelectChannel

1-2 (210)1-4 (410)1-8 (810)

Channel to be configured is selected here.

CopyChannel

-- Copies the Voice/FAX attributes of onechannel to another channel. Attributescan be copied to multiple channels or allchannels at once.

Voice Gain -- Signal amplification (or attenuation) indB.

Input Gain +31dB to –31dB

Modifies audio level entering voicechannel before it is sent over the networkto the remote VOIP. The default &recommended value is 0 dB.

Output Gain +31dB to –31dB

Modifies audio level being output to thedevice attached to the voice channel. Thedefault and recommended value is 0 dB.

DTMF ParametersDTMF Gain -- The DTMF Gain (Dual Tone Multi-

Frequency) controls the volume level ofthe DTMF tones sent out for Touch-Tonedialing.

DTMF Gain,High Tones



DTMF Gain,Low Tones




328


DTMF ParametersDuration(DTMF)

60 – 3000ms

When DTMF: Out of Band is selected, thissetting determines how long each DTMFdigit ‘sounds’ or is held. Default = 100 ms.Not supported in 5.02c BRI software.

DTMF In/Outof Band

Out ofBand, orInband

When DTMF Out of Band is selected, theMultiVOIP detects DTMF tones at its inputand regenerates them at its output. WhenDTMF Inband is selected, the DTMFdigits are passed through the MultiVOIPunit as they are received. In 502c BRIsoftware, “DTMF Out of Band” can bechecked or unchecked.

FAX ParametersFax Enable Y/N Enables or disables fax capability for a

particular channel.Max BaudRate(Fax)

2400, 4800,7200, 9600,12000,14400 bps

Set to match baud rate of fax machineconnected to channel (see Fax machine’suser manual).Default = 14400 bps.

Fax Volume(Default =-9.5 dB )

-18.5 dBto –3.5 dB

Controls output level of fax tones. To bechanged only under the direction of Multi-Tech’s Technical Support.

Jitter Value(Fax)

Default =400 ms

Defines the inter-arrival packet deviation (inmilliseconds) for the fax transmission. Ahigher value will increase the delay,allowing a higher percentage of packets tobe reassembled. A lower value willdecrease the delay allowing fewer packets tobe reassembled.

Mode (Fax) FRF 11;T.38(T.38 notcurrentlysup-ported)

FRF11 is frame-relay FAX standard using thesecoders: G.711, G.728, G.729, G.723.1.T.38 is an ITU-T standard for storing andforwarding FAXes via email using X.25packets. It uses T.30 fax standards andincludes special provisions to preclude FAXtimeouts during IP transmissions.


329

Voice/Fax Parameter Definitions (cont’d)Coder ParametersCoder Manual or

Auto-maticDetermines whether selection of coderis manual or automatic. WhenAutomatic is selected, the local andremote voice channels will negotiatethe voice coder to be used by selectingthe highest bandwidth coder supportedby both sides without exceeding theMax Bandwidth setting. G.723, G.729,or G.711 are negotiated.

SelectedCoder

G.711 a/ulaw 64kbps;G.726, @16/24/32/40 kbps;G.727, @nine bpsrates;G.723.1 @5.3 kbps,6.3 kbps;G.729,8kbps;Net [email protected], 7.2, 8,8.8, 9.6kbps

Select from a range of coders withspecific bandwidths. The higher the bpsrate, the more bandwidth is used. Thechannel that you are calling must havethe same voice coder selected.

Default = G.723.1 @ 6.3 kbps, asrequired for H.323. Here 64K ofdigital voice are compressed to 6.3K,allowing several simultaneousconversations over the same bandwidththat would otherwise carry only one.

To make selections from the SelectedCoder drop-down list, the Manualoption must be enabled.

Maxbandwidth(coder)

11 – 128kbps

This drop-down list enables you toselect the maximum bandwidth allowedfor this channel. The Max Bandwidthdrop-down list is enabled only if theCoder is set to Automatic.If coder is to be selected automatically(“Auto” setting), then enter a value formaximum bandwidth.


330


Advanced FeaturesSilenceCompression

Y/N Determines whether silence compression isenabled (checked) for this voice channel.

With Silence Compression enabled, theMultiVOIP will not transmit voice packetswhen silence is detected, thereby reducingthe amount of network bandwidth that isbeing used by the voice channel.Default = on.

EchoCancellation

Y/N Determines whether echo cancellation isenabled (checked) for this voice channel.

Echo Cancellation removes echo andimproves sound quality. Default = on.

ForwardErrorCorrection

Y/N Determines whether forward errorcorrection is enabled (checked) for thisvoice channel.

Forward Error Correction enablessome of the voice packets that werecorrupted or lost to be recovered. FEC addsan additional 50% overhead to the totalnetwork bandwidth consumed by the voicechannel. Default = Off


331

Voice/Fax Parameter Definitions (cont’d) Field Name Values Description




The AutoCall option enables the localMultiVOIP to call a remote MultiVOIPwithout the user having to dial a PhoneDirectory Database number. As soon as youaccess the local MultiVOIP voice/faxchannel, the MultiVOIP immediatelyconnects to the remote MultiVOIPidentified in the Phone Number box of thisoption.

If the “Pass Through Enable” field ischecked in the Interface Parameters screen,AutoCall must be used.

The Offhook Alert option applies only toFXS channels.

The Offhook Alert option works like this:if a phone goes offhook and yet no numberis dialed within a specific period of time (asset in the Offhook Alert Timer field), thenthat phone will automatically dial the Alertphone number for the voip channel. (TheAlert phone number must be set in theVoice/Fax Parameters | Phone Numberfield; if the voip system is working withouta gatekeeper unit, there must also be amatching phone number entry in theOutbound Phonebook.). One use of thisfeature would be for emergency use where auser goes off hook but does not dial,possibly indicating a crisis situation. TheOffhook Alert feature uses the InterceptTone, as listed in the Regional Parametersscreen. This tone will be outputted on thephone that was taken off hook but that didnot dial. The other end of the connectionwill hear audio from the “crisis” end as is itwould during a normal phone call.


332

Voice/Fax Parameter Definitions (cont’d) Field Name Values Description




(continued from previous page)

Both functions apply on a channel-by-channel basis. It would not be appropriatefor either of these functions to be applied toa channel that serves in a pool of availablechannels for general phone traffic. Eitherfunction requires an entry in the Outgoingphonebook of the local MultiVOIP and amatched setting in the Inbound Phonebookof the remote voip.

GenerateLocal DialTone

Y/N Used for AutoCall only. If selected, dialtone will be generated locally while the callis being established between gateways. Thecapability to generate dial tone locallywould be particularly useful when there is alengthy network delay.


333



OffhookAlert Timer

0 – 3000seconds

The length of time that must elapse beforethe offhook alert is triggered and a call isautomatically made to the phone numberlisted in the Phone Number field.

PhoneNumber

-- Phone number used for Auto Call functionor Offhook Alert Timer function. Thisphone number must correspond to an entryin the Outbound Phonebook of the localMultiVOIP and in the Inbound Phonebookof the remote MultiVOIP (unless agatekeeper unit is used in the voip system).


334

Voice/Fax Parameter Definitions (cont’d) )Field Name Values Description

Dynamic JitterDynamicJitter Buffer

Dynamic Jitter defines a minimum anda maximum jitter value for voicecommunications. When receiving voicepackets from a remote MultiVOIP,varying delays between packets mayoccur due to network traffic problems.This is called Jitter. To compensate, theMultiVOIP uses a Dynamic JitterBuffer. The Jitter Buffer enables theMultiVOIP to wait for delayed voicepackets by automatically adjusting thelength of the Jitter Buffer betweenconfigurable minimum and maximumvalues. An Optimization Factoradjustment controls how quickly thelength of the Jitter Buffer is increasedwhen jitter increases on the network.The length of the jitter buffer directlyeffects the voice delay betweenMultiVOIP gateways.

MinimumJitter Value

60 to 400ms

The minimum dynamic jitter buffer of60 milliseconds is the minimum delaythat would be acceptable over a lowjitter network.Default = 150 msec


335



Dynamic Jitter

MaximumJitter Value

60 to 400ms

The maximum dynamic jitter buffer of400 milliseconds is the maximum delaytolerable over a high jitter network.Default = 300 msec

Optimizat-ionFactor

0 to 12 The Optimization Factor determineshow quickly the length of the DynamicJitter Buffer is changed based on actualjitter encountered on the network.Selecting the minimum value of 0means low voice delay is desired, butincreases the possibility of jitter-induced voice quality problems.Selecting the maximum value of 12means highest voice quality under jitterconditions is desired at the cost ofincreased voice delay.

Default = 7.

Modem RelayTo place modem traffic onto the voip network (an application called “modem relay”),use Coder G.711 mu-law at 64kbps.


336

Voice/Fax Parameter Definitions (cont’d) )


Auto Disconnect

AutomaticDisconnect-ion

-- The Automatic Disconnection groupprovides four options which can beused singly or in any combination.

Jitter Value 1-65535milli-seconds

The Jitter Value defines the averageinter-arrival packet deviation (inmilliseconds) before the call isautomatically disconnected. The defaultis 300 milliseconds. A higher valuemeans voice transmission will be moreaccepting of jitter. A lower value is lesstolerant of jitter.Inactive by default. When active,default = 300 ms. However, value mustequal or exceed Dynamic MinimumJitter Value.

Call Duration 1-65535seconds

Call Duration defines the maximumlength of time (in seconds) that a callremains connected before the call isautomatically disconnected.Inactive by default.When active, default = 180 sec.This may be too short for mostconfigurations, requiring upwardadjustment.

ConsecutivePackets Lost

1-65535 Consecutive Packets Lost defines thenumber of consecutive packets that arelost after which the call is automaticallydisconnected.Inactive by default.When active, default = 30

NetworkDiscon-nection

1 to 65535seconds;Default =30 sec.

Specifies how long to wait beforedisconnecting the call when IP networkconnectivity with the remote site hasbeen lost.


337

9a. (Analog VOIPs). Set Telephony Interface Parameters. This dialog boxcan be reached by pulldown menu, toolbar icon, keyboard shortcut, or sidebar.

Accessing Telephony Interface ParametersPulldown Icon

Shortcut Sidebar

Ctrl + I


338


The kinds of parameters for which values must be chosen depend on the typeof telephony supervisory signaling or interface used (FXO, E&M, etc.). Wepresent here the various parameters grouped and organized by interface type.


339

Note that Interface parameters are applied on a channel-by-channel basis.However, once you have established a set of Interface parameters for aparticular channel, you can apply this entire set of Voice/FAX parameters toanother channel by using the Copy Channel button and its dialog box. Tocopy a set of Interface parameters to all channels, select “Copy to All” andclick Copy.


340

FXS Loop Start Parameters. The parameters applicable to FXS Loop Startare shown in the figure below and described in the table that follows.

FXS Loop Start Interface: Parameter DefinitionsField Name Values DescriptionFXS (LoopStart)

Y/N Enables FXS Loop Start interfacetype.


341

FXS Loop Start Interface: Parameter Definitions (cont’d)Field Name Values Description

Dialing Options fields

Inter DigitTimer

1 - 10 seconds This is the length of time that theMultiVOIP will wait betweendigits. When the time expires, theMultiVOIP will look in theoutbound phonebook for thenumber entered and place the callaccordingly.Default = 2.

MessageWaitingIndication

-- Not applicable to FXS Loop Startinterface

Inter DigitRegenerationTime

in milliseconds The length of time between theoutputting of DTMF digits.Default = 100 ms.

Ring Count,FXS

1-99 Maximum number of rings that theMultiVOIP will issue before givingup the attempted call.

FXS Options,Current Loss

Y/N When enabled, the MultiVOIP willinterrupt loop current in the FXScircuit to initiate a disconnection.This tells the device connected tothe FXS port to hang up. TheMulti-VOIP cannot drop the call;the FXS device must go on hook.


342


Flash Hook Options fields

Generation -- not applicable to FXS interface

Detection Range for Min. and Max.,50 - 1500milliseconds

For a received flash hook to beregarded as such by theMultiVOIP, its duration must fallbetween the minimum andmaximum values given here.

Pass ThroughEnable

Y/N When enabled, this parametercreates an open audio path throughthe MultiVOIP.

If the Pass-Through feature isenabled, the AutoCall feature mustbe enabled for this voip channel inthe Voice/Fax Parameters screen.

Caller ID fields

Caller ID Type Bellcore The MultiVOIP currently supportsonly one implementation of CallerID. That implementation isBellcore type 1 with Caller IDplaced between the first and secondrings of the call.

Caller ID enable Y/N Caller ID information is adescription of the remote callingparty received by the called party.The description has three parts:name of caller, phone number ofcaller, and time of call. The ‘time-of-call’ portion is always generatedby the receiving MultiVOIP unit(on FXS channel) based on its dateand time setup.


343


Caller ID fields

Caller ID enable(cont’d)

Y/N The forms of the ‘Caller Name’and ‘Caller Phone Number’ differdepending on the IP transmissionprotocol used (H.323, SIP, or SPP)and upon entries in the phonebookscreens of the remote (CIDgenerating) voip unit. The CIDName and Number appearing onthe phone at the terminating FXSend will come either from a centraloffice switch (showing a PSTNphone number), or the phonebookof the remote (CID sending) voipunit.

The Caller ID feature has dependencies on both the telco central office and theMultiVOIP phone book. See discussion after the FXO Parameters sectionbelow.


344

FXO Parameters. The parameters applicable to the FXO telephony interfacetype are shown in the figure below and described in the table that follows.


345

FXO Interface: Parameter DefinitionsField Name Values DescriptionInterface, FXO Y/N Enables FXO functionality

Dialing OptionsRegeneration Pulse, DTMF Determines whether digits

generated and sent out will bepulse tones or DTMF.

Inter DigitTimer

1 to 10 seconds This is the length of time that theMultiVOIP will wait betweendigits. When the time expires, theMultiVOIP will look in thephonebook for the number entered.Default = 2.


-- Not applicable to FXO interface.


346

FXO Interface: Parameter Definitions (cont’d)

Field Name Values DescriptionDialing Options (cont’d)

Inter DigitRegenerationTime

50 to 20,000milliseconds

The length of time between theoutputting of DTMF digits.Default = 100 ms.

FXO Disconnect On There are three possible criteria fordisconnection under FXO: currentloss, tone detection, and silencedetection. Disconnection can betriggered by more than one of thethree criteria.

Current Loss Y/N Disconnection to be triggered byloss of current. That is, whenCurrent Loss is enabled (“Y”), theMultiVOIP will hang up the callwhen it detects a loss of currentinitiated by the attached device.

Current LossDetect Timer

integer values(in milliseconds )

The minimum time required fordetecting the current loss signal onthe FXO interface. In other words,this is the minimum length of timethe current must be absent tovalidate ‘current loss’ as adisconnection criterion. Default =500 ms.

Tone Detection Y/N Disconnection to be triggered by atone sequence.


347



FXO Disconnect On (cont’d)

DisconnectTone Sequence

1st tone pair+2nd tone pair

These are DTMF tone pairs.

Values for first tone pair are:*, #, 0, 1-9, and A-D.

Values for second tone pair are:none, 0, 1-9, A-D, *, and #.The tone pairs 1-9, 0, *, and # arethe standard DTMF pairs found onphone sets. The tone pairs A-D are“extended DTMF” tones, whichare used for various PBXfunctions.

Low Tones697Hz770Hz852Hz941Hz

High Tones 1209Hz 1336Hz 1447Hz 1633Hz

DTMF Tone Pairs

147*

2580

369#

ABCD

SilenceDetection

One-Way orTwo-Way

Disconnection to be triggered bysilence in one direction only or inboth directions simultaneously.

Silence Timer inseconds

integer value Duration of silence required totrigger disconnection.

Disconnect onCall ProgressTone

Y/N Allows call on FXO port to bedisconnected when a PBX issues acall-progress tone denoting that thephone station on the PBX that hasbeen involved in the call has beenhung up.

Ring Count,FXO

1-99 Number of rings required beforethe MultiVOIP answers theincoming call.


348



Flash Hook Options fields

Generation 50 - 1500milliseconds

Length of flash hook that will begenerated and sent out when theremote end initiates a flash hookand it is regenerated locally.Default = 600 ms.

Detection Range -- Not applicable to FXO.

Caller ID fieldsCaller ID Type Bellcore The MultiVOIP currently supports

only one implementation of CallerID. That implementation isBellcore type 1 with caller IDplaced between the first and secondrings of the call.

Caller ID enable Y/N Caller ID information is adescription of the remote callingparty received by the called party.The description has three parts:name of caller, phone number ofcaller, and time of call. The ‘time-of-call’ portion is always generatedby the receiving MultiVOIP unit(on FXS channel) based on its dateand time setup. The forms of the‘Caller Name’ and ‘Caller PhoneNumber’ differ depending on theIP transmission protocol used(H.323, SIP, or SPP) and uponentries in the phonebook screens ofthe remote (CID generating) voipunit. The CID Name and Numberappearing on the phone at theterminating FXS end will comeeither from a central office switch(showing a PSTN phone number),or the phonebook of the remote(CID sending) voip unit.


349

The Caller ID feature has dependencies on both the telco central office and theMultiVOIP phone book. See discussion below.

IPNetwork

xxxyyyzzzzJ.Q. Public


Central Officewith

standard telephonyCaller ID service

TerminatingVoIP

GeneratingVoIP

H.323 or SPPProtocol

Call is receivedhere.

Call originates hereat 1:42pm, May 31.

phone of:Melvin Jones763-555-8794

Display shows:

CID Number: 763-555-8794CID Name: Melvin JonesTime Stamp: Date: 05/31 Time:1:42pm

Clock:5-31,

1:42pm

FXS FXO

CID Flow

CID CID

In x.06 release, when SIP protocol is used,CID Name field will duplicate value inCID Number field.

*

*

Figure 6-1: Voip Caller ID Case #1 – Call, through telcocentral office with standard CID, enters voip system

IPNetwork



Central Officewithout


TerminatingVoIP

GeneratingVoIP

H.323 Protocol


Call originates hereat 4:19pm, July 10.

phone of:Wilda Jameson763-555-4071

Display shows:

CID Number: 423CID Name: Anoka-Whse-VP3Time Stamp: Date: 7/10 Time: 4:19pm

Clock:7/10, 4:19pm

FXS FXO

CID Flow

Phone Book Configuration

Gatekeeper RAS Parameters

Q.931 Parameters

Gateway Name: Anoka-Whse-VP3

Inbound Phone Book

423Remove Prefix Add Prefix Forward/Addr

{Channel 2}

748

CIDCID

Ch4

Ch3

Ch2

Ch1


*

*

Figure 6-2: Voip Caller ID Case #2 – Call, through telcocentral office without standard CID, enters H.323 voip system


350

IPNetwork xx xyy yz zz z

J.Q. Publ icx xxy yy zz zzJ.Q. Publ ic



TerminatingVoIP

GeneratingVoIP

SPP Protocol


Call originates hereat 5:47pm, Sept 27.

phone of:Henry Brampton763-555-4077

Display shows:

Clock:15:26, 5-31

FXS FXO

CID Flow

Inbound Phone Book


{Channel 2}

748Phone Book Configuration


Q.931 Parameters


Add/Edit Inbound Phone BookUse as default entry

Remove Prefix:Add Prefix:

Channel Number:Description:

Channel 2

Shipping Dept

CID Number: 423CID Name: Shipping DeptTime Stamp: Date: 0927 Time: 1747

... if “Description” field in Add/EditInbound Phone Book is used

CID Number: 423CID Name: Anoka-Whse-VP3Time Stamp: Date: 0927 Time: 1747

... if “Description” in Add/EditInbound Phone Book is blank

OR

Ch1

Ch2

Ch4

Ch3

Figure 6-3: Voip Caller ID Case #3 – Call, through telcocentral office without standard CID, enters SPP voip system

IPNetworkxxxyyyzzzz

J.Q. Public

TerminatingVoIP

GeneratingVoIP

H.323 Protocol


Call originates hereat 4:51pm, Oct 3.

phone of: Nigel Thurston763-555-9401

Display shows:


Clock:10/03, 4:51pm

FXSFXS

CID Flow



Q.931 Parameters


Inbound Phone Book


{Channel 2}

748

CIDCID

Ch4

Ch3

Ch2

Ch1


*

*


401

402

403

404

Figure 6-4: Voip Caller ID Case #4 – Remote FXS call onH.323 voip system


351

IPNetwork





TerminatingVoIP

GeneratingVoIP

H.323 Protocol


Call originates hereat 6:17pm, Nov 15.

phone of:Edwin Smith763-743-5873

Display shows:


Clock:11/15, 6:17pm

FXS DID

CID Flow



Q.931 Parameters


Inbound Phone Book


{Channel 2}

748

CIDCID

Ch4

Ch3

Ch2

Ch1


*

*

Figure 6-5: Voip Caller ID Case #5 – Call through telco centraloffice without standard CID enters DID channel in H.323 voip

system


352

E&M Parameters. The parameters applicable to the E&M telephonyinterface type are shown in the figure below and described in the table thatfollows.


353

E&M Interface Parameter DefinitionsField Name Values DescriptionInterface E&M enables E&M functionalityType Types 1-5. Refers to the type of E&M

interface being used.

Mode 2-wire or 4-wire Each E&M interface type can beeither 2-wire or 4-wire audio.

Signal Dial Tone orWink

When Dial Tone is selected, nowink is required on the E lead or Mlead in the call initiation or setup.When Wink is selected, a wink isrequired during call setup.

Wink Timer(in ms)

integer values,in milliseconds

This is the length of the wink forwink signaling.Applicable only when Signalparameter is set to “Wink.”

Pass ThroughEnable

Y/N When enabled (“Y”), this feature isused to create an open audio pathfor 2- or 4-wire. The E&M leadsare passed through the voiptransparently.Applicable only for E&MSignaling with Dial Tone.


354

E&M Interface Parameter Definitions (cont’d)Field Name Values Description

Dialing OptionsInter DigitTimer

integer values,in seconds

This is the length of time that theMultiVOIP will wait betweendigits. When the time expires, theMultiVOIP will look in thephonebook for the number entered.Default = 2.


Light or None Allows MultiVOIP to pass mode-code sequences between AvayaMagix PBXs to turn on and off themessage-waiting light on a PBXextension phone.

Mode codes:

*53 + PBX extension turns message light on.

#53 + PBX extension turns message light off.

Signals to turn message-waitinglights on/off are not sent to phonesconnected directly to theMultiVOIP on FXS channels, notto other non- Avaya Magix PBXphone stations on the voip network.

Inter DigitRegenerationTimer

milliseconds The length of time between theoutputting of DTMF digits.Default = 100 ms.


355

E&M Interface Parameter Definitions (cont’d)Field Name Values Description

Dialing Options (cont’d)

Disconnect onCall ProgressTone

Y/N Allows call on FXO port to bedisconnected when a PBX issues acall-progress tone denoting that thephone station on the PBX that hasbeen involved in the call has beenhung up.

Flash Hook Options fieldsGeneration integer values, in

millisecondsLength of flash hook that will begenerated and sent out when theremote end initiates a flash hookand it is regenerated locally.Default = 600 ms.

Detection Range for Min. and Max.,50 1500milliseconds

For a received flash hook to beregarded as such by theMultiVOIP, its duration must fallbetween the minimum andmaximum values given here.


356

DID Parameters. The parameters applicable to the Direct Inward Dial (DID)telephony interface type are shown in the figure below and described in thetable that follows. The DID interface allows one phone line to direct incomingcalls to any one of several extensions without a switchboard operator. Ofcourse, one DID line can handle only one call at a time. The parametersdescribed here pertain to the customer-premises side of the DID connection(DID-DPO, dial-pulse originating); the network side of the DID connection(DID-DPT, dial-pulse terminating) is not supported.

DID Interface Parameter DefinitionsField Name Values DescriptionInterface DID-DPO Enables the customer-premises

side of DID functionalityDID Options MultiVOIP’s use of DID applies

only for incoming DID calls. TheStart Mode used by the MultiVOIPmust match that used by theoriginating telephony equipment,else DID calls cannot becompleted.

Start Modes Immediate Start,Wink Start,Delay Dial

For Immediate Start, the voipdetects the off-hook conditioninitiated by the telco central-officecall and becomes ready to receivedial digits immediately.


357

DID Interface Parameter Definitions (cont’d)Field Name Values Description

DID Options (cont’d)Start Modes Immediate Start,

Wink Start,Delay Dial

For Wink Start, the voip detectsthe off-hook condition. Then thevoip reverses battery polarity for aspecified time (140-290 ms; a“wink”) and then becomes ready toreceive dial digits.For Delay Dial, the voip detectsdetects the off-hook condition.Then the voip reverses batterypolarity for a specified time(reverse polarity duration has wideracceptable range than for WinkStart) and then becomes ready toreceive dial digits.

Wink Timer(in ms)


This is the length of the wink forWink Start and Delay Dialsignaling modes..Applicable only when Start Modeparameter is set to “Wink Start” or“Delay Dial.”

Dialing OptionsInter DigitTimer

integer values,in seconds

This is the length of time that theMultiVOIP will wait betweendigits. When the time expires, theMultiVOIP will look in thephonebook for the number entered.Default = 2.


-- Not applicable to DID-DPOinterface.

Inter-DigitRegenerationTimer


This parameter is applicable whendigits are dialed onto a DID-DPOchannel after the connection hasbeen made. The length of timebetween the outputting of DTMFdigits.Default = 100 ms.


358

9b. (for ISDN-BRI MultiVOIP units). Set ISDN Parameters. This dialog boxcan be reached by pulldown menu, toolbar icon, keyboard shortcut, or sidebar.

Accessing ISDN (BRI) ParametersPulldown Icon

Shortcut Sidebar

Ctrl + T


359

In the ISDN BRI Parameters screen, select one of the BRI interfaces andconfigure it for the particular implementation of ISDN that you will use.Configure each BRI interface per the requirements of your voip system. TheMVP410ST has two ISDN-BRI interfaces and four channels; the MVP810SThas four ISDN-BRI interfaces and eight channels.


360

Note that ISDN BRI parameters are applied on an interface-by-interface basis.However, once you have established a set of ISDN BRI parameters for aparticular interface, you can apply this entire set of parameters to anotherinterface by using the Copy Interface button and its dialog box. To copy a setof ISDN BRI parameters to all interfaces, select “Copy to All” and click Copy.


361

ISDN-BRI Parameter DefinitionsField Name Values DescriptionSelect BRIInterface

ISDNnfor n= 1-2 (410ST)for n=1-4 (810ST)

In this field, you will choose whichISDN port you are configuring.The 410ST has two ISDN –BRIports (or “interfaces”); the 810SThas four ISDN-BRI ports (or“interfaces”). Each port has twochannels.

Layer 1Interface


When “Terminal” is selected, itindicates that the MultiVOIPshould emulate the subscriber(terminal) side of the digitalconnection. When “Network” isselected, it indicates that theMultiVOIP should emulate thecentral office (network) side of thedigital connection.If connecting to a telco or PBXthen choose “Terminal.” Ifconnecting to an ISDN phone orterminal adapter, then choose“Network.” Default = Terminal.

Dialing Options Inter Digit Timer(value inmilliseconds)

Dialing options are relevant whenthe MultiVOIP provides dial toneeither during an overlap receivingmode or providing a second dialtone. Default is 2000, which is 2seconds. Range 250 ms to 10000ms (1/4 sec to 10 sec).

Switch InformationCountry see table below Country in which MultiVOIP is

operating with ISDN.Operator see table below Indicates phone switch

manufacturer/model or refers totelco so as to specify the switchingsystem in question.ISDN is implemented somewhatdifferently in different switches(different software stacks are used).


362

ISDN-BRI Parameter Definitions (continued)Field Name Values Description

Switch InformationPCM Law a-law or mu-law “A-law” is an analog-to-digital

compression/expansionstandard used in Europe.“Mu-law” is the North Americanstandard. See the table below ofPCM-Law defaults based oncountry and operator.

TEI n Assignment(for n= 0-7)

Automatic orPoint-to-Point

SPID 0 numeric,3 to 20 digits

SPID 1 numeric,3 to 20 digits

“Copy Interface”button

Copies the ISDN-BRI attributesof one interface to anotherinterface. Attributes can becopied to multiple interfaces orto all interfaces at once.


363

Country and Operator options for the MVP-410ST/810ST voip units are listedbelow.

Australia

Europe ETSI--A-lawECMA_QSIG--A-lawFT_VN6--A-law

France FT_VN6--A-law

Hong Kong HK_TELA/mu, switch depndntdefault = mu-law

Italy ETSI--A-law

Japan NTT--mu-lawKDD--mu-law

Korea KOREAN_OPA/mu, switch depndntdefault = mu-law

USA N_ISDN1--mu-lawN_ISDN2--mu-lawATT_5E10--mu-lawNT_DMS100--mu-law

ETSI--A-lawAUSTEL_1--A-law


364

10. Set SNMP Parameters (Remote Voip Management). This dialog box canbe reached by pulldown menu, keyboard shortcut, or sidebar. To make theMultiVOIP controllable by a remote PC running the MultiVoipManagersoftware, check the “Enable SNMP Agent” box on the SNMP Parametersscreen.

Accessing “SNMP Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + M


365



366

The SNMP Parameter fields are described in the table below.

SNMP Parameter Definitions


Enable SNMPAgent

Y/N Enables the SNMP code in the firmwareof the MultiVOIP. This must beenabled for the MultiVOIP tocommunicate with and be controllableby theMultiVoipManager software.Default: disabled

Trap Manager Parameters

Address 4 places; n.n.n.nn = 0-255

IP address of MultiVoipManager PC.

CommunityName

-- A “community” is a group of VOIPendpoints that can communicate witheach other. Often “public” is used todesignate a grouping where all end usershave access to entire VOIP network.However, calling permissions can beconfigured to restrict access as needed.

Port Number 162 The default port number of the SNMPmanager receiving the traps is thestandard port 162.

CommunityName 1


First community grouping.


Read/Write


CommunityName 2


Second community grouping


Read/Write



367

11. Set Regional Parameters (Phone Signaling Tones & Cadences). ).This dialog box can be reached by pulldown menu, keyboard shortcut, orsidebar.

Accessing “Regional Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + R


368

The Regional Parameters screen will appear. For the country selected, thestandard set of frequency pairs will be listed for dial tone, busy tone,‘unobtainable’ tone (fast busy or trunk busy), and ring tone.

Remote Configuration/Command Modem. In the Country Selection forBuilt-In Modem field (drop-down list), select the country that best fits yoursituation. This may not be the same as your selection for theCountry/Region field. The selections in the Country Selection for Built-In Modem field entail more detailed groupings of telephony parameters thando the Country/Region values.



369

The Regional Parameters fields are described in the table below.

“Regional Parameter” DefinitionsField Name Values DescriptionCountry/Region

USA, Japan, UK,Custom

Name of a country or region that uses acertain set of tone pairs for dial tone,ring tone, busy tone, unobtainabletone (fast busy tone), survivabilitytone (tone heard briefly, 2 seconds,after going offhook denotingsurvivable mode of VOIP unit), re-order tone (a tone pattern indicatingthe need for the user to hang up thephone), and intercept tone (a tone thatwarns an a party that has gone off hookbut has not begun dialing, within aprescribed time, that an automaticemergency or attendant number will becalled; the automatic call can be usedto direct an attendant’s attention to adisabled or distressed caller, allowingan appropriate response to be made).In some cases, the tone-pair schemedenoted by a country name may alsobe used outside of that country. The“Custom” option (button) assures thatany tone-pairing scheme worldwidecan be accommodated.Note: Intercept tone is applicableonly when the FXS telephonyinterface has been chosen in theInterface screen and when theAutoCall / OffHook Alert field is setto OffHook Alert in the Voice/FaxParameters screen. The timeallowed for dialing before theautomatic calling process begins isset in the Offhook Alert Timer fieldof the Voice/Fax Parametersscreen.


370

“Regional Parameter” DefinitionsField Name Values DescriptionCountry/Region

USA, Japan, UK,CustomNote:“Survivability”tone indicates aspecial type ofcall-routingredundancy &applies toMultiVantage voipunits only.

Name of a country or region that uses acertain set of tone pairs for dial tone,ring tone, busy tone, and‘unobtainable’ tone (fast busy tone),survivability tone (tone heard briefly, 2seconds, after going offhook denotingsurvivable mode of voip unit) and re-order tone (a tone pattern indicatingthe need for the user to hang up thephone). In some cases, the tone-pairscheme denoted by a country namemay also be used outside of thatcountry. The “Custom” option(button) assures that any tone-pairingscheme worldwide can beaccommodated.

Type column dial tone,ring tone,busy tone,unobtainable tone(fast busy),survivability tone,re-order tone

Type of telephony tone-pair for whichfrequency, gain, and cadence are beingpresented.

Frequency 1 freq. in Hertz Lower frequency of pair.Frequency 2 freq. in Hertz Higher frequency of pair.


Amplification factor of lowerfrequency of pair.This applies to the dial, ring, busy and‘unobtainable’ tones that theMultiVOIP outputs as audio to theFXS, FXS, or E&M port. Default: -16dB


Amplification factor of higherfrequency of pair.This applies to the dial, ring, busy, and‘unobtainable’ (fast busy) tones thatthe MultiVOIP outputs as audio to theFXS, FXO, or E&M port. Default: -16dB


371

“Regional Parameter” Definitions (cont’d)

Field Name Values DescriptionCadence(msec) On/Off

n/n/n/nfour integer timevalues inmilli-seconds; zerovalue for dial-toneindicatescontinuous tone

On/off pattern of tone durations usedto denote phone ringing, phone busy,connection unobtainable (fast busy),dial tone (“0” indicates continuoustone), survivability, and re-order.Default values differ for differentcountries/regions. Although mostcadences have only two parts (an “on”duration and an “off” duration), sometelephony cadences have four parts.Most cadences, then, are expressed astwo iterations of a two-part sequence.Although this is redundant, it isnecessary to allow for expression of 4-part cadences.

Custom (button) -- Click on the “Custom” button to bringup the Custom Tone Pair Settingsscreen. (The “Custom” button is activeonly when “Custom” is selected in theCountry/Region field.) This screenallows the user to specify tone pairattributes that are not found in any ofthe standard national/regionaltelephony toning schemes.

CountrySelection forBuilt-In Modem(not applicableto MVP130 orMVP210)

country name MultiVOIP units operating with theX.06 software release (and above)include a built-in modem. Theadministrator can dial into this modemto configure the MultiVOIP unitremotely. The country name values inthis field set telephony parameters thatallow the modem to work in the listedcountry. This value may be differentthan the Country/Region value. Forexample, a user may need to choose“Europe” as the Country/Region valuebut “Denmark” as the Country-Selection-for-Built-In-Modem value.


372

12. Set Custom Tones and Cadences (optional). The Regional Parametersdialog box has a secondary dialog box that allows you to customize DTMFtone pairs to create unique ring-tonesdial-tones, busy-tones or“unobtainable” tones (fast busy signal) or “re-order” tones (telling the userthat she must hang up an off-hook phone) or “survivability” tones (anindication of call-routing redundancy) for your system. This screen allowsthe user to specify tone-pair attributes that are not found in any of thestandard national/regional telephony toning schemes. To access thiscustomization feature, click on the Custom button on the RegionalParameters screen. (The “Custom” button is active only when “Custom” isselected in the Country/Region field.)


373

The Custom Tone-Pair Settings fields are described in the table below.



Tone Pair dial tone,busy tone,ring tone,‘unobtainable’tone,survivability tone,re-order tone

Identifies the type of telephonysignaling tone for which frequenciesare being specified.

TONE PAIR VALUES About Defaults: US telephonyvalues are used as defaults on thisscreen. However, since this dialogbox is provided to allow customtone-pair settings, default values areessentially irrelevant.

Frequency 1 frequency in Hertz Frequency of lower tone of pair.This outbound tone pair enters theMultiVOIP at the input port.

Frequency 2 frequency in Hertz Frequency of higher tone of pair.This outbound tone pair enters theMultiVOIP at the input port.


Amplification factor of lowerfrequency of pair. This figuredescribes amplification that theMultiVOIP applies to outboundtones entering the MultiVOIP atthe input port. Default = -16dB


Amplification factor of higherfrequency of pair. This figuredescribes amplification that theMultiVOIP applies to outboundtones entering the MultiVOIP atthe input port. Default = -16dB


374



Cadence 1 integer time valueinmilli-seconds; zerovalue for dial-toneindicatescontinuous tone

On/off pattern of tone durationsused to denote phone ringing, phonebusy, dial tone (“0” indicatescontinuous tone) survivability andre-order. Cadence 1 is duration offirst period of tone being “on” in thecadence of the telephony signal(which could be ring-tone, busy-tone, unobtainable-tone, or dialtone).


Cadence 2 is duration of first “off”period in signaling cadence.


Cadence 3 is duration of second“on” period in signaling cadence.


Cadence 4 is duration of second“off” period in the signalingcadence, after which the 4-partcadence pattern of the telephonysignal repeats.


375

13. Set SMTP Parameters (Log Reports by Email). The SMTP Parametersscreen is applicable when the VOIP administrator has chosen to receive logreports by email (this is done by selecting the “SMTP” checkbox in theOthers screen and selecting “Enable SMTP” in the SMTP Parametersscreen.). The SMTP Parameters screen can be reached by pulldown menu,keyboard shortcut, or sidebar.

Accessing “SMTP Parameters”Pulldown Icon

Shortcut Sidebar

Ctrl + Alt + S

MultiVOIP as Email Sender. When SMTP is used, the MultiVOIP willactually be given its own email account (with Login Name and Password) onsome mail server connected to the IP network. Using this account, theMultiVOIP will then send out email messages containing log reportinformation. The “Recipient” of the log report email is ordinarily the VoIPadministrator. Because the MultiVOIP cannot receive email, a “Reply-To”address must also be set up. Ordinarily, the “Reply-To” address is that of atechnician who has access to the mail server or MultiVOIP or both, and theVoIP administrator might also be designated as the “Reply-To” party. Themain function of the Reply-To address is to receive error or failure messagesregarding the emailed reports.


376

The SMTP Parameters screen is shown below.

“SMTP Parameters” DefinitionsField Name Values DescriptionEnable SMTP Y/N In order to send log reports by email,

this box must be checked. However, toenable SMTP functionality, you mustalso select “SMTP” in the Logs screen.

Login Name alpha-numeric,per emaildomain

This is the User Name for theMultiVOIP unit’s email account.

Password alpha-numeric Login password for MultiVOIP unit’semail account.

Mail Server IPAddress

n.n.n.nfor n= 0 to 255

This is the mail server’s IP address.This mail server must be accessible onthe IP network to which the MultiVOIPis connected.

Port Number 25 25 is a standard port number for SMTP.


377

......“SMTP Parameters” Definitions (cont’d)


Mail Type text or html Mail type in which log reports will besent.

Subject text User specified. Subject line that willappear for all emailed log reports forthis MultiVOIP unit.

Reply-ToAddress

email address User specified. This email addressfunctions as a source email identifier forthe MultiVOIP, which, of course,cannot usefully receive email messages.The Reply-To address provides adestination for returned messagesindicating the status of messages sent bythe MultiVOIP (esp. to indicate whenlog report email was undeliverable orwhen an error has occurred).

RecipientAddress

email address User specified. Email address at whichVOIP administrator will receive logreports.

Mail Criteria Criteria for sending log summary byemail.The log summary email will be sent outeither when the user-specified numberof log messages has accumulated, oronce every day or multiple days, whichever comes first.

Number ofRecords

integer This is the number of log records thatmust accumulate to trigger the sendingof a log-summary email.

Number of Days integer This is the number of days that mustpass before triggering the sending of alog-summary email.


378

The SMTP Parameters dialog box has a secondary dialog box, CustomFields, that allows you to customize email log messages for the MultiVOIP.The MultiVOIP software logs data about many aspects of the call traffic goingthrough the MultiVOIP. The Custom Fields screen lets you pick which aspectswill be included in the email log reports.

“Custom Fields” DefinitionsField Description Field DescriptionSelect All Log report to

include all fieldsshown.

ChannelNumber

Data channelcarrying call.

Start Date,Time

Date and time the phonecall began.

Duration Length of call. Call Mode Voice or fax.PacketsSent

Total packets sent incall.

PacketsReceived

Total packets receivedin call.

BytesSent

Total bytes sent incall.

BytesReceived

Total bytes received incall.

PacketsLost

Packets lost in call. Coder Voice Coder/Compression Rate usedfor call will be listed inlog.


379

“Custom Fields” Definitions (cont’d)Field Description Field DescriptionOutboundDigits

Digits put out byMultiVOIP onto thephone line.

PrefixMatched

When selected, thephonebook prefixmatched in processingthe call will be listed inlog.

CallStatus

Successful orunsuccessful.

From Details To DetailsGatewayNumber

Originating gateway Gatew N. Completing oranswering gateway

IP Addr IP address wherecall originated.

IP Addr IP address where callwas completed oranswered.

Descript Identifier of sitewhere calloriginated.

Descript Identifier of site wherecall was completed oranswered.

Options When selected, logwill not use/non-useof SilenceCompression andForward ErrorCorrection by calloriginator.

Options When selected, log willnot use/non-use ofSilence Compressionand Forward ErrorCorrection by partyanswering call.


380


381

14. Set Log Reporting Method. The Logs screen lets you choose how theVoIP administrator will receive log reports about the MultiVOIP’sperformance and the phone call traffic that is passing through it. Log reportscan be received in one of three ways:

A. in the MultiVOIP program (GUI),

B. via email (SMTP), or

C. at the MultiVoipManager remote voip system managementprogram (SNMP).

Accessing “Logs” ScreenPulldown Icon

Shortcut Sidebar

Ctrl + Alt + O


382

If you enable console messages, you can customize the types of messages tobe included/excluded in log reports by clicking on the “Filters” button andusing the Console Messages Filter Settings screen (see subsequent page).If you use the logging function, select the logging option that applies to yourVoIP system design. If you intend to use a SysLog Server program forlogging, click in that Enable check box. The common SysLog logical portnumber is 514. If you intend to use the MultiVOIP web browser GUI forconfiguration and control of MultiVOIP units, be aware that the web browserGUI does not support logs directly. However, when the web browser GUI isused, log files can still be sent to the voip administrator via email (whichrequires activating the SMTP logging option in this screen).


383

“Logs” Screen DefinitionsField Name Values DescriptionEnable ConsoleMessages

Y/N Allows MultiVOIP debugging messages to be readvia a basic terminal program like HyperTerminal™ or equivalent. Normally, this should bedisabled because it uses MultiVOIP processingresources. Console messages are meant for techsupport personnel.

Filters (button) Click to access secondary screen on where consolemessages can be included/excluded by categoryand on a per-channel basis. (See the ConsoleMessages Filter Settings screen on subsequentpage.) Not supported in BRI 5.02c software.

Turn Off Logs Y/N Check to disable log-reporting function. Notsupported in BRI 5.02c software.

Logs Buttons Only one of these three log reporting methods,GUI, SMTP, or SNMP, may be chosen.

GUI Y/N User must view logs at the MultiVOIPconfiguration program.

SNMP Y/N Log messages will be delivered to theMultiVoipManager application program.

SMTP Y/N Log messages will be sent to user-specified emailaddress.

SysLog ServerEnable

Y/N This box must be checked if logging is to be donein conjunction with a SysLog Server program. Formore on SysLog Server, see Operation &Maintenance chapter. Not supported in BRI 5.02csoftware.

IP Address n.n.n.nfor n=0-255

IP address of computer, connected to voipnetwork, on which SysLog Server program isrunning. Not supported in BRI 5.02c software.

Port 514 Logical port for SysLog Server. 514 is commonlyused. Not supported in BRI 5.02c software.

Online StatisticsUpdation Interval

integer Set the interval (in seconds) at which logginginformation will be updated. Not supportedin BRI 5.02c software.


384

To customize console messages by category and/or by channel, click on“Filters” and use the Console Messages Filters Settings screen.


385

15. Set Supplementary Services Parameters. This dialog box can be reachedby pulldown menu, keyboard shortcut, or sidebar. This screen is not supportedin BRI 5.02c software.

Accessing “Supplementary Services” Parameters

Pulldown Icon

Shortcut Sidebar

Ctrl + Alt +H

Supplementary Services features derive from the H.450 standard, which bringsto voip telephony functionality once only available with PSTN or PBXtelephony. Supplementary Services features can be used under H.323 only andnot under SIP.


386


Of the features implemented under Supplementary Services, three are veryclosely related: Call Transfer, Call Hold, and Call Waiting. Call NameIdentification is similar but not identical to the premium PSTN featurecommonly known as Caller ID.

Call Transfer. Call Transfer allows one party to re-connect the party withwhom they have been speaking to a third party. The first party isdisconnected when the third party becomes connected. Feature is invokedby a programmable phone keypad sequence (for example, #7).

Call Hold. Call Hold allows one party to maintain an idle (non-talking)connection with another party while receiving another call (Call Waiting),while initiating another call (Call Transfer), or while performing someother call management function. Invoked by keypad sequence.

Call Waiting. Call Waiting notifies an engaged caller of an incoming calland allows them to receive a call from a third party while the party withwhom they have been speaking is put on hold. Invoked by keypadsequence.

Call Name Identification. When enabled for a given voip unit (the ‘home’voip), this feature gives notice to remote voips involved in calls.Notification goes to the remote voip administrator, not to individual phonestations. When the home voip is the caller, a plain English descriptor willbe sent to the remote (callee) voip identifying


387

the channel over which the call is being originated (for example, “CallingParty - Omaha Sales Office Line 2”). If that voip channel is dedicated to acertain individual, the descriptor could say that, as well (for example“Calling Party - Harold Smith in Omaha”). When the home voip receives acall from any remote voip, the home voip sends a status message back tothat caller. This message confirms that the home voip’s phone channel iseither busy or ringing or that a connection has been made (for example,“Busy Party - Omaha Sales Office Line 2”). These messages appear in theStatistics – Call Progress screen of the remote voip.

Note that Supplementary Services parameters are applied on a channel-by-channel basis. However, once you have established a set of supplementaryparameters for a particular channel, you can apply this entire set of parametersto another channel by using the Copy Channel button and its dialog box. Tocopy a set of Supplementary Services parameters to all channels, select “Copyto All” and click Copy.


388

The Supplementary Services fields are described in the tables below.

Supplementary Services Parameter Definitions(Not supported in BRI 5.02c software.)


SelectChannel

1-2 (210);1-4 (410);1-8 (810)

The channel to be configured is selectedhere.

Call TransferEnable

Y/N Select to enable the Call Transfer functionin the voip unit.This is a “blind” transfer and the sequenceof events is as follows:Callers A and B are having aconversation.Caller A wants to put B into contact withC.Caller A dials call transfer sequence.Caller A hears dial tone and dials numberfor caller C.Caller A gets disconnected while Caller Bgets connected to caller C.A brief musical jingle is played for thecaller on hold.

TransferSequence

any phonekeypadcharacter

The numbers and/or symbols that thecaller must press on the phone keypad toinitiate a call transfer.The call-transfer sequence can be 1 to 4characters in length using anycombination of digits or characters(* or #).The sequences for call transfer, call hold,and call waiting can be from 1 to 4 digitsin length consisting of any combinationof digits 1234567890*#.


389


Call HoldEnable

Y/N Select to enable Call Hold function invoip unit.Call Hold allows one party to maintain anidle (non-talking) connection with anotherparty while receiving another call (CallWaiting), while initiating another call(Call Transfer), or while performing someother call management function.

HoldSequence

phonekeypadcharacters

The numbers and/or symbols that thecaller must press on the phone keypad toinitiate a call hold.The call-hold sequence can be 1 to 4characters in length using anycombination of digits or characters(* or #).

Call WaitingEnable

Y/N Select to enable Call Waiting function invoip unit.

RetrieveSequence

phonekeypadcharacters,twocharactersin length

The numbers and/or symbols that thecaller must press on the phone keypad toinitiate retrieval of a waiting call.The call-waiting retrieval sequence can be1 to 4 characters in length using anycombination of digits or characters(* or #).This is the phone keypad sequence that auser must press to retrieve a waiting call.Customize-able. Sequence should bedistinct from sequence that might be usedto retrieve a waiting call via the PBX orPSTN.


390


Call NameIdentificationEnable

Enables CNI function. Call NameIdentification is not the same as Caller ID.When enabled on a given voip unitcurrently being controlled by theMultiVOIP GUI (the ‘home voip’), CallName Identification sends an identifierand status information to theadministrator of the remote voip involvedin the call. The feature operates on achannel-by-channel basis (each channelcan have a separate identifier).If the home voip is originating the call,only the Calling Party field isapplicable. If the home voip is receivingthe call, then the Alerting Party, BusyParty, and Connected Party fields arethe only applicable fields (and any or allof these could be enabled for a given voipchannel). The status information confirmsback to the originator that the callee (thehome voip) is either busy, or ringing, orthat the intended call has been completedand is currently connected.The identifier and status information aremade available to the remote voip unitand appear in the Caller ID field of itsStatistics – Call Progress screen. (This ishow MultiVOIP units handle CNImessages; in other voip brands, H.450may be implemented differently and thenthe message presentation may vary.)


391


Calling Party,AllowedName Type(CNI)

If the ‘home’ voip unit is originating thecall and Calling Party is selected, thenthe identifier (from the Caller Id field)will be sent to the remote voip unit beingcalled. The Caller Id field gives theremote voip administrator a plain-language identifier of the party that isoriginating the call occurring on a specificchannel.

This field is applicable only when the ‘home’voip unit is originating the call.

Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip in thisexample), Call Name Identification hasbeen enabled, Calling Party has beenenabled as an Allowed Name Type, and“Omaha Sales Office Voipchannel 2” hasbeen entered in the Caller Id field.When channel 2 of the Omaha voip isused to make a call to any other voipphone station (for example, the Denveroffice), the message“Calling Party - Omaha Sales OfficeVoipchannel 2” will appear in the “CallerId” field of theStatistics - Call Progress screenof the Denver voip.


392


AlertingParty,AllowedName Type(CNI)

If the ‘home’ voip unit is receiving thecall and Alerting Party is selected, thenthe identifier (from the Caller Id field)will tell the originating remote voip unitthat the call is ringing.


Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip unit inthis example), Call Name Identificationhas been enabled, Alerting Party hasbeen enabled as an Allowed Name Type,and “Omaha Sales Office Voipchannel 2”has been entered in the Caller Id field ofthe Supplementary Services screen.

When channel 2 of the Omaha voipreceives a call from any other voip phonestation (for example, the Denver office),the message “Alerting Party - OmahaSales Office Voipchannel 2” will be sentback and will appear in the Caller Id fieldof the Statistics – Call Progress screenof the Denver voip. This confirms to theDenver voip that the phone is ringing inOmaha.


393


Busy Party,AllowedName Type(CNI)

If the ‘home’ voip unit is receiving a calldirected toward an already engagedchannel or phone station and Busy Partyis selected, then the identifier (from theCaller Id field) will tell the originatingremote voip unit that the channel or calledparty is busy.


Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip unit inthis example), Call Name Identificationhas been enabled, Busy Party has beenenabled as an Allowed Name Type, and“Omaha Sales Office Voipchannel 2” hasbeen entered in the Caller Id field of theSupplementary Services screen.

When channel 2 of the Omaha voip isbusy but still receives a call attempt fromany other voip phone station (forexample, the Denver office), the message“Busy Party - Omaha Sales OfficeVoipchannel 2” will be sent back and willappear in the Caller Id field of theStatistics – Call Progress screen of theDenver voip. This confirms to the Denvervoip that the channel or phone station isbusy in Omaha.


394


ConnectedParty,AllowedName Type(CNI)

If the ‘home’ voip unit is receiving a calland Connected Party is selected, then theidentifier (from the Caller Id field) willtell the originating remote voip unit thatthe attempted call has been completed andthe connection is made.


Example. Suppose a voip system hasoffices in both Denver and Omaha. In theOmaha voip unit (the ‘home’ voip unit inthis example), Call Name Identificationhas been enabled, Connected Party hasbeen enabled as an Allowed Name Type,and “Omaha Sales Office Voipchannel 2”has been entered in the Caller Id field ofthe Supplementary Services screen.

When channel 2 of the Omaha voipcompletes an attempted call from anyother voip phone station (for example, theDenver office), the message “ConnectParty - Omaha Sales Office Voipchannel2” will be sent back and will appear in theCaller Id field of the Statistics – CallProgress screen of the Denver voip. Thisconfirms to the Denver voip that the callhas been completed to Omaha.


395


Caller ID This is the identifier of a specific channelof the ‘home’ voip unit. The Caller Idfield typically describes a person, office,or location, for example, “Harry Smith,”or “Bursar’s Office,” or “BarnesvilleFactory.”

Default -- When this button is clicked, allSupplementary Service parameters are setto their default values.

CopyChannel

-- Copies the Supplementary Serviceattributes of one channel to anotherchannel. Attributes can be copied tomultiple channels or all channels at once.


396

16. Set Baud Rate. The Connection option in the sidebar menu has a“Settings” item that includes the baud-rate setting for the COM port of thecomputer running the MultiVOIP software.

First, it is important to note that the default COM port established by theMultiVOIP program is COM1. Do not accept the default value until youhave checked the COM port allocation on your PC. To do this, check forCOM port assignments in the system resource dialog box(es) of yourWindows operating system. If COM1 is not available, you must change theCOM port setting to COM2 or some other COM port that you haveconfirmed as being available on your PC.

The default baud rate is 115,200 bps.


397

17. View System Information screen and set updating interval (optional). TheSystem Information screen is not supported in BRI 5.02c software.

This dialog box can be reached by pulldown menu, keyboard shortcut, orsidebar.

Accessing “System Information” Screen

Pulldown Icon

Shortcut Sidebar

Ctrl + Alt +Y


398

This screen presents vital system information at a glance. Its primary use is introubleshooting.

System Information Parameter Definitions


Boot CodeVersion

nn.nn Indicates the version of the code that isused at the startup (booting) of the voip.The boot code version is independent ofthe software version.





FirmwareVersion

alpha-numeric

Indicates the version of the MultiVOIPfirmware.


399


18. Saving the MultiVOIP Configuration. When values have been set for allof the MultiVOIP’s various operating parameters, click on Save Setup in thesidebar.


400

19. Creating a User Default Configuration. When a “Setup” (completegrouping of parameters) is being saved, you will be prompted aboutdesignating that setup as a “User Default” setup. A User Default setup maybe useful as a baseline of site-specific values to which you can easily revert.Establishing a User Default Setup is optional.

304

Chapter 7: T1 PhonebookConfiguration

(North American Telephony Standards)

MultiVOIP User Guide T1 PhoneBook Configuration

305

Configuring the MVP2410MultiVOIP Phonebooks

When a VoIP serves a PBX system, it’s important that the operation of theVoIP be transparent to the telephone end user. That is, the VoIP should notentail the dialing of extra digits to reach users elsewhere on the network thatthe VoIP serves. On the contrary, VOIP service more commonly reducesdialed digits by allowing users (served by PBXs in facilities in distant cities) todial their co-workers with 3-, 4-, or 5-digit extensions as if they were in thesame facility.

Furthermore, the setup of the VoIP generally should allow users to make callson a non-toll basis to any numbers accessible without toll by users at all otherlocations on the VoIP system. Consider, for example, a company with VOIP-equipped offices in New York, Miami, and Los Angeles, each served by itsown PBX. When the VOIP phone books are set correctly, personnel in theMiami office should be able to make calls without toll not only to thecompany’s offices in New York and Los Angeles, but also to any numberthat’s local in those two cities.

To achieve transparency of the VoIP telephony system and to give full accessto all types of non-toll calls made possible by the VOIP system, the VoIPadministrator must properly configure the “Outbound” and “Inbound” phone-books of each VoIP in the system.

The “Outbound” phonebook for a particular VoIP unit describes the dialingsequences required for a call to originate locally (typically in a PBX in aparticular facility) and reach any of its possible destinations at remote VoIPsites, including non-toll calls completed in the PSTN at the remote site.

The “Inbound” phonebook for a particular VoIP unit describes the dialingsequences required for a call to originate remotely from any other VOIP sitesin the system, and to terminate on that particular VOIP.

Briefly stated, the MultiVOIP’s Outbound phonebook lists the phone stations itcan call; its Inbound phonebook describes the dialing sequences that can beused to call that MultiVOIP and how those calls will be directed. (Of course,the phone numbers are not literally “listed” individually, but are, instead,described by rule.)

Consider two types of calls in the three-city system described above: (1) callsoriginating from the Miami office and terminating in the New York(Manhattan) office, and (2) calls originating from the Miami office andterminating in New York City but off the company’s premises in an adjacentarea code, an area code different than the company’s office but still a local callfrom that office (e.g., Staten Island).

T1 Phonebook Configuration MultiVOIP User Guide

306

The first type of call requires an entry in the Outbound PhoneBook of theMiami VOIP and a coordinated entry in the Inbound phonebook of the NewYork VOIP. These entries would allow the Miami caller to dial the New Yorkoffice as if its phones were extensions on the Miami PBX.

The second type of call similarly requires an entry in the Outbound PhoneBookof the Miami VOIP and a coordinated entry in the Inbound Phonebook of theNew York VOIP. However, these entries will be longer and more complicated.Any Miami call to New York City local numbers will be sent through theVOIP system rather than through the regular toll public phone system (PSTN).But the phonebook entries can be arranged so that the VOIP system istransparent to the Miami user, such that even though that Miami user dials theNew York City local number just as they would through the public phonesystem, that call will still be completed through the VOIP system.

This PhoneBook Configuration procedure is brief, but it is followed by anexample case. For many people, the example case may be easier to grasp thanthe procedure steps. Configuration is not difficult, but all phone numbersequences and other information must be entered exactly; otherwiseconnections will not be made.


307

Phonebook configuration screens can be accessed using icons or the sidebarmenu.

Phonebook Icons Description

Phonebook Configuration

Inbound Phonebook EntriesList

Add Inbound PhonebookEntry

Edit selected InboundPhonebook Entry

Outbound Phonebook EntriesList

Add Outbound PhonebookEntry

Edit selected OutboundPhonebook Entry


308

Phonebook Sidebar Menu


309

1. Go to the PhoneBook Configuration screen (using either the sidebar ordrop-down menu).


310

In consultation with your VOIP administrator, enter the Gateway Name andvalues for Q.931 parameters and Gatekeeper RAS parameters. Determinewhether your voip system will operate with a proxy server. Determinewhich H.323 version 4 functions you will implement. (They are not alwaysapplicable. See field description for each parameter.) If the SPP protocol isused, values for another group of parameters must be specified, as well.


311

The table below describes all fields in the general PhoneBook Configurationscreen.

PhoneBook Configuration Parameter DefinitionsField Name Values DescriptionGateway Name Y/N This field allows you to specify a

name for this MultiVOIP. Whenplacing a call, this name is sent tothe remote MultiVOIP for displayin Call Progress listings, Logs,etc.

Q.931 Parameters

Use Fast Start Y/N Enables the H.323 Fast Startprocedure. May need to beenabled/disabled for compatibilitywith third-party VOIP gateways.

Call SignalingPort

port number Default: 1720 (H.323)

GateKeeper RAS ParametersGatekeeper /Clear ChannelIP Address

IP address of the GateKeeper.

Port Number Well-known port number forGateKeepers.Must match port number ofGateKeeper, 1719.

Gateway Prefix This number becomes registeredwith the GateKeeper. Callrequests sent to the gatekeeperand preceded by this prefix willbe routed to the VOIP gateway.

GatekeeperName

alpha-numericstring

Optional. The name of theGateKeeper with which thisMultiVOIP is trying to register.

Gateway H.323ID

The H.323 ID is used to registerthis particular MultiVOIP withthe GateKeeper.


312

PhoneBook Configuration Parameter Definitions

(cont’d)Field Name Values Description

SIP Proxy Parameters

Enable Proxy Y/N Allows the MultiVOIP to work inconjunction with a proxy server.

Proxy Server IPAddress

n.n.n.nwheren=0-255

Network address of the proxyserver that the voip is using.

Port Number Logical port number for proxycommunications.

User Name Values: alphnumeric

Description: Identifier used when proxy server isused in network. If a proxy server is used in a SIPvoip network, all clients must enter both a UserName and a Password before being allowed tomake a call.

Password Values: alphanumeric

Description: Password for proxy server function.See “User Name” description above.


313

PhoneBook Configuration Parameter Definitions(cont’d)Field Name Values DescriptionH.323 Version 4 Parameters

Q.931Multiplexing(Mux)

Y/N Signaling for multiple phone callscan be carried on a single portrather than opening a separatesignaling port for each call. Thisconserves bandwidth resources.

H.245Tunneling (Tun)

Values: Y/N

Description: H.245 messages are encapsulatedwithin the Q.931 call-signaling channel. Amongother things, the H.245 messages let the twoendpoints tell each other what their technicalcapabilities are and determine who, during thecall, will be the client and who the server.Tunneling is the process of transmitting theseH.245 messages through the Q.931 channel. Thesame TCP/IP socket (or logical port) already beingused for the Call Signaling Channel is then alsoused by the H.245 Control Channel. Thisencapsulation reduces the number of logical ports(sockets) needed and reduces call setup time.


314

PhoneBook Configuration Parameter Definitions(cont’d)

Field Name Values DescriptionH.323 Version 4 Parameters

Parallel H.245(FS + Tun)

Values: Y/NDescription: FS (Fast Start or Fast Connect) is aQ.931 feature of H.323v2 to hasten call setup aswell as ‘pre-opening’ the media channel beforethe CONNECT message is sent. This pre-openingis a requirement for certain billing activities.Under Parallel H.245 FS + Tun, this Fast Connectfeature can operate simultaneously with H.245Tunneling (see description above).

Annex –E (AE) Values: Y/NDescription: Multiplexed UDP call signalingtransport. Annex E is helpful for high-volumevoip system endpoints. Gateways with lesservolume can afford to use TCP to establish calls.However, for larger volume endpoints, the callsetup times and system resource usage under TCPcan become problematic. Annex E allowsendpoints to perform call-signaling functionsunder the UDP protocol, which involvessubstantially streamlined overhead. (This featureshould not be used on the public Internet becauseof potential problems with security and bandwidthusage.)


315


Field Name Values DescriptionSingle Port Protocol (SPP)

Mode Direct,Client, orRegistrar

SPP voip systems can operate intwo modes:in the direct mode, where allvoip gateways have static IPaddresses assigned to them; or inthe registrar/client mode, whereone voip gateway serves asregistrar and all other gateways,being its clients, point to thatregistrar. The registrar assigns IPaddresses dynamically.

General Options

Port The UDP port on which datatransmission will occur. Eachclient voip has its own port. Iftwo client voips are both behindthe same firewall, then they musthave different ports assigned tothem.If there are two clients and each isbehind a different firewall, thenthe clients could have differentport numbers or the same portnumber.(Default port number = 10000.)

Re-trans-mission(in ms)

If packets are lost (as indicated byabsence of an acknowledgment)then the endpoint will retransmitthe lost packets after thisdesignated time duration haselapsed. (Default value = 2000milliseconds.)

MaxRe-trans-mission

Number of times the voip will re-transmit a lost packet (if noacknowledgment has beenreceived). (Default value = 3)


316



[continued]

Client Options Client Option fields are activeonly in registrar/client mode andonly for client voip units.

Registrar IPAddress

This is the IP address of theregistrar voip to which this clientis assigned. (Default value =0.0.0.0; effectively, there is nouseful default value.)

Registrar Port This is the port number of theregistrar voip to which this clientis assigned. (Default port number= 10000.)

Registrar Options Registrar Option fields are activeonly in registrar/client mode andonly for registrar voip units.

Keep Alive(in sec.)

Time-out duration before aregistrar will unregister a clientthat does not send its “I’m here”signal. Client normally sends its“I’m here” signal every 20seconds. Timeout default = 60seconds.

Proxy/NAT DeviceParameters

Enable SPPProxy/NATdevice

Y/N Enables MultiVOIP (running inSPP Registrar mode) to operate‘behind’ a proxy/NAT device(NAT = Network AddressTranslation).

Proxy/NATDeviceParameters –Public IPAddress

n.n.n.nwheren=0-255

The public IP address of theproxy/NAT device which theMultiVOIP is behind.


317

About SPP Proxy/NAT Device Parameters

IPNetwork

SPP Client/Registrar System

ClientVoip

Public

ClientVoip

Public

ClientVoip

Public

ClientVoip

Public

Proxy/NATDevice

Public IP

Private IP

IP

IP

IP

IP

RegistrarVoip

A Proxy/NAT device is sometimes usedin a Client/Registrar SPP voip systemwhere the registrar voip is in a privatenetwork but serves client voips on apublic network. The Proxy/NAT deviceisolates (protects) the registrar voipfrom the public network.

In such cases, you must check the“Enable SPP Proxy/NAT device” checkboxin the Phonebook Configurat ion screenof the Registrar voip. The private registrarvoip can then function with the client voipsusing the public IP address of the Proxy/NATdevice. You must enter this address in thePublic IP Address field.


318

2. Select PhoneBook Modify and then selectOutbound Phone Book/List Entries.

Click Add.


319

3. The Add/Edit Outbound PhoneBook screen appears.

Enter Outbound PhoneBook data for your MVP2410. Note that the Advancedbutton gives access to the Alternate IP Routing feature, if needed. Alternate IPRouting can be implemented in a secondary screen (as described after theprimary screen field definitions below).


320

The fields of the Add/Edit Outbound Phone Book screen are described in thetable below.

Add/Edit Outbound Phone Book: Field Definitions


DestinationPattern

prefixes,area codes,exchanges,line numbers,extensions

Defines the beginning of dialingsequences for calls that will beconnected to another VOIP inthe system. Numbers beginningwith these sequences arediverted from the PTSN andcarried on Internet or other IPnetwork.

Total Digits as needed number of digits the phone usermust dial to reach specifieddestination

Remove Prefix dialed digits portion of dialed number to beremoved before completing callto destination

Add Prefix dialed digits digits to be added beforecompleting call to destination

IP Address n.n.n.nforn = 0-255

the IP address to which the callwill be directed if it begins withthe destination pattern given

Description alpha-numeric Describes the facility orgeographical location at whichthe call will be completed.

Protocol Type SIP or H.323or SPP

Indicates protocol to be used inoutbound transmission. SinglePort Protocol (SPP) is a non-standard protocol designed byMulti-Tech.


321


(cont’d)


H.323 fieldsUse Gatekeepr Y/N Indicates whether or not

gatekeeper is used.

H.323 ID The H.323 ID assigned to thedestination MultiVOIP. Onlyvalid if “Use Gatekeeper” isenabled for this entry.


Q.931 PortNumber

1720 Q.931 is the call signalingprotocol for setup andtermination of calls (aka ITU-TRecommendation I.451). H.323employs only one “well-known”port (1720) for Q.931 signaling.If Q.931 message-orientedsignaling protocol is used, theport number 1720 must bechosen.


322


(cont’d)Field Name Values DescriptionSIP FieldsUse Proxy Y/N Select if proxy server is used.

TransportProtocol

TCP or

UDP

Voip administrator must choosebetween UDP and TCP transmissionprotocols. UDP is a high-speed,low-overhead connectionlessprotocol where data is transmittedwithout acknowledgment,guaranteed delivery, or guaranteedpacket sequence integrity. TCP isslower connection-oriented protocolwith greater overhead, but havingacknowledgment and guaranteesdelivery and packet sequenceintegrity.

SIP PortNumber

5060 or other

*See RFC3087(“Control ofService Contextusing SIPRequest-URI,”by the NetworkWorkingGroup).

The SIP Port Number is aUDP logical port number.The voip will “listen” for SIPmessages at this logical port.If SIP is used, 5060 is thedefault, standard, or “wellknown” port number to beused. If 5060 is not used,then the port number used isthat specified in the SIPRequest URI (UniversalResource Identifier).

SIP URL sip.userphone@hostserver,where“userphone” isthe telephonenumber and“hostserver”isthe domainname or anaddress on thenetwork

Looking similar to an emailaddress, a SIP URLidentifies a user's address.In SIP communications, eachcaller or callee is identifiedby a SIP url:sip:user_name@host_name.The format of a sip url is verysimilar to an email address,except that the “sip:“ prefix isused.


323

Add/Edit Outbound Phone Book: Field Def’ns (cont’d)Field Name Values DescriptionSPP FieldsUse Registrar Values: Y/N

Description: Select this checkbox to use registrarwhen voip system is operating in the “Registrar/Client”SPP mode. In this mode, one voip (the registrar, as set inPhonebook Configuration screen) has a static IP addressand all other voips (clients) point to the registar’s IPaddress as functionally their own. However, if yourvoip system overall is operating in “Registrar/Client”mode but you want to make an exception and use Directmode for the destination pattern of this particularAdd/Edit Phonebook entry, leave this checkboxunselected.

Leave this checkbox unselected if your overall voipsystem is operating in the “Direct" SPP mode. In thismode, all voips in system are peers and each has its ownstatic IP address.

Port Number Values: numericDescription: When operating in “Registrar/Client”mode, this is the port by which the gateway receives allSPP data and control messages from the registrargateway. (This ability to receive all data and messagesvia one port allows the voip to operate behind a firewallwith only one port open.)When operating in “Direct” mode, this is the Port bywhich peer voips receive data and messages.

Alternate PhoneNumber

numeric Phone number associated withalternate IP routing.

Advancedbutton

Values: N/A

Description: Gives access to secondary screenwhere an Alternate IP Route can be specified forbackup or redundancy of signal paths. Seediscussion on next page. For SIP & H.323operation only.


324

Clicking on the Advanced button brings up the Alternate Routing secondary screen.This feature provides an alternate path for calls if the primary IP network cannot carrythe traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN.However, this feature could also be used to divert traffic to a redundant (backup) unitin case one voip unit fails. The user must specify the IP address of the alternate routefor each destination pattern entry in the Outbound Phonebook.


325

Alternate Routing Field Definitions

FieldName

Values Description

AlternateIPAddress

n.n.n.nwheren= 0-255

Alternate destination for outbound data traffic incase of excessive delay in data transmission.

RoundTripDelay

milliseconds The Round Trip Delay is the criterion for judgingwhen a data pathway is considered blocked. Whenthe delay exceeds the threshold specified here, thedata stream will be diverted to the alternatedestination specified as the Alternate IP Address.

The Alternate Routing function facilitates PSTN Failover protection, that is, it allowsyou to re-route voip calls automatically over the PSTN if the voip system fails. TheMultiVOIP can be programmed to respond to excessive delays in the transmission ofvoice packets, which the MultiVOIP interprets as a failure of the IP network. Upondetecting an excessive delay in transmission of voice packets (overly high “latency”in the network) the MultiVOIP diverts the call to another IP address, which itself isconnected to the PSTN (for example, via an FXO port on the self-same MultiVOIPcould be connected to the PSTN).

IPNETWORK

FXS

FXO

VOIPVOIPPBX

PSTN Line

1. Call originates.2. IP network fails.

3. Call diverts toAlt IP address in voipaccessing PSTN line.

4. Call completedvia PSTN.

PSTN Failover Feature. The MultiVOIP can be programmed to divert calls to thePSTN temporarily in case the IP network fails.


326

4. Select PhoneBook Modify and then select Inbound PhoneBook | List Entries.


327

5. The Add/Edit Inbound PhoneBook screen appears.


328

Enter Inbound PhoneBook data for your MultiVOIP. The fields of theAdd/Edit Inbound PhoneBook screen are described in the table below.

Add/Edit Inbound Phone Book: Field DefinitionsField Name Values DescriptionRemove Prefix dialed digits portion of dialed number to be

removed before completing callto destination(often a local PBX)

Add Prefix dialed digits digits to be added beforecompleting call to destination(often a local PBX)

ChannelNumber

1-24, or“Hunting”

T1 channel number to which thecall will be assigned as it entersthe local telephony equipment(often a local PBX). “Hunting”directs the call to any availablechannel.

Description -- Describes the facility orgeographical location at whichthe call originated.

Call Forward Parameters

Enable Y/N Click the check-box to enablethe call-forwarding feature.


329

Add/Edit Inbound Phone Book: Field Definitions

(cont’d)

Field Name Values DescriptionCall Forward Parameters

ForwardCondition

Uncondit.;BusyNo Resp.

Unconditional. When selected,all calls received will beforwarded.Busy. When selected, calls willbe forwarded when station isbusy.No Response. When selected,calls will be forwarded if calledparty does not answer after aspecified number of rings, asspecified in Ring Count field.

Forwarding can be conditionedon both “Busy” and “NoResponse.”

ForwardDestinationIP address,phone number,port number,etc.

Phone number or IP address to which calls will bedirected.

For H.323 calls, the Forward Destination can beeither a Phone Number of an IP Address.

For SIP calls, the Forward Destination can be oneof the following:(a) phone number, (b) IP address,(c) IP address: port number,(d) phone number:IP addr: port number,(e) SIP URL, or (f) phone #: IP address.

Ring Count 0, 1, 2, 3, etc. When “No Response” iscondition for forwarding calls, thisdetermines how many unanswered rings areneeded to trigger the forwarding.

RegistrationOptionParameters

In an H.323 voip system, gateways can registerwith the system using one of these identifiers: (a)an E.164 identifier, (b) a Tech Prefix identifier, or(c) an H.323 ID identifier.

In a SIP voip system, gateways can register withthe SIP Proxy.

In an SPP voip system, gateways can register withthe SPP Registrar voip unit.


330

6. When your Outbound and Inbound PhoneBook entries are completed, clickon Save Setup in the sidebar menu to save your configuration.

You can change your configuration at any time as needed for your system.

Remember that the initial MVP2410 setup must be done locally or via thebuilt-in Remote Configuration/Command Modem using the MultiVOIPprogram. After the initial configuration is complete, all of the MVP2410 unitsin the VOIP system can be configured, re-configured, and updated from onelocation using the MultiVOIP web GUI software program or the MultiVOIPprogram (in conjunction with the built-in modem).

T1 Phonebook ExamplesThe following example demonstrates how Outbound and Inbound PhoneBookentries work in a situation of multiple area codes. Consider a company withoffices in Minneapolis and Baltimore.

3 Sites, All-T1 ExampleNotice first the area code situation in those two cities: Minneapolis’s localcalling area consists of multiple adjacent area codes; Baltimore’s local callingarea consists of a base area code plus an overlay area code.

Mpls612

NWSuburbs

763 St. Paul& Suburbs

651

SW Suburbs952

CompanyVOIP/PBX

SIte

Baltimore410

Baltimore/Outstate MD

Overlay443

CompanyVOIP/PBX

SIte...


331

An outline of the equipment setup in both offices is shown below.

IPNetwork

DigitalVoIP

200.2.10.3

T1

717-5170

PBX

-5171

-5172

-5173

-5174

Company HQ.MinneapolisNorth Sub.area 763

Local-CallArea Codes:

612, 651,952

325-7001

Router

DigitalVoIP

200.2.9.7

T1

BaltimoreSales Ofc.area 410

PBX

OverlayArea Code:

443

-7002

-7003


332

The screen below shows Outbound PhoneBook entries for the VOIP located inthe company’s Baltimore facility.

The entries in the Minneapolis VOIP’s Inbound PhoneBook match theOutbound PhoneBook entries of the Baltimore VOIP, as shown below.


333

To call the Minneapolis/St. Paul area, a Baltimore employee must dial elevendigits. (In this case, we are assuming that the Baltimore PBX does not requirean “8” or “9” to seize an outside phone line.)

If a Baltimore employee dials any phone number in the 612 area code, the callwill automatically be handled by the company’s voip system. Upon receivingsuch a call, the Minneapolis voip will remove the digits “1612”. But before thesuburban-Minneapolis voip can complete the call to the PSTN of theMinneapolis local calling area, it must dial “9” (to get an outside line from thePBX) and then a comma (which denotes a pause to get a PSTN dial tone) andthen the 10-digit phone number which includes the area code (612 for the cityof Minneapolis; which is different than the area code of the suburb where thePBX is actually located -- 763).

A similar sequence of events occurs when the Baltimore employee callsnumber in the 651 and 952 area codes because number in both of these areacodes are local calls in the Minneapolis/St. Paul area.

The simplest case is a cal from Baltimore to a phone within theMinneapolis/St. Paul area code where the company’s voip and PBX arelocated, namely 763. In that case, that local voip removes 1763 and dials 9 todirect the call to its local 7-digit PSTN.

Finally, consider the longest entry in the Minneapolis Inbound Phonebook,“17637175. Note that the main phone number of the Minneapolis PBX is 763-717-5170. The destination pattern 17637175 means that all calls toMinneapolis employees will stay within the suburban Minneapolis PBX andwill not reach or be carried on the local PSTN.


334

Similarly, the Inbound PhoneBook for the Baltimore VOIP (shown first below)generally matches the Outbound PhoneBook of the Minneapolis VOIP (shownsecond below).

Notice the extended prefix to be removed: 14103257. This entry allowsMinneapolis users to contact Baltimore co-workers as though they were in theMinneapolis facility, using numbers in the range 7000 to 7999.

Note also that a comma (as in the entry 9,443) denotes a delay in dialing. Aone-second delay is commonly used to allow a second dial tone to be generatedfor calls going outside of the facility’s PBX system.


335

The Outbound PhoneBook for the Minneapolis VOIP is shown below. Thethird destination pattern, “7” facilitates reception of co-worker calls usinglocal-appearing-extensions only. In this case, the “Add Prefix” field value forthis phonebook entry would be “1410325” .


336

Configuring Mixed Digital/Analog VOIP SystemsThe MVP2410 digital MultiVOIP unit is compatible with analog VOIPs. Inmany cases, digital and analog VOIP units will appear in the sametelephony/IP system. In addition to MVP-210/410/810 MultiVOIP units(Series II units), legacy analog VOIP units (Series I units made by MultiTech)may be included in the system, as well. When legacy VOIP units are included,the VOIP administrator must handle two styles of phonebooks in the sameVOIP network. The diagram below shows a small-scale system of this kind:one digital VOIP (the MVP2410) operates with two Series II analog VOIPs(an MVP210 and an MVP410), and two Series I legacy VOIPs (two MVP200units).

IPNetwork

PBXDigitalVoIP

MVP2410

200.2.9.9

T1

PSTN

Site D:Pierre, SD

Area Code 615

615-492-3100

EXAMPLE:Digital & Analog VOIPs

in Same System

Router

Other extensionsx3101 - x3199

Unit#200

201

FXSCH1

Series #1 Analog MultiVOIP(Server/Client Phonebook)

MVP200

200.2.9.7Client

Site A:Bismarck, ND

Area Code 701

200.2.9.8Host

(Holds phonebook for bothSeries #1 analog VOIPs.)

Site B:Rochester, MNArea Code 507

CH2 FXS

717-5000FXO

Unit#100


MVP200

507-717-5662Site C:

Suburban Rochester

CH1

FXS

Site E:Cheyenne, WYArea Code 307

KeySystem

402-263-7400


CO Port

FXS Ports

CO Ports

FXS Port

PSTN200.2.9.5

200.2.9.6

FXO

PSTN

102

Site F:Lincoln, NE

Area Code 402

421

Port #4

CH1

Series #2 Analog MultiVOIPMVP210



337

The Series I analog VOIP phone book resides in the “Host” VOIP unit at SiteB. It applies to both of the Series I analog VOIP units.

Each of the Series II analog MultiVOIPs (the MVP210 and the MVP410)requires its own inbound and outbound phonebooks. The MVP2410 digitalMultiVOIP requires its own inbound and outbound phonebooks, as well.


338

These seven phone books are shown below.

Phone Book for Series I Analog VOIP Host Unit (Site B)

VOIP Dir #-OR-DestinationPattern

IP Address Channel Comments

102 200.2.9.8 2 Site B, FXS channel.

101 200.2.9.8 1 Site B, FXO channel.

421 200.2.9.6 0 Site E FXS channel.

201 200.2.9.7 1 Site A, FXS channel.

1615xxxxxxx

200.2.9.9 0(Note 2.)

Gives remote voipusers access to localPSTN of Site D(Pierre, SD, area code615).

3xxx

(Note 1.)

200.2.9.9 0 Allows remote voipusers to call all PBXextensions at Site D(Pierre, SD) usingonly four digits.

1402 200.2.9.5 0 Gives remote voipusers access to localPSTN of Site F(Lincoln, NE; areacode 402).

140226374(Note 1)(Note 3)

200.2.9.5 0 Gives remote voipusers access to keyphone systemextensions at Site F(Lincoln).


339

Note 1. The “x” is a wildcard character.

Note 2. By specifying “Channel 0,” we instruct theMVP2400/2410 to choose any available data channel tocarry the call.

Note 3. Note that Site F key system has only 30 extensions(x7400-7429). This destination pattern (140226374)actually directs calls to 402-263-7430 through402-263-7499 into the key system, as well.This means that such calls, which belong on the PSTN,cannot be completed. In some cases, this might beinconsequential because an entire exchange (fully usedor not) might have been reserved for the company or itmight be unnecessary to reach those numbers. However,to specify only the 30 lines actually used by the keysystem, the destination pattern 140226374 would have tobe replaced by three other destination patterns, namely1402263740, 1402263741, and 1402263742. In thisway, calls to 402-263-7430 through 402-263-7499would be properly directed to the PSTN. In the Site Doutbound phonebook, the 30 lines are defined exactly,that is, without making any adjacent phone numbersunreachable through the voip system.


340

Outbound Phone Book for MVP2410 Digital VOIP(Site D)

Destin.Pattern

RemovePrefix

AddPrefix

IPAddress

Comment

201 200.2.9.7 To originate calls toSite A (Bismarck).

1507 1507 101#Note 3.

200.2.9.8 To originate calls toRochester localPSTN using theFXO channel(channel #1) of theSite B VOIP.

102 200.2.9.8 To originate calls tophone connected toFXS port (channel#2) of the Site BVOIP.

421 200.2.9.6 Calls to Site E(Cheyenne).

1402 200.2.9.5 Calls to Lincoln arealocal PSTN (viaFXO channel, CH4,of the Site F VOIP).

1402263740

200.2.9.5

1402263741

200.2.9.5

1402263742

200.2.9.5

Calls to extensions(thirty) of keysystem at Site F(Lincoln). Humanoperator or auto-attendant is neededto complete thesecalls.

Note 3. The pound sign (“#”) is a delimiter separating the VOIPnumber from the standard telephony phone number.


341

Inbound Phonebook for MVP2410 Digital VOIP (Site D)RemovePrefix

AddPrefix

ChannelNumber

Comment

1615 9,Note 4.Note 5.

0 Allows phone users at remote voipsites to call non-toll numberswithin the Site D area code (615;Pierre, SD) over the VOIPnetwork.

161549231

31 0 Allows voip calls directly toemployees at Site D (at extensionsx3101 to x3199).

Note 4. “9” gives PBX station users access to outside line.

Note 5. The comma represents a one-second pause, the timerequired for the user to receive a dial tone on the outsideline (PSTN). The comma is only allowed in the Inboundphonebook.


342

Outbound Phone Book for MVP410 Analog VOIP(Site F)

Destin.Pattern

RemovePrefix

AddPrefix

IPAddress

Comment

201 200.2.9.7 To originate calls toSite A (Bismarck).

1507 1507 101#Note 3.

200.2.9.8 To originate calls toany PSTN phone inRochester area usingthe FXO channel(channel #1) of theSite B VOIP.

102 200.2.9.8 To originate calls tophone connected toFXS port (channel#2) of the Site BVOIP (Rochester).

421 200.2.9.6 Calls to Site E(Cheyenne).

1615 200.2.9.9 Calls to Pierre areaPSTN via Site DPBX.

31 1615492

200.2.9.9 Calls to Pierre PBXextensions with fourdigits.



343

Inbound Phonebook for MVP410 Analog VOIP (Site F)RemovePrefix

AddPrefix

ChannelNumber

Comment

1402 4 Access to Lincoln local PSTN byusers at remote VOIP locations viaFXO port at Site F.

1402263740

740 0

1402263741

741 0

1402263742

742 0

Gives remote voip users access toextension of key phone system atSite F (Lincoln). Because call iscompleted at key system,abbreviated dialing (4 digits) is notworkable. Human operator orauto-attendant is needed tocomplete these calls.


344

Outbound Phone Book for MVP210 Analog VOIP(Site E)

Destin.Pattern

RemovePrefix

AddPrefix

IPAddress

Comment

201 200.2.9.7 To originate calls toSite A.

1507 1507 101#Note 3.

200.2.9.8 To originate calls toany PSTN phone inRochester area usingthe FXO channel(channel #1) of theSite B VOIP.

102 200.2.9.8 To originate calls tophone connected toFXS port (channel#2) of the Site BVOIP.

1402 200.2.9.5 Calls to Lincoln areaPSTN (via FXOchannel, CH4, of theSite F VOIP).

7 1402263

200.2.9.5 Calls to Lincoln keyextensions with fourdigits.

1615 200.2.9.9 Calls to Pierre areaPSTN via Site DPBX.

31 1615492

200.2.9.9 Calls to Pierre PBXextensions with fourdigits.



345

Inbound Phonebook for MVP210 Analog VOIP (Site E)RemovePrefix

AddPrefix

ChannelNumber

Comment

421 1

Call Completion Summaries

Site A calling Site C, Method 1

1. Dial 101.

2. Hear dial tone from Site B.

3. Dial 7175662.

4. Await completion. Talk.


1. Dial 101#7175662


Note: Some analog VOIP gateways will allowcompletion by Method 2. Others will not.

Site C calling Site A

1. Dial 7175000.

2. Hear dial tone from Site B VOIP.

3. Dial 201.



346

Site D calling Site C

1. Dial 9,15077175662.

2. “9” gets outside line. On some PBXs, an “8” may be used to directcalls to the VOIP, while “9” directs calls to the PSTN. However,some PBX units can be programmed to identify the destinationpatterns of all calls to be directed to the VOIP.

3. PBX at Site D is programmed to divert all calls made to the 507 areacode and exchange 717 into the VOIP network. (It would also bepossible to divert all calls to all phones in area code 507 into the VOIPnetwork, but it may not be desirable to do so.)

4. The MVP2410 removes the prefix “1507” and adds the prefix “101#”for compatibility with the analog MultiVOIP’s phonebook scheme.The “#” is a delimiter separating the analog VOIP’s phone numberfrom the digits that the analog VOIP must dial onto its local PSTN tocomplete the call. The digits “101#7175662” are forwarded to the SiteB analog VOIP.

5. The call passes through the IP network (in this case, the Internet).

6. The call arrives at the Site B VOIP. This analog VOIP receives thisdialing string from the MVP2410: 101#7175662. The analog VOIP,seeing the “101” prefix, uses its own channel #1 (an FXO port) toconnect the call to the PSTN. Then the analog VOIP dials its localphone number 7175662 to complete the call.


347

Site D calling Site FA voip call from Pierre PBX to extension 7424 on the key telephone system in Lincoln,Nebraska.

A. The required entry in the Pierre Outbound Phonebook to facilitate origination ofthe call, would be 1402263742. The call would be directed to the Lincoln voip’s IPaddress, 200.2.9.5.(Generally on such a call, the caller would have to dial an initial “9.” But typicallythe PBX would not pass the initial “9” to the voip. If the PBX did pass along that “9”however, its removal would have to be specified in the local Outbound Phonebook.)

B. The corresponding entry in the Lincoln Inbound Phonebook to facilitatecompletion of the call would be

1402263742 for calls within the office at Lincoln

1402 for calls to the Lincoln local calling area (PSTN).

Call Event Sequence

1. Caller at Pierre dials 914022637424.

2. Pierre PBX removes “9” and passes 14022637424 to voip.

3. Pierre voip passes remaining string, 14022637424 on to the Lincoln voipat IP address 200.2.9.5.

4. The dialed string matches an inbound phonebook entry at the Lincolnvoip, namely 1402263742.

5. The Lincoln voip rings one of the three FXS ports connected to theLincolnkey phone system.

6. The call will be routed to extension 7424 either by a human receptionist/operator or to an auto-attendant (which allows the caller to specify theextension to which they wish to be connected).


348

Site F calling Site DA voip call from a Lincoln key extension to extension 3117 on the PBX in Pierre, SouthDakota.

A. The required entry in the Lincoln Outbound Phonebook to facilitate origination ofthe call, would be “31”. The string “1615492” would have to be added as a prefix.The call would be directed to the Pierre voip’s IP address, 200.2.9.9.

B. The corresponding entry in the Pierre Inbound Phonebook to facilitate completionof the call would be 1615492.

1. Caller at Lincoln picks up phone receiver, presses button on key phoneset. This button has been assigned to a particular voip channel (any one ofthe three FXS ports).

2. The caller at Lincoln hears dial tone from the Lincoln voip.

3. The caller at Lincoln dials 3117.

4. The Lincoln voip adds the prefix 1615492 and sends the entire dialingstring, 16154923117, to the Pierre voipat IP address 200.2.9.9.

5. The Pierre voip matches the called digits 16154923117 to its InboundPhonebook entry “1615492” .

6. The Pierre PBX dials extension 3117 in the office at Pierre.

Variations in PBX CharacteristicsThe exact dialing strings needed in the Outbound and Inbound Phonebooks ofthe MVP2410 will depend on the capabilities of the PBX. Some PBXs requiretrunk access codes (like an “8” or “9” to access an outside line or to access theVOIP network). Other PBXs can automatically distinguish between intra-PBXcalls, PSTN calls, and VOIP calls.Some PBX units can also insert digits automatically when they receive certaindialing strings from a phone station. For example, a PBX may beprogrammable to insert automatically the three-digit VOIP identifier stringsinto calls to be directed to analog VOIPs.The MVP2410 offers complete flexibility for inter-operation with PBX unitsso that a coherent dialing scheme can be established to connect a company’smultiple sites together in a way that is convenient and intuitive for phone users.When working together with modern PBX units, the presence of the MVP2410can be completely transparent to phone users within the company.

349

Chapter 8: E1 PhonebookConfiguration

(European Telephony Standards)

E1 Phonebook Configuration MultiVOIP User Guide

350

MVP3010 Inbound and Outbound MultiVOIPPhonebooks

ImportantDefinition:

The MultiVOIP’s Outbound phonebooklists the phone stations it can call;its Inbound phonebook describes thedialing sequences that can be used tocall that MultiVOIP and how those callswill be directed.

When a VOIP serves a PBX system, the operation of the VOIP should betransparent to the telephone end user and savings in long-distance callingcharges should be enjoyed. Use of the VOIP should not require the dialing ofextra digits to reach users elsewhere on the VOIP network. On the contrary,VOIP service more commonly reduces dialed digits by allowing users (servedby PBXs in facilities in distant cities) to dial their co-workers with 3-, 4-, or 5-digit extensions -- as if they were in the same facility. More importantly, theVOIP system should be configured to maximize savings in long-distancecalling charges. To achieve both of these objectives, ease of use andmaximized savings, the VOIP phonebooks must be set correctly.

NOTE: VOIPs are commonly used foranother reason, as well: VOIPsallow an organization to integratephone and data traffic onto a singlenetwork. Typically these are privatenetworks.

MultiVOIP User Guide E1 PhoneBook Configuration

351

Free Calls: One VOIP Site to AnotherThe most direct use of the VOIP system is making calls between the officeswhere the VOIPs are located. Consider, for example, the Wren ClothingCompany. This company has VOIP-equipped offices in London, Paris, andAmsterdam, each served by its own PBX. VOIP calls between the threeoffices completely avoid international long-distance charges. These calls arefree. The phonebooks can be set up to allow all Wren Clothing employees tocontact each other using 3-, 4-, or 5-digit numbers, as though they were all inthe same building.

France

TheNetherlands

Wren Clothing Co.VOIP/PBX Site

LondonWren Clothing Co.

VOIP/PBX SiteAmsterdam


Paris

United Kingdom

Free VOIP Calls


352

Local Rate Calls: Within Local Calling Area of RemoteVOIP

In the second use of the VOIP system, the local calling area of each VOIPlocation becomes accessible to all of the VOIP system’s users. As a result,international calls can be made at local calling rates. For example, supposethat Wren Clothing buys its zippers from The Bluebird Zipper Company in thewestern part of metropolitan London. In that case, Wren Clothing personnel inboth Paris and Amsterdam could call the Bluebird Zipper Company withoutpaying international long-distance rates. Only London local phone rates wouldbe charged. This applies to calls completed anywhere in London’s localcalling area (which includes both Inner London and Outer London). Generally,local calling rates apply only within a single area code, and, for all calls outsidethat area code, national rates apply. There are, however, some European caseswhere local calling rates extend beyond a single area code. Local ratesbetween Inner and Outer London are one example of this. (It is also possible,in some locations, that calls within an area code may be national calls. But thisis rare.)

France

TheNetherlands


LondonWren Clothing Co.

VOIP/PBX SiteAmsterdam


Paris

United Kingdom

Calls at London local rates

Local Calling Area

Bluebird Zipper Co.London


353

Similarly, the VOIP system allows Wren Clothing employees in London andAmsterdam to call anywhere in Paris at local rates; it allows Wren Clothingemployees in Paris and London to call anywhere in Amsterdam at local rates.

France

TheNetherlands


London


Amsterdam


Paris

United Kingdom

Calls at Paris local rates

Local Calling Areas

Calls at Amsterdam local rates


354

National Rate Calls: Within Nation of Remote VOIP SiteIn the third use of the VOIP system, the national calling area of each VOIPlocation becomes accessible to all of the VOIP system’s users. As a result,international calls can be made at national calling rates. Again, significantsavings are possible. For example, suppose that the Wren Clothing Companybuys its buttons from the Chickadee Button Company in the Dutch city ofRotterdam. In that case, Wren Clothing personnel in both London and Pariscould call the Chickadee Button Company without paying international long-distance rates; only Dutch national calling rates would be charged. Thisapplies to calls completed anywhere in The Netherlands.

France

TheNetherlands


London Wren Clothing Co.VOIP/PBX Site

Amsterdam


Paris

United Kingdom

Calls at DutchNational Rates

Chickadee Button Co.Rotterdam


355

Similarly, the VOIP system allows Wren Clothing employees in London andAmsterdam to call anywhere in France at French national rates; it allows WrenClothing employees in Paris and Amsterdam to call anywhere in the UnitedKingdom at its national rates.

France

TheNetherlands


London Wren Clothing Co.VOIP/PBX Site

Amsterdam


Paris

United Kingdom

Calls at FrenchNational Rates

Calls at UKNational Rates

Inbound versus Outbound PhonebooksTo make the VOIP system transparent to phone users and to allow all possiblefree and reduced-rate calls, the VOIP administrator must configure the“Outbound” and “Inbound” phone-books of each VoIP in the system.

The “Outbound” phonebook for a particular VOIP unit describes the dialingsequences required for a call to originate locally (typically in a PBX in aparticular facility) and reach any of its possible destinations at remote VOIPsites, including calls terminating at points beyond the remote VOIP site.

The “Inbound” phonebook for a particular VOIP unit describes the dialingsequences required for a call to originate remotely from any other VOIP sitesin the system, and to terminate on that particular VOIP.

Briefly stated, the MultiVOIP’s Outbound phonebook lists the phone stations itcan call; its Inbound phonebook lists the dialing sequences that can be used tocall that MultiVOIP. (Of course, the phone numbers are not literally “listed”individually.) The phone stations that can originate or complete calls over theVOIP system are described by numerical rules called “destination patterns.”These destination patterns generally consist of country codes, area codes orcity codes, and local phone exchange numbers.

In order for any VOIP phone call to be made, there must be both an InboundPhonebook entry and an Outbound Phonebook entry that describe the end-to-end connection. The phone station originating the call must be connected to


356

the VOIP system. The Outbound Phonebook for that VOIP unit must have adestination pattern entry that includes the ‘called’ phone (that is, the phonecompleting the call). The Inbound Phonebook of the VOIP where the call iscompleted must have a destination pattern entry that includes the digitsequence dialed by the originating phone station.

The PhoneBook Configuration procedure below is brief, but it is followed byan example case. For many people, the example case may be easier to graspthan the procedure steps. Configuration is not difficult, but all phone numbersequences, destination patterns, and other information must be entered exactly;otherwise connections will not be made.


357

Phonebook configuration screens can be accessed using icons or the sidebarmenu.

Phonebook Icons Description

Phonebook Configuration

Inbound Phonebook EntriesList

Add Inbound PhonebookEntry

Edit selected InboundPhonebook Entry

Outbound Phonebook EntriesList

Add Outbound PhonebookEntry

Edit selected OutboundPhonebook Entry


358

Phonebook Sidebar Menu


359

Phonebook Configuration Procedure1. Go to the PhoneBook Configuration screen (using either the sidebar

menu, drop-down menu, or icon).


360


361

In consultation with your VOIP administrator, enter the Gateway Name andvalues for Q.931 parameters and Gatekeeper RAS parameters. Determinewhether your voip system will operate with a proxy server. Determinewhich H.323 version 4 functions you will implement. (They are not alwaysapplicable. See field description for each parameter.) If the SPP protocol isused, values for another group of parameters must be specified, as well.


362

The table below describes all fields in the PhoneBook Configuration screen.

PhoneBook Configuration Parameter Definitions


Gateway Name Y/N This field allows you to specify a name for thisMultiVOIP. When placing a call, this name issent to the remote MVP3000 for display in CallProgress listings, Logs, etc.

Q.931 ParametersUse Fast Start Y/N Enables the H.323 Fast Start procedure. May

need to be enabled/disabled for compatibilitywith third-party VOIP gateways.

Call SignalingPort

portnumber

Default: 1720 (H.323)

GateKeeper RASParameters

Gatekeeper /Clear ChannelIP Address

IP address of the GateKeeper.

Port Number Well-known port number for GateKeepers.Must match port number of GateKeeper, 1719.

Gateway Prefix This number becomes registered with theGateKeeper. Call requests sent to the gatekeeperand preceded by this prefix will be routed to theVOIP gateway.

GatekeeperName

alpha-numericstring

Optional. The name of the GateKeeper withwhich this MultiVOIP is trying to register.

Gateway H.323ID

The H.323 ID is used to register this particularMultiVOIP with the GateKeeper. H.323 ID is analias entry sent to the GateKeeper, made ofalphanumeric characters. For NetMeetingendpoints, numbers are preferred over letters.The H.323 ID identifies the IP calling sequencethat the GateKeeper must ‘dial’ to contact theremote VOIP.


363


Field Name Values DescriptionSIP Proxy Parameters

Enable Proxy Y/N

Proxy Server IPAddress

n.n.n.nwheren=0-255

Network address of the proxyserver that the voip is using.

Port Number Logical port number for proxycommunications.

User Name Identifier used when proxy serveris used in network. If a proxyserver is used in a SIP voipnetwork, all clients must enterboth a User Name and a Passwordbefore being allowed to make acall.

Password Password for proxy serverfunction. Password for proxyserver function. See “UserName” description above.


364

PhoneBook Configuration Parameter Definitions(cont’d)Field Name Values DescriptionH.323 Version 4 Parameters

Q.931Multiplexing(Mux)

Y/N Signaling for multiple phone callscan be carried on a single portrather than opening a separatesignaling port for each call. Thisconserves bandwidth resources.

H.245Tunneling (Tun)

Y/N H.245 messages are encapsulatedwithin the Q.931 call-signalingchannel. Among other things, theH.245 messages let the twoendpoints tell each other whattheir technical capabilities are anddetermine who, during the call,will be the client and who theserver. Tunneling is the process oftransmitting these H.245 messagesthrough the Q.931 channel. Thesame TCP/IP socket (or logicalport) already being used for theCall Signaling Channel is thenalso used by the H.245 ControlChannel. This encapsulationreduces the number of logicalports (sockets) needed andreduces call setup time.


365


Field Name Values DescriptionH.323 Version 4 Parameters

Parallel H.245(FS + Tun)

Y/N FS (Fast Start or Fast Connect) isa Q.931 feature of H.323v2 tohasten call setup as well as ‘pre-opening’ the media channelbefore the CONNECT message issent. This pre-opening is arequirement for certain billingactivities. Under Parallel H.245FS + Tun, this Fast Connectfeature can operatesimultaneously with H.245Tunneling (see descriptionabove).

Annex –E (AE) Y/N Multiplexed UDP call signalingtransport. Annex E is helpful forhigh-volume voip systemendpoints. Gateways with lesservolume can afford to use TCP toestablish calls. However, forlarger volume endpoints, the callsetup times and system resourceusage under TCP can becomeproblematic. Annex E allowsendpoints to perform call-signaling functions under the UDPprotocol, which involvessubstantially streamlinedoverhead. (This feature shouldnot be used on the public Internetbecause of potential problemswith security and bandwidthusage.)


366



Mode Direct,Client, orRegistrar

SPP voip systems can operate intwo modes:in the direct mode, where allvoip gateways have static IPaddresses assigned to them; or inthe registrar/client mode, whereone voip gateway serves asregistrar and all other gateways,being its clients, point to thatregistrar. The registrar assigns IPaddresses dynamically.

General Options

Port The UDP port on which datatransmission will occur. Eachclient voip has its own port. Iftwo client voips are both behindthe same firewall, then they musthave different ports assigned tothem.If there are two clients and each isbehind a different firewall, thenthe clients could have differentport numbers or the same portnumber.(Default port number = 10000.)

Re-trans-mission(in ms)

If packets are lost (as indicated byabsence of an acknowledgment)then the endpoint will retransmitthe lost packets after thisdesignated time duration haselapsed. (Default value = 2000milliseconds.)

MaxRe-trans-mission

Number of times the voip will re-transmit a lost packet (if noacknowledgment has beenreceived). (Default value = 3)


367



[cont’d]

Client Options Client Option fields are activeonly in registrar/client mode andonly for client voip units.

Registrar IPAddress

This is the IP address of theregistrar voip to which this clientis assigned. (Default value =0.0.0.0; effectively, there is nouseful default value.)

Registrar Port This is the port number of theregistrar voip to which this clientis assigned. (Default port number= 10000.)

Registrar Options Registrar Option fields are activeonly in registrar/client mode andonly for registrar voip units.

Keep Alive(in sec.)

Time-out duration before aregistrar will unregister a clientthat does not send its “I’m here”signal. Client normally sends its“I’m here” signal every 20seconds. Timeout default = 60seconds.


368

2. Select PhoneBook Modify and then selectOutbound Phone Book/List Entries.

Click Add.


369

3. The Add/Edit Outbound PhoneBook screen appears.

Enter Outbound PhoneBook data for your MVP3010. Note that the Advancedbutton gives access to the Alternate IP Routing feature, if needed. Alternate IPRouting can be implemented in a secondary screen (as described after theprimary screen field definitions below).


370

The fields of the Add/Edit Outbound Phone Book screen are described in thetable below.



DestinationPattern

prefixes,area codes,exchanges,line numbers,extensions

Defines the beginning of dialingsequences for calls that will beconnected to another VOIP inthe system. Numbers beginningwith these sequences arediverted from the PTSN andcarried on Internet or other IPnetwork.

Total Digits as needed number of digits the phone usermust dial to reach specifieddestination

Remove Prefix dialed digits portion of dialed number to beremoved before completing callto destination

Add Prefix dialed digits digits to be added beforecompleting call to destination

IP Address n.n.n.nfor = 0-255

the IP address to which the callwill be directed if it begins withthe destination pattern given

Description alpha-numeric Describes the facility orgeographical location at whichthe call will be completed.

Protocol Type SIP, H.323, orSPP

Indicates protocol to be used inoutbound transmission.


371

Add/Edit Outbound Phone Book: Field Definitions(cont’d)


H.323 fieldsUse Gatekeepr Y/N Indicates whether or not

gatekeeper is used.

H.323 ID The H.323 ID assigned to thedestination MultiVOIP. Onlyvalid if “Use Gatekeeper” isenabled for this entry.


Q.931 PortNumber Q.931Port Number

1720 Q.931 is the call signalingprotocol for setup andtermination of calls (aka ITU-TRecommendation I.451). H.323employs only one “well-known”port (1720) for Q.931 signaling.If Q.931 message-orientedsignaling protocol is used, theport number 1720 must bechosen.


372


(cont’d)Field Name Values DescriptionSIP FieldsUse Proxy Y/N Select if proxy server is used.

TransportProtocol

TCP or

UDP

Voip administrator must choosebetween UDP and TCP transmissionprotocols. UDP is a high-speed,low-overhead connectionlessprotocol where data is transmittedwithout acknowledgment,guaranteed delivery, or guaranteedpacket sequence integrity. TCP isslower connection-oriented protocolwith greater overhead, but havingacknowledgment and guaranteesdelivery and packet sequenceintegrity.

SIP PortNumber

5060 or other

*See RFC3087(“Control ofService Contextusing SIPRequest-URI,”by the NetworkWorkingGroup).

The SIP Port Number is aUDP logical port number.The voip will “listen” for SIPmessages at this logical port.If SIP is used, 5060 is thedefault, standard, or “wellknown” port number to beused. If 5060 is not used,then the port number used isthat specified in the SIPRequest URI (UniversalResource Identifier).

SIP URL sip.userphone@hostserver,where“userphone” isthe telephonenumber and“hostserver” isthe domainname or anaddress on thenetwork

Looking similar to an emailaddress, a SIP URLidentifies a user's address.In SIP communications, eachcaller or callee is identifiedby a SIP url:sip:user_name@host_name.The format of a sip url is verysimilar to an email address,except that the “sip:“ prefix isused.


373

Add/Edit Outbound Phone Book: Field Def’ns (cont’d)Field Name Values DescriptionSPP FieldsUse Registrar Values: Y/N

Description: Select this checkbox to use registrarwhen voip system is operating in the “Registrar/Client”SPP mode. In this mode, one voip (the registrar, as set inPhonebook Configuration screen) has a static IP addressand all other voips (clients) point to the registar’s IPaddress as functionally their own. However, if yourvoip system overall is operating in “Registrar/Client”mode but you want to make an exception and use Directmode for the destination pattern of this particularAdd/Edit Phonebook entry, leave this checkboxunselected.Leave this checkbox unselected if your overall voipsystem is operating in the “Direct" SPP mode. In thismode, all voips in system are peers and each has its ownstatic IP address.

Port Number Values: numericDescription: When operating in “Registrar/Client”mode, this is the port by which the gateway receives allSPP data and control messages from the registrargateway. (This ability to receive all data and messagesvia one port allows the voip to operate behind a firewallwith only one port open.)When operating in “Direct” mode, this is the Port bywhich peer voips receive data and messages.

Alternate PhoneNumber

numeric Phone number associated withalternate IP routing.

Advancedbutton

Values: N/A

Description: Gives access to secondary screenwhere an Alternate IP Route can be specified forbackup or redundancy of signal paths. Seediscussion on next page. For SIP & H.323operation only.


374

Clicking on the Advanced button brings up the Alternate Routing secondary screen.This feature provides an alternate path for calls if the primary IP network cannot carrythe traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN.However, this feature could also be used to divert traffic to a redundant (backup) unitin case one voip unit fails. The user must specify the IP address of the alternate routefor each destination pattern entry in the Outbound Phonebook.


375

Alternate Routing Field Definitions

FieldName

Values Description

AlternateIPAddress

n.n.n.nwheren= 0-255

Alternate destination for outbound data traffic incase of excessive delay in data transmission.

RoundTripDelay

milliseconds The Round Trip Delay is the criterion for judgingwhen a data pathway is considered blocked. Whenthe delay exceeds the threshold specified here, thedata stream will be diverted to the alternatedestination specified as the Alternate IP Address.

4. Select PhoneBook Modify and then select Inbound PhoneBook/ListEntries.


376

5. The Add/Edit Inbound PhoneBook screen appears.

Enter Inbound PhoneBook data for your MVP3010. The fields of the Add/EditInbound PhoneBook screen are described in the table below.

Add/Edit Inbound Phone Book: Field Definitions

FieldName

Values Description

RemovePrefix

dialed digits portion of dialed number to be removed beforecompleting call to destination(often a local PBX)

AddPrefix

dialed digits digits to be added before completing call todestination(often a local PBX)


377

Add/Edit Inbound Phone Book: Field Definitions(cont’d)

Field Name Values DescriptionChannelNumber

1-30, or“Hunting”

E1 channel number to which thecall will be assigned as it entersthe local telephony equipment(often a local PBX). “Hunting”directs the call to any availablechannel.

Description -- Describes the facility orgeographical location at whichthe call originated.

Call Forward ParametersEnable Y/N Click the check-box to enable

the call-forwarding feature.

ForwardCondition

Uncondit.;BusyNo Resp.

Unconditional. When selected,all calls received will beforwarded.Busy. When selected, calls willbe forwarded when station isbusy.No Response. When selected,calls will be forwarded if calledparty does not answer after aspecified number of rings, asspecified in Ring Count field.

Forwarding can be conditionedon both “Busy” and “NoResponse.”

ForwardDestinationIP address,phone number,port number,etc.

Phone number or IP address to which calls will bedirected.

For H.323 calls, the Forward Destination can beeither a Phone Number of an IP Address.

For SIP calls, the Forward Destination can be oneof the following:(a) phone number, (b) IP address,(c) IP address: port number,(d) phone number:IP addr: port number,(e) SIP URL, or (f) phone #: IP address.


378

Add/Edit Inbound Phone Book: Field Definitions(cont’d)

Field Name Values DescriptionRing Count integer When No Response is

condition for forwardingcalls, this determines howmany unanswered rings areneeded to trigger theforwarding.

RegistrationOptionParameters

In an H.323 voip system, gateways can registerwith the system using one of these identifiers: (a)an E.164 identifier, (b) a Tech Prefix identifier, or(c) an H.323 ID identifier.

In a SIP voip system, gateways can register withthe SIP Proxy.

In an SPP voip system, gateways can register withthe SPP Registrar voip unit.

6. When your Outbound and Inbound PhoneBook entries are completed, clickon Save Setup in the sidebar menu to save your configuration.

You can change your configuration at any time as needed for your system.

Remember that the initial MVP3010 setup must be done locally or via thebuilt-in Remote Configuration/Command Modem using the MultiVOIPprogram. However, after the initial configuration is complete, all of theMVP3010 units in the VOIP system can be configured, re-configured, andupdated from one location using the MultiVOIP web GUI software program orthe MultiVOIP program (in conjunction with the built-in modem).


379

E1 Phonebook ExamplesTo demonstrate how Outbound and Inbound PhoneBook entries work in aninternational VOIP system, we will re-visit our previous example in greaterdetail. It’s an international company with offices in London, Paris, andAmsterdam. In each office, a MVP3010 has been connected to the PBXsystem.

3 Sites, All-E1 ExampleThe VOIP system will have the following features:

1. Employees in all cities will be able tocall each other over the VOIP systemusing 4-digit extensions.

2. Calls to Outer London and InnerLondon, greater Amsterdam, and greaterParis will be accessible to all companyoffices as local calls.

3. Vendors in Guildford, Lyon, andRotterdam can be contacted as nationalcalls by all company offices.

Note that the phonebook entries for Series II analog MultiVOIP used in Euro-type telephony settings will be the same in format as entries for the MVP3010.


380

LilleReims

Strasbourg

Paris: Area 01

Rouen

Nantes

Lyon

Marseille

Toulouse

Bordeaux

France Country Code: 33


381

010Rotterdam

020 Amsterdam

050Groningen

040Eindhoven

0118Middelburg

038 Zwolle

058Leeuwarden

053Enschede070

The Hague

Haarlem 023

0294 Weesp

Beverwijk 0251

Den Helder 0223

0299 Purmerend

Aalsmeer0297

Texel 0222

043Maastricht

The NetherlandsCountry Code: 31

026Arnhem


382

An outline of the equipment setup in these three offices is shown below.

IPNetwork

DigitalVoIP

200.2.10.3

E1

979-5170

PBX

-5171

-5172

-5173

-5174

Wren Clothing Co.London OfficeCountry Code: +44Area Code: 0208

PBX

Wren Clothing Co.Amsterdam OfficeCountry Code: +31Area/City Code: 020

-4804

-4803

-4802

-4801

688-4800

200.2.8.5E1

Router

DigitalVoIP

200.2.9.7

E1

Wren Clothing Co.Paris OfficeCountry Code: +33Area Code: 01

PBX

-29 83

-29 82

74 71 29 81

DigitalVoIP


383

The screen below shows Outbound PhoneBook entries for the VOIP located in thecompany’s London facility

The Inbound PhoneBook for the London VOIP is shown below.

NOTE: Commas are allowed in the Inbound Phonebook, but not in theOutbound Phonebook. Commas denote a brief pause for a dial tone,allowing time for the PBX to get an outside line.


384

The screen below shows Outbound PhoneBook entries for the VOIP located inthe company’s Paris facility.

The Inbound PhoneBook for the Paris VOIP is shown below.


385

The screen below shows Outbound PhoneBook entries for the VOIP in thecompany’s Amsterdam facility.

The Inbound PhoneBook for the Amsterdam VOIP is shown below.


386

Configuring Digital & Analog VOIPs in Same System

The MVP3010 digital MultiVOIP unit is compatible with analog VOIPs. Inmany cases, digital and analog VOIP units will appear in the sametelephony/IP system. In addition to MVP-210/410/810 MultiVOIP units(Series II units), legacy analog VOIP units (Series I units made by MultiTech)may be included in the system, as well. When legacy VOIP units are included,the VOIP administrator must handle two styles of phonebooks in the sameVOIP network. The diagram below shows a small-scale system of this kind:one digital VOIP (the MVP3010) operates with two Series II analog VOIPs(an MVP210 and an MVP410), and two Series I legacy VOIPs (two MVP200units).

IPNetwork

PBXDigitalVoIP

MVP3010

200.2.9.9

PSTN

Router

Unit#200

201

FXSCH1


MVP200

200.2.9.7Client

200.2.9.8Host

(Holds phonebook for bothSeries #1 analog VOIPs.)

CH2 FXS

FXO

Unit#100


MVP200

CH1

CH1FXS

Site E:Carlisle, UK

Area Code 0122 8

KeySystem

263-7400


CO Port

FXS Ports

CO Ports

FXS Port

PSTN200.2.9.5

200.2.9.6

FXO

PSTN

102

Site F:Tavistock, UK

Area Code 0182

421

Site B:Reading, Berkshire, UK

Area Code 0118

943-6161

Site A:Birmingham, W. Midlands, UK

Area Code 0121

Site D:Inner London, UKArea Code 0207

020-7398-8300


EXAMPLE:Digital & Analog VOIPs

in Same System

118-943-5632Site C:

Reading Area Residential

E1

Port #4 Series #2 Analog MultiVOIPMVP410



387

The Series I analog VOIP phone book resides in the “Host” VOIP unit at SiteB. It applies to both of the Series I analog VOIP units.

Each of the Series II analog MultiVOIPs (the MVP210 and the MVP410)requires its own inbound and outbound phonebooks. The MVP3010 digitalMultiVOIP requires its own inbound and outbound phonebooks, as well.


388

These seven phone books are shown below.

Phone Book for Analog VOIP Host Unit (Site B)VOIP Dir #-OR-DestinationPattern


102 200.2.9.8 2 Site B, FXS channel.(Reading, UK)

101 200.2.9.8 1 Site B, FXO channel.(Reading, UK)

201 200.2.9.7 1 Site A, FXS channel.(Birmingham)

421 200.2.9.6 0 Site E, FXS channel.(Carlisle, UK)

018226374

Note 3.

200.2.9.5 0 Gives remote voip usersaccess to key phone systemextensions at Tavistockoffice (Site F). The keysystem might be arrangedeither so that calls gothrough a human operator orthrough an auto-attendant(which prompts user to dialthe desired extension).

0182 200.2.9.5 4 Gives remote voip usersaccess to Tavistock PSTNvia FXO port (#4) at Site F.

3xx 200.2.9.9 0(Note 1.)

Allows remote voip users tocall all PBX extensions atSite D (Inner London) usingonly three digits.


389

Phone Book for Analog VOIP Host Unit (Site B)(continued)

VOIP Dir #-OR-DestinationPattern


0207xxxxxxx

200.2.9.9 0(Note 2.)

Gives remote voip usersaccess to phone numbers in0207 area code (InnerLondon) in which Site D islocated.

0208xxxxxxx

200.2.9.9 0(Note 2.)

Gives remote voip usersaccess to phone numbers in0208 area code (OuterLondon) for which calls arelocal from Site D (InnerLondon).

Note 1. The “x” is a wildcard character.

Note 2. By specifying “Channel 0,” we instruct the MVP3010 tochoose any available data channel to carry the call.

Note 3. Note that Site F key system has only 30 extensions (x7400-7429). This destination pattern (018226374) actually directs calls to402-263-7430 through402-263-7499 into the key system, as well.This means that such calls, which belong on the PSTN, cannot becompleted. In some cases, this might be inconsequential because anentire exchange (fully used or not) might have been reserved for thecompany or it might be unnecessary to reach those numbers. However,to specify only the 30 lines actually used by the key system, thedestination pattern 018226374 would have to be replaced by three otherdestination patterns, namely 0182263740, 0182263741, and0182263742. In this way, calls to 0182-263-7430 through 0182-263-7499 would be properly directed to the PSTN. In the Site D outboundphonebook, the 30 lines are defined exactly, that is, without making anyadjacent phone numbers unreachable through the voip system.


390

The Outbound PhoneBook of the MVP3010 is shown below.

Outbound Phone Book for MVP3010 Digital VOIP (Site D)Destin.Pattern

RemovePrefix

AddPrefix

IPAddress

Comment

201 200.2.9.7 To originate calls to Site A(Birmingham).

901189 901189 101#Note 3.

200.2.9.8 To originate calls to any PSTNphone in Reading area usingthe FXO channel (channel #1)of the Site B VOIP (Reading,UK).

421 -- -- 200.2.9.6 Calls to Site E (Carlisle).90182 Calls to Tavistock local PSTN

(Site F) could be arranged byoperator or possibly by auto-attendant.

90182263740

9 -- 200.2.9.5

90182263741

9 -- 200.2.9.5

90182263742

9 -- 200.2.9.5

Calls to extensions of keyphone system at Tavistockoffice.

102 200.2.9.8 To originate calls to phoneconnected to FXS port(channel #2) of the Site BVOIP (Reading).

Note 3. The pound sign (“#”) is a delimiter separating the VOIP numberfrom the standard telephony phone number.


391

The Inbound PhoneBook of the MVP3010 is shown below.

Inbound Phone Book for MVP3010 Digital VOIP (Site D)RemovePrefix

AddPrefix

ChannelNumber

Comments

0207 9,7Note 4.Note 5.

0 Allows phone users at remote voip sites tocall local numbers (those within the Site Darea code, 0207, Inner London) over theVOIP network.

0208 9,8Note 4.Note 5.

0 Allows phone users at remote voip sites tocall local numbers (those in Outer London)over the VOIP network.

020739883

3 0 Allows phone users at remote voip sites tocall extensions of the Site D PBX usingthree digits, beginning with “3” .

Note 4. “9” gives PBX station users access to outside line.Note 5. The comma represents a one-second pause, the time required forthe user to receive a dial tone on the outside line (PSTN). Commas can beused in the Inbound Phonebook, but not in the Outbound Phonebook.


392

Outbound Phone Book for MVP410 Analog VOIP(Site F)

Destin.Pattern

RemovePrefix

AddPrefix

IPAddress

Comment

201 200.2.9.7 To originate calls toSite A(Birmingham).

01189 0118 101#Note 3.

200.2.9.8 To originate calls toany PSTN phone inReading area usingthe FXO channel(channel #1) of theSite B VOIP.

102 200.2.9.8 To originate calls tophone connected toFXS port (channel#2) of the Site BVOIP (Reading).

421 200.2.9.6 Calls to Site E(Carlisle).

0207 200.2.9.9 Calls to InnerLondon area PSTNvia Site D PBX.

0208 200.2.9.9 Calls to InnerLondon area PSTNvia Site D PBX.

3 -- 02073988

200.2.9.9 Calls to InnerLondon PBXextensions withthree digits.



393

Inbound Phonebook for MVP410 Analog VOIP (Site F)RemovePrefix

AddPrefix

ChannelNumber

Comment

01822 2 4 Calls to Tavistock local PSTNthrough FXO port (Port #4) atSite F.

0182263740

740. 0

0182263741

741. 0

0182263742

742 0

Gives remote voip users, access toextensions of key phone systematTavistock office.Because call is completed at keysystem, abbreviated dialing (3-digits) is not workable.Human operator or auto-attendant is needed to completethese calls.


394

Outbound Phone Book for MVP210 Analog VOIP(Site E)

Destin.Pattern

RemovePrefix

AddPrefix

IPAddress

Comment

201 200.2.9.7 To originate calls toSite A(Birmingham).

01189 0118 101#Note 3.

200.2.9.8 To originate calls toany PSTN phone inReading area usingthe FXO channel(channel #1) of theSite B VOIP.

102 200.2.9.8 To originate calls tophone connected toFXS port (channel#2) of the Site BVOIP (Reading).

01822 01822 -- 200.2.9.5 Calls to Tavistockarea PSTN (viaFXO channel of theSite F VOIP).

018226374

200.2.9.5 Calls to Tavistockkey system operatoror auto-attendant.

0207 0207 200.2.9.9 Calls to London areaPSTN via Site DPBX.

8 0207398

200.2.9.9 Calls to LondonPBX extensionswith four digits.



395

Inbound Phonebook for MVP210 Analog VOIP (Site E)RemovePrefix

AddPrefix

ChannelNumber

Comment

421 1

Call Completion SummariesSite A calling Site C, Method 1

1. Dial 101.2. Hear dial tone from Site B.3. Dial 9435632.4. Await completion. Talk.


5. Dial 101#94356326. Await completion. Talk.

Note: Some analog VOIP gateways will allow completion by Method 2.Others will not.

Site C calling Site A

1. Dial 9436161.2. Hear dial tone from Site B VOIP.3. Dial 201.4. Await completion. Talk.


396

Site D calling Site C1. Dial 901189435632.2. “9” gets outside line. On some PBXs, an “8” may be used to direct calls

to the VOIP, while “9” directs calls to the PSTN. However, some PBXunits can be programmed to identify the destination patterns of all callsto be directed to the VOIP.

3. PBX at Site D is programmed to divert all calls made to the 118 areacode and exchange 943 into the VOIP network. (It would also bepossible to divert all calls to all phones in area code 118 into the VOIPnetwork, but it may not be desirable to do so.)

4. The MVP3010 removes the prefix “0118” and adds the prefix “101#”for compatibility with the analog MultiVOIP’s phonebook scheme.The “#” is a delimiter separating the analog VOIP’s phone numberfrom the digits that the analog VOIP must dial onto its local PSTN tocomplete the call. The digits “101#9435632” are forwarded to the SiteB analog VOIP.

5. The call passes through the IP network (in this case, the Internet).6. The call arrives at the Site B VOIP. This analog VOIP receives this

dialing string from the MVP3010: 101#9435632. The analog VOIP,seeing the “101” prefix, uses its own channel #1 (an FXO port) toconnect the call to the PSTN. Then the analog VOIP dials its localphone number 9435632 to complete the call.

NOTE: In the case of Reading, Berkshire,, England,both “1189” and “1183” are considered localarea codes. This is, in a sense however, amatter of terminology. It simply means thatnumbers of the form 9xx-xxxx and3xx-xxxx are both local calls for users atother sites in the VOIP network.


397

Site D calling Site FA voip call from Inner London PBX to extension 7424 on the key telephone system inTavistock, UK.

A. The required entry in the London Outbound Phonebook to facilitate origination ofthe call, would be 90182263742. The call would be directed to the Tavistock voip’sIP address, 200.2.9.5. (Generally on such a call, the caller would have to dial aninitial “9”. But typically the PBX would not pass the initial “9” dialed to the voip. Ifthe PBX did pass along that “9” however, its removal would have to be specified inthe local Outbound Phonebook.)

B. The corresponding entry in the Tavistock Inbound Phonebook to facilitatecompletion of the call would be

0182263742 for calls within the office at Tavistock

01822 for calls to the Tavistock local calling area (PSTN).

Call Event Sequence

1. Caller in Inner London dials 901822637424.

2. Inner London voip removes “9” .

3. Inner London voip passes remaining string, 01822637424on to theTavistock voipat IP address 200.2.9.5.

4. The dialed string matches an inbound phonebook entry at the Tavistockvoip, namely 0182263742.

5. The Tavistock voip rings one of the three FXS ports connected to theTavistockkey phone system.

6. The call will be routed to extension 7424 either by a human receptionist/operator or to an auto-attendant (which allows the caller to specify theextension to which they wish to be connected).


398

Site F calling Site DA voip call from a Tavistock key extension to extension 3117 on the PBX in InnerLondon.

A. The required entry in the Tavistock Outbound Phonebook to facilitate originationof the call, would be “3”. The string 02073988 is added, preceding the “3”. The callwould be directed to the Inner London voip’s IP address, 200.2.9.9.

B. The corresponding entry in the Inner-London Inbound Phonebook to facilitatecompletion of the call would be 020739883.

1. The caller in Tavistock picks up the phone receiver, presses a button onthe key phone set. This button has been assigned to a particular voipchannel.

2. The caller in Tavistock hears dial tone from the Tavistock voip.

3. The caller in Tavistock dials 02073983117.

4. The Tavistock voip sends the entire dialed string to the Inner-London voipat IP address 200.2.9.9.

5. The Inner-London voip matches the called digits 02073983117to itsInbound Phonebook entry “020739883, ” which it removes. Then it addsback the “3” as a prefix.

6. The Inner-London PBX dials extension 3117 in the office in InnerLondon.

Variations in PBX CharacteristicsThe exact dialing strings needed in the Outbound and Inbound Phonebooks ofthe MVP3010 will depend on the capabilities of the PBX. Some PBXs requiretrunk access codes (like an “8” or “9” to access an outside line or to access theVOIP network). Other PBXs can automatically distinguish between intra-PBXcalls, PSTN calls, and VOIP calls.

Some PBX units can also insert digits automatically when they receive certaindialing strings from a phone station. For example, a PBX may beprogrammable to insert automatically the three-digit VOIP identifier stringsinto calls to be directed to analog VOIPs.

The MVP3010 offers complete flexibility for inter-operation with PBX units sothat a coherent dialing scheme can be established to connect a company’smultiple sites together in a way that is convenient and intuitive for phone users.When working together with modern PBX units, the presence of the MVP3010can be completely transparent to phone users within the company.


399

International Telephony Numbering Plan ResourcesDue to the expansion of telephone number capacity to accommodate pagers,fax machines, wireless telephony, and other new phone technologies,numbering plans have been changing worldwide. Many new area codes havebeen established; new service categories have been established (for example, toaccommodate GSM, personal numbering, corporate numbering, etc.). Belowwe list several web sites that present up-to-date information on the telephonynumbering plans used around the world. While we find these to be generallygood resources, we would note that URLs may change or becomenonfunctional, and we cannot guarantee the quality of information on thesesites.

URL Descriptionhttp://phonebooth.interocitor.net

/wtngThe World TelephoneNumbering Guidepresents excellentinternational numberinginfo that is both broadand detailed. Thisincludes info on re-numbering plans carriedout worldwide in recentyears to accommodatenew technologies.

http://www.oftel.gov.uk/numbers/number.htm

UK numbering planfrom the Office ofTelecommunications,the UK telephonyauthority.

http://www.itu.int/home/index.html The InternationalTelecommunicationsUnion is an excellentsource and authority oninternational telecomregulations andstandards. National andinternational numberplans are listed on thissite.

URL Descriptionhttp://kropla.com/phones.htm Guide to international

http://phonebooth.interocitor.net/

http://www.oftel.gov.uk/numbers

http://www.itu.int/home/index.html

http://kropla.com/phones.htm


400

use of modems.

http://www.numberplan.org/ National andinternational numberingplans based on directinput from regulatorsworldwide. Includeslists of telecom carriersper country.

http://www.eto.dk/ EuropeanTelecommunicationsOffice. Primarilyconcerned withmobile/wirelessradiotelephony, GSM,etc.

http://www.eto.dk/ETNS.htm European TelephonyNumbering Space.Resources for pan-European telephonyservices, standards, etc.Part of ETO site.

http://www.regtp.de/en/reg_tele/start/fs_05.html

List of Europeantelecom regulatoryagencies by country(from German telecomauthority).

http://www.numberplan.org/

http://www.eto.dk/

http://www.eto.dk/ETNS.htm



403

Chapter 9: Analog/BRI PhonebookConfiguration

Analog/BRI Phonebook Configuration MultiVOIP User Guide

404

Phonebooks for Series II analog MultiVOIP units (MVP130, MVP210,MVP410, and MVP810) and BRI MultiVOIP units (MVP410ST/810ST) are,in principle, configured the same as phonebooks for digital MultiVOIPproducts that would operate in the same environment (under either NorthAmerican or European telephony standards, T1 or E1).

Therefore, if you are operating an analog MultiVOIP unit in a North Americantelephony environment, you will find useful phonebook instructions andexamples in Chapter 7: T1 Phonebook Configuration. If you are operating ananalog MultiVOIP unit in a European telephony environment, you will finduseful phonebook instructions and examples in Chapter 8: E1 PhonebookConfiguration.

Most of the examples in Chapters 7 and 8 describe systems containing bothdigital and analog MultiVOIP units.

You will also find useful information in Chapter 2: Quick Start Guide. Seeespecially these sections:

Phonebook Starter ConfigurationPhonebook TipsPhonebook Example (One Common Situation)

Chapter 2 also contains a “Phonebook Worksheet” section. You may want toprint out several worksheet copies. Paper copies can be very helpful incomparing phonebooks at multiple sites at a glance. This will assist you inmaking the phonebooks clear and consistent and will reduce ‘surfing’ betweenscreens on the configuration program.

403

Chapter 10: Operation andMaintenance

Operation and Maintenance MultiVOIP User Guide

404

Operation and MaintenanceAlthough most Operation and Maintenance functions of the software are in theStatistics group of screens, an important summary appears in the SystemInformation of the Configuration screen group.

System Information screenThis screen presents vital system information at a glance. Its primary use is introubleshooting. This screen is accessible via the Configuration pulldownmenu, the Configuration sidebar menu, or by the keyboard shortcut Ctrl + Alt+ Y. However, the System Information screen is not supported in the BRI5.02c software.

System Information Parameter Definitions(not supported in BRI 5.02c software)


Boot Version nn.nn Indicates the version of the code that isused at the startup (booting) of the voip.The boot code version is independent ofthe software version.





FirmwareVersion

alpha-numeric

Indicates the version of the MultiVOIPfirmware.

MultiVOIP User Guide Operation & Maintenance

405



406

Statistics ScreensOngoing operation of the MultiVOIP, whether it is in a MultiVOIP/PBXsetting or MultiVOIP/telco-office setting, can be monitored for performanceusing the Statistics functions of the MultiVOIP software.

About Call ProgressAccessing Call-Progress StatisticsChannel Icons (Main Screen Lower Left)

Channel icons are green when datatraffic is present, red when idle.

In the web GUI, call progress details can be viewed byclicking on an icon (one for each channel) arrangedsimilarly on the web-browser screen.

Pulldown Icon

Shortcut Sidebar

Alt + A


407

The Call Progress Details Screen


408

Call Progress Details: Field DefinitionsField Name Values DescriptionChannel 1-n Number of data channel or time slot

on which the call is carried. This isthe channel for which call-progressdetails are being viewed.

Call Details

Duration Hours: Minutes:Seconds

The length of the call in hours,minutes, and seconds (hh:mm:ss).

Mode Voice or FAX Indicates whether the call beingdescribed was a voice call or a FAXcall.

Voice Coder G.723, G.729,G.711, etc.

The voice coder being used on thiscall.

Packets Sent integer value The number of data packets sent overthe IP network in the course of thiscall.

Packets Rcvd integer value The number of data packets receivedover the IP network in the course ofthis call.

Bytes Sent integer value The number of bytes of data sent overthe IP network in the course of thiscall.

Bytes Rcvd integer value The number of bytes of data receivedover the IP network in the course ofthis call.

Packets Lost integer value The number of voice packets fromthis call that were lost after beingreceived from the IP network.

OutboundDigits

0-9, #, * The digits transmitted by theMultiVOIP to the PBX/telco for thiscall.

Prefix Matched Displays the dialed digits that werematched to a phonebook entry.


409

Call Progress Details: Field Definitions (cont’d)

From – To Details Description

Gateway Name alphanumericstring

Identifier for the VOIP gateway thathandled this call.

IP Address x.x.x.x,where x has arange of 0 to255

IP address from which the call wasreceived.

Options SC, FEC Displays VOIP transmission optionsin use on the current call. These mayinclude Forward Error Correction orSilence Compression.

SilenceCompression

SC “SC” stands for Silence Compression.With Silence Compression enabled,the MultiVOIP will not transmitvoice packets when silence isdetected, thereby reducing theamount of network bandwidth that isbeing used by the voice channel.

Forward ErrorCorrection

FEC “FEC” stands for Forward ErrorCorrection. Forward ErrorCorrection enables some of thevoice packets that were corrupted orlost to be recovered. FEC adds anadditional 50% overhead to the totalnetwork bandwidth consumed by thevoice channel.Default = Off


410



Supplementary ServicesStatus

Call on Hold alphanumeric Describes held call by its IP addresssource, location/gateway identifier,and hold duration. Location/gatewayidentifiers comes from GatewayName field in Phone BookConfiguration screen of remote voip.

Call Waiting alphanumeric Describes waiting call by its IPaddress source, location/gatewayidentifier, and hold duration.Location/gateway identifiers comesfrom Gateway Name field in PhoneBook Configuration screen ofremote voip.


411



Supplementary ServicesStatus

Caller ID There are fourvalues:“Calling Party +identifier”;“Alerting Party+ identifier”;“Busy Party+ identifier”;and“ConnectedParty +identifier”

This field shows the identifier andstatus of a remote voip (which hasCall Name Identification enabled)with which this voip unit is currentlyengaged in some voip transmission.The status of the engagement(Connected, Alerting, Busy, orCalling) is followed by the identifierof a specific channel of a remote voipunit. This identifier comes from the“Caller Id” field in theSupplementary Services screen ofthe remote voip unit.

Status hangup, active Shows condition of current call.Call ControlStatus

Tun, FS + Tun,AE, Mux

Displays the H.323 version 4 featuresin use for the selected call. Theseinclude tunneling (Tun), Fast Startwith tunneling (FS + Tun), Annex Emultiplexed UDP call signalingtransport (AE), and Q.931Multiplexing (Mux). See PhonebookConfiguration Parameters (in T1 orE1 chapters) for more on H.323v4features.


412

About Logs

The Logs

Accessing “Statistics: Logs”Pulldown Icon

Shortcut Sidebar

Alt + L

The Logs Screen


413

Logs Screen Details: Field Definitions


Event # column 1 or higher All calls are assigned an eventnumber in chronological order, withthe most recent call having thehighest event number.

Start Date,Timecolumn

dd:mm:yyyyhh:mm:ss

The starting time of the call (event).The date is presented as a dayexpression of one or two digits, amonth expression of one or twodigits, and a four-digit year. This isfollowed by a time-of-dayexpression presented as a two-digithour, a two-digit minute, and a two-digit seconds value. (statistics, logs)field

Duration column hh:mm:ss This describes how long the call(event) lasted in hours, minutes, andseconds.

Status column success or failure Displays the status of the call, i.e.,whether the call was completedsuccessfully or not.

Mode column voice or FAX Indicates whether the (event) beingdescribed was a voice call or a FAXcall.

From column gateway name Displays the name of the voicegateway that originates the call.

To column gateway name Displays the name of the voicegateway that completes the call.

Special ButtonsLast Displays last log entry.Delete File Deletes selected log file.

Call DetailsPackets sent integer value The number of data packets sent

over the IP network in the course ofthis call.

Bytes sent integer value The number of bytes of data sentover the IP network in the course ofthis call.


414

Logs Screen Details: Field Definitions (cont’d)Field Name Values Description

Call Details (cont’d)Packets loss(lost)

integer value The number of voice packets fromthis call that were lost after beingreceived from the IP network.

Voice coder G.723, G.729,G.711, etc.

The voice coder being used on thiscall.

Packets received integer value The number of data packetsreceived over the IP network in thecourse of this call.

Bytes received integer value The number of bytes of datareceived over the IP network in thecourse of this call.

Outbound digits 0-9, #, * The digits transmitted by theMultiVOIP to the PBX/telco forthis call.

FROM DetailsGateway Name alphanumeric

stringIdentifier for the VOIP gatewaythat originated this call.

IP Address x.x.x.x,where x has arange of 0 to 255

IP address of the VOIP gatewayfrom which the call was received.

Options FEC, SC Displays VOIP transmissionoptions used by the VOIP gatewayoriginating the call. These mayinclude Forward Error Correctionor Silence Compression.

TO DetailsGateway Name alphanumeric

stringIdentifier for the VOIP gatewaythat completed (terminated) thiscall.

IP Address x.x.x.x,where x has arange of 0 to 255

IP address of the VOIP gateway atwhich the call was completed(terminated).

Options Displays VOIP transmissionoptions used by the VOIP gatewayterminating the call. These mayinclude Forward Error Correctionor Silence Compression.


415

Logs Screen Details: Field Definitions (cont’d)

Supplementary Services Info(Not supported in BRI 502c software.)

Call TransferredTo

phone numberstring

Number of party called in transfer.

Call Forwarded To phone numberstring

Number of party called inforwarding.

CT Ph# phone numberstring

Call Transfer phone number.


416

About IP Statistics

Accessing IP StatisticsPulldown Icon

Shortcut Sidebar

Alt + I

IP Statistics Screen


417

IP Statistics: Field DefinitionsFieldName

Values Description

UDP versus TCP. (User DatagramProtocol versus Transmission ControlProtocol). UDP provides unguaranteed,connectionless transmission of data acrossan IP network. By contrast, TCP providesreliable, connection-oriented transmissionof data.Both TCP and UDP split data into packetscalled “datagrams.” However, TCPincludes extra headers in the datagram toenable retransmission of lost packets andreassembly of packets into their correctorder if they arrive out of order. UDP doesnot provide this. Lost UDP packets areunretrievable; that is, out-of-order UDPpackets cannot be reconstituted in theirproper order..Despite these obvious disadvantages, UDPpackets can be transmitted much faster thanTCP packets -- as much as three timesfaster. In certain applications, like audio andvideo data transmission, the need for highspeed outweighs the need for verified dataintegrity. Sound or pictures often remainintelligible despite a certain amount of lostor disordered data packets (which appear asstatic).

“Clear”button

-- Clears packet tallies from memory.

Total Packets Sum of data packets of all types.Transmitted

integervalue

Total number of packets transmitted by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.

Received integervalue

Total number of packets received by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.


418

IP Statistics: Field Definitions (cont’d)

FieldName

Values Description

Total Packets(cont’d)

Sum of data packets of all types.

ReceivedwithErrors

integervalue

Total number of error-laden packetsreceived by this VOIP gateway since thelast “clearing” or resetting of the counterwithin the MultiVOIP software.

UDP Packets User Datagram Protocol packets.

Transmitted

integervalue

Number of UDP packets transmitted by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.


Number of UDP packets received by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.

ReceivedwithErrors

integervalue

Number of error-laden UDP packetsreceived by this VOIP gateway since thelast “clearing” or resetting of the counterwithin the MultiVOIP software.

TCP Packets Transmission Control Protocol packets.

Transmitted

integervalue

Number of TCP packets transmitted by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.


Number of TCP packets received by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.

ReceivedwithErrors

integervalue

Number of error-laden TCP packetsreceived by this VOIP gateway since thelast “clearing” or resetting of the counterwithin the MultiVOIP software.


419

IP Statistics: Field Definitions (cont’d)

RTP Packets Voice signals are transmitted in RealtimeTransport Protocol packets. RTP packetsare a type or subset of UDP packets.

Transmitted

integervalue

Number of RTP packets transmitted by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.


Number of RTP packets received by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.

ReceivedwithErrors

integervalue

Number of error-laden RTP packetsreceived by this VOIP gateway since thelast “clearing” or resetting of the counterwithin the MultiVOIP software.

RTCP Packets Realtime Transport Control Protocolpackets convey control information to assistin the transmission of RTP (voice) packets.RTCP packets are a type or subset of UDPpackets.

Transmitted

integervalue

Number of RTCP packets transmitted bythis VOIP gateway since the last “clearing”or resetting of the counter within theMultiVOIP software.


Number of RTCP packets received by thisVOIP gateway since the last “clearing” orresetting of the counter within theMultiVOIP software.

ReceivedwithErrors

integervalue

Number of error-laden RTCP packetsreceived by this VOIP gateway since thelast “clearing” or resetting of the counterwithin the MultiVOIP software.


420

About Link ManagementThe Link Management screen is essentially an automated utility for pingingendpoints on your voip network. This utility generates pings of variable sizesat variable intervals and records the response to the pings.

Accessing Link ManagementPulldown

none

Shortcut/Icon Sidebar

none/none


421

Link Management screen Field DefinitionsField Name Values Description

Monitor Link fieldsIP Address toPing

a.b.c.d0-255

This is the IP address of the targetendpoint to be pinged.

No. of Pings 1-999 This field determines how manypings will be generated by the StartNow command.

ResponseTimeout


The duration after which a ping willbe considered to have failed.

Ping Size inBytes

32 – 128 bytes This field determines how long orlarge the ping will be.

Timer Intervalbetween Pings

0 or 30 – 6000minutes

This field determines how long of await there is between one ping andthe next.

Start Nowcommandbutton

-- Initiates pinging.

Clearcommandbutton

-- Erases ping parameters in MonitorLink field group and restores defaultvalues.


422

Link Management screen Field Definitions (cont’d)Field Name Values Description

Link Status Parameters These fields summarize the results ofpinging.

IP Addresscolumn

a.b.c.d0-255

Target of ping.

No. of PingsSent

as listed Number of pings sent to targetendpoint.

No. of PingsReceived

as listed Number of pings received by targetendpoint.

Round TripDelay(Min/Max/Avg)

as listed,in milliseconds

Displays how long it took from timeping was sent to time ping responsewas received.

Last Error as listed Indicates when last data erroroccurred.


423

About Packetization Time

You can use the Packetization Time screen to specify definite packetizationrates for coders selected in the Voice/FAX Parameters screen (in the “CoderOptions” group of fields). The Packetization Time screen is accessible underthe “Advanced” options entry in the sidebar list of the main voip softwarescreen. In dealing with RTP parameters, the Packetization Time screen isclosely related to both Voice/FAX Parameters and to IP Statistics. It is locatedin the “Advanced” group for ease of use.

Accessing Packetization TimePulldown

Shortcut/Icon Sidebar

none/none


424

Packetization Time Screen

Packetization rates can be set separately for each channel.

The table below presents the ranges and increments for packetization rates.

Packetization Ranges and IncrementsCoder Types Range (in Kbps);

{default value}Increments (in Kbps)

G711, G726, G727 5-120 {5} 5G723 30-120 {30} 30G729 10-120 {10} 10Netcoder 20-120 {20} 20


425

Once the packetization rate has been set for one channel, it can be copied into otherchannels.


426

About T1/E1 and BRI StatisticsAccessing T1 Statistics

Pulldown Icon

Shortcut Sidebar

Alt + T

The T1 and E1 Statistics screens are only accessible and applicable for the MVP2410,and MVP3010.

The BRI statistics screens are only accessible and applicable for the MVP410ST andMVP810ST .


427

T1 Statistics Screen


428

T1 Statistics: Field Definitions

FieldName

Values Description

Red Alarm Integer tally ofalarms countedsince last reset.

The alarm condition declared when a devicereceives no signal or cannot synchronize to thesignal being received. A Red Alarm isgenerated if the incoming data stream has notransitions for 176 consecutive pulse positions.

Blue Alarm Tally since lastreset.

Alarm signal consisting of all 1’s (including framingbit positions) which indicates disconnection or failureof attached equipment.

Loss of FrameAlignment

Tally since lastreset.

Loss of data frame synchronization.

ExcessiveZeroes


Displayed value will increment if consecutivezeroes beyond a set threshold are detected. I.e.,tally increments if more than 7 consecutivezeroes in the received data stream are detectedunder B8ZS line coding, or if 15 consecutivezeroes are detected under AMI line coding.

Status FreezeSignalingActive

Signaling has been frozen at the mostrecent values due to loss of framealignment, loss of multiframe alignment ordue to a receive slip.

LineLoopbackDeactivationSignal

Line loopback deactivation signal has beendetected in the receive bit stream.

Transmit LineShort

A short exists between the transmit pair forat least 32 consecutive pulses.

Transmit DataOverflow

For use by MTS Technical Supportpersonnel.

Transmit SlipPositive

The frequency of the transmit clock is lessthan the frequency of the transmit systeminterface working clock. A frame isrepeated.


429

T1 Statistics: Field Definitions (cont’d)

FieldName

Values Description

Yellow Alarm Tally since lastreset.

The alarm signal sent by a remote T1/E1device to indicate that it sees no receive signalor cannot synchronize on the receive signal.

Frame SearchRestart Flag

[To be supplied.]

Loss ofMultiFrameAlignment


In D4 or ESF mode, displayed value willincrement if multiframe alignment has beenlost or if loss of frame alignment has beendetected.

Transmit Slip Tally since lastreset.

Slip in transmitted data stream. Slips indicatea clocking mismatch (or lack ofsynchronization) between T1/E1 devices.When slips occur, data may be lost orrepeated.

Pulse DensityViolation

The pulse density of the received data streamis below the requirement defined by ANSIT1.403 or more than 15 consecutive zeros aredetected.

LineLoopbackActivationSignal

The line loopback activation signal hasbeen detected in the received bit stream.

Transmit LineOpen

At least 32 consecutive zeros weretransmitted.

Transmit DataUnderrun

For use by MTS Technical SupportPersonnel.

Transmit SlipNegative

The frequency of the transmit clock isgreater than the frequency of the transmitsystem interface working clock. A frameis skipped.


430

T1 Statistics: Field Definitions (cont’d)

FieldName

Values Description

BipolarViolation

Integer tally ofviolation countsince last reset.

Two successive pulses of the same polarityhave been received and these pulses are notpart of zero substitution. On an AMI-encodedline, this represents a line error. On a B8ZSline, this may represent the substitution for astring of 8 zeroes.

Receive Slip Tally since lastreset.

A receive slip (positive or negative) hasoccurred. Slips indicate a clocking mismatch(or lack of synchronization) between T1/E1devices. When slips occur, data may be lost orrepeated.


431

E1 Statistics Screen

E1 Statistics: Field Definitions

FieldName

Values Description

Red Alarm Integer tally ofalarms countedsince last reset.

The alarm condition declared when a devicereceives no signal or cannot synchronize to thesignal being received. A Red Alarm isgenerated if the incoming data stream has notransitions for 176 consecutive pulse positions.

Blue Alarm Tally since lastreset.

Alarm signal consisting of all 1’s (includingframing bit positions) which indicatesdisconnection or failure of attached equipment.

Loss of FrameAlignment


Loss of data frame synchronization.


432

E1 Statistics: Field Definitions (cont’d)

FieldName

Values Description

ReceiveTimeslot 16AlarmIndicationSignal

Detected alarm indication signal intimeslot 16 according to ITU-T G.775.Indicates the incoming time slot 16contains less than 4 zeros in each of twoconsecutive time slot 16 multiframeperiods.

Transmit LineShort

A short exists between the transmit pairfor at least 32 consecutive pulses.

Transmit DataOverflow

For use by MTS personnel.

Transmit SlipPositive

The frequency of the transmit clock isless than the frequency of the transmitsystem interface working clock. A frameis repeated.

Yellow Alarm Tally since lastreset.

The alarm signal sent by a remote T1/E1device to indicate that it sees no receive signalor cannot synchronize on the receive signal.

Status FreezeSignalingActive

Signaling has been frozen at the most recentvalues due to loss of frame alignment, loss ofmultiframe alignment or due to a receive slip.

Loss ofMultiFrameAlignment


In D4 or ESF mode, displayed value willincrement if multiframe alignment has beenlost or if loss of frame alignment has beendetected.

ReceiveTimeslot 16Loss of Signal

The time slot 16 data stream contains all zerosfor at least 16 contiguously received timeslots.


433

E1 Statistics: Field Definitions (cont’d)

FieldName

Values Description

ReceiveTimeslot 16Loss ofMultiFrameAlignment

The framing pattern '0000' in 2 consecutiveCAS multiframes were not found or in all timeslot 16 of the previous multiframe all bits werereset.

Transmit LineOpen

At least 32 consecutive zeroes weretransmitted.

Transmit DataUnderrun

For use by MTS Technical SupportPersonnel.

Transmit SlipNegative

The frequency of the transmit clock is greaterthan the frequency of the transmit systeminterface working clock. A frame is skipped.

BipolarViolation

Integer tally ofviolation countsince last reset.

Bipolar Violation (or BPV) refers to twosuccessive pulses of the same polarity on theE1 line. On an AMI-encoded line, thisrepresents a line error. On a B8ZS line, thismay represent the substitution for a string of 8zeroes.

ExcessiveZeroes


Displayed value will increment if consecutivezeroes beyond a set threshold are detected.I.e., tally increments if more than 7consecutive zeroes in the received data streamare detected under B8ZS line coding, or if 15consecutive zeroes are detected under AMIline coding.

Transmit Slip Tally since lastreset.

Slip in transmitted data stream. Slips indicatea clocking mismatch (or lack ofsynchronization) between T1/E1 devices.When slips occur, data may be lost orrepeated.

Receive Slip Tally since lastreset.

Slip in received data stream. Slips indicate aclocking mismatch (or lack of synchronization)between T1/E1 devices. When slips occur, data maybe lost or repeated.


434

ISDN BRI Statistics Screen


435

ISDN BRI Statistics: Field Definitions

FieldName

Values Description

Select BRIInterface

ISDNn

For n=1-2 (410ST)

For n-1-4 (810ST)

In this field, you can choose the ISDN portfor which you want to view the status. The410ST has two ISDN –BRI ports (or“interfaces”); the 810ST has four ISDN-BRI ports (or “interfaces”). Each interfacehas two channels.

Layer 1 Interface

Status inactive (F1),sensing (F2),deactivated (F3),awaiting signal (F4),identifying input (F5),synchronized (F6),activated (F7),lost framing (F8),deactive (G1),pending activation(G2),active (G3),pendingdeactivation (G4)

Shows the current Layer 1 status of theISDN connection. Each status description(inactive, sensing, etc.) corresponds to aparticular “state” label (F1-F8 and G1-G4).

State F1-F8 (for Terminalmode ports),G1-G4 (for Networkmode ports)

Shows the I.430 state name for Layer 1. An“F” state name indicates this port is inTerminal mode (F1-F8), as set in the ISDNBRI Parameters screen.

A “G” state name indicates that this port isin Network mode (G1-G4), as set in theISDN BRI Parameters screen.

Loss OfFraming

integer Shows the number of lost-framing events onthe ISDN physical layer.

Loss ofSync

integer Shows the number of lost-synchronizationevents on the ISDN physical layer.


436

ISDN BRI Statistics: Field Definitions(continued)

FieldName

Values Description

Switch Information:TEI Assignment

TEI 0 throughTEI 7

0-63 (point-to-pointassignments)

64-126 (automaticassignments)

Displays the value for each TEI assigned tothe BRI port. The TEI (Terminal EndpointIdentifier) uniquely identifies each deviceconnected to the ISDN physical layer.

Switch Information:D-Channel Information

Tx Packets 0 to 4294967295 Shows the number of packets transmitted onthe channel. When the value exceeds4294967295 packets, it will reset to zeroand continue counting.

Rx Packets 0 to 4294967295 Shows the number of packets received onthe channel. When the value exceeds4294967295 packets, it will reset to zeroand continue counting.

Switch Information:SPID 0

(SPID 0number)

numeric,3 to 20 digits

A SPID (Service Profile Identifier) isassigned by the ISDN provider andpertains to one channel of the BRIinterface (port), in this case channel 0.The SPID identifies an ISDN terminaluniquely. The SPID associates a set ofservices (features) with the terminal.(In Terminal mode the provider is atelco or PBX. In Network modeMultiVOIP is the provider.) A SPID isonly used when the “Country” field isset to “USA” in the ISDN BRIParameters screen.

Status Not Checked,Correct,Incorrect

Indicates whether SPID0 is correct,incorrect, or not being checked.


437

ISDN BRI Statistics: Field Definitions(continued)

FieldName

Values Description

Switch Information:SPID 1

(SPID 1number)

numeric SPID for channel 1 of the BRIinterface. Otherwise, same as SPID0description above.

Status Not Checked,Correct,Incorrect

Indicates whether SPID1 is correct,incorrect, or not being checked.

“Clear”button

Clears (sets to zero) allISDN BRI Statistics fieldswith numeric tally values (these are Loss ofFraming, Loss of Sync, Tx Packets, RxPackets).


438

About Registered Gateway DetailsThe Registered Gateway Details screen presents a real-time display of thespecial operating parameters of the Single Port Protocol (SPP). These areconfigured in the PhoneBook Configuration screen and in the Add/EditOutbound PhoneBook screen.

Accessing Registered Gateway DetailsPulldown Icon

Shortcut Sidebar


439

Registered Gateway Details: Field Definitions

FieldName

Values Description

Column Headings

Description alphanumeric This is a descriptor for a particular voipgateway unit. This descriptor should generallyidentify the physical location of the unit (e.g.,city, building, etc.) and perhaps even itslocation in an equipment rack.

IP Address n.n.n.n,

for n = 0-255

The RAS address for the gateway.

Port Port by which the gateway exchanges H.225RAS messages with the gatekeeper. .

RegisterDuration

The time remaining in seconds before theTimeToLive timer expires. If the gateway failsto reregister within this time, the endpoint isunregistered.

Status The current status of the gateway, eitherregistered or unregistered.

No. of Entries The number of gateways currently registered tothe Registrar. This includes all SPP clientsregistered and the Registrar itself.

Details

Count ofRegisteredNumbers

If a registered gateway is selected (by clicking onit in the screen), The "Count of RegisteredNumbers" will indicate the number of registeredphone numbers for the selected gateway. When aclient registers, all of its inbound phonebook'sphone numbers become registered.

List ofRegisteredNumbers

Lists all of the registered phone numbers for theselected gateway.


440

MultiVoip Program Menu Items After the MultiVoip program is installed on the PC, it can be launched fromthe Programs group of the Windows Start menu ( Start | Programs |MultiVOIP ____ | … ). In this section, we describe the software functionsavailable on this menu.

Several basic software functions are accessible from the MultiVoip softwaremenu, as shown below.

MultiVOIP Program Menu

Menu Selection Description

Configuration Select this to enter the Configurationprogram where values for IP, telephony,and other parameters are set.

Date and Time Setup Select this for access to set calendar/clockused for data logging.

Download CAS Protocol Telephony CAS files are for ChannelAssociated Signaling. There are many CASfiles, some labeled for specific functionality,others for countries or regions where certaintelephony attributes are standard.


441

MultiVOIP Program Menu (cont’d)

Menu Selection Description

Download Factory Defaults Select this to return the configurationparameters to the original factory values.

Download Firmware Select this to download new versions offirmware as enhancements becomeavailable.

Download User Defaults To be used after a full set of parametervalues, values specified by the user, havebeen saved (using Save Setup). Thiscommand loads the saved user defaultsinto the MultiVOIP.

Set Password Select this to create a password for accessto the MultiVOIP software programs(Program group commands, WindowsGUI, web browser GUI, & FTP server).Only the FTP Server function requires apassword for access. The FTP Serverfunction also requires that a username beestablished along with the password.

Uninstall Select this to uninstall the MultiVOIPsoftware (most, but not all components areremoved from computer when thiscommand is invoked).

Upgrade Software Loads firmware (including H.323 stack)and factory default settings from thecontroller PC to the MultiVOIP unit.

“Downloading” here refers to transferring program files from the PC to thenonvolatile “flash” memory of the MultiVOIP. Such transfers are made via thePC’s serial port. This can be understood as a “download” from the perspectiveof the MultiVOIP unit.

When new versions of the MultiVoip software become available, they will beposted on MultiTech’s web or FTP sites. Although transferring updatedprogram files from the MultiTech web/FTP site to the user’s PC can generallybe considered a download (from the perspective of the PC), this type ofdownload cannot be initiated from the MultiVoip software’s Program menucommand set.

Generally, updated firmware must be downloaded from the MultiTechweb/FTP site to the PC before it can be loaded from the PC to the MultiVOIP.


442

Date and Time SetupThe dialog box below allows you to set the time and date indicators of theMultiVOIP system.

Obtaining Updated FirmwareGenerally, updated firmware must be downloaded from the MultiTechweb/FTP site to the user’s PC before it can be downloaded from that PC to theMultiVOIP.

Note that the structure of the MultiTech web/FTP site may change withoutnotice. However, firmware updates can generally be found using standard webtechniques. For example, you can access updated firmware by doing a searchor by clicking on Support.


443

If you conduct a search, for example, on the word “MultiVoip,” you will bedirected to a list of firmware that can be downloaded.

If you choose Support, you can select “MultiVoip” in the Product Supportmenu and then click on Firmware to find MultiVOIP resources.


444

Once the updated firmware has been located, it can be downloaded from theweb/ftp site using normal PC/Windows procedures. While the next 3 screensbelow pertain to the MVP3010, similar screens will appear for any MultiVOIPmodel described in this manual.

� ��

� !�"#$� ��

�%�� &��$��'��("�)"�*��"�&�%�� +,��-./0�)"�� &��$��$1 ��2%��1� ��1��1� ��3� "%�� $��4��-.�%��


445

Generally, the firmware file will be a self-extracting compressed file (with .zipextension), which must be expanded (decompressed, or “unzipped”) on theuser’s PC in a user-specified directory.

$15��67"�1� ��6��


446

Implementing a Software UpgradeBeginning with the 4.03/6.03 software release, MultiVOIP software can beupgraded locally using a single command at the MultiVOIP Windows GUI,namely Upgrade Software. This command downloads firmware (includingthe H.323 stack), and factory default settings from the controller PC to theMultiVOIP unit.

When using the MultiVOIP Windows GUI, firmware and factory defaultsettings can also be transferred from controller PC to MultiVOIP piecemealusing separate commands.

When using the MultiVOIP web browser GUI to control/configure the voipremotely, upgrading of software must be done on a piecemeal basis using theFTP Server function of the MultiVOIP unit.

When performing a piecemeal software upgrade (whether from the WindowsGUI or web browser GUI), follow these steps in order:

1. Identify Current Firmware Version2. Download Firmware3. Download Factory Defaults

When upgrading firmware, the software commands “Download Firmware,”and “Download Factory Defaults” must be implemented in order, else theupgrade is incomplete.

Identifying Current Firmware VersionBefore implementing a MultiVOIP firmware upgrade, be sure to verify thefirmware version currently loaded on it. The firmware version appears in theMultiVoip Program menu. Go to Start | Programs | MultiVOIP ____ x.xx.The final expression, x.xx, is the firmware version number. In the illustrationbelow, the firmware version is 4.00a, made for the E1 MultiVOIP (MVP3010).

When a new firmware version is installed, the MultiVOIP software can beupgraded in one step using the Upgrade Software command, or piecemealusing the Download Firmware command and the Download FactoryDefaults command.


447

Download Firmware transfers the firmware (including the H.323 protocolstack) in the PC’s MultiVOIP directory into the nonvolatile flash memory ofthe MultiVOIP.Download Factory Defaults sets all configuration parameters to the standarddefault values that are loaded at the MultiTech factory.Upgrade Software implements both the Download Firmware command andthe Download Factory Defaults command.

Downloading Firmware1. The MultiVoip Configuration program must be off when invoking the

Download Firmware command. If it is on, the command will not work.

2. To invoke the Download Factory Defaults command, go to Start |Programs | MVP____ x.xx | Download Firmware.


448

3. If a password has been established, the Password Verification screen willappear.

Type in the password and click OK.

4. The MultiVOIP ___- Firmware screen appears saying“MultiVOIP [model number] is up. Reboot to Download Firmware?”

Click OK to download the firmware.

The “Boot” LED on the MultiVOIP will light up and remain lit during thefile transfer process.


449

5. The program will locate the firmware “.bin” file in the MultiVOIP directory.Highlight the correct (newest) “.bin” file and click Open.

6. Progress bars will appear at the bottom of the screen during the file transfer.

The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.

7. The Download Firmware procedure is complete.


450

Downloading CAS Protocols

1. The MultiVoip Configuration program must be off when invoking theDownload CAS Protocol command. If it is on, the command will notwork.

2. To invoke the Download H.323 PDL command, go to Start | Programs |MVP____ x.xx | Download H.323 PDL.


451


Type in password and click OK.

4. The MultiVOIP ____- Firmware screen appears saying “MultiVOIP[model number] is up. Reboot to Download Firmware?”

Click OK to download the CAS Protocol file(s) to the MultiVOIP.

The “Boot” LED on the MultiVOIP will light up and remain lit during thefile transfer process.


452

5. The program will locate the CAS protocol file in the MultiVOIP directory.Highlight the correct (newest) file and click Open.

6. Progress bars will appear at the bottom of the screen during the file transfer.


7. The Download CAS Protocol procedure is complete.

Downloading Factory Defaults1. The MultiVoip Configuration program must be off when invoking the

Download Factory Defaults command. If it is on, the command will notwork.

2.To invoke the Download Factory Defaults command, go to Start |Programs | MVP____ x.xx | Download Factory Defaults.


453


Type in the password and click OK.

4. The MVP____- Firmware screen appears saying “MultiVOIP [modelnumber] is up. Reboot to Download Firmware?”

Click OK to download the factory defaults.The “Boot” LED on the MultiVOIP will light up and remain lit during thefile transfer process.


454

5. After the PC gets a response from the MultiVOIP, the Dialog – IPParameters screen will appear.

The user should verify that the correct IP parameter values are listed on thescreen and revise them if necessary. Then click OK.

6. Progress bars will appear at the bottom of the screen during the data transfer.


7. The Download Factory Defaults procedure is complete.

Setting and Downloading User DefaultsThe Download User Defaults command allows you to maintain a known workingconfiguration that is specific to your VOIP system. You can then experiment withalterations or improvements to the configurations confident that a workingconfiguration can be restored if necessary.

1. Before you can invoke the Download User Defaults command, you mustfirst save a set of configuration parameters by using the Save Setupcommand in the sidebar menu of the MultiVOIP software.


455

2. Before the setup configuration is saved, you will be prompted to save thesetup as the User Default Configuration. Select the checkbox and click OK.

Save Current Setup as User Default Configuration

MultiVOIP _____ will be brought down.

OK Cancel Help

A user default file will be created.

3. The MVP____- Firmware screen appears saying “MultiVOIP [modelnumber] is up. Reboot to Download Firmware?”

Click OK to download the factory defaults. The “Boot” LED on theMultiVOIP will light up and remain lit during the file transfer process.

4. Progress bars will appear during the file transfer process.


456

5. When the file transfer process is complete, the Dialog-- IP Parametersscreen will appear.

6. Set the IP values per your particular VOIP system. Click OK. Progress barswill appear as the MultiVOIP reboots itself. Downloading IFM

FirmwareThe Download IFM Firmware command applies only to the MVP210/410/810and MVP210G/410G/810G models. This command transfers firmware to thetelephony interface modules of each voice channel. These firmware moduleshandle the physical interface (FXS, FXO and E&M) to the attached analogtelephony equipment.


457


458

Setting a Password (Windows GUI)After a user name has been designated and a password has been set, thatpassword is required to gain access to any functionality of the MultiVOIPsoftware. Only one user name and password can be assigned to a voip unit.The user name will be required when communicating with the MultiVOIP viathe web browser GUI.

NOTE: Record your user name and password in a safe place. If thepassword is lost, forgotten, or unretrievable, the user mustcontact MultiTech Tech Support in order to resume use of theMultiVOIP unit.

1. The MultiVoip configuration program must be off when invoking the SetPassword command. If it is on, the command will not work.

2. To invoke the Set Password command, go to Start | Programs | MVP____x.xx | Set Password.

3. You will be prompted to confirm that you want to establish a password,which will entail rebooting the MultiVOIP (which is done automatically).

Click OK to proceed with establishing a password.


459

4. The Password screen will appear. If you intend to use the FTP Serverfunction that is built into the MultiVOIP, enter a user name. (A User Name isnot needed to access the local Windows GUI, the web browser GUI, or thecommands in the Program group.) Type your password in the Passwordfield of the Password screen. Type this same password again in theConfirm Password field to verify the password you have chosen.

NOTE: Be sure to write down your password in a convenient butsecure place. If the password is forgotten, contact MultiTechTechnical Support for advice.

Click OK.

5. A message will appear indicating that a password has been set successfully.

After the password has been set successfully, the MultiVOIP will re-bootitself and, in so doing, its BOOT LED will light up.


460

6. After the password has been set, the user will be required to enter thepassword to gain access to the web browser GUI and any part of theMultiVOIP software listed in the Program group menu. User Name andPassword are both needed for access to the FTP Server residing in theMultiVOIP.

When MultiVOIP program asks for password at launch of program, theprogram will simply shut down if CANCEL is selected.

The MultiVOIP program will produce an error message if an invalid passwordis entered.


461

Setting a Password (Web Browser GUI)Setting a password is optional when using the MultiVOIP web browser GUI.Only one password can be assigned and it works for all MultiVOIP softwarefunctions (Windows GUI, web browser GUI, FTP server, and all Programmenu commands, e.g., Upgrade Software – only the FTP Server functionrequires a User Name in addition to the password). After a password has beenset, that password is required to access the MultiVOIP web browser GUI.

NOTE: Record your user name and password in a safe place. If thepassword is lost, forgotten, or unretrievable, the user mustcontact MultiTech Tech Support in order to resume use of theMultiVOIP web browser GUI.


462

Un-Installing the MultiVOIP Software1. To un-install the MultiVOIP configuration software, go to Start | Programs

and locate the MultiVOIP entry. Select Uninstall MVP____ vx.xx (versionsmay vary).

2. Two confirmation screens will appear. Click Yes and OK when you arecertain you want to continue with the uninstallation process.


463

3. A special warning message similar to that shown below may appear for theMultiVOIP software’s “.bin” file. Click Yes.

5"��"�� 2��%��&��8��%�� %�9��"%� ��&��2��%2%��:��%�9��"�"%� ��&��2��%2%��:��9��5�� &6�"�2��:$1��#� ��%1� ��1!;�� 1�!��9�"�) %��"&�)��"#��"��&�&�� "%��"�2��%2%��3�� ":��(��<�%�9��"=��)�%�:��(�'��


Click Finish.


464

Upgrading SoftwareAs noted earlier (see the section Implementing a Software Upgrade above), theUpgrade Software command transfers, from the controller PC to theMultiVOIP unit, firmware (including the H323 stack) and factory defaultconfiguration settings. As such, Upgrade Software implements the functionsof both Download Firmware and Download Factory Defaults in a singlecommand.


465

FTP Server File Transfers (“Downloads”)With the 4.03/6.03 software release, MultiTech has built an FTP server into theMultiVOIP unit. Therefore, file transfers from the controller PC to the voipunit can be done using an FTP client program or even using a browser (e.g.,Internet Explorer or Netscape, used in conjunction with Windows Explorer).

The terminology of “downloads” and “uploads” gets a bit confusing in thiscontext. File transfers from a client to a server are typically considered“uploads.” File transfers from a large repository of data to machines with lessdata capacity are considered “downloads.” In this case, these metaphors arecontradictory: the FTP server is actually housed in the MultiVOIP unit, andthe controller PC, which is actually the repository of the info to be transferred,uses an FTP client program. In this situation, we have chosen to call thetransfer of files from the PC to the voip “downloads.” (Be aware that someFTP client programs may use the opposite terminology, i.e., they may refer tothe file transfer as an “upload “)

You can download firmware, CAS telephony protocols, default configurationparameters, and phonebook data for the MultiVOIP unit with this FTPfunctionality. These downloads are done over a network, not by a local serialport connection. Consequently, voips at distant locations can be updated froma central control point.

The phonebook downloading feature greatly reduces the data-entry required toestablish inbound and outbound phonebooks for the voip units within a system.Although each MultiVOIP unit will require some unique phonebook entries,most will be common to the entire voip system. After the phonebooks for thefirst few voip units have been compiled, phonebooks for additional voipsbecome much simpler: you copy the common material by downloading andthen do data entry for the few phonebook items that are unique to thatparticular voip unit or voip site.


466

To transfer files using the FTP server functionality in the MultiVOIP, followthese directions.

1. Establish Network Connection and IP Addresses. Both the controller PCand the MultiVOIP unit(s) must be connected to the same IP network. An IPaddress must be assigned for each.

IP Address of Control PC ____ . ____ . ____ . ____

IP Address of voip unit #1 ____ . ____ . ____ . ____

:.

:.

:.

:.

:.

IP address of voip unit #n ____ . ____ . ____ . ____

2. Establish User Name and Password. You must establish a user name and(optionally) a password for contacting the voip over the IP network. (Whenconnection is made via a local serial connection between the PC and the voipunit, no user name is needed.)

As shown above, the username and password can be set in the web GUI as wellas in the Windows GUI.


467

3. Install FTP Client Program or Use Substitute. You should install an FTPclient program on the controller PC. FTP file transfers can be done using aweb browser (e.g., Netscape or Internet Explorer) in conjunction with a localWindows browser a (e.g., Windows Explorer), but this approach is somewhatclumsy (it requires use of two application programs rather than one) and itlimits downloading to only one VOIP unit at a time. With an FTP clientprogram, multiple voips can receive FTP file transmissions in response to asingle command (the transfers may occur serially however).

Although MultiTech does not provide an FTP client program with theMultiVOIP software or endorse any particular FTP client program, we remindour readers that adequate FTP programs are readily available under retail,shareware and freeware licenses. (Read and observe any End-User LicenseAgreement carefully.) Two examples of this are the “WSFTP” client and the“SmartFTP” client, with the former having an essentially text-based interfaceand the latter having a more graphically oriented interface, as of this writing.User preferences will vary. Examples here show use of both programs.

4. Enable FTP Functionality. Go to the IP Parameters screen and click onthe “FTP Server: Enable” box.


468

5. Identify Files to be Updated. Determine which files you want to update.Six types of files can be updated using the FTP feature. In some cases, the fileto be transferred will have “Ftp” as the part of its filename just before thesuffix (or extension). So, for example, the file “mvpt1Ftp.bin” can betransferred to update the bin file (firmware) residing in the MultiVOIP.Similarly, the file “fxo_loopFtp.cas” could be transferred to enable use of theFXO Loop Start telephony interface in one of the analog voip units and the file“r2_brazilFtp.cas” could be transferred to enable a particular telephonyprotocol used in Brazil.

File Type File Names Description

firmware“bin” file

mvpt1Ftp.bin This is the MultiVOIPfirmware file. Only onefile of this type will be inthe directory.

factory defaults fdefFtp.cnf This file contains factorydefault settings for user-changeable configurationparameters. Only one fileof this type will be in thedirectory.

CAS file fxo_loopFtp.cas,em_winkFtp.cas,r2_brazilFtp.casr2_chinaFtp.cas

These telephony files are forChannel AssociatedSignaling. The directorycontains many CAS files,some labeled for specificfunctionality, others forcountries or regions wherecertain attributes arestandard.

H323 PDL file This file is specific to theparticular version of theH.323 standard beingused. This file rarelyneeds to be updated.

inboundphonebook

InPhBk.tmr This file updates theinbound phonebook in theMultiVOIP unit.

outboundphonebook

OutPhBk.tmr This file updates theoutbound phonebook inthe MultiVOIP unit.


469

6. Contact MultiVOIP FTP Server. You must make contact with the FTPServer in the voip using either a web browser or FTP client program. Enter theIP address of the MultiVOIP’s FTP Server. If you are using a browser, theaddress must be preceded by “ftp://” (otherwise you’ll reach the web GUIwithin the MultiVOIP unit).


470

7. Log In. Use the User Name and password established in item #2 above.The login screens will differ depending on whether the FTP file transfer is tobe done with a web browser (see first screen below) or with an FTP clientprogram (see second screen below).


471

8. Invoke Download. Downloading can be done with a web browser or withan FTP client program.

8A. Download with Web Browser.

8A1. In the local Windows browser, locate the directory holding theMultiVOIP program files. The default location will beC:\Program Files \Multi-Tech Systems \MultiVOIP xxxx yyyy(where x and y represent MultiVOIP model numbers andsoftware version numbers).

8A2. Drag-and-drop files from the local Windows browser (e.g.,Windows Explorer) to the web browser.


472

You may be asked to confirm the overwriting of files on the MultiVOIP. Doso.

File transfer between PC and voip will look like transfer within voipdirectories.


473

8B. Download with FTP Client Program.

8B1. In the local directory browser of the FTP client program, locatethe directory holding the MultiVOIP program files. The defaultlocation will be C:\Program Files \Multi-Tech Systems\MultiVOIP xxxx yyyy (where x and y represent MultiVOIPmodel numbers and software version numbers).

8B2. In the FTP client program window, drag-and-drop files from thelocal browser pane to the pane for the MultiVOIP FTP server.FTP client GUI operations vary. In some cases, you can choosebetween immediate and queued transfer. In some cases, theremay be automated capabilities to transfer to multipledestinations with a single command.


474

Some FTP client programs are more graphically oriented (see previous screen),while others (like the “WS-FTP” client) are more text oriented.

9. Verify Transfer. The files transferred will appear in the directory of theMultiVOIP.

10. Log Out of FTP Session. Whether the file transfer was done with a webbrowser or with an FTP client program, you must log out of the FTP sessionbefore opening the MultiVOIP Windows GUI.


475

Web Browser Interface

Beginning with the 4.03/6.03 software release, you can control the MultiVOIPunit with a graphic user interface (GUI) based on the common web browserplatform. Qualifying browsers are InternetExplorer6 and Netscape6.

MultiVOIP Web Browser GUI Overview

Function Remote configuration and control ofMultiVOIP units.

ConfigurationPrerequisite

Local Windows GUI must be used toassign IP address to MultiVOIP.

Browser VersionRequirement

Internet Explorer 6.0 or higher; orNetscape 6.0 or higher

Java Requirement Java Runtime Environmentversion 1.4.0_01 or higher(this application program is includedwith MultiVOIP)

Video Usability large video monitor recommended


476

The initial configuration step of assigning the voip unit an IP address must stillbe done locally using the Windows GUI. However, all additionalconfiguration can be done via the web GUI.

The content and organization of the web GUI is directly parallel to theWindows GUI. For each screen in the Windows GUI, there is a correspondingscreen in the web GUI. The fields on each screen are the same, as well.

The Windows GUI gives access to commands via icons and pulldown menuswhereas the web GUI does not.

The web GUI, however, cannot perform logging in the same direct mode donein the Windows GUI. However, when the web GUI is used, logging can bedone by email (SMTP).


477

The graphic layout of the web GUI is also somewhat larger-scale than that ofthe Windows GUI. For that reason, it’s helpful to use as large of a videomonitor as possible.


In order to use the web GUI, you must also install a Java application programon the controller PC. This Java program is included on the MultiVOIP productCD. ). Java is needed to support drop-down menus and multiple windows inthe web GUI.

To install the Java program, go to the Java directory on the MultiVOIPproduct CD. Double-click on the EXE file to begin the installation. Followthe instructions on the Install Shield screens.


478

During the installation, you must specify which browser you’ll use in theSelect Browsers screen.

When installation is complete, the Java program becomes accessible in yourStart | Programs menu (Java resources are readily available via the web).However, the Java program runs automatically in the background as a plug-insupporting the MultiVOIP web GUI. No overt user actions are required.


479

After the Java program has been installed, you can access the MultiVOIP usingthe web browser GUI. Close the MultiVOIP Windows GUI. Start the webbrowser. Enter the IP address of the MultiVOIP unit. Enter a password whenprompted. (A password is needed here only if password has been set for thelocal Windows GUI or for the MultiVOIP’s FTP Server function. See “Settinga Password -- Web Browser GUI” earlier in this chapter.) The web browserGUI offers essentially the same control over the voip as can be achieved usingthe Windows GUI. As noted earlier, logging functions cannot be handled viathe web GUI. And, because network communications will be slower thandirect communications over a serial PC cable, command execution will besomewhat slower over the web browser GUI than with the Windows GUI.


480

SysLog Server FunctionsBeginning with the 4.03/6.03 software release, we have built SysLog serverfunctionality into the software of the MultiVOIP units. SysLog is a de factostandard for logging events in network communication systems.

The SysLog Server resides in the MultiVOIP unit itself. To implement thisfunctionality, you will need a SysLog client program (sometimes referred to asa “daemon”). SysLog client programs, both paid and freeware, can be obtainedfrom Kiwi Enterprises, among other firms. Read the End-User LicenseAgreement carefully and observe license requirements. Seewww.kiwisyslog.com. SysLog client programs essentially give you a means ofstructuring console messages for convenience and ease of use.

MultiTech Systems does not endorse any particular SysLog client program.SysLog client programs by qualified providers should suffice for use withMultiVOIP units. Kiwi’s brief description of their SysLog program is asfollows:

“Kiwi Syslog Daemon is a freeware SyslogDaemon for the Windows platform. It receives,logs, displays and forwards Syslog messages fromhosts such as routers, switches, Unix hosts andany other syslog enabled device. There are manycustomizable options available.”



481

Before a SysLog client program is used, the SysLog functionality must beenabled within the MultiVOIP in the Logs menu under Configuration.

The IP Address used will be that of the MultiVOIP itself.

In the Port field, entered by default, is the standard (‘well-known’) logicalport, 514.


482

Configuring the SysLog Client Program. Configure the SysLog clientprogram for your own needs. In various SysLog client programs, you candefine where log messages will be saved/archived, opt for interaction with anSNMP system (like MultiVoipManager), set the content and format of logmessages, determine disk space allocation limits for log messages, andestablish a hierarchy for the seriousness of messages (normal, alert, critical,emergency, etc.). A sample presentation of SysLog info in the Kiwi daemon isshown below. SysLog programs will vary in features and presentation.

483

Chapter 11 Warranty, Service, andTech Support

Warranty, Service, & Tech Support MultiVOIP User Guide

484

Limited WarrantyMulti-Tech Systems, Inc. (“MTS”) warrants that its products will be free fromdefects in material or workmanship for a period of two years from the date ofpurchase, or if proof of purchase is not provided, two years from date ofshipment. MTS MAKES NO OTHER WARRANTY, EXPRESSED ORIMPLIED, AND ALL IMPLIED WARRANTIES OF MERCHANTABILITYAND FITNESS FOR A PARTICULAR PURPOSE ARE HEREBYDISCLAIMED. This warranty does not apply to any products which havebeen damaged by lightning storms, water, or power surges or which have beenneglected, altered, abused, used for a purpose other than the one for which theywere manufactured, repaired by the customer or any party without MTS’swritten authorization, or used in any manner inconsistent with MTS’sinstructions.

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 thewarranty period, or, at MTS’s option, issuance of a refund of the purchaseprice. Defective products must be returned by Customer to MTS’s factory—transportation prepaid.

MTS WILL NOT BE LIABLE FOR CONSEQUENTIAL DAMAGES ANDUNDER NO CIRCUMSTANCES WILL ITS LIABILITY EXCEED THEPURCHASE PRICE FOR DEFECTIVE PRODUCTS.

Repair Procedures for U.S. and CanadianCustomers

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 DriveMounds View, MN 55112Attn: Repairs, Serial # ________________

A Returned Materials Authorization (RMA) is not required. Return shippingcharges (surface) will be paid by MTS.

Please include, inside the shipping box, a description of the problem, a returnshipping address (it must be a street address, not a P.O. Box number), yourtelephone number, and if the product is out of warranty, a check or purchaseorder for repair charges.

MultiVOIP User Guide Warranty, Service, & Tech Support

485

For out-of-warranty repair charges, go to www.multitech.com/documents/warranties

Extended two-year overnight replacement service agreements are available forselected products. Please call MTS at (888) 288-5470, extension 5308, or visitour web site at www.multitech.com/programs/orcfor details on rates and coverages.

Please direct your questions regarding technical matters, product configuration,verification that the product is defective, etc., to our Technical Supportdepartment at (800) 972-2439 or [email protected]. Please direct your questions regarding repairexpediting, receiving, shipping, billing, etc., to our Repair Accountingdepartment at (800) 328-9717 or (763) 717-5631, or [email protected].

Repairs for damages caused by lightning storms, water, power surges, incorrectinstallation, physical abuse, or used-caused damages are billed on a time-plus-materials basis.

http://www.multitech.com/programs/orc

mailto:[email protected]


Warranty, Service, & Tech Support MultiVOIP User Guide

486

Technical SupportMulti-Tech Systems has an excellent staff of technical support personnelavailable to help you get the most out of your Multi-Tech product. If you haveany questions about the operation of this unit, or experience difficulty duringinstallation you can contact Tech Support via the following:

Contacting Technical SupportCountry By E-mail By telephone

France [email protected] (33) 1-64 61 0981

India [email protected]

(91) 124-340778

U.K. [email protected]

(44) 118 959 7774

U.S. &Canada

[email protected]

(800) 972-2439

Rest ofWorld

[email protected]

(763) 785-3500

Internet: http://www.multitech.com/ _forms/email_tech_support.htm

Please have your product information available, including model and serialnumber.



487

Chapter 12: Regulatory Information

Regulatory Information MultiVOIP User Guide

488

EMC, Safety, and R&TTE Directive ComplianceThe CE mark is affixed to this product to confirm compliance with thefollowing European Community Directives:

Council Directive 89/336/EEC of 3 May 1989 on the approximation of the laws ofMember States relating to electromagnetic compatibility,andCouncil Directive 73/23/EEC of 19 February 1973 on the harmonization of the lawsof Member States relating to electrical equipment designed for use within certainvoltage limits,andCouncil Directive 1999/5/EC of 9 March 1999 on radio equipment andtelecommunications terminal equipment and the mutual recognition of theirconformity.

FCC DeclarationNOTE: This equipment has been tested and found to comply with the limitsfor a Class A digital device, pursuant to Part 15 of the FCC Rules. Theselimits are designed to provide reasonable protection against harmfulinterference when the equipment is operated in a commercial environment.This equipment generates, uses and can radiate radio frequency energy, and ifnot installed and used in accordance with the instructions, may cause harmfulinterference to radio communications. Operation of this equipment in aresidential area is likely to cause harmful interference in which case the userwill 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 undesiredoperation.

Warning: Changes or modifications to this unit not expressly approved by theparty responsible for compliance could void the user’s authority to operate theequipment.

MultiVOIP User Guide Regulatory Information

489

Industry CanadaThis Class A digital apparatus meets all requirements of the CanadianInterference-Causing Equipment Regulations.Cet appareil numérique de la classe Arespecte toutes les exigences duReglement Canadien sur le matériel brouilleur.

FCC Part 68 Telecom1. This equipment complies with part 68 of the Federal Communications

Commission Rules. On the outside surface of this equipment is a label thatcontains, among other information, the FCC registration number. Thisinformation must be provided to the telephone company.

2. As indicated below, the suitable jack (Universal Service Order Codeconnecting arrangement) for this equipment is shown. If applicable, thefacility interface codes (FIC) and service order codes (SOC) are shown.

3. An FCC compliant telephone cord and modular plug is provided with thisequipment. This equipment is designed to be connected to the telephonenetwork or premises wiring using a compatible modular jack that is Part 68compliant. See installation instructions for details.

4. If this equipment causes harm to the telephone network, the telephonecompany will notify you in advance that temporary discontinuance of servicemay be required. If advance notice is not practical, the telephone companywill notify the customer as soon as possible.

5. The telephone company may make changes in its facilities, equipment,operation, or procedures that could affect the operation of the equipment. Ifthis happens, the telephone company will provide advance notice to allowyou to make necessary modifications to maintain uninterrupted service.

6. If trouble is experienced with this equipment (the model of which isindicated below), please contact Multi-Tech Systems, Inc. at the addressshown below for details of how to have repairs made. If the equipment iscausing harm to the network, the telephone company may request you toremove the equipment form t 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 andwarranty.

8. Manufacturer: Multi-Tech Systems, Inc. Trade name: MultiVOIP Model number: MVP-2410/3010/810/410/210/130 FCC registration number: US: AU7DDNAN46050

Regulatory Information MultiVOIP User Guide

490

Modular jack (USOC): RJ-48C Service center in USA: Multi-Tech Systems, Inc.

2205 Woodale DriveMounds View, MN 55112Tel: (763) 785-3500

FAX: (763) 785-9874

Canadian Limitations NoticeNotice: The Industry Canada label identifies certified equipment. Thiscertification means that the equipment meets certain telecommunicationsnetwork protective, operational and safety requirements. The Department doesnot guarantee the equipment will operate to the user’s satisfaction.

Before installing this equipment, users should ensure that it is permissible to beconnected to the facilities of the local telecommunications company. Theequipment must also be installed using an acceptable method of connection.The customer should be aware that compliance with the above conditions maynot prevent degradation of service in some situations.

Repairs to certified equipment should be made by an authorized Canadianmaintenance facility designated by the supplier. Any repairs or alterationsmade by the user to this equipment, or equipment malfunctions, may give thetelecommunications company cause to request the user to disconnect theequipment.

Users should ensure for their own protection that the electrical groundconnections of the power utility, telephone lines and internal metallic waterpipe system, if present, are connected together. This precaution may beparticularly important in rural areas.

Caution: Users should not attempt to make such connections themselves, butshould contact the appropriate electric inspection authority, or electrician, asappropriate.

491

Appendix A: Expansion CardInstallation

(MVP24-48 & MVP30-60)

T1/E1 Expansion Cards MultiVOIP User Guide

492

InstallationBoth the MVP2410 and the MVP3010 use the same mechanical chassis. Thischassis accommodates a second MultiVOIP circuit card or motherboardmodule. The add-on module for the MVP2410 is the MVP24-48 product; theadd-on module for the MVP3010 is the MVP30-60 product. The MVP2410Gwill not accept an expansion card because its second card slot is occupied bygatekeeper circuitry.

To install an expansion card into an MVP2410 or MVP3010, you must:

1. Power down and unplug the MVP2410/3010 unit.

2. Using a Phillips or star-bit screwdriver, remove the blank plate at the rear ofthe MVP2410/3010 chassis (see Figure A-1). Save the screw.

Figure A-1: Remove Plate Covering Expansion Slot

3. A power cable for the expansion card (+5V) is already present within theMVP2410/3010 unit. This power cable has a two-pin “molex” connector.When the rear cover plate has been removed, the cable is accessible from therear at the right side of the expansion slot. Locate this connector within theMVP2410/3010. See Figure A-2.

MultiVOIP User Guide T1/E1 Expansion Cards

493

Power Cable

Molex Connector

Figure A-2: MVP2410/3010 Chassis (top/rear view)

4. While keeping the power cable out of the way, fit the MVP24-48 orMVP30-60 card into the grooves of the expansion slot. Push it in far enough toallow connection of the power cable to the receptacle on the vertical plate ofthe expansion card. (See Figure A-2.) Connect the power cable.

5. Push the expansion card fully into the chassis. See Figure A-3.

Figure A-3: Sliding Expansion Card into Chassis

Secure the vertical plate of the expansion card to the chassis with a screw.

T1/E1 Expansion Cards MultiVOIP User Guide

494

OperationThe MVP2410/3010 front panel has two sets of identical LEDs. In theMVP2410/3010 without an expansion card, only the left-hand set of LEDs isfunctional. However, when the MultiVOIP unit has been upgraded with anMVP24-48 or MVP30-60 expansion card, the right-hand set of LEDs will alsobecome active.

Remember that the expansion card must be configured as though it weresimply another complete MultiVOIP unit: it requires its own T1/E1 line; itrequires its own connection to a computer running the MultiVOIPconfiguration software. All of the procedures and operations that apply to theoriginal motherboard of the MVP2410/3010 will also apply to the expansioncard. See applicable User Guide chapters for details.

495

Appendix B: Cable Pinouts

Cable Pinouts MultiVOIP User Guide

496

Appendix B: Cable Pinouts

Command Cable

RJ-45 Connector End-to-End Pin Info

��;��>��4��+��,

PIN NO.

4

7

8

3

2

6

1

5

1

2

3

4

5

6

7

8

To DTEDevice(e.g., PC)

CLEAR TO SEND

TRANSMIT DATA

RECEIVE DATA

SIGNAL GROUND

PIN NO.

To CommandPort Connector

DB9FRJ-45

RJ-45 connector plugs into Command Port ofMultiVOIP.DB-9 connector plugs into serial port of commandPC (which runs MultiVOIP configurationsoftware).

Ethernet ConnectorThe functions of the individual conductors of the MultiVOIP’s Ethernet port areshown on a pin-by-pin basis below.

RJ-45 Ethernet Connector Pin Circuit Signal Name

��;��>��4��+��,1236

TD+ Data Transmit PositiveTD- Data Transmit NegativeRD+ Data Receive PositiveRD- Data Receive Negative

MultiVOIP User Guide Cable Pinouts

497

T1/E1 ConnectorT1/E1 Connector

��;��>��4��+��,

Transmit Pair (to line)}45

Receive Pair (from line)}12

Voice/Fax Channel Connectors

��;��>��4��+��, ��;

Pin Functions (E&M Interface)Pin Descr Function

1 M Input

2 E Output

3 T1 4-Wire Output

4 R 4-Wire Input, 2-Wire Input

5 T 4-Wire Input, 2-Wire Input

6 R1 4-Wire Output

7 SG Signal Ground (Output)

8 SB Signal Battery (Output)


498

Pin Functions (FXS/FXO Interface)FXS Pin Description FXO Pin Description

2 N/C 2 N/C

3 Ring 3 Tip

4 Tip 4 Ring

5 N/C 5 N/C

MultiVOIP User Guide Cable Pinouts

499

ISDN BRI RJ-45 Pinout InformationThe S/T interface uses an 8-conductor modular cable terminated with an 8-pinRJ-45 plug. An 8-pin RJ-45 jack located on the terminal is used to connect theterminal to the DSL (Digital Subscriber Loops) using this modular cable.

The table below shows the Pin Number, Terminal Pin Signal Name andNetwork Pin Signal name for the S/T interface.

Pin TE Signal NT Signal Pin

1 Not used Not used 12 Not used Not used 23 Tx+ Rx+ 34 Rx- Tx- 45 Rx+ Tx+ 56 Tx- Rx- 67 Not used Not used 78 Not used Not used 8

��;��>��4��+��,

TE=Terminal Equipment

NT=Network


500

ISDN Interfaces: “ST” and “U”

The MVP410ST and MVP810ST are ISDN-BRI voip units that use an S/Toutlet interface. You will need an NT1 device to connect these units to anynetwork equipment that has the “U” ISDN interface. In the UK, and in manyEuropean countries, the telco supplies an NT1 device for ISDN-BRI service.

An ISDN Basic Rate (BRI) U-Loop consists of two conductors from the telcocentral office to the customer premises. The equipment on both sides of the U-loop accommodates the extensive length of the U-loop and the noisyenvironment in which it may operate. At the customer premises, the U-loop isterminated by an NT1 (network termination 1 ) device. An NT1 device makesan end-user’s 4-wire terminal equipment compatible with the telco’s 2-wiretwisted pair ISDN-BRI line.

The NT1 drives an S/T bus. The S/T bus is usually made up of 4 wires, but insome cases may be 6 or 8 wires.

“S” and “T” refer to connection points in the ISDN specification.

When a PBX is present, S refers to the connection between the PBX and theterminal. (“Terminal” can mean any sort of end-user ISDN device: dataterminals, telephones, FAX machines, voip units, etc.)

Point T refers to the connection between the NT1 device and customer suppliedequipment. Terminals can connect directly to the NT1 device at point T, orthere may be a PBX (private branch exchange, i.e., a customer-ownedtelephone exchange). The figure below shows “S” and “T” connection pointsin an ISDN network.

TelcoCentralOffice

Point “S”

Point “T”4-8 Wires

Point “S”4-8 Wires

Point “U”2 Wires

Point “S”

NT1

NT2(PBX)

TerminalTerminal

Terminal

501

Appendix C: TCP/UDP PortAssignments

TCP/UDP Port Assignments MultiVOIP User Guide

502

Well Known Port Numbers

The following description of port number assignments for Internet Protocol (IP)communication is taken from the Internet Assigned Numbers Authority (IANA) website (www.iana.org).

“The Well Known Ports are assigned bythe IANA and on most systems can onlybe used by system (or root) processes orby programs executed by privilegedusers. Ports are used in the TCP[RFC793] to name the ends of logicalconnections which carry long termconversations. For the purpose ofproviding services to unknown callers, aservice contact port is defined. This listspecifies the port used by the serverprocess as its contact port. The contactport is sometimes called the "well-known port". To the extent possible,these same port assignments are usedwith the UDP [RFC768]. The range forassigned ports managed by the IANA is0-1023.”

Well-known port numbers especially pertinent to MultiVOIP operation are listedbelow.

Port Number Assignment List

Well-Known Port Numbers

Function Port Numbertelnet 23tftp 69snmp 161snmp tray 162gatekeeper registration 1719H.323 1720SIP 5060SysLog 514

503

Appendix D: InstallationInstructions for MVP428

Upgrade Card

8-Channel Analog Expansion Card MultiVOIP User Guide

Installation Instructions for MVP428 Upgrade Card

In this procedure, you will install an additional circuit board into the MVP410,converting it from a 4-channel voip to an 8-channel voip.

at the rear of the

Summary: (A) Attach four standoffs to main circuit card.(B) Mate the 60-pin connectors (male connector

on main circuit card; female on upgrade card).(C) Attach upgrade card to main circuit card (4

screws).

* *(A)

Replace main card screws

(C)ttach upgrade cardrews into standoffs

-- 4 places).

Procedure in Detail1. Power down and unplug the MVP410 unit.

2. Using a Phillips driver, remove the blank cover plate MVP410 chassis. Save the screws.

screws on blank cover plate (2)

* Add standoffs here (2 places).

with standoffs here(2 places).

(B)Mate 60-pinconnectors.

A(sc

Figure D-1. Installation Summary

504

Figure D-2: Removing screws from blank cover plate

MultiVOIP User Guide 8-Channel Analog Expansion Card

505

3. Using a Phillips driver, remove the three screws that secure the main circuitboard and back panel assembly to the chassis.

NOTE:Follow standard ESDprecautions to protect thecircuit board from staticelectricity damage.

back panel screws (3)

Figure D-3: Removing screws from back panel

4. Slide the main circuit board out of the chassis far enough to unplug thepower connector.

power connector

Figure D-4: Accessing power connector

5. Unplug the power connector from the main circuit board.6. Slide the main circuit board completely out of the chassis and place on a

non-conductive, static-safe tabletop surface.7. Remove mounting hardware (2 screws, 2 nuts, and 4 standoffs) from its

package.

8-Channel Analog Expansion Card MultiVOIP User Guide

506

8. On the phone-jack side of the circuit card, three screws attach the circuitcard to the back panel. Two of these screws are adjacent to the four phone-jack pairs. Remove these two screws.

Screw locations (2)at phone-jack edgeof board.

Figure D-5: Screws to be removed and replaced with standoffs(phone-jack edge of board; top view)

9. Replace these two screws with standoffs.10. There are two copper-plated holes at the LED edge of the circuit card.

Place a nut beneath each hole (lockwasher side should be in contact withboard) and attach a standoff to each location).

Standoff locations (2) at LED edgeof board (top view).

Standoff/nut attachment(rear bottom view)

Figure D-6: Standoffs at LED edge of board (top view)

MultiVOIP User Guide 8-Channel Analog Expansion Card

507

11. Locate the male 60-pin vertical connector near the LED edge of the maincircuit card. Check that pins are straight and evenly spaced. If not, thencorrect for straightness and spacing. Locate the 60-pin female connectoron the upgrade circuit card.

12. Set the upgrade circuit card on top of the main circuit card. Align theupgrade card’s 4 pairs of phone-jacks with the 4 pairs of holes in thebackplane of the main card. Slide the phone jacks into the holes.

13. Mate the upgrade card’s 60-pin female connector with the main card’s 60-pin male connector.

**

*

*

60-pin connectors

*These screws (4 places)attach upgrade cardto main card.

Figure D-7. Attaching upgrade card to main circuit card(secure 4 Phillips screws; mate 60-pin connectors)

14. There are four copper-plated attachment holes, two each at the front andrear edges of the upgrade card. Attach the upgrade card to the main cardusing 4 Phillips screws. The upgrade card should now be firmly attachedto the main card.

15. Slide the main circuit card back into the chassis far enough to allow re-connection of power cable.

16. Re-connect power cable.17. Slide the main circuit card fully into the chassis.18. Re-attach the backplane of the main circuit card to the chassis with 3

screws.

Index MultiVOIP User Guide

508

Index

MultiVOIP User Guide Index

509

INDEX(

Alternate Phone Number field, SPPE1.............................................. 373

Alternate Phone Number, SPPT1.............................................. 323

A

abbreviated dialing, inter-officeE1.............................................. 351T1.............................................. 306

access codes, PBX .......................... 65access codes, types

PBX ............................................ 70PSTN .......................................... 70special ......................................... 70

access digits, PBX65. See phonebookdigits, types used

access to networkanalog........................................ 366T1/E1 ........................................ 169

access to remote PSTNE1................................................ 18T1................................................ 11

accessing Statistics, Logs screen . 412accessing Call Progress (Statistics)

screen ........................................ 406accessing configuration parameter

groupsanalog........................................ 316T1/E1 ........................................ 133

accessing interface parameters...... 337accessing IP Parameters screen

analog........................................ 317T1/E1 ........................................ 134

accessing IP Statistics screen........ 416accessing Logs (Statistics) screen. 412accessing logs screen

analog........................................ 381T1/E1 ........................................ 184

accessing Regional Parametersanalog........................................ 367T1/E1 ........................................ 170

accessing Registered Gateway Details(Statistics) screen ......................439

accessing Registered Gateway Detailsscreen ................................ 438, 439

accessing RTP Parameters screen .423accessing SMTP parameters

analog........................................375T1/E1 ........................................178

accessing SNMP parametersanalog........................................364T1/E1 ........................................167

accessing Supplementary Servicesscreenanalog........................................385T1/E1 ........................................188

accessing System Information screenanalog........................................397T1/E1 ........................................200

accessing T1 Statistics screen .......426accessing T1/E1/ISDN Parameters

screen ........................................152accessing Voice/FAX Parameters

screen ................................ 140, 324Add Inbound Phonebook Entry icons

E1 ..............................................357T1 ..............................................307

Add Outbound Phonebook Entry iconE1 ..............................................357T1 ..............................................307

Add Prefix (inbound) fieldE1 ..............................................376T1 ..............................................328

Add Prefix (outbound) fieldE1 ..............................................370T1 ..............................................320

Add/Edit Inbound Phonebook fielddefinitionsE1 .............................. 376, 377, 378T1 ...................................... 328, 329

Add/Edit Inbound Phonebook screenE1 ..............................................376T1 ..............................................328


510

Add/Edit Inbound Phonebook screenfields (E1)Add Prefix................................. 376Channel Number....................... 377Description (callee location) ..... 377Enable (Call Forwarding) ......... 377Forward Condition.................... 377Forward Destination ................. 377Registration Option Parameters 378Remove Prefix .......................... 376Ring Count................................ 378

Add/Edit Inbound Phonebook screenfields (T1)Add Prefix................................. 328Channel Number....................... 328Description (callee location) ..... 328Enable (Call Forwarding) ......... 328Forward Condition.................... 329Forward Destination ................. 329Registration Option Parameters 329Remove Prefix .......................... 328Ring Count................................ 329

Add/Edit Outbound Phonebook fielddefinitionsE1...................... 370, 371, 372, 373T1...................... 320, 321, 322, 323

Add/Edit Outbound Phonebook fields(E1)Add Prefix................................. 370Advanced button....................... 372Description................................ 370destination pattern..................... 370Gateway Prefix ......................... 371H.323 ID ................................... 371IP Address................................. 370Protocol Type ........................... 370Q.931 Port Number................... 371Remove Prefix .......................... 370SIP Port Number....................... 372SIP URL ................................... 372Total Digits ............................... 370Transport Protocol (SIP)........... 372Use Gatekeeper................. 371, 373Use Proxy (SIP) ........................ 372

Add/Edit Outbound Phonebook fields(T1)Add Prefix................................. 320Advanced button....................... 322

Description................................320destination pattern .....................320Gateway Prefix..........................321H.323 ID ...................................321IP Address.................................320Protocol Type............................320Q.931 Port Number ...................321Remove Prefix ..........................320SIP Port Number .......................322SIP URL....................................322Total Digits ...............................320Transport Protocol (SIP) ...........322Use Gatekeeper ................. 321, 323Use Proxy (SIP) ........................322

Add/Edit Outbound Phonebook screenE1 ..............................................369T1 ..............................................319

Add/Edit Outbound Phonebook SPPFieldsE1 ..............................................373T1 ..............................................323

add-on module (analog, 4-to-8channel), installation .................504

add-on module (T1/E1)operation ...................................494

add-on module (T1/E1), installation..................................................492

Address (SNMP) fieldanalog........................................366T1/E1 ........................................169

Advanced button, OutboundPhonebookE1 ..............................................373T1 ..............................................323

Advanced Features field groupanalog........................................330T1/E1 ........................................146

airflow...........................................117Alerting Party

Supplementary Services (analog).............................. 392, 393, 394

Supplementary Services (T1/E1).............................. 195, 196, 197

Allowed Name Type (analog)Alerting Party............ 392, 393, 394Calling Party .............................391

Allowed Name Type (T1/E1)Alerting Party............ 195, 196, 197


511

Calling Party ............................. 194Allowed Name Types, Call Name ID

(analog)Alerting Party............................ 392Busy Party................................. 393Calling Party ............................. 391Connected Party........................ 394

Allowed Name Types, Call Name ID(T1/E1)Alerting Party............................ 195Busy Party................................. 196Calling Party ............................. 194Connected Party........................ 197

Alternate IP Address fieldE1.............................................. 375T1.............................................. 325

Alternate IP RoutingE1.............................................. 369T1.............................................. 319

Alternate Phone Number, SPP(Add/Edit Outbound Phonebook)E1.............................................. 373T1.............................................. 323

Alternate RoutingPSTN failover feature, and ....... 325

Alternate Routing field definitionsE1.............................................. 375T1.............................................. 325

Alternate Routing field definitions(E1)Alternate IP Address................. 375Round Trip Delay ..................... 375

Alternate Routing field definitions(T1)Alternate IP Address................. 325Round Trip Delay ..................... 325

analog phonebook......................... 404using T1 & E1 examples for ..... 404

analog phonebook examples ......... 304analog telephony interface parameters

.................................................. 308Annex E field

E1.............................................. 365T1.............................................. 314

area codes ....................................... 69Auto Disconnect field group

analog........................................ 336T1/E1 ........................................ 151

AutoCallanalog/BRI ................................331T1/E1 ........................................147

AutoCall (Voice/Fax Params - analog)and Pass Through Enable (FXSLoop Start) ................................331

AutoCall/Offhook Alert fieldanalog/BRI ........................331, 332T1/E1 ........................................147

Automatic Disconnection fieldanalog........................................336

B

bandwidthcoder (analog) ...........................329coder (T1/E1) ............................145

battery caution...............................111baud rate, default (MultiVOIP

software connection)T1/E1 ....................................199analog....................................396

baud rate, faxanalog........................................328T1/E1 ........................................144

baud rate, settinganalog........................................396T1/E1 ........................................199

Bipolar Violation (E1 stats) field ..433Bipolar Violation (T1 stats) field ..430Blue Alarm (E1 stats) field ...........431Blue Alarm (T1 stats) field ...........428Boot Code Version

System Info (analog).................398Boot LED

analog models .............................32analog, MVP130 .........................33BRI models .................................39MVP-210 ..................................133MVP-410/810 ...........................125MVP-410ST/810ST ..................129on MVP-2410/3010...................120

Boot VersionSystem Info (T1/E1).......... 201, 404

booting timeanalog..........................................32analog, MVP130 .........................33BRI..............................................39E1 ................................................24


512

T1................................................ 17box contents

verifying.................................... 112BRI connector pinout.................... 499BRI interface types

ST and U................................... 500built-in modem

setup in Regional Parametersscreen ...................................... 59

built-in modem (analog)setup in Regional Parameters

screen ............................ 311, 368built-in modem (T1/E1)

setup in Regional Parametersscreen .................... 128, 170, 171

busy & no-response (T1)forwarding, dual conditions329,

377busy tone, custom

analog........................................ 373T1/E1 ................................ 175, 176

busy-tonesanalog........................................ 372T1/E1 ........................................ 175

Bytes Received (call progress) field.................................................. 408

Bytes Received (SMTP logs) fieldanalog........................................ 378T1/E1 ........................................ 181

Bytes received (statistics, logs) field.................................................. 414

Bytes Sent (call progress) field ..... 408Bytes Sent (SMTP logs) field

analog........................................ 378T1/E1 ........................................ 181

Bytes sent (statistics, logs) field ... 413

C

cable length, maximum spanE1.............................................. 161T1.............................................. 155

cabling diagram, quickanalog models ........... 52, 53, 55, 56BRI models ................................. 54E1 models ................................... 52MVP130...................................... 56MVP210...................................... 55MVP2410.................................... 52

MVP3010....................................52MVP410......................................53MVP-410ST/810ST ....................54MVP810......................................53T1 models....................................52

cabling problem, fixinganalog models ...........................316T1/E1 models ............................133

cabling procedureMVP130....................................134MVP210....................................130MVP2410..................................119MVP3010..................................119MVP410....................................121MVP-410ST..............................126MVP810....................................121MVP-810ST..............................126

Cadence 1 (custom) fieldanalog........................................374T1/E1 ........................................177




Cadence fieldanalog........................................371T1/E1 ........................................174

cadences, customT1.E1................................. 177, 374T1/E1 ........................................175

cadences, signalinganalog........................................367T1/E1 ........................................170

Call Control PHB fieldanalog........................................319T1/E1 ........................................136

Call Control StatusCall Progress Details (statistics)

field .......................................411Call Control Status (call progress)

field ...........................................411Call Duration field

analog........................................336


513

T1/E1 ........................................ 151Call Forward Parameters (inbound

phonebook)E1.............................................. 377T1.............................................. 329

Call Forwarded Tologs (statistics) field.................. 415

Call HoldANALOG ................................... 30BRI ............................................. 38E1................................................ 23T1................................................ 16

Call Hold (analog) ........................ 386Call Hold (T1/E1) ......................... 189Call Hold Enable

analog........................................ 389T1/E1 ........................................ 192

Call Mode (SMTP logs) fieldanalog........................................ 378T1/E1 ........................................ 181

Call Name IdentificationANALOG ................................... 30BRI ............................................. 38E1................................................ 23T1................................................ 16

Call Name Identification (analog)Alerting Party............ 392, 393, 394Calling Party ............................. 391

Call Name Identification (T1/E1)Alerting Party............ 195, 196, 197Calling Party ............................. 194

Call Name Identification (analog) 386Call Name Identification (T1/E1) . 189Call On Hold

Call Progress Details (statistics)field............................... 408, 410

Call on Hold (call progress) field . 410Call Progress (Statistics)............... 406Call Progress Details (statistics)

screen fieldCall On Hold......................... 408Call Waiting.......................... 408Caller ID ............................... 408Call On Hold......................... 410Call Waiting.......................... 410Caller ID ............................... 411

Call Progress Details (statistics)screen fields

Channel .................................408Duration ................................408Mode .....................................408Voice Coder ..........................408Packets Sent ..........................408Packets Received...................408Bytes Sent .............................408Bytes Received......................408Packets Lost ..........................408Outbound Digits....................408Prefix Matched......................408Gateway Name......................409IP Address.............................409Options..................................409Silence Compression.............409Forward Error Correction......409Status.....................................411Call Control Status ................411

Call Signalling Port fieldE1 ..............................................362T1 ..............................................311

Call Status (SMTP logs) fieldanalog........................................379T1/E1 ........................................182

Call TransferANALOG....................................30BRI..............................................38E1 ................................................23T1 ................................................16

Call Transfer (analog) ...................386Call Transfer (T1/E1)....................189Call Transfer Enable

analog........................................388T1/E1 ........................................191

Call Transfer music jingle during holdanalog........................................388T1/E1 ........................................191

Call Transferred Tologs (statistics) field ..................415

Call WaitingANALOG....................................30BRI..............................................38Call Progress Details (statistics)

field ............................... 408, 410E1 ................................................23T1 ................................................16

Call Waiting (analog)....................386Call Waiting (call progress) field..410


514

Call Waiting (T1/E1) .................... 189Call Waiting Enable

analog........................................ 389T1/E1 ........................................ 192

Caller IDCall Progress Details (statistics)

field............................... 408, 411Caller ID (analog) ......................... 386Caller ID (call progress) field ....... 411Caller ID (Supplementary Services)

fieldanalog........................................ 395T1/E1 ........................................ 198

Caller ID (T1/E1).......................... 190Caller ID enable

FXO .......................................... 348FXS Loop Start ................. 342, 343

Caller ID examples ....... 349, 350, 351Caller ID fields

FXO .......................................... 348Caller ID Type

FXO .......................................... 348FXS Loop Start ......................... 342

Caller Name Identification Enableanalog........................................ 390T1/E1 ........................................ 193

calling area codes............................ 69Calling Party

Supplementary Services (analog).............................................. 391

Supplementary Services (T1/E1).............................................. 194

Canadian Class A requirements .... 489Canadian Limitations Notice

(regulatory) ............................... 490CAS Protocol field

E1.............................................. 162T1.............................................. 156

CAS Protocols, downloading........ 450CAS vs. CCS

T1...................................... 156, 162CCS vs. CAS

T1...................................... 156, 162CD

MultiVOIP .................................. 44Channel (call progress) field......... 408channel capacity................................ 9

analog.......................................... 26

BRI..............................................34E1 ................................................18T1 ................................................11

Channel Number (inbound) fieldE1 ..............................................377T1 ..............................................328

Channel Number (SMTP logs) fieldanalog........................................378T1/E1 ........................................181

channel tracing on/off (logging)analog........................................384T1/E1 ........................................187

city codes ........................................69Clear (button), ISDN BRI Statistics

screen ........................................437Clear (IP Statistics) button ............417Clear command (Link Management)

button ........................................421Client Options fields

E1 ..............................................367T1 ..............................................316

Clocking fieldE1 ..............................................165T1 ..............................................159

coder (analog)bandwidth, max.........................329G.711.........................................329G.723.1......................................329G.726.........................................329G.727.........................................329G.729.........................................329Net Coder ..................................329

Coder (SMTP logs) fieldanalog........................................378T1/E1 ........................................181

coder (T1/E1)bandwidth, max.........................145G.711.........................................145G.723.1......................................145G.726.........................................145G.727.........................................145G.729.........................................145Net Coder ..................................145

Coder fieldanalog........................................329T1/E1 ........................................145

coder optionspacketization rates and ..............423


515

Coder Parameters field groupanalog........................................ 329T1/E1 ........................................ 145

coder types (voice/fax, RTPpacketization)T1/E1 ........................................ 424

COM porton command PC........................ 116

COM port (analog models)conflict, resolving ..................... 315error message ............................ 315

COM port (T1/E1 models)conflict, resolving ..................... 132error message ............................ 132

COM port allocationanalog........................................ 396T1/E1 ........................................ 199

COM port assignmentsanalog........................................ 396T1/E1 ........................................ 199

COM port conflicterror message ............................ 116

COM Port Setup screen ................ 116COM Port Setup screen (analog

models) ..................................... 315COM Port Setup screen (T1/E1

models) ..................................... 132comma

meaning/use in phonebook ......... 71comma use

and second dial tone.................... 71command cable pinout .................. 496command modem

and Regional Parameters screen . 59and Regional Parameters screen

(analog) ......................... 311, 368and Regional Parameters screen

(T1/E1).................. 128, 170, 171Command Modem

setup for ...................................... 59Command Modem (analog)

setup for ............................ 311, 368Command Modem (T1/E1)

setup for .................... 128, 170, 171command PC

COM port assignment (detailed)116COM port requirement................ 51demands upon ............................. 51

non-dedicated use........................51operating system .........................51settings ........................................51specifications...............................51

Command PCCOM port requirement................40non-dedicated use of ...................40operating system .........................40

community (voip) definedanalog........................................366T1/E1 ........................................169

Community Name 1 (SNMP) fieldanalog........................................366T1/E1 ........................................169

compatibility, Fast StartE1 ..............................................362T1 ..............................................311

compatibility, H.450 with H.323, notwith SIPanalog.................................. 27, 385BRI..............................................35E1 ................................................19T1 ................................................12T1/E1 ........................................188

compression standardE1 ..............................................165T1 ..............................................159

compression, silenceanalog........................................330T1/E1 ........................................146

Compression, Silence (SMTP logs)analog........................................379T1/E1 ........................................182

computer requirements....................40configuration of voip (analog)

local versus remote....................305configuration of voip (T1/E1)

local versus remote............122, 123Configuration option (MultiVOIP

program menu)..........................440Configuration Parameter Groups,

accessinganalog........................................316T1/E1 ........................................133

configuration procedure, localdetailed, analog .........................312detailed, T1/E1..........................129summary, analog .......................311


516

summary, T1/E1 ....................... 128configuration, local

analog/BRI................................ 307T1/E1 ........................................ 124

configuration, phonebookE1.............................................. 356starter .......................................... 62T1.............................................. 306

configuration, savinganalog........................................ 399T1/E1 ........................................ 202user ........................................... 455

configuration, starterphone/IP...................................... 58

configuration, user defaultanalog........................................ 400T1/E1 ........................................ 203

Configuring MultiVOIP phonebooks,generalE1.............................................. 350T1.............................................. 305

confirming connectivity.................. 80conflicts

COM port.................................. 116Connection Problems, Solving

analog........................................ 315T1/E1 ........................................ 132

connectivityconfirmation of ........................... 80confirming with remote voip 50, 62pinging and ................................. 81

connectivity test .............................. 77Consecutive Packets Lost field

analog........................................ 336T1/E1 ........................................ 151

Console Message Settings, Filters foranalog........................................ 384T1/E1 ........................................ 187

console messages .......... 60, 77, 79, 80console messages, enabling

analog........................................ 382T1/E1 ........................................ 185

console parameters trackedanalog........................................ 384T1/E1 ........................................ 187

contacting technical support ......... 486coordinated phonebook entries

E1.............................................. 356

T1 ..............................................306Copy Channel command (Interface

Parameters)analog........................................339

Copy Channel command (Voice/FaxParameters)analog........................................326T1/E1 ........................................142

Copy Channel fieldanalog........................................327

Copy Channel field (Voice/FaxParameters)T1/E1 ........................................143

Copy Channel, SupplementaryServices commandanalog........................................387T1/E1 ........................................190

Copy Channel, SupplementaryServices fieldanalog........................................395T1/E1 ........................................198

Copy Interface commandBRI............................................360

Count of Registered Numbers field(Registered Gateway Details) ...439

countryISDN type and...........................166switch type and ISDN ...............166

Country (ISDN) fieldE1/ISDN....................................164

country codes ..................................69Country definitions

ISDN-BRI .................................363Country field

ISDN-BRI .................................361Country field (ISDN)

T1/ISDN....................................158Country Selection for Built-In Modem

fieldanalog (except MVP130) ..........371T1/E1 ........................................174

Country/Region (tone schemes) fieldanalog................................ 369, 370T1/E1 ........................................172

CRC and ESF frame format (T1) ..155CRC Check field

T1 ..............................................155Creating a User Default Configuration


517

analog........................................ 400T1/E1 ........................................ 203

CT Ph#logs (statistics) field.................. 415

Current Loss (FXO disconnectcriteria) field ............................. 346

Current Loss Detect Timer (FXO)field........................................... 346

Current Loss fieldFXS Loop Start ......................... 341

Custom (tones, Regional)fieldanalog........................................ 371T1/E1 ........................................ 174

custom cadencesanalog........................................ 374T1/E1 ........................................ 177

custom DTMFanalog........................................ 373T1/E1 ................................ 175, 176

Custom Fields (SMTP) definitionsanalog................................ 378, 379T1/E1 ................................ 181, 182

Custom Fields, SMTP log email(analog)Bytes Received ......................... 378Bytes Sent ................................. 378Call Mode ................................. 378Call Status................................. 379Channel Number....................... 378Coder ........................................ 378Options...................................... 379Options...................................... 379Description (callee)................... 379Description (caller) ................... 379Duration .................................... 378From Gateway Number ............ 379From IP Address ....................... 379Outbound Digits........................ 379Packets Lost .............................. 378Packets Received ...................... 378Packets Sent .............................. 378Prefix Matched ......................... 379Select All .................................. 378Start Date, Time........................ 378To Gateway Number................. 379To IP Address ........................... 379

Custom Fields, SMTP log email(T1/E1)

Bytes Received..........................181Bytes Sent .................................181Call Mode..................................181Call Status .................................182Channel Number .......................181Coder.........................................181Options......................................182Options......................................182Description (callee) ...................182Description (caller) ...................182Duration ....................................181From Gateway Number.............182From IP Address .......................182Outbound Digits........................182Packets Lost ..............................181Packets Received ......................181Packets Sent ..............................181Prefix Matched..........................182Select All...................................181Start Date, Time ........................181To Gateway Number.................182To IP Address ...........................182

Custom Tone-Pair Settings (analog)fieldsCadence 1..................................374Cadence 2..................................374Cadence 3..................................374Cadence 4..................................374

Custom Tone-Pair Settings (T1/E1)fieldsCadence 1..................................177Cadence 2..................................177Cadence 3..................................177Cadence 4..................................177

Custom Tone-Pair Settings definitionsanalog................................ 373, 374T1/E1 ................................ 176, 177

Custom Tone-Pair Settings fields(analog)Frequency 1...............................373Frequency 2...............................373Gain 1........................................373Gain 2........................................373Tone Pair...................................373

Custom Tone-Pair Settings fields(T1/E1)Frequency 1...............................176Frequency 2...............................176


518

Gain 1 ....................................... 176Gain 2 ....................................... 176Tone Pair................................... 176

custom tones, settingT1/E1 ........................................ 175

customized log emailanalog................................ 378, 379T1/E1 ................................ 181, 182

D

D Channel Information fields (ISDNBRI Statistics)........................... 436

data capacity ..................................... 9analog.......................................... 26BRI ............................................. 34E1................................................ 18T1................................................ 11

data compressionanalog.......................................... 27BRI ............................................. 35E1................................................ 19T1................................................ 12

Date & Time Setup (program menuoption), command ..................... 442

Date and Time Setup option(MultiVOIP program menu) ..... 440

debugging messagesanalog........................................ 383T1/E1 ........................................ 185

Default (Supplementary Services)fieldanalog........................................ 395T1/E1 ........................................ 198

Default (Voice/FAX) fieldanalog........................................ 327T1/E1 ........................................ 143

default baud rate (MultiVOIPsoftware connection)analog........................................ 396T1/E1 ........................................ 199

default configuration, useranalog........................................ 400T1/E1 ........................................ 203

default values, software ................ 452delay, packets

analog........................................ 334T1/E1 ........................................ 149

delay, versus voice quality

analog........................................335T1/E1 ........................................150

Delete File buttonLogs (Statistics) screen .............413

Description (callee location)E1 ..............................................377T1 ..............................................328

Description (callee, outboundphonebook)E1 ..............................................370T1 ..............................................320

Description field (Registered GatewayDetails)......................................439

Description, From Details (SMTPlogs) fieldanalog........................................379T1/E1 ........................................182

Description, To Details (SMTP logs)fieldanalog........................................379T1/E1 ........................................182

Destination Pattern (outbound) fieldE1 ..............................................370T1 ..............................................320

destination patternsdigits used ...................................69tips about.....................................69

destination patterns, discussionE1 ..............................................355T1 ..............................................305

Detection Flash Hook fieldE1 ..............................................163T1 ..............................................157

Detection Range, Flash Hook OptionsfieldE&M .........................................355FXO ..........................................348FXS Loop Start .........................342

Detection TimeE1..............................................163T1 ..............................................157

Detection Time fieldE1 ..............................................163T1 ..............................................157

dial tone, customanalog........................................373T1/E1 ................................ 175, 176

dial tone, second


519

and comma use ........................... 71pausing for .................................. 71

Dialing Options (E&M) fields ...... 354Dialing Options (FXO) fields345, 346Dialing Options field

ISDN-BRI................................. 361dialing patterns

digits used ................................... 69inbound/outbound matching ....... 71tips about..................................... 69

dial-tonesanalog........................................ 372T1/E1 ........................................ 175

DID interface (MVP210)uses of ....................................... 133

DID interface (MVP-410/810)uses of ....................................... 124

DID Interface Parameter definitions.................................................. 357

DID Interface Parameter fieldsMessage Waiting Indication ..... 357

DID Interface Parameters ............. 356DID jumper

MVP210.................................... 130MVP-410/810 ........................... 122

DID lines (MVP210)polarity sensitivity and.............. 133

DID lines (MVP-410/810)polarity sensitivity and.............. 124

DID-DPO Interface Parameterdefinitions ................................. 356

DID-DPO Interface Parameter fieldsInter Digit Timer (dialing) ........ 357Start Modes............................... 356Wink Timer............................... 356

DID-DPO Parameter fieldsInter-Digit Regeneration Timer

(dialing) ................................ 357DID-DPO vs. DID-DPT ............... 356DiffServ and IP datagram

analog........................................ 320T1/E1 ........................................ 137

DiffServ PHB (Per Hop Behavior)valueanalog........................................ 319T1/E1 ........................................ 136

digits in phonebookspecialized codes ........................ 70

types ............................................69dimensions

analog models .............................43E1 models....................................42

Disconnect on Call Progress Tone(E&M) field ..............................355

Disconnect on Call Progress Tone(FXO) field................................347

Disconnect Tone Sequence (FXO)field ...........................................347

disconnection criteria, FXO ..346, 347DNS Server IP Address

T1/E1 ................................138, 321Download CAS Protocol (program

menu option) , command ..........450Download CAS Protocol option

(MultiVOIP program menu) .....440Download Factory Defaults (program

menu option) , command ..........452Download Factory Defaults option

(MultiVOIP program menu) .....441Download Firmware (program menu

option), command ............. 446, 447Download Firmware option

description (MultiVOIP programmenu) ........................................441

Download User Defaults (programmenu option) , command ..........454

Download User Defaults optiondescription (MultiVOIP programmenu) ........................................441

downloading firmware, machineperspective ........................ 441, 465

downloading user defaults ............454downloads vs. uploads (FTP)........465dropping digits, in phonebook.........70DTMF

extended ....................................347standard .....................................347

DTMF frequency chart..................347DTMF Gain (High Tones) field

analog........................................327T1/E1 ........................................143

DTMF Gain (Low Tones) fieldanalog........................................327T1/E1 ........................................143

DTMF Gain fieldanalog........................................327


520

T1/E1 ........................................ 143DTMF In/Out of Band field

analog........................................ 328T1/E1 ........................................ 144

DTMF inbandanalog........................................ 328T1/E1 ........................................ 144

DTMF out of bandanalog........................................ 328T1/E1 ........................................ 144

DTMF ParametersT1/E1 ........................................ 143

DTMF, custom tone pairsanalog........................................ 373T1/E1 ................................ 175, 176

Duration (call progress) field........ 408Duration (DTMF) field

analog........................................ 328T1/E1 ........................................ 144

Duration (SMTP logs) fieldanalog........................................ 378T1/E1 ........................................ 181

Duration (statistics, logs) field...... 413Dynamic Jitter Buffer field

analog........................................ 334T1/E1 ........................................ 149

Dynamic Jitter field groupanalog........................................ 334T1/E1 ........................................ 149

Dynamic Jitter fieldsanalog........................................ 335T1/E1 ........................................ 150

E

E&M interface (MVP210)matching telco trunk line .......... 132uses of ....................................... 132

E&M interface (MVP-410/810)matching telco trunk line .......... 124uses of ....................................... 124

E&M Interface Parameter fieldsDetection Range (flash hook) ... 355Disconnect on Call Progress Tone

.............................................. 355Flash Hook................................ 355Inter Digit Timer (dialing) ........ 354Interface .................................... 353Message Waiting Indication ..... 354

Pass Through.............................353Regeneration (dialing)...............354Signal ........................................353Type ..........................................353Wink Timer...............................353

E&M Parameter definitions353, 354,355

E&M Parameters...........................352E1 Parameter definitions161, 162,

163, 165Clocking....................................165Line Build-Out ..........................165Line Coding ..............................165PCM Law..................................165Pulse Shape Level .....................165

E1 Parameter fieldsCAS Protocol ........................162

CRC Check ...............................162Detect Flash Hook.....................163Frame Format............................162Long-Haul Mode.......................162

E1 Parameters screen ....................160E1 Statistics field definitions431, 432,

433E1 Statistics fields

Bipolar Variation ......................433Blue Alarm................................431Excessive Zeroes.......................433Loss of Frame Alignment..........431Loss of MultiFrame Alignment.432Receive Slip ..............................433Receive Timeslot 16 Alarm

Indication Signal ...................432Receive Timeslot 16 Loss of

MultiFrame Alignment..........433Receive Timeslot 16 Loss of Signal

..............................................432Red Alarm.................................431Status Freeze Signalling Active 432Transmit Data Overflow ...........432Transmit Data Underrun ...........433Transmit Line Open ..................433Transmit Line Short ..................432Transmit Slip.............................433Transmit Slip Negative .............433Transmit Slip Positive...............432Yellow Alarm............................432

E1 telephony parameters ...............126


521

E1/ISDN Parameter definitions .... 164E1/ISDN Parameter fields

Country ..................................... 164Enable ISDN-PRI ..................... 164Operator .................................... 164Terminal Network..................... 164

Echo Cancellation fieldanalog........................................ 330T1/E1 ........................................ 146

echo, removinganalog........................................ 330T1/E1 ........................................ 146

Edit selected Inbound PhonebookEntry iconE1.............................................. 357T1.............................................. 307

Edit selected Outbound PhonebookEntry iconE1.............................................. 357T1.............................................. 307

email account for voip unitanalog........................................ 376T1/E1 ........................................ 179

email address for voipanalog................................ 310, 375quick ........................................... 50T1/E1 ................................ 127, 178

email log reportsanalog........................................ 375quick ........................................... 60recipient ...................................... 60reply-to address........................... 60subject line.................................. 60T1/E1 ........................................ 178

email logs, illustrationanalog........................................ 380T1/E1 ........................................ 183

EMC, Safety, R&TTE DirectiveCompliance ............................... 488

emergency phone numberscaution about............................... 71

Enable (Call Fwdg)E1.............................................. 377T1.............................................. 328

Enable Call Holdanalog........................................ 389T1/E1 ........................................ 192

Enable Call Transfer

analog........................................388T1/E1 ........................................191

Enable Call Waitinganalog........................................389T1/E1 ........................................192

Enable Caller Name Identificationanalog........................................390T1/E1 ........................................193

Enable Console Messages fieldanalog........................................383T1/E1 ........................................185

Enable DNS fieldanalog........................................321T1/E1 ................................138, 321

Enable ISDN-PRI fieldE1/ISDN....................................164T1/ISDN....................................158

Enable Proxy fieldE1 ..............................................363T1 ..............................................312

Enable SMTP fieldanalog........................................376T1/E1 ........................................179

Enable SNMP Agent............. 167, 364Enable SNMP Agent field

analog........................................366T1/E1 ........................................169

Enable SPP Proxy/NAT deviceT1 ..............................................316

enabling SMTPanalog........................................375T1/E1 ........................................178

enabling web browser GUIanalog.................................. 58, 323T1/E1 ........................................139

Error Correction (SMTP logs)analog........................................379T1/E1 ........................................182

error correction, forwardanalog........................................330T1/E1 ........................................146

error messageCOM port conflict.....................116COM port conflict (analog models)

..............................................315error message (analog models)

MultiVOIP Not Found ..............316Phone Database Not Read.........316


522

error message (T1/E1 models)MultiVOIP Not Found.............. 133Phone Database Not Read......... 133

ESF and CRC frame format (T1).. 155ethernet cable pinout..................... 496Ethernet interface

analog.......................................... 26BRI ............................................. 34

European Community Directives.. 488Event # (statistics, logs) field........ 413Excessive Zeroes (E1 stats) field .. 433Excessive Zeroes (T1 stats) field .. 428exchanges, phone

dedicated..................................... 70institutional ................................. 70local ............................................ 70non-local ..................................... 70organizational ............................. 70

expansion card (analog, 4-to-8channel) installation.................. 504

expansion card (T1/E1) installation.................................................. 492

expansion card (T1/E1)operation.. 494

F

factory default software settings ... 452factory defaults, downloading....... 452factory repair for customers U.S. &

Canada ...................................... 484failover (PSTN)

analog models ............................. 27BRI models ................................. 35E1 models ................................... 19T1 models ................................... 12

failover (PSTN) feature ................ 325FAQ for MultiVOIPs...................... 10fast busy (unobtainable) tones

analog................................ 175, 372Fast ConnectSee Fast Start. See Fast

StartE1.............................................. 365T1.............................................. 314

Fast Start compatibilityE1.............................................. 362T1.............................................. 311

Fast Start plus H.245 Tunneling fieldE1.............................................. 365T1.............................................. 314

fax baud rate, defaultanalog........................................328T1/E1 ........................................144

Fax Enable fieldanalog........................................328T1/E1 ........................................144

fax machineconnecting to analog voip

(MVP130) .............................134connecting to analog voip

(MVP210) .....................132, 133connecting to analog voip (MVP-

410/810) ........................123, 124FAX Parameters

analog........................................328T1/E1 ........................................144

fax tones, output levelanalog........................................328T1/E1 ........................................144

Fax Volume fieldanalog........................................328T1/E1 ........................................144

FCC Declaration ...........................488FCC Part 68 Telecom rules ...........489FCC registration number...............490FCC rules, Part 15.........................488FDX LED

analog models .............................32analog, MVP130 .........................33BRI models .................................39E1 ................................................24T1 ................................................17

Filters (Console Message Settings)analog........................................384T1/E1 ........................................187

Filters button (Console MessageSettings)analog........................................383T1/E1 ........................................186

firmware upgrade, implementing ..446Firmware Version

(analog) .....................................398Firmware Version (System Info)

T1/E1 ........................................201firmware version, identifying........446firmware, downloading .................447firmware, obtaining updated .........442Flash Hook Options fields


523

E&M ......................................... 355FXO .......................................... 348

forgotten password................ 458, 461Forward Address/Number

T1...................................... 329, 377Forward Condition (Call Fwdg)

E1.............................................. 377T1.............................................. 329

Forward Error Correction (callprogress) field ........................... 409

Forward Error Correction (SMTPlogs)analog........................................ 379T1/E1 ........................................ 182

Forward Error Correction fieldanalog........................................ 330T1/E1 ........................................ 146

forward on busyT1...................................... 329, 377

Forward upon No ResponseE1.............................................. 377T1.............................................. 329

forwarding, dual conditions (T1)busy & no-response .......... 329, 377

Frame Format fieldE1.............................................. 161T1.............................................. 155

frame relay, and faxanalog........................................ 328T1/E1 ........................................ 144

Frame Search Restart Flag (T1 stats)field........................................... 429

Frame Type fieldanalog........................................ 319T1/E1 ........................................ 136

free callsE1.............................................. 351T1.............................................. 305

frequencies, touch tone ................. 347Frequency 1 (custom tone) field

analog........................................ 373T1/E1 ........................................ 176

Frequency 1 (tone pair scheme)analog........................................ 370T1/E1 ........................................ 173

Frequency 2 (custom tone) fieldanalog........................................ 373T1/E1 ........................................ 176

Frequency 2 (tone pair scheme)analog........................................370T1/E1 ........................................173

frequency, poweranalog models .............................43E1 models....................................42

FRF11analog........................................328T1/E1 ........................................144

From (gateway, statistics, logs) field..................................................413

front panelanalog models .............................32BRI models .................................39E1 ................................................24MVP2400....................................16MVP2410....................................16MVP3010....................................24T1 ................................................16

FTP client program .......................465FTP client program, obtaining ......467FTP client programs

graphic vs. textual orientation ...474FTP file transfers

using FTP client program..........467using web browser ....................467

FTP Server Enable fieldanalog........................................322T1/E1 ........................................138

FTP Server functionas added feature.........................465enabling.....................................467

FTP Server, contacting..................469FTP Server, invoking

download/transferusing FTP client program..........473using web browser ....................471

FTP Server, logging in ..................470FTP Server, logging out ................474FTP transfers

file types............................ 465, 468phonebooks ...............................465server location...........................465

function tracing on/off (logging)analog........................................384T1/E1 ........................................187

FXO Disconnect On fields .... 346, 347FXO disconnection criteria ...........346


524

FXO disconnection, triggering of346,347

FXO interface (MVP130)uses of ....................................... 134

FXO Interface Parameter definitions.......................................... 345, 346

FXO Interface Parameter Definitions.................................................. 347

FXO Interface Parameter fieldsCurrent Loss ............................. 346Current Loss Detect Timer ....... 346Detection Range (flash hook) ... 348Disconnect on Call Progress Tone

.............................................. 347Disconnect Tone Sequence....... 347Flash Hook................................ 348Inter Digit Regeneration Timer. 346Inter Digit Timer (dialing) ........ 345Message Waiting Indication ..... 345Regeneration (dialing) .............. 345Ring Count................................ 347Silence Detection ...................... 347Silence Timer............................ 347Tone Detection.......................... 346

FXO interface(MVP210)uses of ....................................... 132

FXO interface(MVP-410/810)uses of ....................................... 123

FXO Parameter fieldsCaller ID enable........................ 348Caller ID Type .......................... 348FXO Current Detect Timer ....... 346Tone Detection.......................... 346

FXO Parameters............................ 344FXS interface(MVP130)

uses of ....................................... 134FXS interface(MVP210)

uses of ....................................... 132FXS interface(MVP-410/810)

uses of ....................................... 123FXS Loop Start Interface parameter

definitions ................................. 340FXS Loop Start Interface Parameter

fieldsCaller ID enable................ 342, 343Caller ID Type .......................... 342Current Loss ............................. 341Detection Range (flash hook) ... 342

Inter Digit Regeneration Timer .341Inter Digit Timer .......................341Message Waiting Indication......341Pass Through Enable.................342Ring Count ................................341

FXS Loop Start Parameter fieldsInter Digit Timer .......................340Message Waiting Light .............340

FXS Loop Start Parameters...........340FXS/FXO connector

MVP130....................................134MVP210....................................132MVP-410/810 ...........................123

G

G711 coders (RTP packetization,voice/fax)T1/E1 ........................................424





Gain 1 (custom tone) fieldanalog........................................373T1/E1 ........................................176

Gain 1 (tone pair scheme)analog........................................370T1/E1 ........................................173

Gain 2 (custom tone) fieldanalog........................................373T1/E1 ........................................176

Gain 2 (tone pair scheme)analog........................................370T1/E1 ........................................173

Gatekeeper / Clear Channel IPAddress (Gatekeeper RAS) fieldE1 ..............................................362T1 ..............................................311

gatekeeper interactionanalog models .............................27


525

BRI models ................................. 35E1 models ............................. 19, 20T1 models ............................. 12, 13

Gatekeeper Name (Gatekeeper RAS)fieldE1.............................................. 362T1.............................................. 311

GateKeeper RAS ParametersT1.............................................. 311

Gateway (IP Parameters) fieldanalog........................................ 321T1/E1 ........................................ 138

Gateway H.323 ID (Gatekeeper RAS)fieldE1.............................................. 362T1.............................................. 311

Gateway Name (call progress) field.................................................. 409

Gateway Name (callee, statistics,logs) field.................................. 414

Gateway Name (caller, statistics, logs)field........................................... 414

Gateway Name fieldE1.............................................. 362T1.............................................. 311

Gateway Number, From Details(SMTP logs) fieldanalog........................................ 379T1/E1 ........................................ 182

Gateway Number, To Details (SMTPlogs) fieldanalog........................................ 379T1/E1 ........................................ 182

Gateway Prefix (Gatekeeper RAS)fieldE1.............................................. 362T1.............................................. 311

Gateway Prefix (outboundphonebook) fieldE1.............................................. 371T1.............................................. 321

General Options fieldsE1.............................................. 366T1.............................................. 315

Generate Local Dial Tone(Voice/FAX – AutoCall/OffhookAlert) fieldanalog/BRI................................ 332

Generation Flash-Hook Options fieldE&M .........................................355FXO ..........................................348

Generation TimeE1..............................................163T1 ..............................................157

Generation Time fieldE1 ..............................................163T1 ..............................................157

groundingin rack installations ...................117MVP210....................................133MVP410ST ...............................129MVP810ST ...............................129

grounding screw, diagrams(MVP-2410/3010).......................52(MVP-410/810)...........................53(MVP-410ST/810ST)..................54

GUI (log reporting type) buttonanalog........................................383T1/E1 ........................................186

H

H.245 Tunneling fieldE1 ..............................................364T1 ..............................................313

H.323compatibility (analog models).....26compatibility (BRI models).........35compatibility (E1 models)...........19compatibility (T1 models)...........12

H.323 Annex E fieldE1 ..............................................365T1 ..............................................314

H.323 coderanalog........................................329T1/E1 ........................................145

H.323 fields (Outbound Phonebook)E1 ..............................................371T1 ..............................................321

H.323 ID (Outbound Phonebook)fieldT1 ...................................... 321, 371

H.323 version 4 featuresanalog..........................................27BRI..............................................35E1 ................................................19T1 ................................................12


526

H.323 Version 4 ParametersE1...................................... 364, 365T1...................................... 313, 314

H.450 features, incompatible with SIPanalog.................................. 27, 385BRI ............................................. 35E1................................................ 19T1................................................ 12T1/E1 ........................................ 188

H.450 functionalitylogs for...................................... 415

H.450 standardANALOG ................................... 30BRI ............................................. 38E1................................................ 23T1................................................ 16

Hold Sequenceanalog........................................ 389T1/E1 ........................................ 192

Hold Sequence (analog)................ 386Hold Sequence (T1/E1) ................ 189hold, caller on (analog)

musical jingle for ...................... 388hold, caller on (T1/E1)

musical jingle for ...................... 191hookup

MVP130...................................... 56MVP210...................................... 55MVP2410.................................... 52MVP3010.................................... 52MVP410...................................... 53MVP-410ST/810ST.................... 54MVP810...................................... 53

HyperTerminal programand connectivity testing .............. 78

I

IANA ............................................ 502icon

variable version................... 10, 113icons, phonebook

E1.............................................. 357T1.............................................. 307

identifying current firmware version.................................................. 446

implementing firmware upgrade... 446in band, DTMF

analog........................................ 328

T1/E1 ........................................144inbound phonebook

example .......................................72Inbound Phonebook Entries List icon

E1 ..............................................357T1 ..............................................307

Inbound Phonebook entries, listE1 ..............................................375T1 ..............................................326

inbound phonebook examplequick............................................66

inbound vs. outbound phonebooksE1 ..............................................355T1 ..............................................305

Industry Canada requirements.......489info sources

analog telephony details...... 48, 308BRI telephony details..................49E1 details.....................................48E1 telephony details ..................126IP details......................................47IP details (analog system) .........307IP details (T1/E1 system) ..........124ISDN-BRI telephony details .....309SMTP details...............................50T1 details.....................................47T1 telephony details ..................125voip email account ......................50

info sources (analog models)SMTP details.............................310voip email account ....................310

info sources (T1/E1 models)SMTP details.............................127voip email account ....................127

Input Gain fieldanalog........................................327T1/E1 ........................................143

installationairflow.......................................117analog prerequisites...........307, 308BRI prerequisites.........................49E1 prerequisites................... 48, 126expansion card (analog, 4-to-8

channel).................................504expansion card (T1/E1) .............492full summary ...............................46in a nutshell .................................44in rack .......................................116


527

IP prerequisites ........................... 47ISDN-BRI prerequisites............ 309log reports by email (analog

models) ................................. 310log reports by email (T1/E1

models) ................................. 127software (detailed) .................... 111T1 prerequisites .................. 47, 125T1/E1 prerequisites ................... 124upgrade card (analog, 4-to-8

channel) ................................ 504upgrade card (T1/E1) ................ 492voip email account(analog models)

.............................................. 310voip email account(T1/E1 models)

.............................................. 127installation preparations (optional)

log reports by email .................... 50voip email account...................... 50

installation, mechanicalanalog models ............................. 26BRI models ................................. 34E1 models ................................... 18T1 models ................................... 11

installation, quicklog reports by email .................... 50voip email account...................... 50

installing Java vis-a-vis web GUI . 477integrated phone/data networks .... 350Inter Digit Regeneration Time (FXO)

field........................................... 346Inter Digit Regeneration Time (FXS

Loop Start) field........................ 341Inter Digit Regeneration Timer

(E&M) field .............................. 354Inter Digit Timer (dialing) field

DID-DPO.................................. 357E&M ......................................... 354FXO .......................................... 345FXS Loop Start ......................... 341

Intercept Tone (Regional Params) andOffhook Alert (Voice/Fax Params).................................................. 369

Intercept Tone and required Interface& Voice/Fax settingsanalog........................................ 369

Interface field (DID-DPO)............ 356Interface field (E&M) ................... 353

interface parameters, accessing.....337interface parameters, setting..........337interface types, BRI

ST and U ...................................500interfaces

BRI telephony .............................54inter-office dialing

E1 ..............................................351T1 ..............................................306

inter-operation (analog)with T1/E1 voips.........................25

inter-operation (BRI)with T1/E1/BRI voips .................34

inter-operation with phone systemanalog models .............................26BRI models .................................34E1 models....................................18T1 models....................................11

IP Address (call progress) field.....409IP Address (callee, statistics, logs)

field ...........................................414IP Address (caller, statistics, logs)

field ...........................................414IP Address (outbound phonebook)

E1 ..............................................370T1 ..............................................320

IP Address fieldanalog........................................321T1/E1 ........................................138

IP Address field (Registered GatewayDetails)......................................439

IP Address to Ping (LinkManagement) field ....................421

IP Address, From Details (SMTPlogs) fieldanalog........................................379T1/E1 ........................................182

IP address, SysLog Serveranalog........................................383T1/E1 ........................................186

IP Address, To Details (SMTP logs)fieldanalog........................................379T1/E1 ........................................182

IP datagram and DiffServanalog........................................320T1/E1 ........................................137

IP Mask field


528

analog........................................ 321T1/E1 ........................................ 138

IP parameter definitionsanalog........................................ 319T1/E1 ........................................ 136

IP Parameter fields (analog)Diffserv Parameters .................. 319DNS Server IP Address ............ 321Enable DHCP ........................... 321Enable DNS .............................. 321Frame Type............................... 319FTP Server Enable.................... 321Gateway .................................... 321IP Address................................. 321IP Mask..................................... 321

IP Parameter fields (T1/E1)Enable DHCP ........................... 138

IP Parameters screen, accessinganalog........................................ 317T1/E1 ........................................ 134

IP startup configuration .................. 58IP Statistics field definitions . 416, 418IP Statistics fields

Clear.......................................... 416Received (RTCP Packets)......... 419Received (RTP Packets) ........... 419Received (TCP Packets) ........... 418Received (Total Packets) .......... 416Received (UDP Packets)........... 418Received with errors (RTCP

Packets)................................. 419Received with errors (RTP Packets)

.............................................. 419Received with errors (TCP Packets)

.............................................. 418Received with errors (Total

Packets)................................. 418Received with errors (UDP

Packets)................................. 418Transmitted (RTCP Packets) .... 419Transmitted (RTP Packets) ....... 419Transmitted (TCP Packets) ....... 418Transmitted (Total Packets)...... 416Transmitted (UDP Packets) ...... 418

IP Statistics function ..................... 416ISDN BRI Interface screen fields

Status, Layer 1 Interface ........... 435Status, SPID0............................ 436

Status, SPID1 ............................437ISDN BRI Parameters

TEI n Assignment .....................362ISDN BRI Parameters fields

A-Law .......................................362Country .....................................361Dialing Options.........................361Inter Digit Timer .......................361Layer 1 Interface .......................361MU-Law....................................362Operator ....................................361PCM Law..................................362Select BRI Interface ..................361SPID 0.......................................362SPID 1.......................................362Switch Information ...................361

ISDN BRI Statistics screen fieldsClear (button) ............................437D Channel Information (field

group)....................................436Layer 1 Interface (field group)..435Loss of Framing ........................435Loss of Sync..............................435Rx Packets.................................436Select BRI Interface ..................435SPID0........................................436SPID1........................................437State ..........................................435Switch Information (field group)

..............................................436Tx Packets.................................436

ISDN parameters, setting ..............166ISDN-BRI operating modes

MVP-410ST/810ST).................128ISDN-BRI Parameter definitions ..361ISDN-BRI telephony interfaces

uses of .......................................128ISDN-BRI telephony parameters ..309ISDN-PRI

types supported .........................166ISDN-PRI implementations ..........166

J

Javainstalling....................................477web GUI and .............................477

jitter bufferanalog........................................334


529

T1/E1 ........................................ 149Jitter Value (Fax) field

analog........................................ 328T1/E1 ........................................ 144

Jitter Value fieldanalog........................................ 336T1/E1 ........................................ 151

jitter, dynamicanalog........................................ 334T1/E1 ........................................ 149

jumper, DIDMVP210.................................... 130MVP-410/810 ........................... 122

K

Keep Alive fieldE1.............................................. 367T1.............................................. 316

key systemconnecting to analog voip

(MVP130) ............................. 134connecting to analog voip

(MVP210) ............................. 132connecting to analog voip (MVP-

410/810)................................ 123Knowledge Base (online, for

MultiVOIPs) ............................... 10

L

lab voip networkuse in setup ................................. 71

Last buttonLogs (Statistics) screen ............. 413

Last Error (Link Management) field.................................................. 422

Layer 1 InterfaceISDN-BRI................................. 361

Layer 1 Interface fields (ISDN BRIStatistics) .................................. 435

LED definitionsanalog models ............................. 32BRI models ................................. 39E1................................................ 24MVP2400.................................... 16MVP2410.................................... 17MVP3010.................................... 24T1................................................ 17

LED definitions (analog models)

FDX ............................................32LNK ............................................32

LED definitions (analog)Boot.............................................32Ethernet .......................................32Power ..........................................32RCV (channel) ............................32RSG.............................................32XMT (channel)............................32XSG ............................................32

LED definitions (analog, E1)LNK ............................................24

LED definitions (analog, MVP130)Boot.............................................33Ethernet .......................................33FDX ............................................33LNK ............................................33Power ..........................................33RCV (channel) ............................33RSG.............................................33XMT (channel)............................33XSG ............................................33

LED definitions (analog, T1)LNK ............................................17

LED definitions (BRI models)FDX ............................................39LNK ............................................39

LED definitions (BRI)Boot.............................................39Ethernet .......................................39Power ..........................................39RCV (channel) ............................39XMT (channel)............................39

LED definitions (E1)Boot.............................................24E1 ................................................24FDX ............................................24IC ................................................24LC ...............................................24LS................................................24ONL ............................................24Power ..........................................24PRI ..............................................24

LED definitions (T1)Boot.............................................17FDX ............................................17IC ................................................17LC ...............................................17


530

LS ............................................... 17ONL............................................ 17Power .......................................... 17PRI.............................................. 17

LED indicatorsE1................................................ 23T1................................................ 16

LED indicators (analog)channel operation........................ 30general operation ........................ 30

LED indicators (BRI)channel operation........................ 38general operation ........................ 38

LED indicators, activeanalog.......................................... 30E1................................................ 23T1................................................ 16

LED sets (T1/E1), left and right ... 494LED types

analog models ............................. 30BRI models ................................. 38

liftingprecaution about........................ 111

limitations notice (regulatory),Canadian ................................... 490

limited warranty............................ 484Line Build Out field

E1.............................................. 165T1.............................................. 159

Line Coding fieldE1.............................................. 165T1.............................................. 159

Line Loopback Activation Signal (T1stats) field ................................. 429

Line Loopback Deactivation Signal(T1 stats) field........................... 428

Link Management (Statistics) fieldsClear command button.............. 421IP Address column.................... 422IP Address to Ping .................... 421Last Error .................................. 422No. of Pings .............................. 421No. of Pings Received .............. 422No. of Pings Sent ...................... 422Ping Size in Bytes..................... 421Response Timeout .................... 421Round Trip Delay ..................... 422Start Now command button ...... 421

Timer Interval between Pings ...421Link Management (Statistics) screen

field definitions ................. 421, 422IP Address (ping target .................422Link Status fields

Link Management (Statistics)screen ....................................422

List of Registered Numbers field(Registered Gateway Details) ...439

lithium battery caution ..................111LNK LED

analog models .............................32analog, MVP130 .........................33BRI models .................................39E1 ................................................24T1 ................................................17

loading of weight in rack ..............117local configuration

analog/BRI ................................307T1/E1 ........................................124

local configuration proceduredetailed, analog .........................312detailed, T1/E1..........................129summary, analog .......................311summary, T1/E1........................128

local exchange numbers ..................70local voip configuration (analog) ..305local voip configuration (T1/E1)...122local Windows GUI vs. web GUI

comparison................................476local-rate access (E1)

to remote PSTN...........................18local-rate calls to remote voip sites

E1 ..............................................352log report email, customizing

analog................................ 378, 379T1/E1 ................................ 181, 182

log report email, triggeringanalog........................................377T1/E1 ........................................180

log reporting method, settinganalog........................................381T1/E1 ........................................184

log reportsanalog models ...........................310T1/E1 models ............................127

log reports & SMTPanalog........................................375


531

T1/E1 ........................................ 178log reports and SMTP

quick ........................................... 60log reports by email

analog........................................ 375quick ........................................... 60T1/E1 ........................................ 178

log reports, quick ............................ 50logging options

analog........................................ 382T1/E1 ........................................ 185

logging update intervalanalog........................................ 382T1/E1 ........................................ 185

logging, web GUI and................... 476Login Name (SMTP) field

analog........................................ 376T1/E1 ........................................ 179

Logs (Statistics) fieldsBytes received........................... 414Bytes Sent ................................. 413Call Forwarded to ..................... 415Call Transferred to .................... 415CT Ph#...................................... 415Duration .................................... 413Event #...................................... 413From (gateway)......................... 413Gateway Name (callee)............. 414Gateway Name (caller) ............. 414H.450 functionality ................... 415IP Address (callee).................... 414IP Address (caller) .................... 414Mode......................................... 413Options (caller) ......................... 414Options callee ........................... 414Outbound digits ........................ 414Packets Lost .............................. 414Packets received........................ 414Packets Sent .............................. 413Start Date, Time........................ 413Status ........................................ 413Supplementary Services info .... 415To (gateway)............................. 413Voice coder............................... 414

Logs (Statistics) function .............. 412Logs (Statistics) screen

Delete File button ..................... 413Last button ................................ 413

logs and web browser GUIanalog........................................382T1/E1 ........................................185

logs by email, illustrationanalog........................................380T1/E1 ........................................183

Logs screen definitionsanalog........................................382T1/E1 ........................................185

Logs screen field definitionsanalog........................................383T1/E1 ........................................186

Logs screen parameters (analog)Enable Console Messages .........383Filters ........................................383GUI ...........................................383IP Address (SysLog Server)......383Online Statistics Updation Interval

..............................................383Port (SysLog Server).................383SMTP ........................................383SNMP........................................383SysLog Server Enable...............383Turn Off Logs ...........................383

Logs screen parameters (T1/E1)Console Message Settings.........186Enable Console Messages .........185Filters ........................................186GUI ...........................................186IP Address (SysLog Server)......186Online Statistics Updation Interval

..............................................186Port (SysLog Server).................186SMTP ........................................186SNMP........................................186SysLog Server Enable...............186Turn Off Logs ...........................186

logs screen, accessinganalog........................................381T1/E1 ........................................184

long distance call savingsT1 ..............................................305

long-distance call savingsE1 ..............................................350

Long-Haul Mode fieldE1 ..............................................161T1 ..............................................155


532

Loss of Frame Alignment (E1 stats)field........................................... 431

Loss of Frame Alignment (T1 stats)field........................................... 428

Loss Of Framing field (ISDN BRIStatistics, Layer 1 Interface) ..... 435

Loss of MultiFrame Alignment (E1stats) field ................................. 432

Loss of MultiFrame Alignment (T1stats) field ................................. 429

Loss of Sync field (ISDN BRIParameters, Layer 1 Interface).. 435

lost packets, consecutiveanalog........................................ 336T1/E1 ........................................ 151

lost password ........................ 458, 461

M

Mac AddressSystem Info (analog)................. 398System Info (T1/E1) ......... 201, 404

mail criteria (SMTP), recordsanalog........................................ 377T1/E1 ........................................ 180

Mail Server IP Address (SMTP) fieldanalog........................................ 376T1/E1 ........................................ 179

Mail Type (SMTP logs) fieldanalog........................................ 377T1/E1 ........................................ 180

mains frequencyanalog models ............................. 43E1 models ................................... 42

management (E1 models)local ............................................ 20remote (SNMP)........................... 20remote (web browser GUI) ......... 20

management of voips, remoteanalog........................................ 364T1/E1 ........................................ 167

Max bandwidth (coder)analog........................................ 329T1/E1 ........................................ 145

Max Baud Rate fieldanalog........................................ 328T1/E1 ........................................ 144

Max Retransmission (SPP, GeneralOptions) field

E1 ..............................................366T1 ..............................................315

maximum cable spanE1 ..............................................161T1 ..............................................155

Maximum Jitter Value fieldanalog........................................335T1/E1 ........................................150

Message Waiting Indication (DID-DPO) .........................................357

Message Waiting Indication (E&M)and DID.....................................354

Message Waiting Indication fieldDID-DPO ..................................357E&M .........................................354FXO ..........................................345FXS Loop Start .........................341

Minimum Jitter Value fieldanalog........................................334T1/E1 ........................................149

Mode (call progress) field .............408Mode (Fax) field

analog........................................328T1/E1 ........................................144

Mode (SPP) fieldE1 ..............................................366T1 ..............................................315

Mode (statistics, logs) field ...........413model descriptions

E1 ................................................18modem relay

analog........................................335T1/E1 ........................................150

modem traffic on voip networkanalog........................................335T1/E1 ........................................150

modem, commandand Regional Parameters Country

Selection..................................59modem, command (analog)

and Regional Parameters CountrySelection........................ 311, 368

modem, command (T1/E1)and Regional Parameters Country

Selection................ 128, 170, 171modem, remote

configuration/commandsetup for ......................................59


533

modem, remoteconfiguration/command (analog)setup for ............................ 311, 368

modem, remoteconfiguration/command (T1/E1)setup for .................... 128, 170, 171

Monitor Link fieldsLink Management (Statistics)

screen .................................... 421mounting

analog models ............................. 26BRI models ................................. 34E1 models ................................... 18T1 models ................................... 11

mounting in rack ........................... 116procedure for............................. 118safety................................. 111, 117

mounting options .............................. 9Multiplexed UDP field

E1.............................................. 365T1.............................................. 314

MultiVOIP configuration software . 57E1 models ................................... 20T1 models ................................... 13

MultiVOIP FAQ (on MTS web site).................................................... 10

MultiVOIP Program Menu items.. 440MultiVOIP Program Menu options

Configuration............................ 440Date & Time Setup ................... 440Download CAS Protocol .......... 440Download Factory Defaults ...... 441Download Firmware ................. 441Set Password............................. 441Uninstall.................................... 441Upgrade Software ..................... 441

MultiVOIP program menu, optiondescriptions ....................... 440, 441

MultiVOIP softwareinstalling ................................... 111location of files ......................... 114program icon location ............... 115uninstalling ....................... 118, 462

MultiVOIP software (analog)moving around in...................... 316

MultiVOIP software (T1/E1)moving around in...................... 133

MultiVoipManager ......................... 10

analog........................................305T1/E1 ........................................123

MultiVoipManager softwareE1 models....................................20T1 models....................................13

musical jingle during call transferanalog........................................388T1/E1 ........................................191

MVP130cabling procedure......................134Introduction.................................25unpacking..................................115

MVP210grounding ..................................133

MVP210xcabling procedure......................130unpacking..................................114

MVP2410cabling procedure......................119unpacking..................................112

MVP-2410remote configuration modem ....120

MVP3010cabling procedure......................119remote configuration modem ....120unpacking..................................112

MVP410cabling procedure......................121remote configuration modem ....125

MVP410STgrounding ..................................129

MVP-410STcabling procedure......................126

MVP410xunpacking..................................113

MVP810cabling procedure......................121remote configuration modem ....125

MVP810STgrounding ..................................129

MVP-810STcabling procedure......................126

MVP810xunpacking..................................113

N

national-rate calls to foreign voip sitesE1 ..............................................354


534

Netcoder coders (RTP packetization,voice/fax)T1/E1 ........................................ 424

network accessanalog........................................ 366T1/E1 ........................................ 169

Network Disconnection fieldanalog........................................ 336T1/E1 ........................................ 151

network/terminal settings, voip andPBXE1/ISDN ................................... 164ISDN-BRI................................. 361T1/ISDN ................................... 158

No. of Entries field (RegisteredGateway Details) ...................... 439

No. of Pings (Link Management) field.................................................. 421

No. of Pings Received (LinkManagement) field.................... 422

No. of Pings Sent (Link Management)field........................................... 422

no-response & busy(T1)forwarding, dual conditions329,

377NT1 device

when required for MVP410ST . 127when required for MVP810ST . 127

NT1 device, use ofBRI voip units..................... 49, 127

Number of Days (email log criteria)analog........................................ 377T1/E1 ........................................ 180

Number of Records (email logcriteria)analog........................................ 377T1/E1 ........................................ 180

numbering plan resources ............. 399

O

obtaining updated firmware .......... 442Offhook alert................................. 331Offhook Alert (Voice/Fax Params)

and Intercept Tone (RegionalParams) ..................................... 331

Offhook Alert Timer (Voice/FAX --AutoCall/Offhook Alert) fieldanalog/BRI................................ 333

T1/E1 ........................................148official phone numbers

caution about...............................71Online Statistics Updation Interval

field (Logs)analog........................................383T1/E1 ........................................186

operating system .............................40operating temperature ...................117operating voltage

analog models .............................43T1 models....................................42

operationexpansion card (T1/E1) .............494

Operator (ISDN) fieldE1/ISDN....................................164T1/ISDN....................................158

Operator definitionsISDN-BRI .................................363

Operator fieldISDN-BRI .................................361

Optimization Factor fieldanalog........................................335T1/E1 ........................................150

Options (call progress) field..........409Options (callee, statistics, logs) field

..................................................414Options, From Details (SMTP logs)

fieldanalog........................................379T1/E1 ........................................182

Options, To Details (SMTP logs) fieldanalog........................................379T1/E1 ........................................182

out of band, DTMFanalog........................................328T1/E1 ........................................144

Outbound Digits (call progress) field..................................................408

Outbound Digits (SMTP logs) fieldanalog........................................379T1/E1 ........................................182

Outbound digits (statistics, logs) field..................................................414

outbound phonebookexample .......................................72

Outbound Phonebook Entries Listicon


535

E1.............................................. 357T1.............................................. 307

Outbound Phonebook entries, listE1.............................................. 368T1.............................................. 318

outbound phonebook examplequick ........................................... 62

outbound vs. inbound phonebooksE1.............................................. 355T1.............................................. 305

Output Gain fieldanalog........................................ 327T1/E1 ........................................ 143

output level, fax tonesanalog........................................ 328T1/E1 ........................................ 144

outside line, access to ............... 70, 72

P

packet priority and DiffServanalog........................................ 320T1/E1 ........................................ 137

packetization (RTP), ranges &incrementsT1/E1 ........................................ 424

packetization ratescoder options and...................... 423

Packets Lost (call progress) field.. 408Packets Lost (SMTP logs) field

analog........................................ 378T1/E1 ........................................ 181

Packets lost (statistics, logs) field . 414Packets Received (call progress) field

.................................................. 408Packets Received (SMTP logs) field

analog........................................ 378T1/E1 ........................................ 181

Packets received (statistics, logs) field.................................................. 414

Packets Sent (call progress) field.. 408Packets Sent (SMTP logs) field

analog........................................ 378T1/E1 ........................................ 181

Packets sent (statistics, logs) field 413packets, consecutive lost

analog........................................ 336T1/E1 ........................................ 151

Parallel H.245 field

E1 ..............................................365T1 ..............................................314

parameters tracked by consoleanalog........................................384T1/E1 ........................................187

Pass Through (E&M) field............353Pass Through Enable (FXS Loop Start

interface) and AutoCall (Voice/FaxParams) .....................................342

Password (proxy server) fieldE1 ..............................................363T1 ..............................................312

Password (SMTP) fieldanalog........................................376T1/E1 ........................................179

password, lost/forgotten ........ 458, 461password, setting...........................458

web browser GUI ......................461patents ...............................................2patterns, destination

tips about.....................................69PBX characteristics, variations in

E1 ..............................................398T1 ..............................................348

PBX interactionanalog models .............................26BRI models .................................34E1 models....................................18T1 models....................................11

PC, commandCOM port assignment (detailed)116COM port requirement................51demands upon .............................51non-dedicated use........................51operating system .........................51settings ........................................51specifications...............................51

PCM Law fieldE1 ..............................................165ISDN-BRI .................................362T1 ..............................................159

Permissions (SNMP) fieldanalog........................................366T1/E1 ........................................169

personnel requirementfor rack installation ...................117to lift during installation............118to lift unit during installation.....111


536

phone exchangesdedicated..................................... 70institutional ................................. 70local ............................................ 70non-local ..................................... 70organizational ............................. 70

Phone Number (Voice/FAX –AutoCall/Offhook Alert) fieldanalog/BRI................................ 333T1/E1 ........................................ 148

Phone Signaling Tones & Cadencesanalog........................................ 367T1/E1 ........................................ 170

phone startup configuration ............ 58phone switch types

ISDN implementations in ......... 166phone/IP details

importance of writing down........ 46importance of writing down

(analog) ................................. 307importance of writing down

(T1/E1).................................. 124phonebook

FTP remote file transfers .......... 465phonebook configuration

starter .......................................... 62phonebook configuration (analog)304,

404phonebook configuration (remote) 465phonebook configuration (T1/E1). 122Phonebook Configuration icon

E1.............................................. 357T1.............................................. 307

Phonebook Configuration ParameterdefinitionsE1...................... 362, 363, 364, 365T1...................... 311, 312, 313, 314

Phonebook Configuration procedureT1.............................................. 306

Phonebook Configuration ProcedureE1.............................................. 356

Phonebook Configuration screenE1.............................................. 359T1.............................................. 306

Phonebook Configuration screen (E1)Mode (SPP Protocol) ................ 366

Phonebook Configuration screen (T1)Mode (SPP Protocol) ................ 315

Phonebook Configuration screenfields (E1)Annex E (H.323, UDP

multiplexing).........................365Call Signalling Port...................362Client Options ...........................366Enable Proxy.............................363Gatekeeper Name......................362Gatekeeper/Clear Channel IP

Address .................................362Gatekeeper/Clear-Channel IP

Address .................................362Gateway H.323 ID ....................362Gateway Name..........................362Gateway Prefix..........................362General Options ........................366H.245 Tunneling .......................364Keep Alive ................................366Max Retransmission (SPP, General

Options).................................366Parallel H.245 (Tunneling with Fast

Start)......................................365Port (SPP, General Options) .....366Port Number (Gatekeeper) ........362Port Number (proxy server) ......363Proxy Server IP Address ...........363Q.931 Multiplexing...................364Register with GateKeeper .........362Registrar IP Address .................366Registrar Options ......................366Registrar Port ............................366Retransmission (SPP, General

Options).................................366Use Fast Start ............................362User Name (proxy server).........363

Phonebook configuration screen fields(T1)Password (proxy server)............312

Phonebook Configuration screenfields (T1)Annex E (H.323, UDP

multiplexing).........................314Call Signalling Port...................311Client Options ...........................315Enable Proxy.............................312Gatekeeper Name......................311Gatekeeper/Clear Channel IP

Address .................................311


537

Gateway H.323 ID .................... 311Gateway Name.......................... 311Gateway Prefix ......................... 311General Options ........................ 315H.245 Tunneling ....................... 313Keep Alive................................ 315Max Retransmission (SPP, General

Options) ................................ 315Parallel H.245 (Tunneling with Fast

Start) ..................................... 314Password (proxy server) ........... 363Port (SPP, General Options) ..... 315Port Number (Gatekeeper)........ 311Port Number (proxy server) ...... 312Proxy Server IP Address........... 312Q.931 Multiplexing................... 313Register with GateKeeper......... 311Registrar IP Address ................. 315Registrar Options ...................... 315Registrar Port ............................ 315Retransmission (SPP, General

Options) ................................ 315Use Fast Start............................ 311User Name (proxy server)......... 312

phonebook destination patterns ...... 69phonebook dialing patterns............. 69phonebook digits

dropping...................................... 70leading ........................................ 70non-PSTN type ........................... 70specialized codes ........................ 70types used ................................... 69

phonebook entries, coordinatingE1.............................................. 356T1.............................................. 306

phonebook examplesanalog........................................ 304mixed digital/analog ................... 72

phonebook iconsE1.............................................. 357T1.............................................. 307

phonebook objectives &considerationsE1.............................................. 355

phonebook sidebar menuE1.............................................. 358T1.............................................. 308

phonebook tips................................ 69

phonebook worksheet ...............75, 76phonebook, analog voips...............404phonebook, inbound

example .......................................72example, quick ............................66

phonebook, outboundexample .......................................72example, quick ............................62

phonebooks, inbound vs. outboundE1 ..............................................355T1 ..............................................305

phonebooks, objectives &considerationsT1 ..............................................305

Phonebooks, objectives &considerationsE1 ..............................................350

phonebooks, sample ........................74Ping Size in Bytes (Link

Management) field ....................421pinging and connectivity .................81pinout

BRI connector ...........................499command cable .........................496ethernet cable ............................496T1/E1 connector........................497Voice/FAX connector ...............497

polarity sensitivityDID lines and (MVP210) ..........133DID lines and (MVP-410/810)..124

Port (SPP, General Options) fieldE1 ..............................................366T1 ..............................................315

Port field (Registered GatewayDetails)......................................439

Port field, SysLog Serveranalog........................................383T1/E1 ........................................186

Port Number (Gatekeeper RAS) fieldE1 ..............................................362T1 ..............................................311

Port Number (proxy server)E1 ..............................................363

Port Number (proxy server) fieldT1 ..............................................312

Port Number (SMTP) fieldanalog........................................376T1/E1 ........................................179


538

port number (SNMP) fieldanalog........................................ 366T1/E1 ........................................ 169

Port Number field, SPP (OutboundPhonebook)E1.............................................. 373T1.............................................. 323

power consumptionanalog models ............................. 43E1 models ................................... 42

power frequencyanalog models ............................. 43E1 models ................................... 42

Power LEDanalog models ....................... 32, 33BRI models ................................. 39

powering of ISDN-BRI phonesMVP-410ST/810ST.................. 128

Prefix Matched (call progress) field.................................................. 408

Prefix Matched (SMTP logs) fieldanalog........................................ 379T1/E1 ........................................ 182

prerequisitesfor technical configuration (analog)

.............................................. 307for technical configuration (T1/E1)

.............................................. 124prerequisites for installation

BRI info ...................................... 49E1 info ........................................ 48IP info ......................................... 47T1 info ........................................ 47

PRIISDN implementations ............. 166

product CD ..................................... 44use in software installation . 57, 111

Product CDE1 models ................................... 20T1 models ................................... 13

product family............................. 9, 10product groups .................................. 8Program Menu items..................... 440Protocol Type (outbound phonebook)

E1.............................................. 370T1.............................................. 320

Proxy Server IP AddressE1.............................................. 363

Proxy Server IP Address fieldT1 ..............................................312

Proxy/NAT Device Parameters –Public IP AddressT1 ..............................................316

PSTN failover featureAlternate Routing, and ..............325analog models .............................27BRI models .................................35E1 models....................................19T1 models....................................12

Pulse Density Violation (T1 stats)field ...........................................429

Pulse Shape Level fieldE1 ..............................................165T1 ..............................................159

Q

Q.931 Multiplexing fieldE1 ..............................................364T1 ..............................................313

Q.931 ParametersT1 ..............................................311

Q.931 Port Number (outboundphonebook) fieldE1 ..............................................371T1 ..............................................321

quality-of-serviceanalog..........................................27BRI..............................................35E1 ................................................19T1 ................................................12

R

rack mountinggrounding ..................................117safety................................. 111, 117

rack mounting instructions............116rack mounting procedure ..............118rack, equipment

weight capacity of .....................117rack-mountable voip models .........111RCV (channel) LED

analog models .............................32analog, MVP130 .........................33BRI models .................................39

Receive Slip (E1 Stats) field .........433Receive Slip (T1 Stats) field .........430


539

Receive Timeslot 16 Alarm IndicationSignal (E1 stats) field................ 432

Receive Timeslot 16 Loss ofMultiFrame Alignment (E1 stats)field........................................... 433

Receive Timeslot 16 Loss of Signal(E1 stats) field........................... 432

Received (RTCP Packets, IP Stats)field........................................... 419

Received (RTP Packets, IP Stats) field.................................................. 419

Received (TCP Packets, IP Stats) field.................................................. 418

Received (Total Packets, IP Stats)field........................................... 417

Received (UDP Packets, IP Stats)field........................................... 418

Received with Errors (RTCP Packets,IP Stats) field ............................ 419

Received with Errors (RTP Packets,IP Stats) field ............................ 419

Received with Errors (TCP Packets,IP Stats) field ............................ 418

Received with Errors (Total Packets,IP Stats) field ............................ 418

Received with Errors (UDP Packets,IP Stats) field ............................ 418

Recipient Address (email logs) fieldT1/E1 ........................................ 180

Recipient Address (email logs)fieldanalog........................................ 377

recovering voice packetsanalog........................................ 330T1/E1 ........................................ 146

Red Alarm (E1 stats) field ............ 431Red Alarm (T1 stats) field ............ 428Regeneration (dialing, FXO) field 345Regional Parameter definitions

analog........................ 369, 370, 371T1/E1 ........................ 172, 173, 174

Regional Parameter fields (analog)Cadence .................................... 371Country/Region (tone schemes) 369Custom (tones).......................... 371Frequency 1 .............................. 370Frequency 2 .............................. 370Gain 1 ....................................... 370Gain 2 ....................................... 370

Pulse Generation Ratio..............371type (of tone).............................370

Regional Parameter fields (T1/E1)Cadence.....................................174Country/Region (tone schemes) 172Custom (tones) ..........................174Frequency 1...............................173Frequency 2...............................173Gain 1........................................173Gain 2........................................173type (of tone).............................172

Regional Parameters fields (analog,except MVP130)Country Selection for Built-In

Modem ..................................371Regional Parameters fields (T1/E1)

Country Selection for Built-InModem ..................................174

regional parameters, settinganalog........................................367T1/E1 ........................................170

Register Duration field (RegisteredGateway Details).......................439

Registered Gateway Details(Statistics) screen, accessing .....439

Registered Gateway Details‘Statistics’ function ........... 438, 439

Registered Gateway Details screen439Registered Gateway Details screen

fieldsDescription................................439IP Address.................................439No. of Entries ............................439Port............................................439Register Duration ......................439Status.........................................439

Registered Gateway Details screenfields: ........................................439

Registrar IP Address fieldE1 ..............................................367T1 ..............................................316

Registrar Options fieldsE1 ..............................................367T1 ..............................................316

Registrar Port fieldE1 ..............................................367T1 ..............................................316


540

Registration Option Parameters(Inbound Phone Book)E1.............................................. 378T1.............................................. 329

remote configuration modemMVP-2410 ................................ 120MVP3010.................................. 120MVP410.................................... 125MVP810.................................... 125

Remote Configuration/CommandModemsetup for ...................................... 59

Remote Configuration/CommandModem (analog)setup for ............................ 311, 368

Remote Configuration/CommandModem (T1/E1)setup for .................... 128, 170, 171

remote control/configurationweb GUI and............................. 477

remote phonebook configuration .. 465remote voip

using to confirm configuration50,62

remote voip configuration (analog).................................................. 305

remote voip configuration (T1/E1)122Remote Voip Management

analog........................................ 364T1/E1 ........................................ 167

Remove Prefix (inbound) fieldE1.............................................. 376T1.............................................. 328

Remove Prefix (outbound) fieldE1.............................................. 370T1.............................................. 320

re-order tone, customT1/E1 ........................................ 175

repair procedures for customers U.S.& Canada .................................. 484

Reply-To Address (email logs) fieldT1/E1 ........................................ 180

Reply-To Address (email logs)fieldanalog........................................ 377

Resolutions (MultiVOIPtroubleshooting).......................... 10

Response Timeout (LinkManagement) field.................... 421

Retransmission (SPP, GeneralOptions) fieldE1 ..............................................366T1 ..............................................315

Retrieve Sequenceanalog........................................389T1/E1 ........................................192

Retrieve Sequence (analog) ..........386Retrieve Sequence (T1/E1) ...........189RFC768.........................................502RFC793.........................................502ring cadences, custom

analog........................................374T1/E1 ................................ 175, 177

Ring Count (FXO) field ................347Ring Count field

FXS Loop Start .........................341Ring Count forwarding condition

E1 ..............................................378T1 ..............................................329

ring tone, customanalog........................................373T1/E1 ................................ 175, 176

ring-tonesanalog........................................372T1/E1 ........................................175

Round Trip Delay (LinkManagement) field ....................422

Round Trip Delay fieldE1 ..............................................375T1 ..............................................325

RSG LEDanalog models .............................32analog, MVP130 .........................33

RTP packetization, ranges &increments .................................424

RTP Parameters screen .................424Rx Packets field (ISDN BRI

Statistics, D-Channel Information)..................................................436

S

Safety Recommendations for RackInstallations ...............................117

safety warnings .............................111Safety Warnings Telecom...........111sample phonebooks .........................74Save Setup command


541

analog........................................ 399T1/E1 ........................................ 202

saving configurationanalog........................................ 399T1/E1 ........................................ 202user ........................................... 455

Saving the MultiVOIP Configurationanalog........................................ 399T1/E1 ........................................ 202

savings on toll callsE1.............................................. 350T1.............................................. 305

scale-abilityE1................................................ 18T1................................................ 11

second dial toneand comma use ........................... 71

Select All (SMTP logs) fieldanalog........................................ 378T1/E1 ........................................ 181

Select BRI Interface field ............. 435Select BRI Interface ISDN-BRI field

BRI ........................................... 361Select Channel field

analog........................................ 327T1/E1 ........................................ 143

Select Channel, SupplementaryServices fieldanalog........................................ 388T1/E1 ........................................ 191

Selected Coder fieldanalog........................................ 329T1/E1 ........................................ 145

Set Baud Rateanalog........................................ 396T1/E1 ........................................ 199

Set Custom Tones & CadencesT1/E1 ........................................ 175

Set ISDN Parameters .................... 166Set Log Reporting Method

analog........................................ 381T1/E1 ........................................ 184

Set Password (program menu option) ,command .................................. 458

Set Password (web browser GUI) ,command .................................. 461

Set Password option description(MultiVOIP program menu) ..... 441

Set Regional Parametersanalog........................................367T1/E1 ........................................170

Set SMTP Parametersanalog........................................375T1/E1 ........................................178

Set SNMP Parametersanalog........................................364T1/E1 ........................................167

Set Supplementary ServicesParametersanalog........................................385T1/E1 ........................................188

Set T1/E1/ISDN Parameters .........152Set Telephony Interface Parameters

..................................................337Set Voice/FAX Parameters

analog........................................324T1/E1 ........................................140

setting IP parametersanalog........................................317T1/E1 ........................................134

setting password............................458web browser GUI ......................461

setting RTP Parameters .................424setting user defaults.......................454setup, saving

analog........................................399T1/E1 ........................................202user............................................455

setup, saving user values...............454Signal (type, E&M) field ..............353signaling cadences

analog........................................367T1/E1 ........................................170

signaling parameters (analogtelephony) .................................337

signaling tonesanalog........................................367T1/E1 ........................................170

signaling typesanalog telephony (MVP130) .....134analog telephony (MVP210)132,

133analog telephony (MVP-410/810)

......................................123, 124Silence Compression (call progress)

field ...........................................409


542

Silence Compression (SMTP logs)analog........................................ 379T1/E1 ........................................ 182

Silence Compression fieldanalog........................................ 330T1/E1 ........................................ 146

Silence Detection (FXO) field ...... 347Silence Timer (FXO) field............ 347simulated voip network

use in startup............................... 71Single-Port Protocol, general

descriptionanalog.......................................... 27BRI ............................................. 35E1................................................ 19T1................................................ 12

SIPcompatibility

analog models ......................... 27BRI models ............................. 35E1 models ............................... 19T1 models ............................... 12

SIP Fields (Outbound Phonebook)E1.............................................. 372T1.............................................. 322

SIP incompatibility with H.450Supplementary Servicesanalog.................................. 27, 385BRI ............................................. 35E1................................................ 19T1................................................ 12T1/E1 ........................................ 188

SIP Port Number fieldE1.............................................. 372T1.............................................. 322

SIP port number, standardE1.............................................. 372T1.............................................. 322

SIP Proxy ParametersE1.............................................. 363T1.............................................. 312

SIP URL fieldE1.............................................. 372T1.............................................. 322

SMTPquick setup.................................. 60

SMTP (log reporting type) buttonanalog........................................ 383

T1/E1 ........................................186SMTP logs by email, illustration

analog........................................380T1/E1 ........................................183

SMTP Parameters definitionsanalog........................................376T1/E1 ........................................179

SMTP Parameters fields (analog)Mail Server IP Address .............376Mail Type..................................377Number of Days........................377Number of Records ...................377Port Number..............................376Recipient Address .....................377Reply-To Address .....................377Subject ......................................377

SMTP Parameters fields (T1/E1)Enable SMTP............................179Login Name ..............................179Mail Server IP Address .............179Mail Type..................................180Number of Days........................180Number of Records ...................180Password ...................................179Port Number..............................179Recipient Address .....................180Reply-To Address .....................180Subject ......................................180

SMTP parameters, accessinganalog........................................375T1/E1 ........................................178

SMTP parameters,settinganalog........................................375T1/E1 ........................................178

SMTP port, standardanalog........................................376T1/E1 ........................................179

SMTP prerequisitesanalog models ...........................310quick............................................50T1/E1 models ............................127

SMTP, enablinganalog........................................375T1/E1 ........................................178

SNMP (log reporting type) buttonanalog........................................383T1/E1 ........................................186

SNMP agent program


543

analog........................................ 305T1/E1 ........................................ 123

SNMP agent, enablinganalog........................................ 364T1/E1 ........................................ 167

SNMP Parameter DefinitionsT1/E1 ........................................ 169

SNMP Parameter fields (analog)Address ..................................... 366Community Name (2) ............... 366Community Name 1.................. 366Enable SNMP Agent................. 366Permissions (1) ......................... 366Permissions (2) ......................... 366Port Number ............................. 366

SNMP Parameter fields (T1/E1)Address ..................................... 169Community Name (2) ............... 169Community Name 1.................. 169Enable SNMP Agent................. 169Permissions (1) ......................... 169Permissions (2) ......................... 169Port Number ............................. 169

SNMP Parameters, settinganalog........................................ 364T1/E1 ........................................ 167

softwarecontrol......................................... 57uninstalling (detailed) ............... 118updates (analog)........................ 305updates (T1/E1) ........................ 123

software (MultiVOIP)uninstalling ............................... 462

software configurationsummary ................................... 111

software installationdetailed...................................... 111quick ........................................... 57

software loading............................ 111software loading, quick................... 57software version numbers ............. 113software, MultiVOIP (analog)

screen-surfing in ....................... 316software, MultiVOIP (T1/E1)

moving around in...................... 133screen-surfing in ....................... 133

software, MultiVOIP(analog)moving around in...................... 316

software, on command PC ..............57Solving Common Connection

Problemsanalog........................................315T1/E1 ........................................132

sound quality, improvinganalog........................................330T1/E1 ........................................146

specialized codes, in dialing............70specifications

E1 models....................................42T1 models....................................41

SPID 0ISDN-BRI .................................362

SPID 1ISDN-BRI .................................362

SPID0 field (ISDN BRI Statistics,Switch Information) ..................436

SPID1 field (ISDN BRI Statistics,Switch Information) ..................437

SPP Fields (Outbound Phonebook)E1 ..............................................373T1 ..............................................323

SPP Fields (Phonebook Configurationscreen)T1 ..............................................315

SPP Fields (PhoneBook Configurationscreen)E1 ..............................................367

SPP, general descriptionanalog..........................................27BRI..............................................35E1 ................................................19T1 ................................................12

SPP, strengths & compatibilities ofanalog..........................................27BRI..............................................35E1 ................................................19T1 ................................................12

ST interface (ISDN-BRI)description.................................500

Start Date, Time (SMTP logs) fieldanalog........................................378T1/E1 ........................................181

Start Date,Time (statistics, logs) field..................................................413

Start Modes (DID-DPO) field356, 357


544

Start Now command (LinkManagement) button................. 421

starter configurationinbound phonebook .................... 66outbound phonebook .................. 62phone/IP...................................... 58

startup tasks .................................... 46State field (ISDN BRI Statistics,

Layer 1 Interface) ..................... 435Options (caller .............................. 414Status (call progress) field ............ 411Status (statistics, logs) field .......... 413Status field (ISDN BRI Statistics,

Layer 1 Interface) ..................... 435Status field (ISDN BRI Statistics,

SPID0) ...................................... 436Status field (ISDN BRI Statistics,

SPID1) ...................................... 437Status field (Registered Gateway

Details)...................................... 439Status Freeze Signalling Active (E1

stats) field ................................. 432Status Freeze Signalling Active (T1

stats) field ................................. 428Subject (email logs) field

analog........................................ 377T1/E1 ........................................ 180

supervisory signaling (analog)...... 338supervisory signaling parameters

(analog telephony) .................... 337supervisory signaling types

MVP130.................................... 134MVP210............................ 132, 133MVP-410/810 ................... 123, 124

Supplementary (Telephony) ServicesANALOG ................................... 30BRI ............................................. 38E1................................................ 23T1................................................ 16

Supplementary Services (analog)Alerting Party............ 392, 393, 394Call Hold................................... 386Call Hold Enable....................... 389Call Name Identification........... 386Call Transfer ............................. 386Call Transfer Enable ................. 388Call Waiting.............................. 386Call Waiting Enable.................. 389

Caller Name Identification Enable..............................................390

Calling Party .............................391Enable Call Hold.......................389Enable Call Transfer .................388Enable Call Waiting ..................389Enable Caller Name Identification

..............................................390Hold Sequence ..........................389Retrieve Sequence.....................389Select Channel ..........................388Transfer Sequence.....................388

Supplementary Services (T1/E1)Alerting Party............ 195, 196, 197Call Hold...................................189Call Hold Enable.......................192Call Name Identification...........190Call Transfer .............................189Call Transfer Enable .................191Call Waiting ..............................189Call Waiting Enable ..................192Caller Name Identification Enable

..............................................193Calling Party .............................194Enable Call Hold.......................192Enable Call Transfer .................191Enable Call Waiting ..................192Enable Caller Name Identification

..............................................193Hold Sequence ..........................192Retrieve Sequence.....................192Select Channel ..........................191Transfer Sequence.....................191

Supplementary Services Infologs for ......................................415

Supplementary Services Parameterbuttons (analog)Copy Channel............................395Default ......................................395

Supplementary Services Parameterbuttons (T1/E1)Copy Channel............................198Default ......................................198

Supplementary Services ParameterDefinitionsanalog388, 389, 390, 391, 392, 393,

394, 395


545

T1/E1191, 192, 193, 194, 195, 196,197, 198

Supplementary Services Parameterfields (analog)Call Transfer Enable ................. 388Call Waiting Enable.................. 389Hold Sequence .......................... 389Retrieve Sequence .................... 389Transfer Sequence .................... 388

Supplementary Services Parameterfields (analog)Alerting Party............................ 392Allowed Name Types391, 392,

393, 394Busy Party................................. 393Call Hold Enable....................... 389Call Name Identification Enable390Caller ID ................................... 395Calling Party ............................. 391Connected Party........................ 394Select Channel .......................... 388

Supplementary Services Parameterfields (T1/E1)Call Transfer Enable ................. 191Call Waiting Enable.................. 192Hold Sequence .......................... 192Retrieve Sequence .................... 192Transfer Sequence .................... 191

Supplementary Services Parameterfields (T1/E1)Alerting Party............................ 195Allowed Name Types194, 195,

196, 197Busy Party................................. 196Call Hold Enable....................... 192Call Name Identification Enable193Caller ID ................................... 198Calling Party ............................. 194Connected Party........................ 197Select Channel .......................... 191

Supplementary Services Parametersscreen, accessinganalog........................................ 385T1/E1 ........................................ 188

Supplementary Services parameters,settinganalog........................................ 385T1/E1 ........................................ 188

Supplementary Services, incompatiblewith SIPanalog.................................. 27, 385BRI..............................................35E1 ................................................19T1 ................................................12T1/E1 ........................................188

support, technical ..........................486Switch Information fields (ISDN BRI

Statistics)...................................436switch types (phone) and ISDN-PRI

..................................................166SysLog client

ANALOG....................................29BRI..............................................37E1 ................................................22T1 ................................................15

SysLog client programsavailability.................................480features & presentation types ....482

SysLog functionalityANALOG....................................29BRI..............................................37E1 ................................................22T1 ................................................15

SysLog serverANALOG....................................29BRI..............................................37E1 ................................................22T1 ................................................15

SysLog Server Enable fieldanalog........................................383T1/E1 ........................................186

SysLog Server functionas added feature.........................480capabilities of ............................482enabling.....................................481location of .................................480

SysLog Server IP Address fieldanalog........................................383T1/E1 ........................................186

SysLog Server, enablinganalog........................................382T1/E1 ........................................185

System Information screenfor op & maint...........................404

System Information screen, accessinganalog........................................397


546

T1/E1 ........................................ 200System Information update interval,

settinganalog........................................ 397for op & maint .......................... 405T1/E1 ........................................ 200

T

T1 model descriptions..................... 11T1 Parameter definitions155, 156,

157, 159Clocking.................................... 159Line Build-Out.......................... 159Line Coding .............................. 159PCM Law.................................. 159Pulse Shape Level..................... 159Yellow Alarm Format............... 159

T1 Parameter fieldsCAS Protocol ........................ 156

CRC Check ............................... 155Detect Flash Hook .................... 157Frame Format............................ 155Long-Haul Mode ...................... 155T1/E1/ISDN.............................. 155

T1 Parameters screen.................... 154T1 Statistics field definitions 429, 430T1 Statistics fields

Bipolar Violation ...................... 430Frame Search Restart Flag........ 429Line Loopback Activation Signal

.............................................. 429Loss of MultiFrame Alignment 429Pulse Density Violation ............ 429Receive Slip .............................. 430Transmit Data Underrun ........... 429Transmit Line Open.................. 429Transmit Slip ............................ 429Transmit Slip Negative ............. 429Yellow Alarm ........................... 429

T1 telephony parameters............... 125T1/E1 connector pinout ................ 497T1/E1 Statistics function............... 426T1/E1/ISDN field

E1.............................................. 161T1.............................................. 155

T1/E1/ISDN Parameters screen,accessing................................... 152

T1/E1/ISDN parameters, setting... 152

T1/ISDN Parameter definitions.....158T1/ISDN Parameter fields

Country .....................................158Enable ISDN-PRI......................158Operator ....................................158Terminal Network .....................158

table-top voip models....................111TCP/UDP compared

E1 ..............................................372IP Statistics context...................417T1 ..............................................322

technical configurationstartup..........................................58

technical configuration (analog)prerequisites to ..........................307summary....................................304

technical configuration (T1/E1)prerequisites to ..........................124summary....................................122

technical configuration proceduredetailed, analog .........................312detailed, T1/E1..........................129summary, analog .......................311summary, T1/E1........................128

technical support ...........................486TEI Assignment fields (ISDN BRI

Statistics, Switch Information)..436TEI n Assignment

ISDN-BRI .................................362TEIn fields (ISDN BRI Statistics,

Switch Information) ..................436telco authorities and ISDN............166telecom safety warnings..............111telephony interface parameters,

setting........................................337telephony interfaces

uses of ....... 123, 124, 132, 133, 134telephony interfaces, BRI................54telephony signaling cadences

analog........................................367T1/E1 ........................................170

telephony signaling tonesanalog........................................367T1/E1 ........................................170

telephony startup configuration.......58telephony toning schemes

analog........................................372T1/E1 ........................................175


547

temperatureoperating ................................... 117

terminal mode (ISDN-BRI) & D-channel supportMVP-410ST/810ST.................. 127

Terminal Network fieldE1/ISDN ................................... 164T1/ISDN ................................... 158

terminal/network settings, voip andPBXE1/ISDN ................................... 164ISDN-BRI................................. 361T1/ISDN ................................... 158

Timer Interval between Pings (LinkManagement) field.................... 421

tips, phonebook............................... 69To (gateway, statistics, logs) field 413toll-call savings

E1.............................................. 350T1.............................................. 305

toll-free access (T1)to remote PSTN .......................... 11within voip network.................... 11

toll-free access (within voip network)E1................................................ 18T1................................................ 11

Tone Detection (FXO disconnectcriteria) field ............................. 346

Tone Pair (custom) fieldanalog........................................ 373T1/E1 ........................................ 176

tone pairs, customT1/E1 ........................................ 175

tones, signalinganalog........................................ 367T1/E1 ........................................ 170

Total Digits (outbound) fieldE1.............................................. 370T1.............................................. 320

touch tone frequencies .................. 347trace on/off (logging)

analog........................................ 384T1/E1 ........................................ 187

Transfer Sequenceanalog........................................ 388T1/E1 ........................................ 191

Transfer Sequence (analog) .......... 386Transfer Sequence (T1/E1)........... 189

Transmit Data Overflow (E1 stats)field ...........................................432

Transmit Data Overflow (T1 stats)field ...........................................428

Transmit Data Underrun (E1 stats)field ...........................................433

Transmit Data Underrun (T1 stats)field ...........................................429

Transmit Line Open (E1 stats) field..................................................433

Transmit Line Open (T1 stats) field..................................................429

Transmit Line Short (E1 stats) field..................................................432

Transmit Line Short (T1 stats) field..................................................428

Transmit Slip (E1 stats) field ........433Transmit Slip (T1 stats) field ........429Transmit Slip Negative (E1 stats) field

..................................................433Transmit Slip Negative (T1 stats) field

..................................................429Transmit Slip Positive (E1 stats) field

..................................................432Transmit Slip Positive (T1 stats) field

..................................................428Transmitted (RTCP Packets, IP Stats)

field ...........................................419Transmitted (RTP Packets, IP Stats)

field ...........................................419Transmitted (TCP Packets, IP Stats)

field ...........................................418Transmitted (Total Packets, IP Stats)

field ...........................................417Transmitted (UDP Packets, IP Stats)

field ...........................................418Transport Protocol (SIP) field

E1 ..............................................372T1 ..............................................322

trap manager parameters (SNMP)T1/E1 ........................................169

triggering log report emailanalog........................................377T1/E1 ........................................180

troubleshooting ...............................81Troubleshooting Resolutions for

MultiVOIPs.................................10Turn Off Logs field


548

analog........................................ 383T1/E1 ........................................ 186

Tx Packets field (ISDN BRI Statistics,D-Channel Information) ........... 436

Type (E&M type) field ................. 353Type (of tone) field

analog........................................ 370T1/E1 ........................................ 172

Type-of-Service IP header field &DiffServanalog........................................ 320T1/E1 ........................................ 137

U

U interface (ISDN-BRI)description ................................ 500

UDP multiplexed (H.323 Annex E)fieldE1.............................................. 365T1.............................................. 314

UDP/TCP comparedE1.............................................. 372IP Statistics context................... 417T1.............................................. 322

unconditional forwardingE1.............................................. 377T1.............................................. 329

Uninstall (program menu option) ,command .................................. 462

Uninstall option description(MultiVOIP program menu) ..... 441

uninstalling MultiVOIP software118,462

unobtainable tone, customanalog........................................ 373T1/E1 ................................ 175, 176

unobtainable tonesanalog................................ 175, 372

unpackingMVP130.................................... 115MVP210x.................................. 114MVP2410.................................. 112MVP3010.................................. 112MVP410x.................................. 113MVP810x.................................. 113

Up TimeSystem Info (analog)................. 398System Info (T1/E1) ......... 201, 404

update interval (logging)analog........................................382T1/E1 ........................................185

updated firmware, obtaining .........442upgrade

E1 ................................................18T1 ................................................11

upgrade card (analog, 4-to-8 channel)installation.................................504

upgrade card (T1/E1) installation..492Upgrade Software option description

MultiVOIP program menu ........441upgrade, firmware .........................446uploads vs. downloads (FTP)........465Use Fast Start (Q.931) field

E1 ..............................................362T1 ..............................................311

Use Gatekeeper (OutboundPhonebook) fieldE1 ..............................................371T1 ..............................................321

Use Proxy (SIP) fieldE1 ..............................................372T1 ..............................................322

Use Registrar field (OutboundPhonebook)E1 ..............................................373T1 ..............................................323

user default configuration, creatinganalog........................................400T1/E1 ........................................203

user defaults, downloading ...........454user defaults, setting......................454user name

Windows GUI ...........................458User Name (proxy server) field

E1 ..............................................363T1 ..............................................312

user values (software), saving .......454

V

variations in PBX characteristicsE1 ..............................................398T1 ..............................................348

version numbers ..............................10version numbers (software)...........113version, firmware ..........................446Voice Coder (call progress) field ..408


549

Voice coder (statistics, logs) field. 414voice delay

analog................................ 334, 335T1/E1 ................................ 149, 150

Voice Gain fieldanalog........................................ 327T1/E1 ........................................ 143

voice packets (analog)recovering lost/corrupted .......... 330

voice packets (T1/E1)recovering lost/corrupted .......... 146

voice packets, consecutive lostanalog........................................ 336T1/E1 ........................................ 151

voice packets, delayedanalog................................ 334, 335T1/E1 ................................ 149, 150

voice packets, re-assemblinganalog........................................ 328

voice packets, re-assemblyT1/E1 ........................................ 144

voice quality, improvinganalog........................................ 330T1/E1 ........................................ 146

voice quality, versus delayanalog........................................ 335T1/E1 ........................................ 150

Voice/FAX connector pinout........ 497Voice/FAX Parameter definitions

analog................................ 335, 336T1/E1 ................................ 150, 151

Voice/FAX Parameter Definitionsanalog........ 327, 328, 329, 330, 334T1/E1 .........143, 144, 145, 146, 149

Voice/FAX Parameter fields (analog)Automatic Disconnection ......... 336Call Duration ............................ 336Consecutive Packets Lost ......... 336Copy Channel ........................... 327Default ...................................... 327DTMF Gain .............................. 327DTMF Gain (High Tones) ........ 327DTMF Gain (Low Tones)......... 327DTMF In/Out of Band .............. 327Duration (DTMF) ..................... 327Dynamic Jitter Buffer ............... 334Echo Cancellation..................... 330Fax Enable ................................ 328

Fax Volume...............................328Forward Error Correction..........330Input Gain .................................327Jitter Value ................................336Jitter Value (Fax) ......................328Max Baud Rate (Fax)................328Maximum Jitter Value ..............335Minimum Jitter Value ...............334Mode (Fax)................................328Network Disconnection.............336Optimization Factor ..................335Output Gain...............................327Select Channel ..........................327Silence Compression.................330Voice Gain ................................327

Voice/FAX Parameter fields(analog/BRI)AutoCall/Offhook Alert ....331, 332AutoCall/Offhook Alert fields331,

332Generate Local Dial Tone .........332Offhook Alert Timer .................333Phone Number (Auto Call/Offhook

Alert) .....................................333Voice/FAX Parameter fields (T1/E1)

AutoCall/Offhook Alert ............147AutoCall/Offhook Alert fields ..147Automatic Disconnection..........151Call Duration.............................151Consecutive Packets Lost..........151Copy Channel............................143Default ......................................143DTMF Gain...............................143DTMF Gain (High Tones) ........143DTMF Gain (Low Tones) .........143DTMF In/Out of Band ..............143Duration (DTMF)......................143Dynamic Jitter Buffer................149Echo Cancellation .....................146Fax Enable ................................144Fax Volume...............................144Forward Error Correction..........146Input Gain .................................143Jitter Value ................................151Jitter Value (Fax) ......................144Max Baud Rate .........................144Maximum Jitter Value ..............150Minimum Jitter Value ...............149


550

Mode (Fax) ............................... 144Network Disconnection ............ 151Offhook Alert Timer................. 148Optimization Factor .................. 150Output Gain .............................. 143Phone Number .......................... 148Phone Number (AutoCall/Offhook

Alert)..................................... 148Select Channel .......................... 143Silence Compression ................ 146Voice Gain................................ 143

Voice/FAX Parameters screen,accessinganalog........................................ 324T1/E1 ........................................ 140

Voice/FAX parameters, settinganalog........................................ 324T1/E1 ........................................ 140

Voip Caller ID Case #1 –telcostandard CID enters voip system.................................................. 349

Voip Caller ID Case #2 – H.323 voipsystem, no telco CID................. 349

Voip Caller ID Case #3 –SPP ....... 350Voip Caller ID Case #4 – Remote

FXS call on H.323 voip system 350Voip Caller ID Case #5 –DID channel

in H.323 voip system ................ 351voip dialing digits

non-PSTN type ........................... 70types used ................................... 69

voip email accountanalog........................................ 376T1/E1 ........................................ 179

voip management, remoteanalog........................................ 364T1/E1 ........................................ 167

Voip Media PHB fieldanalog........................................ 319T1/E1 ........................................ 136

voip network, lab/simulateduse in startup............................... 71

voip softwarehost PC.................................. 40, 51

voip software (analog)host PC...................................... 305

voip software (T1/E1)host PC...................................... 123

voip system example, conceptual (E1)calls to remote PSTN ................352foreign calls, national rates .......354voip site to voip site ..................351

voip system example, digital &analog, with phonebook detailsE1 ..............................................386T1 ..............................................336

voip system example, digital only,with phonebook detailsE1 ..............................................379T1 ..............................................330

voip(E1)basic functions of ........................19

voip(T1)basic functions of ........................12

voltage, operatinganalog models .............................43E1 models....................................42

W

warnings, safety ............................111warranty ........................................484web browser GUI and logs

analog........................................382T1/E1 ........................................185

web browser GUI, enablinganalog.................................. 58, 323T1/E1 ........................................139

web browser interfacebrowser version requirement475,

478general.......................................475Java requirement .......................475prerequisite local assigning of IP

address...................................476video useability .........................475

web GUIJava and.....................................477remote control/configuration and

..............................................477web GUI vs. local Windows GUI

comparison................................476web GUI vs. Windows GUI

BRI..............................................36web GUI, logging and...................476weight

analog models .............................43


551

E1 models ................................... 42T1 models ................................... 41

weight loadingin rack ....................................... 117

weight of unitlifting precaution....................... 111personnel requirement............... 111

Well Known Ports......................... 502well-known port number, SMTP

analog........................................ 376T1/E1 ........................................ 179

well-known port, gatekeeperregistrationE1.............................................. 362T1.............................................. 311

well-known port, Q.931 params,H.323E1...................................... 362, 371T1...................................... 311, 321

well-known port, SIPE1.............................................. 372T1.............................................. 322

well-known port, SNMPanalog........................................ 366

T1/E1 ........................................169Windows GUI vs. web GUI

BRI..............................................37wink signaling (DID-DPO)...........357wink signaling (E&M) ..................353Wink Timer (DID-DPO) field.......357Wink Timer (E&M) field..............353worksheet

phonebook.............................75, 76

X

XMT (channel) LEDanalog models .............................32analog, MVP130 .........................33BRI models .................................39

XSG LEDanalog models .............................32analog, MVP130 .........................33

Y

Yellow Alarm (E1 stats) field .......432Yellow Alarm (T1 stats) field .......429Yellow Alarm Format field (T1)...159

553

S000249I

Date post:	27-Nov-2014
Category:	Documents
Upload:	starre3
View:	912 times
Download:	4 times