MultiVOIP®
Voice over IP gateways
User Guide
Digital Models:
MVP-2410/3010
AnalogModels: MVP-130/130FXS
& MVP-210/410/810
BRI Models: MVP-410ST/810ST
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CONTENTS
CHAPTER 1: OVERVIEW.......................................................................................7
ABOUT THIS MANUAL...............................................................................................8
T1 Front Panel LEDs..........................................................................................16
E1 Front Panel LEDs .........................................................................................23
E1 LED Descriptions..........................................................................................24
INTRODUCTION TO ANALOG MULTIVOIPS (MVP-130/130FXS, MVP-210/410/810
& MVP428) ............................................................................................................25
COMPUTER REQUIREMENTS ....................................................................................40
SPECIFICATIONS ......................................................................................................41
Specs for Digital T1 MultiVOIP Units................................................................41
Specs for Digital E1 MultiVOIP Units................................................................42
Specs for Analog/BRI MultiVOIP Units..............................................................43
INSTALLATION AT A GLANCE ..................................................................................44
RELATED DOCUMENTATION....................................................................................44
INTRODUCTION........................................................................................................46
MULTIVOIP STARTUP TASKS .................................................................................46
Gather IP Information...................................................................................................47
Gather Telephone Information (T1) .............................................................................47
Gather Telephone Information (E1) .............................................................................48
Gather Telephone Information (Analog) ......................................................................48
Gather Telephone Information (ISDN BRI).................................................................49
Obtain Email Address for VOIP (for email call log reporting).....................................50
Identify Remote VOIP Site to Call...............................................................................50
Identify VOIP Protocol to be Used...............................................................................50
Placement ...........................................................................................................51
Quick Hookups....................................................................................................52
Phone/IP Starter Configuration..........................................................................59
Outbound Phonebook ...................................................................................................63
Inbound Phonebook......................................................................................................67
Phonebook Tips ..................................................................................................70
Phonebook Example ...........................................................................................73
Connectivity Test ................................................................................................78
Troubleshooting..................................................................................................82
3
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Contents
MultiVOIP User Guide
INTRODUCTION........................................................................................................85
SAFETY WARNINGS .................................................................................................85
Lithium Battery Caution .....................................................................................85
Safety Warnings Telecom....................................................................................85
UNPACKING YOUR MULTIVOIP..............................................................................86
Unpacking the MVP2410/3010...........................................................................86
Unpacking the MVP-410/810 .............................................................................87
Unpacking the MVP210......................................................................................88
Unpacking the MVP-130/130FXS.......................................................................89
CABLING .................................................................................................................93
Cabling Procedure for MVP-410/810.................................................................95
Cabling Procedure for MVP210.......................................................................104
INTRODUCTION......................................................................................................111
CHAPTER 5: TECHNICAL CONFIGURATION
FOR DIGITAL T1/E1
LOCAL CONFIGURATION........................................................................................124
Pre-Requisites...................................................................................................124
IP Parameters..............................................................................................................124
T1 Telephony Parameters (for MVP2410) .................................................................125
E1 Telephony Parameters (for MVP3010) .................................................................126
SMTP Parameters (for email call log reporting).........................................................127
Modem Relay ....................................................................................................150
CHAPTER 6: TECHNICAL CONFIGURATION FOR ANALOG/BRI
MULTIVOIPS (MVP-130/130FXS, MVP-210, MVP-410, MVP-810 & MVP-
410ST/810ST)..........................................................................................................206
LOCAL CONFIGURATION........................................................................................210
Pre-Requisites...................................................................................................210
IP Parameters..............................................................................................................210
SMTP Parameters (for email call log reporting).........................................................213
4
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MultiVOIP User Guide
ContentsVOIP
Modem Relay ....................................................................................................238
T1 PHONEBOOK EXAMPLES...................................................................................336
3 Sites, All-T1 Example.....................................................................................336
Call Completion Summaries .............................................................................351
Variations in PBX Characteristics....................................................................354
Free Calls: One VOIP Site to Another.............................................................357
E1 PHONEBOOK EXAMPLES...................................................................................387
3 Sites, All-E1 Example ....................................................................................387
Call Completion Summaries.......................................................................................402
Variations in PBX Characteristics....................................................................405
OPERATION AND MAINTENANCE ...........................................................................411
System Information screen................................................................................411
Statistics Screens ..............................................................................................414
About Call Progress..........................................................................................414
About Logs........................................................................................................420
About IP Statistics.............................................................................................425
About Link Management...................................................................................429
About Packetization Time .................................................................................432
About T1/E1 and BRI Statistics.........................................................................435
About Registered Gateway Details ...................................................................447
MULTIVOIP PROGRAM MENU ITEMS .....................................................................449
Configuration Option........................................................................................451
Configuration Port Setup..................................................................................451
Date and Time Setup.........................................................................................452
Obtaining Updated Firmware...........................................................................452
Implementing a Software Upgrade ...................................................................456
Identifying Current Firmware Version .......................................................................456
Downloading Firmware..............................................................................................457
Downloading Factory Defaults...................................................................................460
Setting a Password (Windows GUI) .................................................................468
5
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Contents
MultiVOIP User Guide
Un-Installing the MultiVOIP Software .............................................................473
Upgrading Software..........................................................................................475
WEB BROWSER INTERFACE ...................................................................................486
SYSLOG SERVER FUNCTIONS ................................................................................491
FCC DECLARATION...............................................................................................495
Industry Canada ...............................................................................................496
FCC Part 68 Telecom.......................................................................................496
Canadian Limitations Notice............................................................................497
APPENDIX A: EXPANSION CARD INSTALLATION (MVP24-48 & MVP30-
60).............................................................................................................................498
INSTALLATION.......................................................................................................499
OPERATION............................................................................................................501
APPENDIX B: CABLE PINOUTS ..............................................................................503
Command Cable ...............................................................................................503
Ethernet Connector...........................................................................................503
T1/E1 Connector...............................................................................................504
Voice/Fax Channel Connectors ........................................................................504
ISDN BRI RJ-45 Pinout Information ................................................................506
ISDN Interfaces: “ST” and “U” .....................................................................507
WELL KNOWN PORT NUMBERS.............................................................................509
PORT NUMBER ASSIGNMENT LIST.........................................................................509
APPENDIX D: INSTALLATION INSTRUCTIONS FOR MVP428 UPGRADE
CARD.......................................................................................................................510
INDEX .....................................................................................................................515
6
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Overview
MultiVOIP User Guide
About This Manual
This 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 the MVP2410 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 & MVP130FXS.
4. ISDN-BRI MultiVOIP units, models MVP410ST & MVP810ST.
These MultiVOIP units can inter-operate with the earlier generation of
MultiVOIP products (the MVP200, MVP400, MVP800, MVP120, etc.)
In this manual, an “x” suffix to a model number means the assertion
applies to all suffix types of that model number. E.g., “MVP410x”
refers collectively to MVP410 and MVP410ST.
The table below describes the vital characteristics of the various models
described in this manual.
8
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MultiVOIP User Guide
Overview
MultiVOIP Product Family
MVP-
2410
MVP
24-48
MVP
3010
MVP
30-60
Description
Model
Function
T1
T1
E1
E1
digital digital
digital
VOIP
unit
digital
VOIP
add-on
card
VOIP
unit
VOIP
add-on
card
Capacity
24
24
30
30
channels added
channels
channels added
channels
Chassis/
Mounting
19” 1U circuit
19” 1U
rack
mount
circuit
card
only
rack
card
only
mount
Description
Model
MVP
810
MVP
428
MVP
410
MVP
210
MVP-
130/
130FXS
analog
voip
Function
Capacity
analog
voip
add-on analog
analog
voip
card
voip
8
4 added
4
2
1
channels channels
channels
channels
channel
table
top
Chassis/
19” 1U
Mounting rack
mount
circuit
card
19” 1U
rack
mount
Table
top
only
Description
Model
MVP810ST
MVP410ST
Function
Capacity
ISDN-BRI voip ISDN-BRI voip
4 ISDN lines
2 ISDN lines
(8 B-channels)
(4 B-channels)
Chassis/
Mounting
19” 1U rack mount 19” 1U rack mount
1. “BRI” means Basic Rate Interface.
9
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Overview
MultiVOIP User Guide
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 two things: information on a very specific software setting or
technical parameter (about telephony or IP) or they need help when
setting up phonebooks for their voip 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 wade through examples like those
in our chapters on T1 (North American standard) Phonebooks and E1
(Euro standard) Phonebooks. Also, the quick setup info of the printed
Quick Start Guide is replicated in this manual for your convenience.
Finally, this manual is meant to be comprehensive. If you notice that
something important is lacking, please let us know.
Additional Resources. The MultiTech web site (www.multitech.com)
offers both a list of Frequently Asked Questions (the MultiVOIP FAQ)
and a collection 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
product family is a coordinated set of products that can operate with
each other in a seamless fashion. For example, both the digital and
analog MultiVOIP units use the same graphic user interface (GUI) in
the MultiVOIP configuration software and both operate under a single
GUI in the MultiVoipManager remote management software. Because
this is the case, the various model numbers and version numbers of
MultiVOIP family products will each appear in various dialog boxes
and commands. But instead of showing these dialog boxes once for
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
except the very model and version numbers themselves will be the
same regardless of the MultiVOIP model used. Also, in some cases, we
will use other typographic devices, like blank underlining
(“MultiVOIP ____”) to denote information that applies to any
and all of the products in this product family.
10
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Overview
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-on
expansion card that doubles the capacity of the MVP2410 without
adding another chassis. These voice-over-IP products have fax
capabilities. These models adhere to the North American standard of
T1 trunk telephony using digital 24-channel time-division multiplexing,
which allows 24 phone conversations to occur on the T1 line
simultaneously. They can also accommodate 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 a few voice-over-IP lines, but not needing carrier-class equipment.
When expansion is needed, the MVP2410 can be field-upgraded into a
dual T1 unit by installing the MVP24-48 kit, which is essentially a
second MultiVOIP motherboard that fits in an open expansion-card slot
in the MVP2410. The upgraded dual unit then accommodates two T1
lines.
T1 VOIP Traffic. The MVP2410 accepts its outbound traffic from a T1
trunk that’s connected to either a PBX or to a telco/carrier. The
MVP2410 transforms the telephony signals into IP packets for
transmission on LANs, WANs, or the Internet. Inbound IP data traffic
is converted to telephony data and signaling.
When connected to PBX. When connected to a PBX, the MVP2410
creates a network node served by 10/100-Base T connections. Local
PBX phone extensions gain toll-free access to all phone stations directly
connected to the VOIP network. Phone extensions at any VOIP location
also gain toll-free access to the entire local public-switched telephone
network (PSTN) at every other VOIP location in the system.
When connected to PSTN. When the T1 line(s) connected to the
MVP2410 are connected directly to the PSTN, the unit becomes a Point-
of-Presence server dedicated to local calls off-net.
11
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Overview
MultiVOIP User Guide
H.323, SIP & SPP. Being H.323 compatible, the MVP2410 can place
calls to telephone equipment at remote IP network locations that also
contain H.323 compatible voice-over-IP gateways. It will interface with
H.323 software and H.323 gatekeeper units. H.323 specifications also
bring to voip telephony many special features common to conventional
telephony. H.323 features of this kind that have been implemented into
the MultiVOIP include Call Hold, Call Waiting, Call Name
Identification, Call Forwarding (from the H.450 standard), and Call
Transfer (H.450.2 from H.323 Version 2). The fourth version of the
H.323 standard improves system resource usage (esp. logical port or
socket usage) by handling call signaling more compactly and allowing
use of the low-overhead UDP protocol instead of the error-correcting
TCP protocol where possible.
The MultiVOIP is also SIP-compatible. (“SIP” means Session Initiation
Protocol.) However, H.450 Supplementary Services features can be
used under H.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 certain situations, especially when firewalls are used and
when dynamic IP address assignment is needed. However, when SPP
is used, certain features of SIP and H.323 will not be available and SPP
will not inter-operate with voip systems using H.323 or SIP.
Data Compression & Quality of Service. The MultiVOIP MVP2410
comes equipped 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
basic functions: (a) it converts a dialed number into an IP address, (b) it
sends voice over the data network, (c) it establishes a connection with
another VOIP gateway at a remote site, and (d) it receives voice over
the data network. Voice is handled as IP packets with a variety of
compression options. Each T1 connection to the MultiVOIP provides 24
time-slot channels to connect to the telco or to serve phone or fax
stations connected to a PBX.
Ports. The MVP2410 has one 10/100 Mbps Ethernet LAN interface and
one Command port for configuration. An MVP2410 upgraded with the
MVP24-48 kit will have two Ethernet LAN interfaces and two
Command ports.
PSTN Failover Feature. The MultiVOIP can be programmed to divert
calls to the PSTN temporarily in case the IP network fails.
Gatekeeper. T1 voip systems can have gatekeeper functionality by
adding, as an endpoint, a Multi-Tech standalone gatekeeper (special
12
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MultiVOIP User Guide
Overview
software residing in separate 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 standard gatekeepers functions (address
translation, admission control, and bandwidth control) and also
supports many valuable optional functions (call control signaling, call
authorization, bandwidth management, and call management).
Management. Configuration and system management can be done
locally with the MultiVOIP configuration software. After an IP address
has been assigned locally, other configuration can be done remotely
using the MultiVOIP web browser GUI. Remote system management
can be done with the MultiVoipManager SNMP software or via the
MultiVOIP web browser GUI. All of these control software packages
are included on the Product CD.
13
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Overview
MultiVOIP User Guide
While the web GUI’s appearance differs slightly, its content and
organization are essentially the same as that of the Windows GUI
(except for logging).
The primary advantage of the web GUI is remote access for control and
configuration. The controller PC and the MultiVOIP unit itself must
both be connected 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 the
MultiVOIP Windows GUI at any time. However, you must log out of
the web browser GUI before using the MultiVOIP Windows GUI.
14
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Overview
Logging of System Events. MultiTech has built SysLog Server
functionality into the software of the MultiVOIP units. SysLog is a de
facto standard for logging events in network communication systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement
this functionality, you will need a SysLog client program (sometimes
referred to as a “daemon”). SysLog client programs, both paid and
freeware, can be obtained from Kiwi Enterprises, among other firms.
a means of structuring console messages for 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 with MultiVOIP units. Kiwi’s brief description of their
SysLog program indicates the typical scope of such programs. “Kiwi
Syslog Daemon is a freeware Syslog Daemon for the Windows
platform. It receives, logs, displays and forwards Syslog messages from
hosts such as routers, switches, Unix hosts and any other syslog
enabled device. There are many customizable options available.”
15
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Overview
MultiVOIP User Guide
Supplementary Telephony Services. The H.450 standard (an addition
to H.323) brings to voip telephony more of the premium features found
in PSTN and PBX telephony. MultiVOIP units offer five of these H.450
features: Call Transfer, Call Hold, Call Waiting, Call Name
Identification (not the same as Caller ID), and Call Forwarding. (The
first four features are found in the “Supplementary Services” window;
the fifth, Call Forwarding, appears in the Add/Edit Inbound
phonebook screen.) Note that the first three features are closely related.
All of these H.450 features are supported for H.323 operation only; they
are not supported for SIP or SPP.
T1 Front Panel LEDs
The MVP2410 and MVP24-48 both use a common main circuit board or
motherboard. Consequently the LED indicators are the same for both.
Active LEDs. The MVP2410 front panel has two sets of identical LEDs.
In the MVP2410 as shipped (that is, without an expansion card), the
left-hand set of LEDs is functional whereas the right-hand set is not.
When the MVP2410 has been upgraded with an MVP24-48 kit, the
right-hand set of LEDs will also become active.
Figure 1-3: MVP2410 LEDs
T1 LED Descriptions. The descriptions below apply to the digital T1
MultiVOIP units. The MVP2410 has four sets of LEDs plus a lone LED
at its far right end. As viewed from the front of the MVP2410, it is the
two left groups that are active and present feedback about the operation
of the unit. If an MVP24-48 expansion card is added to the MVP2410,
the two LED groups on the right become operational with respect to the
second T1 connection.
16
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MultiVOIP User Guide
Overview
MVP2410 Front Panel LED Definitions
DESCRIPTION
LED NAME
Power
Indicates presence of power.
After power up, the Boot LED will be on for about 10
seconds while the MVP2410 is booting.
Boot
FDX
Full-Duplex & Collision LED. This LED indicates
whether the Ethernet connection is half-duplex or full-
duplex (FDX) and, in half-duplex mode, indicates
occurrence of data collisions. LED is on constantly for
full-duplex mode; LED is off constantly for half-duplex
mode. When operating in half-duplex mode, the LED
will flash during data collisions.
LNK
Link/Activity LED. This LED is lit if Ethernet
connection 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.
T1
When lit, indicates presence of T1 connection.
E1. Not supported.
E1
PRI
ONL
PRI. On if T1 line is of ISDN-Primary-Rate type.
Online. This LED is on when frame synchroni-
zation has been established on the T1/E1 link.
IC
IC LED is on when Internal Clocking is selected in
T1/E1 configuration.
LC
LS
Indicates Loss of Carrier.
Indicates Loss of Signal.
For testing purposes only.
Test
17
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Overview
MultiVOIP User Guide
Introduction to EI MultiVOIPs
(MVP3010 & MVP30-60)
We proudly present MultiTech’s E1 Digital Multi-VOIP products. The
MVP3010 is a rack-mount model and the MVP30-60 is an add-on
expansion card that doubles the capacity of the MVP3010 without
adding another chassis. All of these voice-over-IP products have fax
capabilities. All adhere to the European standard of E1 trunk telephony
using digital 30-channel time-division multiplexing, which allows 30
phone conversations to occur on the E1 line simultaneously. All can
also accommodate E1 lines of the ISDN Primary Rate Interface type
(ISDN-PRI).
Figure 1-4. MultiVOIP MVP3010 Chassis
Scale-ability. The MVP3010 is tailored to companies needing more
than a few voice-over-IP lines, but not needing carrier-class equipment.
When expansion is needed, the MVP3010 can be field-upgraded into a
dual E1 unit by installing the MVP30-60 kit, which is essentially a
second MultiVOIP motherboard that fits into an open expansion-card
slot in the MVP3010. The upgraded dual unit then accommodates two
E1 lines.
E1 VOIP Traffic. The MVP3010 accepts its outbound traffic from an E1
trunk that’s connected to either a PBX or to a telco/carrier. The
MVP3010 transforms the telephony signals into IP packets for
transmission on LANs, WANs, or the Internet. Inbound IP data traffic
is converted to telephony data and signaling.
When connected to PBX. When connected to a PBX, the MVP3010
creates a network node served by 10/100-Base T connections. Local
PBX phone extensions gain toll-free access to all phone stations directly
connected to the VOIP network. Phone extensions at any VOIP location
also gain local-rate access to the entire local public-switched telephone
network (PSTN) at every other VOIP location in the system.
When connected to PSTN. When the E1 line(s) connected to the
MVP3010 are connected directly to the PSTN, the unit becomes a Point-
of-Presence server dedicated to local calls off-net.
18
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MultiVOIP User Guide
Overview
H. 323, SIP, & SPP. Being H.323 compatible, the MVP3010 can place
calls to telephone equipment at remote IP network locations that also
contain H.323 compatible voice-over-IP gateways. It will interface with
H.323 software and H.323 gatekeeper units. H.323 specifications also
bring to voip telephony many special features common to conventional
telephony. H.323 features of this 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 the H.323 standard improves
system resource usage (esp. logical port or socket usage) by handling
call signaling more compactly and allowing use of the low-overhead
UDP protocol instead of the error-correcting TCP protocol where
possible.
The MultiVOIP is also SIP-compatible. (“SIP” means Session Initiation
Protocol.) However, H.450 Supplementary Services features can be
used under H.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 certain situations, especially when firewalls are used and
when dynamic IP address assignment is needed. However, when SPP
is used, certain features of SIP and H.323 will not be available and SPP
will not inter-operate with voip systems using H.323 or SIP.
Data Compression & Quality of Service. The MultiVOIP3010 comes
equipped 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
basic functions: (a) it converts a dialed number into an IP address, (b) it
sends voice over the data network, (c) it establishes a connection with
another VOIP gateway at a remote site, and (d) it receives voice over
the data network. Voice is handled as IP packets with a variety of
compression options. Each E1 connection to the MultiVOIP provides 30
time-slot channels to connect to the telco or to serve phone or fax
stations connected to a PBX.
Ports. The MVP3010 also has a 10/100 Mbps Ethernet LAN interface,
and a Command port for configuration. An MVP3010 upgraded with
the MVP30-60 kit will have two Ethernet LAN interfaces and two
Command ports.
PSTN Failover Feature. The MultiVOIP can be programmed to divert
calls to the PSTN temporarily in case the IP network fails.
19
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Overview
MultiVOIP User Guide
Gatekeeper. E1 voip systems can have gatekeeper functionality by
adding, as an endpoint, a Multi-Tech standalone gatekeeper (special
software residing in separate 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 standard gatekeepers functions (address
translation, admission control, bandwidth control, and zone
management) and also supports many valuable optional functions (call
control signaling, call authorization, and bandwidth management).
Management. Configuration and system management can be done
locally with the MultiVOIP configuration software. After an IP address
has been assigned locally, other configuration can be done remotely
using the MultiVOIP web browser GUI. Remote system management
can be done with the MultiVoipManager SNMP software or via the
MultiVOIP web browser GUI. All of these control software packages
are included on the Product CD.
20
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MultiVOIP User Guide
Overview
While the web GUI’s appearance differs slightly, its content and
organization are essentially the same as that of the Windows GUI
(except for logging).
The primary advantage of the web GUI is remote access for control and
configuration. The controller PC and the MultiVOIP unit itself must
both be connected 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 the
MultiVOIP Windows GUI at any time. However, you must log out of
the web browser GUI before using the MultiVOIP Windows GUI.
21
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Overview
MultiVOIP User Guide
Logging of System Events. MultiTech has built SysLog Server
functionality into the software of the MultiVOIP units. SysLog is a de
facto standard for logging events in network communication systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement
this functionality, you will need a SysLog client program (sometimes
referred to as a “daemon”). SysLog client programs, both paid and
freeware, can be obtained from Kiwi Enterprises, among other firms.
a means of structuring console messages for 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 with MultiVOIP units. Kiwi’s brief description of their
SysLog program indicates the typical scope of such programs. “Kiwi
Syslog Daemon is a freeware Syslog Daemon for the Windows
platform. It receives, logs, displays and forwards Syslog messages from
hosts such as routers, switches, Unix hosts and any other syslog
enabled device. There are many customizable options available.”
22
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MultiVOIP User Guide
Overview
Supplementary Telephony Services. The H.450 standard (an addition
to H.323) brings to voip telephony more of the premium features found
in PSTN and PBX telephony. MultiVOIP units offer five of these H.450
features: Call Transfer, Call Hold, Call Waiting, Call Name
Identification (not the same as Caller ID), and Call Forwarding. (The
first four features are found in the “Supplementary Services” window;
the fifth, Call Forwarding, appears in the Add/Edit Inbound
phonebook screen.) Note that the first three features are closely related.
All of these H.450 features are supported for H.323 operation only; they
are not supported for SIP or SPP.
E1 Front Panel LEDs
Because the MVP3010 and MVP30-60 both use a common main circuit
card or motherboard, 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.
In the MVP3010 as shipped (that is, without an expansion card), the
left-hand set of LEDs is functional whereas the right-hand set is not.
When the MVP3010 has been upgraded with an MVP30-60 kit, the
right-hand set of LEDs will also become active.
23
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Overview
MultiVOIP User Guide
E1 LED Descriptions
MVP3010 Front Panel LED Definitions
DESCRIPTION
LED NAME
Power
Indicates presence of power.
After power up, the Boot LED will be on for
about 10 seconds while the MVP3010 is booting.
Full-Duplex & Collision LED. This LED indicates
whether the Ethernet connection is half-duplex or full-
duplex (FDX) and, in half-duplex mode, indicates
occurrence of data collisions. LED is on constantly for
full-duplex mode; LED is off constantly for half-
duplex mode. When operating in half-duplex mode,
the LED will flash during data collisions.
Boot
FDX
LNK
Link/Activity LED. This LED is lit if Ethernet
connection 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.
T1
E1
T1. Not supported.
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
synchronization has been established on the
T1/E1 link.
IC
IC LED is on when Internal Clocking is selected
in T1/E1 configuration.
LC
LS
Indicates Loss of Carrier.
Indicates Loss of Signal.
For testing purposes only.
Test
24
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MultiVOIP User Guide
Overview
Introduction to Analog MultiVOIPs
(MVP-130/130FXS, MVP-210/410/810 &
MVP428)
VOIP: The Free Ride. We proudly present Multi-Tech's MVP-
130/130FXS and MVP-210/410/810 generation of MultiVOIP Voice-
over-IP Gateways. All of these models allow voice/fax communication
to be transmitted at no additional expense over your existing IP
network, which has ordinarily been data only. To access this free voice
and fax communication, you simply connect the MultiVOIP to your
telephone equipment and your existing Internet connection. These
analog MultiVOIPs inter-operate readily with T1 or E1 MultiVOIP
units.
Voi ce/Fax5
Voci e/ Fax6
Voci eF/ ax7
Voi ec/ Fax8
CR
X
X
M
M
T
T
R
VC
SX
G
SR
SR
G
G
X
X
TM
TM
RC
V
X
S
G
R
GS
GS
MX
XM
T
T
R
C
V
X
GS
SR
G
G
X
X
MT
MT
V
X
S
G
R
GS
GS
Power
Boot
Ether net
Voci e/Fax1
Voci e/ Fax2
Voci eF/ ax
3
Voi ec/ Fax4
CR
R
C
V
X
M
T
C
LO
NL
K
R
VC
SX
G
RC
V
X
S
G
R
R
C
V
X
GS
SR
V
X
S
G
R
Figure 1-6: MVP-410/810 Chassis
Figure 1-7: MVP-210 Chassis
25
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Overview
MultiVOIP User Guide
Figure 1-8: MultiVOIP MVP-130/130FXS Chassis
Capacity. MultiVOIP model MVP810 is an eight-channel unit, the
model MVP410 is a four-channel, the model MVP210 is a two-channel
units, the MV130 is a single-channel unit and the MVP130FXS is a
single-channel unit that supports the FXS telephony interface only. All
of these MultiVOIP units have a 10/100Mbps Ethernet interface and a
command port for configuration. The MVP428 is an expansion circuit
card for the four-channel MVP410 that turns it into an eight-channel
voip.
Mounting. Mechanically, the MVP410 and MVP810 MultiVOIPs are
designed for a one-high industry-standard EIA 19-inch rack enclosure.
By contrast, MVP-130/130FXS and the MVP210 are tabletop units. The
product must be installed by qualified service personnel in a restricted-
access area, in accordance with Articles 110-16, 10-17, and 110-18 of the
National Electrical Code, ANSI/NFPA 70.
Phone System Transparency. These MultiVOIPs inter-operate with a
telephone switch or PBX, acting as a switching device that directs voice
and fax calls over an IP network. The MultiVOIPs have “phonebooks,”
directories that determine to who calls may be made and the sequences
that must be used to complete calls through the MultiVOIP. The
phonebooks allow the phone user to interact with the VOIP system just
as they would with an ordinary PBX or telco switch. When the
phonebooks are set, special dialing sequences are minimized or
eliminated altogether. Once the call destination is determined, the
phonebook settings determine whether the destination VOIP unit must
strip off or add dialing digits to make the call appear at its destination
to be a local call.
26
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MultiVOIP User Guide
Overview
H. 323, SIP, & SPP. Being H.323 compatible, the analog MultiVOIP
unit can place calls to telephone equipment at remote IP network
locations that also contain H.323 compatible voice-over-IP gateways. It
will interface with H.323 software and H.323 gatekeeper units. H.323
specifications also bring to voip telephony many special features
common to conventional telephony. H.323 features of this 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
the H.323 standard improves system resource usage (esp. logical port or
socket usage) by handling call signaling more compactly and allowing
use of the low-overhead UDP protocol instead of the error-correcting
TCP protocol where possible.
The MultiVOIP is also SIP-compatible. (“SIP” means Session Initiation
Protocol.) However, H.450 Supplementary Services features can be
used under H.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 certain situations, especially when firewalls are used and
when dynamic IP address assignment is needed. However, when SPP
is used, certain features of SIP and H.323 will not be available and SPP
will not inter-operate with voip systems using H.323 or SIP.
Data Compression & Quality of Service. The analog MultiVOIP unit
comes equipped with a variety of data compression capabilities,
including G.723, G.729, and G.711 and features DiffServ quality-of-
service (QoS) capabilities.
PSTN Failover Feature. The MultiVOIP can be programmed to divert
calls to the PSTN temporarily in case the IP network fails.
Gatekeepers. For voip systems built with MultiTech’s analog gateway
units, users can have a stand-alone gatekeeper (gatekeeper software
residing in separate 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 standard gatekeepers functions (address translation,
admission control, and bandwidth control) and also supports many
valuable optional functions (call control signaling, call authorization,
bandwidth management, and call management).
27
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Overview
MultiVOIP User Guide
Management. Configuration and system management can be done
locally with the MultiVOIP configuration software. After an IP address
has been assigned locally, other configuration can be done remotely
using the MultiVOIP web browser GUI. Remote system management
can be done with the MultiVoipManager SNMP software or via the
MultiVOIP web browser GUI. All of these control software packages
are included on the Product CD.
While the web GUI’s appearance differs slightly, its content and
organization are essentially the same as that of the Windows GUI
(except for logging).
The primary advantage of the web GUI is remote access for control and
configuration. The controller PC and the MultiVOIP unit itself must
both be connected to the same IP network and their IP addresses must
be known.
28
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MultiVOIP User Guide
Overview
Once you’ve begun using the web browser GUI, you can go back to the
MultiVOIP Windows GUI at any time. However, you must log out of
the web browser GUI before using the MultiVOIP Windows GUI.
Logging of System Events. MultiTech has built SysLog Server
functionality into the software of the MultiVOIP units. SysLog is a de
facto standard for logging events in network communication systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement
this functionality, you will need a SysLog client program (sometimes
referred to as a “daemon”). SysLog client programs, both paid and
freeware, can be obtained from Kiwi Enterprises, among other firms.
a means of structuring console messages for 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 with MultiVOIP units. Kiwi’s brief description of their
SysLog program indicates the typical scope of such programs. “Kiwi
Syslog Daemon is a freeware Syslog Daemon for the Windows
platform. It receives, logs, displays and forwards Syslog messages from
hosts such as routers, switches, Unix hosts and any other syslog
enabled device. There are many customizable options available.”
29
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Overview
MultiVOIP User Guide
Supplementary Telephony Services. The H.450 standard (an addition
to H.323) brings to voip telephony more of the premium features found
in PSTN and PBX telephony. MultiVOIP units offer five of these H.450
features: Call Transfer, Call Hold, Call Waiting, Call Name
Identification (not the same as Caller ID), and Call Forwarding. (The
first four features are found in the “Supplementary Services” window;
the fifth, Call Forwarding, appears in the Add/Edit Inbound
phonebook screen.) Note that the first three features are closely related.
All of these H.450 features are supported for H.323 operation only; they
are not supported for SIP or SPP.
Analog MultiVOIP Front Panel LEDs
LED Types. The MultiVOIPs have two types of LEDs on their front
panels:
(1) general operation LED indicators (for power, booting, and
ethernet functions), and
(2) channel operation LED indicators that describe the data traffic
and performance in each VOIP data channel.
Active LEDs. On both the MVP410 and MVP810, there are eight sets of
channel-operation LEDs. However, on the MVP410, only the lower
four sets of channel-operation LEDs are functional. On the MVP810, all
eight sets are functional.
Figure 1-9. MVP410/810 LEDs
30
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MultiVOIP User Guide
Overview
Similarly, the MVP210 has the general-operation indicator LEDs and
two sets of channel-operation LEDs, one for each channel.
Figure 1-10. MVP210 LEDs
Finally, the MVP130 has the general-operation indicator LEDs and a set
of channel-operation LEDs for its single voip channel.
Figure 1-11. MVP-130/130FXS LEDs
31
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Overview
MultiVOIP User Guide
Analog MultiVOIP LED Descriptions
MVP-210/410/810 Front Panel LED Definitions
LED NAME DESCRIPTION
General Operation LEDs (one set on each MultiVOIP model)
Power
Boot
Indicates presence of power.
After power up, the Boot LED will be on briefly while the
MultiVOIP is booting. It lights whenever the MultiVOIP is
booting or downloading a setup configuration data set.
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
on constantly for full-duplex mode; LED is off
constantly for half-duplex mode. When operating in
half-duplex mode, the LED will flash during data
collisions.
Ethernet
LNK. Link/Activity LED. This LED is lit if Ethernet
connection 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)
Transmit. This indicator blinks when voice packets
are being transmitted to the local area network.
XMT
RCV
XSG
Receive. This indicator blinks when voice packets
are being received from the local area network.
Transmit Signal. This indicator lights when the FXS-
configured channel is off-hook, the FXO-configured
channel is receiving a ring from the Telco, or the M
lead is active on the E&M configured channel. That is,
it lights when the MultiVOIP is receiving a ring from
the PBX.
Receive Signal. This indicator lights when the FXS-
configured channel is ringing, the FXO-configured
channel has taken the line off-hook, or the E lead is
active on the E&M-configured channel.
RSG
32
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MultiVOIP User Guide
Overview
MVP-130/130FXS Front Panel LED Definitions
LED NAME DESCRIPTION
General Operation LEDs
Power
Boot
Indicates presence of power.
After power up, the Boot LED will be on briefly while
the MultiVOIP is booting. It lights whenever the
MultiVOIP is booting or downloading a setup
configuration data set.
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
on constantly for full-duplex mode; LED is off
constantly for half-duplex mode. When operating in
half-duplex mode, the LED will flash during data
collisions.
Ethernet
LNK. Link/Activity LED. This LED is lit if Ethernet
connection 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
Transmit. This indicator blinks when voice packets
are being transmitted to the local area network.
TX
RX
XS
Receive. This indicator blinks when voice packets
are being received from the local area network.
Transmit Signal. This indicator lights when the
FXS-configured channel is off-hook or the FXO-
configured channel (MVP130 only) is receiving a ring
from the Telco or PBX.
Receive Signal. This indicator lights when the FXS-
configured channel is ringing or the FXO-configured
channel (MVP130 only) has taken the line off-hook.
RS
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Overview
MultiVOIP User Guide
Introduction to ISDN-BRI MultiVOIPs
(MVP410ST & MVP810ST)
VOIP: The Free Ride. We proudly present Multi-Tech's MVP-
410ST/810ST generation of MultiVOIP Voice-over-IP Gateways. All of
these models allow voice/fax communication to be transmitted at no
additional expense over your existing IP network, which has ordinarily
been data only. To access this free voice and fax communication, you
simply connect the MultiVOIP to your telephone equipment and your
existing Internet connection. These ISDN Basic Rate Interface (ISDN-
BRI) MultiVOIPs inter-operate readily with T1 or E1 MultiVOIP units
(T1 and E1 MultiVOIP units can operate in ISDN Primary Rate Mode,
ISDN-PRI, as well).
Power
Boot
Ethernet
RCV XMT CO
ISDN
1
ISDN
2
Ch
M T R CV
ISDN
3
ISDN
4
Ch
1
Ch
2
3
Ch
4
Ch
5
Ch
6
Ch
7
Ch
8
L
LNK
X
M
T
R
C
V
X
M
T
R
C
V
X
X
M
T
R
C
V
X
M
T
R
C
V
X
M
T
R
C
V
X
M
T
R
C
V
X
M
T
R
C
V
D
D
D
D
Figure 1-12: MVP-410ST/810ST Chassis
Capacity. MultiVOIP model MVP810ST accommodates four ISDN-BRI
lines (eight B-channels) and model MVP410ST accommodates two
ISDN-BRI channels (four B-channels). Both of these MultiVOIP units
have a 10/100Mbps Ethernet interface and a command port for
configuration.
Mounting. Mechanically, the MVP410ST and MVP810ST MultiVOIPs
are designed for a one-high industry-standard EIA 19-inch rack
enclosure. The product must be installed by qualified service personnel
in a restricted-access area, in accordance with Articles 110-16, 10-17, and
110-18 of the National Electrical Code, ANSI/NFPA 70.
Phone System Transparency. These MultiVOIPs inter-operate with a
telephone switch or PBX, acting as a switching device that directs voice
and fax calls over an IP network. The MultiVOIPs have “phonebooks,”
directories that determine to who calls may be made and the sequences
that must be used to complete calls through the MultiVOIP. The
phonebooks allow the phone user to interact with the VOIP system just
as they would with an ordinary PBX or telco switch. When the
phonebooks are set, special dialing sequences are minimized or
eliminated altogether. Once the call destination is determined, the
phonebook settings determine whether the destination VOIP unit must
strip off or add dialing digits to make the call appear at its destination
to be a local call.
34
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MultiVOIP User Guide
Overview
H. 323, SIP, & SPP. Being H.323 compatible, the BRI MultiVOIP unit
can place calls to telephone equipment at remote IP network locations
that also contain H.323 compatible voice-over-IP gateways. It will
interface with H.323 software and H.323 gatekeeper units. H.323
specifications also bring to voip telephony many special features
common to conventional telephony. H.323 features of this 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
the H.323 standard improves system resource usage (esp. logical port or
socket usage) by handling call signaling more compactly and allowing
use of the low-overhead UDP protocol instead of the error-correcting
TCP protocol where possible.
The MultiVOIP is also SIP-compatible. (“SIP” means Session Initiation
Protocol.) However, H.450 Supplementary Services features can be
used under H.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 certain situations, especially when firewalls are used and
when dynamic IP address assignment is needed. However, when SPP
is used, certain features of SIP and H.323 will not be available and SPP
will not inter-operate with voip systems using H.323 or SIP.
Data Compression & Quality of Service. The BRI MultiVOIP unit
comes equipped with a variety of data compression capabilities,
including G.723, G.729, and G.711 and features DiffServ quality-of-
service (QoS) capabilities.
PSTN Failover Feature. The MultiVOIP can be programmed to divert
calls to the PSTN temporarily in case the IP network fails.
Gatekeeper. At this writing, ISDN-BRI MultiVOIP systems can have
gatekeeper functionality only by adding, as an endpoint, a standalone
gatekeeper (special software residing in separate hardware).
Gatekeepers are optional 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 standard gatekeepers functions (address
translation, admission control, bandwidth control, and zone
management) and also support many valuable optional functions (call
control signaling, call authorization, bandwidth management, and call
management). The stand-alone gatekeeper is, however, slightly more
feature-rich than the embedded gatekeeper. For more details, see the
“Embedded Gatekeeper” chapter of this manual and the manual on
MultiTech’s stand-alone gatekeeper.
35
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Overview
MultiVOIP User Guide
Management. Configuration and system management can be done
locally with the MultiVOIP configuration software. After an IP address
has been assigned locally, other configuration can be done remotely
using the MultiVOIP web browser GUI. Remote system management
can be done with the MultiVOIP web browser GUI. These control
software packages are included on the Product CD.
While the web GUI’s appearance differs slightly, its content and
organization are essentially the same as that of the Windows GUI
(except for logging).
The primary advantage of the web GUI is remote access for control and
configuration. The controller PC and the MultiVOIP unit itself must
both be connected to the same IP network and their IP addresses must
be known.
36
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MultiVOIP User Guide
Overview
Once you’ve begun using the web browser GUI, you can go back to the
MultiVOIP Windows GUI at any time. However, you must log out of
the web browser GUI before using the MultiVOIP Windows GUI.
Logging of System Events. MultiTech has built SysLog Server
functionality into the software of the MultiVOIP units. SysLog is a de
facto standard for logging events in network communication systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement
this functionality, you will need a SysLog client program (sometimes
referred to as a “daemon”). SysLog client programs, both paid and
freeware, can be obtained from Kiwi Enterprises, among other firms.
a means of structuring console messages for 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 with MultiVOIP units. Kiwi’s brief description of their
SysLog program indicates the typical scope of such programs. “Kiwi
Syslog Daemon is a freeware Syslog Daemon for the Windows
platform. It receives, logs, displays and forwards Syslog messages from
hosts such as routers, switches, Unix hosts and any other syslog
enabled device. There are many customizable options available.”
37
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Overview
MultiVOIP User Guide
Supplementary Telephony Services. The H.450 standard (an addition
to H.323) brings to voip telephony more of the premium features found
in PSTN and PBX telephony. MultiVOIP units offer five of these H.450
features: Call Transfer, Call Hold, Call Waiting, Call Name
Identification (not the same as Caller ID), and Call Forwarding. (The
first four features are found in the “Supplementary Services” window;
the fifth, Call Forwarding, appears in the Add/Edit Inbound
phonebook screen.) Note that the first three features are closely related.
All of these H.450 features are supported for H.323 operation only; they
are not supported for SIP or SPP.
ISDN BRI MultiVOIP Front Panel LEDs
LED Types. The MultiVOIPs have two types of LEDs on their front
panels:
(1) general operation LED indicators (for power, booting, and
ethernet functions), and
(2) channel operation LED indicators that describe the data traffic
and performance in each VOIP data channel.
Active LEDs. On the MVP810ST, there are four sets of ISDN-operation
LEDs. On the MVP410ST, there are two sets of ISDN-operation LEDs.
Each set contains one “D” LED and two sets of channel operation LEDs
(XMT and RCV).
Figure 1-13. MVP-410ST/810ST LEDs
38
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MultiVOIP User Guide
Overview
ISDN-BRI MultiVOIP LED Descriptions
MVP-410ST/810ST Front Panel LED Definitions
LED NAME DESCRIPTION
General Operation LEDs (one set on each MultiVOIP model)
Power
Boot
Indicates presence of power.
After power up, the Boot LED will be on briefly while the
MultiVOIP is booting. It lights whenever the MultiVOIP is
booting or downloading a setup configuration data set.
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
on constantly for full-duplex mode; LED is off
constantly for half-duplex mode. When operating in
half-duplex mode, the LED will flash during data
collisions.
Ethernet
LNK. Link/Activity LED. This LED is lit if Ethernet
connection 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.
D-Channel Operation LEDs (one for each ISDN line)
ISDN D-channel & physical layer indicator. One “D”
LED for each ISDN-BRI connection. The “D” LED is
off when the BRI physical layer is de-activated.* It
flashes when a connection is being established on the
physical layer. It is on when the physical layer has
been activated. It flickers to indicate D-channel traffic.
*If the voip is running in terminal mode and its BRI
line is unplugged, the D LED goes off. However, if the
voip is running in network mode and its BRI line is
unplugged, its LED will flash at regular interval.
D
B-Channel Operation LEDs (one for each B-channel)
Transmit. This indicator blinks when voice packets
are being transmitted onto the B-channel.
XMT
Receive. This indicator blinks when voice packets
are being received on the B-channel.
RCV
39
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Overview
MultiVOIP User Guide
Computer Requirements
The computer on which the MultiVOIP’s configuration program is
installed must meet these requirements:
•
must be IBM-compatible PC with MS Windows operating
system;
•
must have an available COM port for connection to the
MultiVOIP.
However, this PC does not need to be connected to the MultiVOIP
permanently. It only needs to be connected when local configuration
and monitoring are done. Nearly all configuration and monitoring
functions can be done remotely via the IP network.
40
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MultiVOIP User Guide
Overview
Specifications
Specs for Digital T1 MultiVOIP Units
Digital T1 MultiVOIP Specifications
Parameter
……/Model
MVP-2410
w/ MVP24-48
Expansion
Card
MVP-2410
Operating
Voltage/Current
100-240 VAC
1.2 - 0.6 A
50/60 Hz
100-240 VAC
1.2 - 0.6 A
50/60 Hz
Mains
Frequencies
Power
17 watts
27 watts
Consumption
Mechanical
Dimensions
1.75”H x
17.4”W x
8.75”D
1.75”H x
17.4”W x
8.75”D
4.5cm H x
44.2 cm W x
22.2 cm D
7.1 lbs.
4.5cm H x
44.2 cm W x
22.2 cm D
7.5 lbs.
Weight
(3.2 kg)
(3.4 kg)
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MultiVOIP User Guide
Specs for Digital E1 MultiVOIP Units
Digital E1 MultiVOIP Specifications
Parameter
……/Model
MVP-3010
MVP-3010
w/ MVP30-60
Expansion
Card
Operating
Voltage/Current
100-240 VAC
1.2 - 0.6 A
50/60 Hz
100-240 VAC
1.2 - 0.6 A
50/60 Hz
Mains
Frequencies
Power
17 watts
27 watts
Consumption
Mechanical
Dimensions
1.75”H x
17.4”W x
8.75”D
1.75”H x
17.4”W x
8.75”D
4.5cm H x
44.2 cm W x
22.2 cm D
7.1 lbs.
4.5cm H x
44.2 cm W x
22.2 cm D
7.5 lbs.
Weight
(3.2 kg)
(3.4 kg)
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MultiVOIP User Guide
Overview
Specs for Analog/BRI MultiVOIP Units
Parameter
/Model
Operating
Voltage/
Current
MVP210
MVP410
MVP810 or
MVP410 + 428
100-240 VAC
1.2 - 0.6 A
100-240 VAC
1.2 - 0.6 A
External
transformer:
3A @5V
50/60 Hz
50/60 Hz
50/60 Hz
Mains
Frequencies
Power
19 watts
29 watts
46 watts
Consumption
Mechanical
Dimensions
6.2” W x
9” D x
1.4” H
1.75” H x
17.4” W x
8.5” D
1.75” H x
17.4” W x
8.5” D
15.8cm W x
22.9cm D x
3.6cm H
1.8lbs (.82kg)
2.6lbs (1.17kg)
with
4.5cm H x
44.2 cm W x
21.6 cm D
7.1 lbs.
4.5cm H x
44.2 cm W x
21.6 cm D
7.7 lbs.
Weight
(3.2 kg)
(3.5 kg)
transformer
Parameter
……/Model
MVP410ST
MVP810ST
MVP-
130/130FXS
100-240VAC
1.0 A
100-240VAC
1.2-0.6 A
100-240VAC
1.2-0.6 A
Operating
Voltage/
Current
50/60 Hz
12 watts
50/60 Hz
50/60 Hz
Mains
Frequencies
Power
Consumption
Mechanical
Dimensions
18 watts
9.7 watts (with
phone off hook)
4.3" W x 5.6" D
1.0" H
Same as
MVP410
Same as MVP810
10.8 cm W X
14.2 cm D X
2.95 cm H
8 oz.
6.61 lbs.
(3.00 kg)
6.75 lbs.
(3.06 kg)
Weight
(23 g)
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Overview
MultiVOIP User Guide
Installation at a Glance
The basic steps of installing your MultiVOIP network involve
unpacking the units, connecting the cables, and configuring the units
using management software (MultiVOIP Configuration software) and
confirming connectivity with another voip site. This process results in a
fully functional Voice-Over-IP network.
Related Documentation
The MultiVOIP User Guide (the document you are now reading) comes
in electronic form and is included on your system CD. It presents in-
depth information on the features and functionality of Multi-Tech’s
MultiVOIP Product Family.
The CD media is produced using Adobe AcrobatTM for viewing and
printing the user guide. To view or print your copy of a user guide,
load Acrobat ReaderTM on your system. The Acrobat Reader is included
on the MultiVOIP CD and is also a free download from Adobe’s Web
Site:
This MultiVOIP User Guide is also available on Multi-Tech’s Web site
at:
Viewing and printing a user guide from the Web also requires that you
have the Acrobat Reader loaded on your system. ToselecttheMultiVOIP
User Guide from the Multi-Tech Systems home page, click Documents and then click
MultiVOIP Family in the product list drop-down window. All documents for this
MultiVOIP Product Family will be displayed. You can then choose User Guide
(MultiVOIP Product Family) to view or download the .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 the field.
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Quick Start Instructions
MultiVOIP User Guide
Introduction
This chapter gets the MultiVOIP up and running quickly. The details
we’ve skipped to make this brief can be found elsewhere in the manual
(see Table of Contents and Index).
MultiVOIP Startup Tasks
Task
Summary
z Collecting Phone/IP
The MultiVOIP must be configured to
interface with your particular phone
system and IP network. To do so,
certain details must be known about
those phone and IP systems.
Details (vital!)
z Placement
Decide where you’ll mount the voip.
z Command/Control
Computer Setup:
Specs & Settings
Some modest minimum specifications
must be met. A COM port must be set
up.
z Hookup
Connect power, phone, and data cables
per diagram.
z Software Installation
This is the configuration program.
It’s a standard Windows software
installation.
z Phone/IP Starter
You will enter phone numbers and IP
addresses. You’ll use default parameter
values where possible to get the system
running quickly.
Configuration
z Phonebook Starter
The phonebook is where you specify
how calls will be routed. To get the
system running quickly, you’ll make
phonebooks for just two voip sites.
Configuration
z Connectivity Test
You’ll find out if your voip system can
carry phone calls between two sites.
That means you’re up and running!
z Troubleshooting
Detect and remedy any problems that
might have prevented connectivity.
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MultiVOIP User Guide
Quick Start Instructions
Phone/IP Details *Absolutely Needed*
Before Starting the Installation
Gather IP Information
Ask your computer network
administrator.
Info needed to operate:
all MultiVOIP models.
9
IP Network Parameters:
Record for each VOIP Site
in System
#
• IP Address
• IP Mask
• Gateway
• Domain Name Server (DNS) Info
(not implemented; for future use)
Gather Telephone Information (T1)
T1 Phone Parameters
Info needed to operate:
9
MVP2410
Ask phone company or
PBX maintainer.
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 the
opposite 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)
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Quick Start Instructions
MultiVOIP User Guide
Phone/IP Details *Absolutely Needed* (cont’d)
Gather Telephone Information (E1)
E1 Phone Parameters
Info needed to operate:
9
MVP3010
Ask phone company or
PBX maintainer.
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
internal or external clocking? _________________
Note that the setting used in the voip unit will be the
opposite of the setting used by the telco/PBX.
• Which line coding is used? AMI___ or HDB3___
• Pulse shape level?: (most commonly 0 to 40 meters)
Gather Telephone Information (Analog)
Needed for:
MVP810
MVP410
MVP210
Analog Phone Parameters
9
Ask phone company or
telecom manager.
MVP130
MVP130FXS
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 a
phone, fax, or KTS (key telephone system)
• If FXO, determine if line will be an analog PBX
extension or an analog line from a telco central office
• If E&M, determine these aspects of the E&M trunk
line 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?
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MultiVOIP User Guide
Quick Start Instructions
Gather Telephone Information (ISDN BRI)
ISDN-BRI Phone Parameters
Needed for:
MVP810ST
MVP410ST
9
Ask phone company or
telecom manager.
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 and
which BRI ports will be terminal side.
• If you are connecting the MultiVOIP to network
equipment with a “U” interface, an NT1 device must
be connected between them.
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Quick Start Instructions
MultiVOIP User Guide
Phone/IP Details Often Needed/Wanted
Obtain Email Address for VOIP (for email call log reporting)
required if log reports of
VOIP call traffic
Optional
are to be sent by email
SMTP Parameters
Preparation Task:
To: I.T. Department
Ask Mail Server
re: email account for VOIP
administrator to set up
email account (with
password) for the
MultiVOIP unit itself.
Be sure to give a unique
identifier to each
individual MultiVOIP
unit.
Get the IP address of the
mail server computer, as
well.
Identify Remote VOIP Site to Call
When you’re done installing the MultiVOIP, you’ll want to confirm that
it is configured and operating properly. To do so, it’s good to have
another voip that you can call for testing purposes. You’ll want to
confirm end-to-end connectivity. 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 the
installation of this MultiVOIP with an installation of another unit at a
remote site.
Identify VOIP Protocol to be Used
Will 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 highly desirable to use the same VOIP protocol for
all VOIP units in the system. SPP 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.
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MultiVOIP User Guide
Quick Start Instructions
Placement
Mount your MultiVOIP in a safe and convenient location where cables
for your network and phone system are accessible. Rack-mounting
instructions are 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
the MultiVOIP are not especially demanding. Still, we recommend
using a reasonably new computer. The computer that you use to
configure your MultiVOIP need not be dedicated to the MultiVOIP
after installation is complete.
COM port on controller PC. You’ll need an available COM port on the
controller PC. You’ll need to know which COM port is available for use
with the MultiVOIP (COM1, COM2, etc.).
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Quick Start Instructions
MultiVOIP User Guide
Quick Hookups
Hookup for MVP2410 & MVP3010
T1/E1 MultiVOIP Hookup
(MVP-2410/3010)
Grounding
Screw
Cabling to your IP network.
RJ-45 connector.
Cabling to computer running
MultiVOIP software.
RJ-45 to serial connector (DB9).
T1/E1/PRI cabling to your PBX,
and/or to the PSTN.
RJ-45 connector.
Digital Voice
Trunk
Command
Ethernet
10 /100
Command
Modem
l
RS-232
O
Cabling to POTS line
(RJ-11 connector) for remote
configuration via the built-in
modem.
Power Cable
Receptacle
On/Off Switch
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MultiVOIP User Guide
Quick Start Instructions
Hookup for MVP410 & MVP810
Analog MultiVOIP Hookup
MVP-410/810
Power Cable
Receptacle
Cabling to computer running
MultiVOIP software.
Connector at MultiVOIP: DB-25.
Connector at computer: DB-9.
MVP810 has 8 connector pairs.
Grounding
Screw:
Connect to
Earth Ground
MVP410 has 4 connector pairs.
Only 1 connector of any pair is
used at a time.
FXS/FXO
FXS/FXO
E&M
E&M
E&M FXS/FXO
E&M FXS/FXO
Command
E&M FXS/FXO
Modem
Ethernet
Command
FXS/FXO
E&M
FXS/FXO
E&M
E&M FXS/FXO
On/Off
Switch
Cabling to phone equipment.
Cabling to your IP network.
RJ-45 connector.
E&M
(RJ-45 connector):
connects to E&M trunk line
from PBX or telco office.
Cabling to POTS line,
for remote configuration.
RJ-11 connector.
FXS
(RJ-11 connector):
FXO
(RJ-11 connector):
connects to analog phone line
or analog PBX extension.
connects to phone, fax,
or key phone system.
DID-DPO
(RJ-11 connector):
connects to DID line
from telco office. DPO functionality
not yet supported.
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Quick Start Instructions
MultiVOIP User Guide
Hookup for MVP410ST & MVP810ST
ISDN MultiVOIP Hookup
MVP-410ST/810ST
Power Cable
Receptacle
Cabling to computer running
MultiVOIP software.
Grounding
Screw:
Connect to
Earth Ground
MVP810ST has 4 ISDN connectors.
MVP410ST has 2 ISDN connectors.
Connector at MultiVOIP: DB-25.
Connector at computer: DB-9.
ISDN1
ISDN2
ISDN4
ISDN3
Command
On/Off
Switch
Cabling to phone equipment.
ISDNn (RJ-45 connector):
connects to ISDN BRI line
from PBX or telco office.
Or connects to ISDN phone
or terminal adapter.
NT1 Device required between
voip interface (ports ISDN1 - ISDN4)
and network equipment with
“U” interface. Not needed for
connection to network equipment
with “S/T” interface.
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MultiVOIP User Guide
Quick Start Instructions
Hookup for MVP210
CH1
CH2
ETHERNET
FXS/FXO
E&M FXS/FXO E&M
RS232
POWER
10/100 COMMAND
10BASET
COMMAND PORT
POWER
Voice/Fax Channel 1 - 2
Connections
E&M FXO/FXS
Power Connection
GND
FXS
E&M
FXO
Command Port Connection
Ethernet Connection
PSTN
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Quick Start Instructions
MultiVOIP User Guide
Hookup for MVP130
Power
Ethernet Command FXS/FXO
Power Connection
FXO
FXS
PBX
ꢀ
Command Port Connection
Telephony Connection
PSTN
Hub
Network Connection
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MultiVOIP User Guide
Quick Start Instructions
Hookup for MVP130FXS
Power
Ethernet Co mmand
VOICE
FX S
Power Connection
Telephony Connection
ꢀ
Command Port Connection
Hub
Network Connection
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Quick Start Instructions
MultiVOIP User Guide
Load MultiVOIP Control Software onto PC
For 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 to
My 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 and click 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.
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MultiVOIP User Guide
Quick Start Instructions
Phone/IP Starter Configuration
Full details here:
Chapter 5: Technical Configuration for
Digital T1/E1 MultiVOIPs
MVP2410
MVP3010
in User Guide.
MVP130
MVP130FXS
MVP210
MVP410
MVP810
Chapter 6: Technical Configuration for
Analog/BRI MultiVOIPs
in 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
web browser GUI? (It has the same functionality as the local
Windows GUI, but offers remote access.)
If NO, skip to step 5.
If YES, continue with step 4.
4. Enable Web Browser GUI (Optional). To do configuration and
operation procedures 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 or above; or Netscape 6.0 or above.)
A. Be sure an IP address has
been assigned to the
E. Open web browser.
(Note: The PC being used must
be connected to and have an IP
address on the same IP network
that the voip is on.)
MultiVOIP unit (this must be
done in the MultiVOIP
Windows GUI).
B. Save Setup in Windows GUI.
F. Browse to IP address of
MultiVOIP unit.
C. Close the MultiVOIP
G. If username and password
have been established, enter
them when prompted by
voip.
Windows GUI.
D. Install Java program from
MultiVOIP product CD.
(Must be Java Runtime
H. Use web browser GUI to
configure or operate voip.
Environment 1.4.2_01 or above.)
NOTE: Required on first use of
Web Browser GUI only.
Need more See “Web Browser Interface” in Operation &
info?
Maintenance chapter of User Guide (on CD).
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Quick Start Instructions
MultiVOIP User Guide
Once you’ve begun using the web browser GUI, you can go back to
the MultiVOIP Windows GUI at any time. However, you must log
out of the web browser GUI before using the MultiVOIP Windows
GUI.
5. Go to Configuration | Voice/Fax. Select Coder | “Automatic.” At
the right-hand side of the dialog box, click Default. If you know any
specific parameter values that will apply to your system, enter them.
Click Copy Channel. Select Copy to All. Click Copy. At main
Voice/Fax Parameters screen, click OK to exit from the dialog box.
6. Enter telephone system information.
Analog MultiVOIPs
MVP-130/130FXS,
MVP-210/410/810
Digital MultiVOIPs
MVP-2410/3010
Go to
Go to
Configuration | Interface.
Enter parameters obtained
from phone company or
PBX administrator.
Configuration | T1/E1/ISDN.
Enter parameters obtained
from phone company or
PBX administrator.
ISDN-BRI MultiVOIPs
MVP-410ST/810ST
Go to Configuration | ISDN BRI.
Enter parameters obtained from phone company
or PBX administrator.
If the voip is connected to BRI extensions of a PBX
or a phone company, then select "Terminal"
in the ISDN BRI Parameters screen.
If the voip is connected to ISDN terminal adapters
and/or ISDN phones, then select "Network"
in the ISDN BRI Parameters screen.
7. Go to Configuration | Regional Parameters. Select the
Country/Region that fits your situation. Click Default and confirm.
8. Go to Configuration | Regional Parameters. In the Country
Selection for Built-In Modem field (drop-down list), select the
country that best fits your situation. (This may not be the same as
your selection for the Country/Region field. The selections in the
Country Selection for Built-In Modem field entail more detailed
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MultiVOIP User Guide
Quick Start Instructions
groupings of telephony parameters than do the Country/Region
values.)
Click OK to exit from the Regional Parameters dialog box.
9. Do you want the phone-call logs produced by the MultiVOIP to be
sent out by 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
by email. 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 the MultiVOIP 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 email
account is located in the “Mail Server IP Address” field.
Typically the email log reports are sent to the Voip Administrator
but they can be sent to any email address. Decide where you want
the email logs sent and enter that email address in the “Recipient
Address” field.
Whenever email log messages are sent out, they must have a
standard Subject line. Something like “Phone Logs for Voip N” is
useful. If you have more than one MultiVoip unit in the building,
you’ll need a unique identifier 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 Voip
Administrator.
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 a SysLog client program. For more info, see the “SysLog
Server Functions” section of the Operation & Maintenance chapter of
the User Guide.
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Quick Start Instructions
MultiVOIP User Guide
Phone/IP Starter Configuration (continued)
12. Enable premium (H.450) telephony features.
Go to Supplementary Services. Select any features to be used.
For Call Hold, Call Transfer, & Call Waiting, specify the key sequence
that the phone user will press to invoke the feature. For Call Name
Identification, specify the allowed name types to be used and a caller-
id descriptor.
If Call Forwarding is to be used, enable this feature in the
Add/Edit Inbound Phone Book screen.
After making changes, click on OK in the current configuration
screen before moving on to the next configuration screen.
13. Go to Save Setup | Save and Reboot. Click OK. This will save the
parameter values that you have just entered.
The MultiVOIP’s “BOOT” LED will light up while the configuration
file is being saved and loaded into the MultiVOIP. Don’t do anything
to the MultiVOIP until the “BOOT “LED is off (a loss of power at this
point could cause the MultiVOIP unit to lose the configuration
settings you have made).
END OF PROCEDURE.
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MultiVOIP User Guide
Quick Start Instructions
Phonebook Starter Configuration (with remote voip)
If the topic of voip phone books is new to you, it may be helpful to read
the PhoneBook Tips section (page 71) before starting this procedure.
To do this part of the quick setup, you need to know of another voip
that you can call to conduct a test. It should be at a remote location,
typically somewhere outside of your building. You must know the
phone number and IP address for that site. We are assuming here that
the MultiVOIP will operate in conjunction with a PBX.
You must configure both the Outbound Phonebook and the Inbound
Phonebook. A starter configuration only means that two voip locations
will be set up to begin the system and establish voip communication.
Outbound Phonebook
1. 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
(area code, 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 Call
Example
Technician in Seattle (area
206) must set up one voip
there, another in Chicago
(area 312, downtown).
Technician in central
London (area 0207) to set
up voip there, another in
Birmingham (area 0121).
Answer: Write down 312.
Answer: write down 0121.
Euro, International Call Example
Technician in Rotterdam (country 31; city 010) to
set up one voip there, another in Bordeaux
(country 33; area 05).
Answer: write down 3305.
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MultiVOIP User Guide
4. Suppose you want to call a phone number outside of your building
using a phone station that is an extension from your PBX system (if
present). What digits must you dial? Often a “9” or “8” must be
dialed to “get an outside line” 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 can dial a remote area code.
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
Seattle voip works with
PBX that uses “8” for all
voip calls. “1” must
immediately precede area
code of dialed number.
London voip works with
PBX that uses “9” for all
out-of-building calls
whether by voip or by
PSTN. “0” must
immediately precede area
code of dialed number.
Answer: write down 81.
Answer: write down 90.
Euro, International Call Example
Rotterdam/Bordeaux system.
Rotterdam voip works with PBX where “9” is
used for all out-of-building calls. “0” must
precede all international calls.
Answer: write down 90.
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Quick Start Instructions
5. In the “Destination Pattern” field of the Add/Edit Outbound
Phonebook screen, enter the digits from step 4 followed by the digits
from step 3.
North America,
Long-Distance Example
Seattle-Chicago system.
Euro, National Call
Example
London/Birming. system.
Answer: enter 81312 as
Destination Pat-
tern in Outbound
Phone book of
Leading zero of
Birmingham area code is
dropped when combined
with national-dialing
access code. (Such
Seattle voip.
practices vary by country.)
Answer: enter 90121 as
Destination Pat-
tern in Outbound
Phonebook of
London voip.
Not 900121.
Euro, International Call Example
Rotterdam/Bordeaux system.
Answer: enter 903305 as Destination Pattern in
Outbound Phonebook of Rotterdam voip.
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Quick Start Instructions
MultiVOIP User Guide
6. Tally up the number of digits that must be dialed to reach the remote
voip site (including prefix digits of all types). Enter this number in
the “Total Digits” field.
North America,
Long-Distance Example
Seattle-Chicago system.
Euro, National Call
Example
London/Birming. system.
To complete Seattle-to-
To complete London-to-
Birmingham call, 90121 must
be followed by the 7-digit
local phone number in
Birmingham.
Chicago call, 81312 must be
followed by the 7-digit local
phone number in Chicago.
Answer: enter 12 as number
of Total Digits in
Outbound Phone
book of Seattle
Answer: enter 12 as number
of Total Digits in
Outbound Phone
book of London
voip.
voip.
Euro, International Call Example
Rotterdam/Bordeaux system.
To complete Rotterdam-to-Bordeaux call, 903305 must
be followed by 8-digit local phone number in Bordeaux.
Answer: enter 14 as number of Total Digits in
Outbound Phonebook of Rotterdam voip.
7. In the “Remove Prefix” field, enter the initial PBX access digit
(“8” or “9”).
North America,
Long-Distance Example
Seattle-Chicago system.
Euro, National Call
Example
London/Birming. system.
Answer: enter 8 in “Remove
Prefix” field of
Answer: enter 9 in “Remove
Prefix” field of
Seattle Outbound
Phonebook.
London Outbound
Phonebook.
Euro, International Call Example
Rotterdam/Bordeaux system.
Answer: enter 9 in “Remove Prefix” field of Outbound
Phonebook for Rotterdam voip.
Some PBXs will not ‘hand off’ the “8” or “9” to the voip. But for those PBX
units that do, it’s important to enter the “8” or “9” in the “Remove Prefix”
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field in the Outbound Phonebook. This precludes the problem of having to
make two inbound phonebook entries at remote voips, one to account for
situations where “8” is used as the 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 Phonebook
1. 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 are required to reach your local site from the remote voip
location (think of it as though the call were being made through the
PSTN – even though it will not be).
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
Seattle is area 206. Chicago
employees must dial 81
before dialing any Seattle
number on the voip system.
Inner London is 0207 area.
Birmingham employees must
dial 9 before dialing any
London number on the voip
system.
Answer: 1206 is prefix to be
removed by local
Answer: 0207 is prefix to be
removed by local
(Seattle) voip.
(London) voip.
Euro, International Call Example
Rotterdam/Bordeaux system.
Rotterdam is country code 31, city code 010. Bordeaux
employees must dial 903110 before dialing any
Rotterdam number on the voip system.
Answer: 03110 is prefix to be removed by local
(Rotterdam) voip.
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4. In the “Add Prefix” field, enter any digits that must be dialed from
your local voip to gain access to the PSTN.
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
On Seattle PBX, “8” is used to
get an outside line.
On London PBX, “9” is used
to get an outside line.
Answer: 8 is the prefix to be
added by local
Answer: 9 is the prefix to be
added by local
(Seattle) voip.
(London) voip.
Euro, International Call Example
Rotterdam/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 unit will assign the call to an available channel. If desired,
specific channels can be assigned to specific incoming calls (i.e., to
any set of calls received with a particular incoming dialing pattern).
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6. In the “Description” field, it is useful to describe the ultimate
destination of the calls. For example, in a New York City voip
system, “incoming calls to Manhattan office,” might describe a
phonebook entry, as might the descriptor “incoming calls to NYC
local calling area.” The description should make the routing of calls
easy to understand. (40 characters max.)
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
Possible Description:.
Free Seattle access, all
employees
Possible Description:.
Local-rate London access,
all employees
Euro, International Call Example
Rotterdam/Bordeaux system.
Possible Description:. Local-rate Rotterdam access, all
employees
7. Repeat steps 2-6 for each inbound phonebook entry. When all entries
are complete, 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.
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Phonebook Tips
Preparing 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 not
generally enter complete phone numbers in the voip phonebook.
Instead, you’ll enter “destination patterns” that involve area codes and
other digits. If the destination pattern is a whole area code, you’ll be
assigning all calls to that area code to go to a particular voip that has a
unique IP address. If your destination pattern includes an area code
plus a particular local phone exchange number, 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 in your
destination pattern, you are asking the voip to handle calls to more
destinations.
2. The Four Types of Phonebook Digits Used. Important!
“Destination patterns” to be entered in your phonebook will generally
consist of:
(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 to remember that all the digits in any phonebook entry must be
of one of these four 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 when making non-local calls. They always precede the phone
number that would be dialed when making a local call.
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(b) access codes. There are digits (PSTN access codes) that must be
dialed to gain 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
connected to PBX systems or key systems. Often a “9” must be dialed
on a PBX phone to gain access to the PSTN (‘to get an outside line’).
Sometimes “8” must be dialed 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 voip system without a special access
code (so that “9” might still be used for all calls outside of the building).
There are also digits (special access codes) that must be dialed to gain
access to a particular discount long-distance carrier or to some other
closed or proprietary telephone system.
(c) local exchange numbers. Within any calling area there will be many
local exchange numbers. A single exchange may be used for an entire
small town. 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 also commonly assigned exchange
numbers for their exclusive use. In some cases, these organizational-
assigned exchanges can become non-localized because the exchange is
assigned to one facility and linked, by the organization’s private
network, to other sometimes distant locations.
(d) specialized codes. Some proprietary voip units assign, to sites and
phone stations, numbers that are not compatible with PSTN
numbering. This can also occur in PBX or key systems. These
specialized numbers must be handled on a case-by-case basis.
3. Knowing When to Drop Digits.
Example
When calling area codes and
access codes are used in
combination, a leading “1” or “0”
must sometimes be dropped.
Area code for Inner London is
listed as “0207.” However, in
international calls the leading
“0” is dropped.
U.K.
Country
Code
Phonebook Entry Æ
International
Access Code
Leading Zero
Dropped from
Area Code
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4. Using a Comma.
Detail
Commas are used in telephone
dialing strings to indicate a pause
to allow a dial tone to appear
(common on PBX and key
systems). Commas may be used
only in the “Add Prefix” field of
the Inbound Phonebook.
,
= 1-second pause
In many PBX systems
(not needed in all)
5. Ease of Use. The phonebook setup determines how easy the voip
system is to use. Generally, you’ll want to make it so dialing a voip call
is very similar to dialing any other number (on the PSTN or through the
PBX).
6. Avoid Unintentional Calls to Official/Emergency Numbers. Dialing a
voip call will typically be somewhat different than ordinary dialing.
Because of this, it’s possible to set up situations, quite unwittingly,
where phone users may be predisposed to call official numbers without
intending to do so. Conversely, a voip/PBX system might also make it
difficult to place an official/emergency call when one intends to do so.
Study your phonebook setup and do some dialing on the system to
avoid these pitfalls.
7. Inbound/Outbound Pattern Matching. In general, the Inbound
Phonebook entries of the local voip unit will match the Outbound
Phonebook entries of the remote voip unit. Similarly, the Outbound
Phonebook entries of the local voip unit will match the Inbound
Phonebook entries of the remote voip unit. There will often be non-
matching entries, but it’s nonetheless useful to notice the matching
between the phonebooks.
8. Simulating Network in-lab/on-benchtop. One common method of
configuring a voip network is to set up a local IP network in a lab,
connect voip units to it, and perhaps have phones connected on channel
banks to make test calls.
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Phonebook Example
One Common Situation
Boise Office
Area: 208
PBX System.
Voip Example
. This company has offices in three
different 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 calls
without using a special access digit (“8” is used in
some systems). Finally, the system operates so that
employees in any office can dial employees in any
other office using only three digits. Here are the
phonebooks needed for that system.
Main Number:
333-2700
PSTN
90 extensions
204.16.49.73
24-Channel
Digital VoIP
(MVP2410)
Inbound Phonebook
Each Inbound Phonebook contains
two entries. The first entry (4 digits)
specifies how incoming calls from the
other voip sites will be handled if
they go out onto the local PSTN.
Essentially, all those calls come to the
receiving voip with a pattern
beginning with 1+area code. The local
voip removes those four digits
because they aren’t needed when
dialing locally. The local voip
attaches a “9” at the beginning of the
number to get an outside line. The
PBX then completes the call to the
PSTN.
Santa Fe Office
Area: 505
204.16.49.74
8-Channel
Analog VoIP
(MVP810)
IP
Network
PBX System.
Main Number:
444-3200
40 extensions
Inbound
The second
Phonebook entry
(1 digit) is for receiving calls from
company employees in the other two
cities. The out-of-town employee
simply dials 3 digits. The first of the
three digits is uniquely used at each
site and so acts as a destination
pattern (Boise extensions are 7xx,
Santa Fe extensions 2xx, Flagstaff
extensions 6xx).
PSTN
The local voip sees the pattern in its
inbound phone book and notes the
first digit (here either 2, 5, or 6).
To make the match, this first digit,
2, 5, or 6 is put in the “Remove Prefix”
field. This first digit must then be
added back once again so that the
voip will send all three digits to the
PBX. The PBX can then dial the
specific extension identified by the
three-digit number.
Each Outbound Phonebook contains two
pairs of entries, two entries for each
remote site. Whenever an out-of-town
employee dials a 12-digit number
beginning with the listed 5-digit
destination pattern (9+1+area code) of
another company location, the PBX
hands the call to the voip system. The
local voip strips off the “9” and directs
the call to the IP address of the remote
voip. The remote voip receives the call
and hands it to its PBX. The PBX then
completes the call to the PSTN.
Flagstaff Office
Area: 520
The one-digit Outbound destination
patterns pertain to 3-digit calling
between company employees.
204.16.49.75
8-Channel
Analog VoIP
(MVP810)
PBX System.
Main Number:
777-5600
PSTN
30 extensions
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Voip Sites with Phonebooks
Boise Voip
Inbound Phonebook
Boise Voip
Outbound Phonebook
Boise Office
Area: 208
Prefix Description
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Remove
91208
Destin.
Pattern
PBX System.
Main Number:
333-2700
to Add Incoming Calls
204.
16.49.
74
9,
7
Incoming calls 91505
to PSTN,
12
none
none
none
none
Outgoing calls
to Santa Fe
area
3-digit calls to
Santa Fe
employees
(extensions
200 to 240)
Outgoing calls
PSTN
Boise Area
204.
16.49.
74
7
i ncoming calls
to extensions
of company’s
PBX system
in Boise
2
3
90 extensions
204.16.49.73
91520
6
12
3
none
none
none
none
204.
24-Channel
Digital VoIP
(MVP2410)
16.49. to Flagstaff
75
204.
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
IP
Network
Santa Fe Office
Area: 505
Santa Fe Voip
Inbound Phonebook
Santa Fe Voip
Outbound Phonebook
204.16.49.74
Prefix Description
to Add Incoming Calls
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Remove
91505
Destin.
Pattern
8-Channel
Analog VoIP
(MVP810)
9,
2
Incoming calls 91208
to PSTN,
Santa Fe local
calls
Incoming calls
to extensions
of company’s
PBX system
in Santa Fe
12
none
none
204.
Outgoing calls
16.49. to Boise area
73
2
7
3
none
none
204.
3-digit calls to
PBX System.
Main Number:
444-3200
16.49. Boise
73
employees
(extensions
700-790)
91520
6
12
3
none
none
none
none
204.
Outgoing calls
16.49. to Flagstaff
75
204.
40 extensions
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
PSTN
Flagstaff Voip
Inbound Phonebook
Flagstaff Voip
Outbound Phonebook
Flagstaff Office
Area: 520
Prefix Description
to Add Incoming Calls
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
204.16
.49.74
91520
9
Incoming calls 91505
to PSTN,
Flagstaff local
12
none
none
none
none
Outgoing calls
to Santa Fe
area
204.16.49.75
8-Channel
Analog VoIP
(MVP810)
calls
204.16
.49.74
6
6
Incoming calls
to extensions
of company’s
PBX system
in Flagstaff
2
3
3-digit calls to
Santa Fe
employees
(extensions
200-240)
Outgoing calls
to Boise area
3-digit calls to
Boise
PBX System.
Main Number:
777-5600
204.16
.49.73
91208
7
12
3
none
none
none
none
PSTN
204.16
.49.73
employees
(extensions
700-790)
30 extensions
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Sample Phonebooks Enlarged
Boise Voip
Boise Voip
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
91208
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
204.
16.49.
74
9,
7
Incoming calls 91505
to PSTN,
12
none
none
none
none
Outgoing calls
to Santa Fe
area
3-digit calls to
Santa Fe
employees
(extensions
200 to 240)
Outgoing calls
Boise Area
204.
16.49.
74
7
i ncoming calls
to extensions
of company’s
PBX system
in Boise
2
3
91520
6
12
3
none
none
none
none
204.
16.49. to Flagstaff
75
204.
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
Santa Fe Voip
Santa Fe Voip
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
91505
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
9,
2
Incoming calls 91208
to PSTN,
12
none
none
none
none
204.
Outgoing calls
16.49. to Boise area
73
Santa Fe local
calls
2
Incoming calls
to extensions
of company’s
PBX system
in Santa Fe
7
3
204.
3-digit calls to
16.49. Boise
73
employees
(extensions
700-790)
91520
6
12
3
none
none
none
none
204.
Outgoing calls
16.49. to Flagstaff
75
204.
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
Flagstaff Voip
Inbound Phonebook
Flagstaff Voip
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Destin.
Pattern
to Add Incoming Calls
Remove
204.16
.49.74
91520
9
Incoming calls 91505
to PSTN,
Flagstaff local
calls
12
none
none
none
none
Outgoing calls
to Santa Fe
area
204.16
.49.74
6
6
Incoming calls
to extensions
of company’s
PBX system
in Flagstaff
2
3
3-digit calls to
Santa Fe
employees
(extensions
200-240)
204.16
.49.73
91208
7
12
3
none
none
none
none
Outgoing calls
to Boise area
3-digit calls to
Boise
204.16
.49.73
employees
(extensions
700-790)
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Phonebook Worksheet
Voip Location/ID:____________________________
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
Other Details:
Voip Location/ID:____________________________
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
Other Details:
Voip Location/ID:____________________________
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
Other Details:
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Enlarged Phonebook Worksheet
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Connectivity Test
The procedures “Phone/IP Starter Configuration” and “Phonebook
Starter Configuration” must be completed before you can do this
procedure.
1. These connections must be made:
Connections
for digital MultiVOIPs
(MVP-2410/3010)
for analog MultiVOIPs
(MVP-130/130FXS
& MVP-210/410/810)
MultiVOIP to local PBX
MultiVOIP to local phone
station
–OR--
MultiVOIP to extension of
key phone system
MultiVOIP to command PC
MultiVOIP to Internet
MultiVOIP to command PC
MultiVOIP to Internet
2. Inbound Phonebook and Outbound Phonebook must both be set up
with at least one entry in each. These entries must allow for
connection between two 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 “Console Messages” checkbox.
4. You now need to free up the COM port connection (currently being
used by the MultiVOIP program) so that the HyperTerminal program
can use it. To do this, you can either (a) click on Connection in the
sidebar and select “Disconnect” from the drop-down box, or (b) close
down the MultiVOIP program altogether.
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5. Open the HyperTerminal program.
6. Use HyperTerminal to receive and record console messages from the
MultiVOIP unit. To do so, set up HyperTerminal as follows (setup
shown is for Windows NT4; details will differ slightly in other MS
operating systems):
y In the upper toolbar of the HyperTerminal screen, click on
the Properties button.
y In the “Connect To” tab of the Connection Properties
dialog box, click on the Configure button.
y In the next dialog box, on the “General” tab, set “Maximum
Speed” to 115200 bps.
y On the “Connection” tab, set connection preferences to:
Data bits: 8
Parity:
none
Stop bits: 1
y Click OK twice to exit settings dialog boxes.
7. Make VOIP call.
for digital MultiVOIPs
(MVP-2410/3010)
for analog MultiVOIPs
(MVP-130/130FXS
& MVP-210/410/810)
Make call from an extension Make call on a local phone
of the local PBX.
line accessing PSTN directly
or through key system
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8. Read console messages recorded on HyperTerminal.
Console Messages from Originating VOIP. The voip unit that
originates the 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:Mounds
View,TEL:17637175662,17637175662
[00033290] H323IF[0]:srcAddr = NAME:New
York,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
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Console Messages from Terminating VOIP. The voip unit connected
to the phone where the call is answered will send back messages like
that shown below.
[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
has been 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
the Troubleshooting procedure below.
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Troubleshooting
If you cannot establish connectivity between two voips in the system,
follow the 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.
y
y
Check cabling. Are connections well seated? To correct receptacle?
Is the ONL LED on?
(If on, ONL indicates that the MultiVOIP is online on the
network.)
y
Are T1/E1/PRI Parameter settings correct?
B. For MVP-130/130FXS, MVP210, MVP410, or MVP810.
y
y
Check cabling. Are connections well seated? To correct receptacle?
Are telephone Interface Parameter settings correct?
C. For MVP410ST or MVP810ST.
y
Check cabling. Are connections well seated? To correct
receptacle?
y
If terminal equipment is connected to the voip, then "Network"
should be selected for that BRI interface in the ISDN BRI
Parameters screen.
Note: Each BRI interface is separately configurable.
If network equipment such as an ISDN BRI PBX or an
ISDN BRI line from a phone company is connected to the voip,
then "Terminal" should be selected for that BRI interface in the
ISDN BRI Parameters screen.
y
y
Was the proper country and operator chosen?
Was the proper type of line coding (A-law or u-law) chosen?
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3. Verify phonebook configuration.
4. Observe console messages while placing a call. Look for error
messages indicating phonebook problems, network problems, voice-
coder mismatches, etc.
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Mechanical Installation & Cabling
Introduction
The MultiVOIP models MVP130, MVP130FXS and MVP210 are tabletop
units and can be handled easily by one person. However, the MVP410,
MVP810, MVP2410, and MVP3010 are somewhat heavier units. When
these units are to be installed into a rack, two able-bodied persons
should participate.
Please read the safety notices before beginning installation.
Safety Warnings
Lithium Battery Caution
A lithium battery on the voice/fax channel board provides backup
power for the timekeeping capability. The battery has an estimated life
expectancy of ten years.
When the battery starts to weaken, the date and time may be incorrect.
If the battery fails, the board must be sent back to Multi-Tech Systems
for battery replacement.
Warning: There is danger of explosion if the battery is incorrectly
replaced.
Safety Warnings Telecom
1. Never install telephone wiring during a lightning storm.
2. Never install a telephone jack in wet locations unless the jack is
specifically designed 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
telephone line has been disconnected at the network interface.
5. Use caution when installing or modifying telephone lines.
6. Avoid using a telephone (other than a cordless type) during an
electrical storm. There may be a remote 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 larger
telecommunication line cord.
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UnpackingYour MultiVOIP
When unpacking your MultiVOIP, check to see that all of the items
shown are included in the box. For the various MultiVOIP models, the
contents of the box will be different. Study the particular illustration
below that is appropriate to the model you have purchased. If any box
contents are missing, contact MultiTech Tech Support at 1-800-972-2439.
Unpacking the MVP2410/3010
Figure 3-1: Unpacking the MVP2410/3010
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Unpacking the MVP-410/810
Voice/Fax over IP Networks
Quick Start
Guide
Voice/Fax
RCV XSG
5
1
Voice/Fax
RCV XSG
6
2
Voice/Fax
RCV XS
7
3
Voice/Fax
RCV XSG
8
4
XMT
RSG
RSG
XMT
XMT
RSG
RSG
XMT
XMT
G
RSG
RSG
XMT
XMT
RSG
RSG
Ethernet
XMT COL
Power
Boot
Voice/Fax
RCV XSG
Voice/Fax
RCV XSG
Voice/Fax
RCV XS
Voice/Fax
RCV XSG
RCV
LNK
XMT
G
Figure 3-2: Unpacking the MVP-410/810
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Rack Mounting Instructions for
MVP-2410/3010 & MVP-410/810
The MultiVOIPs can be mounted in an industry-standard EIA 19-inch
rack enclosure, as shown in Figure 3-5.
Figure 3-5: Rack-Mounting (MVP2410/3010 or MVP410/810)
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Safety Recommendations for Rack Installations
Ensure proper installation of the unit in a closed or multi-unit enclosure
by following the recommended installation as defined by the enclosure
manufacturer. Do not place the unit directly on top of other equipment
or place other equipment directly on top of the unit. If installing the
unit in a closed or multi-unit enclosure, ensure adequate airflow within
the rack so that the maximum recommended ambient temperature is
not exceeded. Ensure that the unit is properly connected to earth
ground by verifying that it is reliably grounded when mounted within
a rack. If a power strip is used, ensure that the power strip provides
adequate grounding of the attached apparatus.
When mounting the equipment in the rack, make sure mechanical
loading is even to avoid a hazardous condition, such as loading heavy
equipment in rack unevenly. The rack used should safely support the
combined weight of all the equipment it supports.
Ensure that the mains supply circuit is capable of handling the load of
the equipment. See the power label on the equipment for load
requirements (full specifications for MultiVOIP models are presented in
chapter 1 of this manual).
Maximum ambient temperature for the unit is 60 degrees Celsius (140
degrees Fahrenheit) at 20-90% non-condensing relative humidity. This
equipment should only be installed by properly qualified service
personnel. Only connect like circuits. In other words, connect SELV
(Secondary Extra Low Voltage) circuits to SELV circuits and TN
(Telecommunications Network) circuits to TN circuits.
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19-Inch Rack Enclosure Mounting Procedure
Attaching the MultiVOIP to a rack-rail of an EIA 19-inch rack enclosure
will certainly require two persons. Essentially, the technicians must
attach the brackets to the MultiVOIP chassis with the screws provided,
as shown in Figure 3-6, and then secure unit to rack rails by the
brackets, as shown in Figure 3-7. Because equipment racks vary, screws
for rack-rail mounting are not provided. Follow the instructions of the
rack manufacturer and use screws that fit.
1. Position the right rack-mounting bracket on the MultiVOIP
using the 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-410/810)
Figure 3-7: Attaching MultiVOIP to Rack Rail
(MVP-2410/3010 & MVP-410/810)
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Cabling
Cabling Procedure for MVP2410/3010
Cabling 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 the associated cable connections. The following
procedure details the steps necessary for cabling your MultiVOIP.
1. Connect the power cord to a live AC outlet, then connect it to the
MultiVOIP’s power receptacle shown at top right in Figure 3-8.
DIGITAL VOICE
ETHERNET COMMAND
TRUNK
10 BASET
RS232
DIGITAL VOICE
ETHERNET COMMAND
COMMAND
MODEM
T1
ꢀ
Command Port Connection
PBX
Hub
PSTN
Network Connection
Telephony Connection
Figure 3-8. Cabling for MVP2410/3010
2. Connect the MultiVOIP to the PC (the computer that will hold the
MultiVOIP software) using the RJ-45 to DB9 (female) cable provided
with your unit. Plug the RJ-45 end of the cable into the Command
port of the MultiVOIP and connect the other end (the DB9 connector)
to the PC serial port you are using (typically COM1 or COM2). See
Figure 3-8.
3. Connect a network cable to the Ethernet connector on the back of the
MultiVOIP. Connect the other end of the cable to your network.
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4. If you intend to configure the MultiVOIP remotely using the
MultiVOIP Windows GUI, connect an RJ-11 phone cable between the
Command Modem connector (at the rear of the MultiVOIP) and a
receptacle served by a telco POTS line. See Figure 3-9.
The Command Modem is built into the MultiVOIP unit. To configure
the MultiVOIP remotely using its Windows GUI, you must call into
the MultiVOIP’s Command Modem. Once a connection is made, the
configuration process is identical to local configuration with the
Windows GUI.
DIGITAL VOICE
ETHERNET COMMAND
10 B ASET
TRUNK
RS232
DIGITAL VOICE
ETHERNET COMMAND
COMMAND
MODEM
Grounding Screw
Telco POTS Line
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 it is reliably grounded when mounted within a rack.
This can be accomplished by connecting a grounding wire between
the chassis grounding screw (see Figure 3-9) and a metallic object that
will provide an electrical ground.
6. Turn on power to the MultiVOIP by setting the power switch on the
right side panel to the ON position. Wait for the Boot LED on the
MultiVOIP to go off before proceeding. This may take a couple of
minutes.
Proceed to Chapter 4 “Software Installation.”
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Cabling Procedure for MVP-410/810
Cabling involves connecting the MultiVOIP to your LAN and telephone
equipment.
1. For DID channels only. If all channels of your MultiVOIP will be
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
must change the jumper on the MultiVOIP circuit card.
a. Disconnect power. Unplug the AC power cord from the wall outlet
or from the receptacle on the MultiVOIP unit.
b. Using a #1 Phillips driver, remove the three screws (at back of unit)
that attach the main circuit card to the chassis of the MultiVOIP.
Screws (3) holding circuit card assembly to chassis.
x
x
MVP410/810
rear panel
x
Figure 3-10. MVP-410/810 Rear Screw Locations
c. Pull the main circuit card out about 5 inches (the power
connection to the board prevents it from being removed entirely
from the chassis).
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d. Identify the channels on which the DID interface will be used.
Jumper Configurations
(enlarged)
Upper Circuit Card
MVP810 only
For DID
Interface type
Ch 7
Ch 6
Ch 8
Ch 5
U10
U7
U9
U8
Jumpers 5-8
For non-DID
Interface type
}
Main Circuit Card
MVP-410/810
Generality:
For channels using the DID
interface, the jumper must
not straddle across the
Ch 1
Ch 2
Ch 3
Ch 4
cross-hatched area between
the jumper posts.
}
For channels using any non-DID
interface, it is acceptable that the
jumper straddles across the
cross-hatched area between
the jumper posts.
Jumpers 1-4
Figure 3-11. MVP-410/810 Channel Jumper Settings
e. Position the jumper for each DID channel so that it does not connect
the two jumper posts. For DID operation of a voip channel, the
MultiVOIP will work properly if you simply remove the jumper
altogether, but that is inadviseable because the jumper might be
needed later if a different telephony interface is used for that voip
channel.
f. Slide the main circuit card back into the MultiVOIP chassis and
replace the three screws.
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2. Connect the power cord supplied with your MultiVOIP to a live AC
outlet and to the power connector on the back of the MultiVOIP as
shown at top right in Figure 3-12.
Command Modem connector
for remote configuration
ETH ERNET
COMMAND
E& M FXS/FX O
E& M FXS/FXO
E &M FXS/FXO
E&M FXSF/ XO
E&M FXS/FXO
E&M FXS /FXO
E&M FXS/FXO
E&M FX S/FXO
COMMAND
MODEM
10 BASET
Voice/Fax Channel Connections
Channels 1-4 Bottom MVP410 /8 10
Channels 5-8 Top MVP810 Only
E&M FXS/FXO
Ethernet Connection
FXS
E&M
FXO
Comm and Port Connec tion
PSTN
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 the MultiVOIP and the other end into the PC serial port. See
Figure 3-12.
4. Connect a network cable to the ETHERNET 10BASET connector on
the back of the MultiVOIP. Connect the other end of the cable to your
network.
5. 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/FXO
connector 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
connector on the back of the MultiVOIP.
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Connect the other end to the trunk line.
Verify that the E&M Type in the E&M Options group of the Interface
dialog 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/FXO
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 line
rings busy consistently, you will need to reverse the polarity of
one end of the connector (swap the connections of the wires to
the two middle pins of one RJ-11 connector).
6. Repeat step 5 to connect the remaining telephone equipment to each
channel on your MultiVOIP. Although a MultiVOIP’s channels are
often all configured identically, each channel is individually
configurable. So, for example, some channels of a MultiVOIP might
use the FXO interface and others the FXS; some might use the DID
interface and others E&M, etc.
7. If you intend to configure the MultiVOIP remotely using the
MultiVOIP Windows GUI, connect an RJ-11 phone cable between the
Command Modem connector (at the rear of the MultiVOIP) and a
receptacle served by a telco POTS line. See Figure 3-13.
The Command Modem is built into the MultiVOIP unit. To configure
the MultiVOIP remotely using its Windows GUI, you must call into
the MultiVOIP’s Command Modem. Once a connection is made, the
configuration process is identical to local configuration with the
Windows GUI.
Command Modem connector
for remote configuration
ETHERNET
COMMAND
MODEM
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
COMMAND
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
10 BASET
MVP-410/810
Rear Panel
Grounding Screw
Telco POTS Line
Figure 3-13. MVP-410/810 Voip Connections for GND &
Remote Config Modem
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8. Ensure that the unit is properly connected to earth ground by
verifying that it is reliably grounded when mounted within a rack.
This can be accomplished by connecting a grounding wire between
the chassis grounding screw (see Figure 3-13) and a metallic object
that will provide an electrical ground.
9. Turn on power to the MultiVOIP by placing the ON/OFF switch on
the back panel to the ON position. Wait for the Boot LED on the
MultiVOIP to go off before proceeding. This may take a few minutes.
Proceed to Chapter 4 to load the MultiVOIP software.
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Cabling Procedure for MVP-410ST/810ST
Cabling involves connecting the MultiVOIP to your LAN and telephone
equipment.
1. Connect the power cord supplied with your MultiVOIP to a live AC
outlet and to the power connector on the back of the MultiVOIP as
shown at top right in Figure 3-14.
Command Modem/TA connector
for remote configuration
E THE RN ET
COMMAND
COMMAND
IS DN 1
ISDN2
IS DN 3
ISDN4
MODEM
10 BASET
ISDN-BRI Connections
ISDN1 & ISDN2 : MVP410ST/810ST
ISDN3 & ISDN4: MVP810ST only
?
NETWORK
MO DE
TERMINAL
MODE
Ethernet Connection
*
IS DN
TA
NT1
Device
CommandPort Connection
PSTN
PBX
is needed
* NT1 Device
if PBX has “U” interface.
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 the MultiVOIP and the other end into the PC serial port. See
Figure 3-14.
3. Connect a network cable to the ETHERNET 10BASET connector on
the back of the MultiVOIP. Connect the other end of the cable to your
network.
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4. Terminal Mode. When a voip ISDN connector is to be connected to a
PBX extension line or to a telco line, select “Terminal” as the
“Layer 1 Interface” in the ISDN Parameters screen. When making
cable connections, an NT1 device will be needed between the
MultiVOIP and the PSTN or between the MultiVOIP and any PBX
with a “U” interface. (For more information, see Appendix B: Cable
Pinouts in this manual.) Connect cables between voip ISDN
connectors and network equipment.
NOTE: In order to operate in Terminal mode, the
network equipment to which you will be
connecting (e.g., PBX) must support D-channel
signaling in its ISDN-S/T interface.
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Network Mode. When a voip ISDN connector is to be connected to an
ISDN phone station or to an ISDN terminal adapter (TA), select
“Network” as the “Layer 1 Interface” in the ISDN Parameters screen
of the MultiVOIP software. 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 for ISDN phone stations.
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5. Repeat the above step to connect the remaining ISDN telephone
equipment to each ISDN connector on your MultiVOIP. Be aware
that you can assign each ISDN line separately and independently to
either Network mode or Terminal mode. That is, all ISDN lines do
not have to be assigned in to the same operating mode.
ETHERNET
COMMAND
IS DN 1
ISDN2
IS DN 3
ISDN4
10 BASET
ISDN-BRI Connections
ISDN1 & ISDN2 : MVP410ST/810ST
ISDN3 & ISDN4: MVP810ST only
Grounding Screw
?
NETWORK
MO DE
TERMINAL
MODE
Ethernet Connection
*
IS DN
TA
NT1
Device
Command Port Connection
PSTN
PBX
is needed
* NT1 Device
if PBX has “U” interface.
Figure 3-15: ISDN/BRI Voip Connections for GND & Remote
Configuration Modem
6. Ensure that the unit is properly connected to earth ground by
verifying that it is reliably grounded when mounted within a rack.
This can be accomplished by connecting a grounding wire between
the chassis grounding screw (see Figure 3-15) and a metallic object
that will provide an electrical ground.
7. Turn on power to the MultiVOIP by placing the ON/OFF switch on
the back panel to the ON position. Wait for the Boot LED on the
MultiVOIP to go off before proceeding. This may take a few minutes.
Proceed to Chapter 4 to load the MultiVOIP software.
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Cabling Procedure for MVP210
Cabling involves connecting the MultiVOIP to your LAN and telephone
equipment.
1. For DID channels only. If both channels of your MVP210 MultiVOIP
will be 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
must change the jumper on the MultiVOIP circuit card.
a. Disconnect power. Unplug the AC power cord from the wall outlet
or from the receptacle on the MultiVOIP unit.
b. Using a #1 Phillips driver, remove the screw (at bottom of unit,
near the back-cover end) that attaches the main circuit card to the
chassis of the MVP210.
c. Pull the main circuit card out about half way.
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d. Identify the channels on which the DID interface will be used.
L
E
D
1
4
L
E
D
1
2
L
E
D
7
L
ED
1
3
L
E
D
11
L
E
D
10
LE
D
9
LE
D
8
L
ED
6
LE
D
5
LE
D
4
LE
D
3
L
E
D
2
L
E
D
1
R
113
R7
2
R74
R114
R58
R
57
R56
R5
5
R2
05
R2
MVP210 Circuit Board
Ch1
Ch2
as configured
for DID Interface
JP4
Ch 1 Jumper
Block
P7
JP7
as shipped,
for non-DID interfaces
JP8
Ch 2 Jumper
Block
JP1
F
B
3
J3
J
7
J5
J9
J
11
J1
S
1
0
J
15
as configured
for DID Interface
Figure 3-16. MVP210 Channel Jumper Settings
e. Position the jumper for each DID channel so that it does not connect
the two jumper posts. For DID operation of a voip channel, the
MultiVOIP will work properly if you simply remove the jumper
altogether, but that is inadviseable because the jumper might be
needed later if a different telephony interface is used for that voip
channel.
f. Slide the main circuit card back into the MultiVOIP chassis and
replace the screw at the bottom of the unit.
2. Connect the power cord supplied with your MultiVOIP to the power
connector on the back of the MultiVOIP and to a live AC outlet as
shown in Figure 3-17.
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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 the MultiVOIP 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
back 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/FXO
connector 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
connector on 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 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.)
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Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO
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 line
rings busy consistently, you will need to reverse the polarity of
one end of the connector (swap the connections of the wires to
the two middle pins of one RJ-11 connector).
6. Repeat the above step to connect the remaining telephone equipment
to the second channel on your MultiVOIP.
7. Ensure that the unit is properly connected to earth ground by
verifying that it is reliably grounded when mounted within a rack.
This can be accomplished by connecting a grounding wire between
the chassis 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 back panel to the ON position. Wait for the BOOT LED on the
MultiVOIP to go off before proceeding. This may take a few minutes.
Proceed to the Software Installation chapter to load the MultiVOIP
software.
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Cabling Procedure for MVP-130/130FXS
Power
Ethernet Command FXS/FXO
Power Connection
FXO
FXS
PBX
ꢀ
Command Port Connection
Telephony Connection
PSTN
Hub
Network Connection
Figure 3-18: Cabling for MVP-130/130FXS
Cabling involves connecting the MultiVOIP to your LAN and telephone
equipment.
1. Connect the power cord supplied with your MultiVOIP to the power
connector on the back of the MultiVOIP and to a live AC outlet as
shown in Figure 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 the MultiVOIP 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
back of the MultiVOIP. Connect the other end of the cable to your
network.
4. Since the MVP130FXS supports the FXS interface only, its connection
options differ from that of the MVP130, which supports both FXS and
FXO.
A. For MVP130. To connect a station device such as an analog telephone,
a fax machine, or a Key Telephone System (KTS) (FXS interface), or a
PBX extension (FXO interface) to your MultiVOIP, attach one end of
an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the
back MultiVOIP and the other end to the device or phone jack. You
will define the interface in the Interface dialog box in the software
when you configure the unit.
B. For MVP130FXS. To connect a station device such as an analog
telephone or fax machine to your MultiVOIP, attach one end of an RJ-
11 phone cord to the VOICE connector on the back MultiVOIP and
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the other end to the device or phone jack. You will define the
interface in the Interface dialog box in the software when you
configure the unit. The MVP130FXS supports only the FXS interface.
Proceed to Chapter 4 to load the MultiVOIP software.
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Software Installation
Introduction
Configuring software for your MultiVOIP entails three tasks:
(1) loading the software onto the PC (this is “Software Installation and
is discussed in this chapter),
(2) setting values for telephony and IP parameters that will fit your
system (this is “Technical Configuration” and it is discussed in Chapter
5 for T1/E1 MultiVOIP units and in Chapter 6 for analog MultiVOIP
units), and
(3) establishing “phonebooks” that contain the various dialing patterns
for VOIP calls made to different locations (this is “Phonebook
Configuration” and it is discussed in Chapters 7, 8, and 9 for T1, E1,
and analog MultiVOIP units respectively).
Loading MultiVOIP Software onto the PC
The software loading procedure does not present every screen or option
in the loading process. It is assumed that someone with a thorough
knowledge of Windows and the software loading process is performing
the installation.
The MultiVOIP software and User Guide are contained on the
MultiVOIP product CD. Because the CD is auto-detectable, it will start
up automatically when you insert it into your CD-ROM drive. When
you have finished loading your MultiVOIP software, you can view and
print the User Guide by clicking on the View Manuals icon.
1. Be sure that your MultiVOIP has been properly cabled and that the
power is turned on.
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2. Insert the MultiVOIP CD into your CD-ROM drive. The CD should
start automatically. It may take 10 to 20 seconds for the Multi-Tech
CD installation 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
Install Software icon.
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4. A ‘welcome’ screen appears.
Press Enter or click Next to continue.
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5. Follow the on-screen instructions to install your MultiVOIP software.
The first screen asks you to choose the folder location of the files of
the MultiVOIP software.
Choose a location and click Next.
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6. At the next screen, you must select a program folder location for the
MultiVOIP software program icon.
Click Next. Transient progress screens will appear while files are
being copied.
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7. On the next screen you can select the COM port that the command
PC will use when communicating with the MultiVoip unit. After
software installation, the COM port can be re-set in the MultiVOIP
Software (from the sidebar menu, select Connection | Settings to
access the COM Port Setup screen or use the keyboard shortcut Ctrl
+ G).
NOTE: If the COM port setting made
here conflicts with the actual COM
port resources available in the
command PC, this error message will
appear when the MultiVOIP program
is launched. If this occurs, you must
reset the COM port.
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8. A completion screen will appear.
Click Finish.
9. When setup of the MultiVOIP software is complete, you will be
prompted to run the MultiVOIP software to configure the VOIP.
Software installation is complete at this point. You may proceed with
Technical Configuration now or not, at your convenience.
Technical Configuration instructions are in the next two chapters of
this manual: Chapter 5 for T1/E1 MultiVOIP units and Chapter 6 for
Analog MultiVOIP units.
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Un-Installing the MultiVOIP Configuration
Software
1. To un-install the MultiVOIP configuration software, go to Start |
Programs and locate the entry for the MultiVOIP program. Select
Uninstall.
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2. Two confirmation screens will appear. Click Yes and OK when you
are certain you want to continue with the uninstallation process.
3. A special warning message similar to that shown below may appear
concerning the MultiVOIP software’s “.bin” file. Click Yes.
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4. A completion screen will appear.
Click Finish.
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Technical Configuration (T1/E1)
MultiVOIP User Guide
Configuring the Digital T1/E1 MultiVOIP
There are two ways in which the MultiVOIP must be configured before
operation: technical configuration and phonebook configuration.
Technical Configuration. First, the MultiVOIP must be configured to
operate with technical parameter settings that will match the
equipment with which it interfaces. 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 or
regions,
(6) its operation with a mail server on the same IP network (per SMTP
parameters) such that log reports about VoIP telephone call traffic can
be sent to 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
seven categories is what we call “technical configuration” and it is
described in this chapter.
Phonebook Configuration. The second type of configuration that is
required for the MultiVOIP pertains to the phone number dialing
sequences that it will receive and transmit when handling calls. Both
the PBX/telephony equipment and the other VOIP devices that the
MultiVOIP unit interacts with will affect dialing patterns. We call this
“Phonebook Configuration,” and it is described in Chapter 7: T1
Phonebook Configuration and Chapter 8: E1 Phonebook Configuration of this
manual. Chapter 2, the Quick Start Instructions, presents additional
examples relevant to the T1/E1 voips.
Local/Remote Configuration. The MultiVOIP must be configured
locally at first (to establish an IP address for the MultiVOIP unit). But
changes to this initial 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 MultiVOIP configuration program is used.
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Remote configuration is done through a connection between the
MultiVOIP’s Ethernet (network) port and a computer connected to the
same network. The computer could be miles or continents away from
the MultiVOIP itself. There are two ways of doing remote configuration
and operation of the MultiVOIP unit: (1) using the MultiVoipManager
SNMP program, or (2) using the MultiVOIP web browser interface
program.
MultiVoipManager. MultiVoipManager is an SNMP agent program
(Simple Network Management Protocol) that extends the capabilities of
the MultiVOIP configuration program: MultiVoipManager allows the
user to manage any number of VOIPs on a network, whereas the
MultiVOIP configuration program 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 a time.
MultiVoipManager may (but does not need to) reside on the same PC
as the MultiVOIP configuration program. The MultiVoipManager
program is on the MultiVOIP Product CD. Updates, when applicable,
may be posted at on the MultiTech FTP site. To download, go to
Web Browser Interface. The MultiVOIP web browser GUI gives access
to the same commands and configuration parameters as are available in
the MultiVOIP Windows GUI except for logging functions. When
using the web browser GUI, logging can be done by email (the SMTP
option).
Functional Equivalence of Interfaces. The MultiVOIP configuration
program is required to do the initial configuration (that is, setting an IP
address for the MultiVOIP unit) so that the VOIP unit can communicate
with the MultiVoipManager program or with the web browser GUI.
Management of the VOIP after that point can be done from any of these
three programs since they all offer essentially the same functionality.
Functionally, either the MultiVoipManager program or the web
browser GUI can replace the MultiVOIP configuration program after
the initial configuration is complete (with minor exceptions, as noted).
WARNING: Do not attempt to interface the MultiVOIP unit with
two control programs simultaneously (that is, by
accessing the MultiVOIP configuration program via
the Command Port and either the
MultiVoipManager program or the web browser
interface via the Ethernet Port). The results of using
two programs to control a single VOIP
simultaneously would be unpredictable.
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Local Configuration
This 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 be handled through the web browser GUI, as well (see
the Operation and Maintenance chapter of this manual). In most aspects
of configuration, the Windows GUI and web-browser GUI differ only
graphically, not functionally. For information on SNMP remote
configuration and management, see the MultiVoipManager
documentation.
Pre-Requisites
To complete the configuration of the
MultiVOIP unit, you must know several
things about the overall system.
Before configuring your MultiVOIP Gateway unit, you must know the
values for several IP and T1/E1 parameters that describe the IP
network system and telephony system (PBX or telco central office
equipment) with which the digital MultiVOIP will interact. If you plan
to receive log reports on phone traffic by email (SMTP), you must
arrange to have an email address assigned to the VOIP unit on the
email server on your IP network.
IP Parameters
The following parameters must be known about the network (LAN,
WAN, Internet, etc.) to which the MultiVOIP will connect:
Ask your computer network
administrator.
Info needed to operate:
all MultiVOIP models.
9
IP Network Parameters:
Record for each VOIP Site
in System
#
• IP Address
• IP Mask
• Gateway
• Domain Name Server (DNS) Info
(not implemented; for future use)
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Write down the values for these IP parameters. You will need to enter
these values in the “IP Parameters” screen in the Configuration section
of the MultiVOIP software. You must have this IP information about
every VOIP in the system.
T1 Telephony Parameters (for MVP2410)
The following parameters must be known about the PBX or telco
central office equipment to which the T1 MultiVOIP will connect:
T1 Phone Parameters
Info needed to operate:
9
MVP2410
Ask phone company or
PBX maintainer.
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 the
opposite of the setting used by the telco/PBX.
• Which line coding is used? AMI___ or B8ZS___
Write down the values for these T1 parameters. You will need to enter
these values in the “T1/E1 Parameters” screen in the Configuration
section of the MultiVOIP software.
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E1 Telephony Parameters (for MVP3010)
The following parameters must be known about the PBX or telco
central office equipment to which the E1 MultiVOIP will connect:
E1 Phone Parameters
Info needed to operate:
9
MVP3010
Ask phone company or
PBX maintainer.
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
internal or external clocking? _________________
Note that the setting used in the voip unit will be the
opposite of the setting used by the telco/PBX.
• 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
these values in the “T1/E1 Parameters” screen in the Configuration
section of the MultiVOIP software.
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SMTP Parameters (for email call log reporting)
required if log reports of
VOIP call traffic
Optional
are to be sent by email
SMTP Parameters
Preparation Task:
To: I.T. Department
Ask Mail Server
re: email account for VOIP
administrator to set up
email account (with
password) for the
MultiVOIP unit itself.
Be sure to give a unique
identifier to each
individual MultiVOIP
unit. .
Get the IP address of the
mail server computer, as
well.
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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 must be 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 access them.
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 remote
management software is used).
12. Set Regional Parameters (Phone Signaling Tones & Cadences and
setup for built-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
Configuration program; SNMP, remotely in MultiVoipManager
program; or SMTP, via email).
16. Set Supplementary Services Parameters. The Supplementary
Services screen allows voip deployment of features that are normally
found in PBX or PSTN 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).
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Technical Configuration (T1/E1)
Local Configuration Procedure (Detailed)
You can begin the configuration process as a continuation of the
MultiVOIP software installation. You can establish your configuration
or modify it at any time by launching the MultiVOIP program from the
Windows Start menu.
1. Check Power and Cabling. Be sure the MultiVOIP is turned on and
connected to the computer via the MultiVOIP’s Command Port (DB9
connector at computer’s COM port; RJ45 connector at MultiVOIP).
You must allow the MultiVOIP to finish booting before you launch
the MultiVOIP Configuration Program. The RED boot LED turns
itself off when the booting process is completed.
2. Start MultiVOIP Configuration Program. Launch the MultiVOIP
program from the Windows Start menu (from the folder location
determined during installation).
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3. Confirm Connection. If the MultiVOIP is set for an available COM
port and 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.)
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In the lower left corner of the screen, the connection status of the
MultiVOIP will be displayed. The messages in the lower left corner
will change as detection occurs. The message “MultiVOIP Found”
confirms that the MultiVOIP is in contact with the MultiVOIP
configuration program. Skip to step 5.
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4. Solving Common Connection Problems.
A. Fixing a COM Port Problem. If the MultiVOIP main screen appears
but is grayed out and seems inaccessible, the COM port that was
specified for its communication with the PC is unavailable and must
be changed. An error message 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
the Connection 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 are currently available, re-allocate COM port resources
in the computer’s MS Windows operating system to make one
available.)
Ctrl + G
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4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by
the computer, 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.
For instructions on MultiVOIP cable connections, see the “Cabling”
section of Chapter 3.
5. Configuration Parameter Groups: Getting Familiar, Learning
About Access. The first part of configuration concerns IP parameters,
Voice/FAX parameters, T1/E1 parameters, SNMP parameters,
Regional parameters, SMTP parameters, Supplementary Services
parameters, Logs, and System Information. In the MultiVOIP software,
these seven types of parameters are grouped together under
“Configuration” and each has its own dialog box for entering values.
Generally, you can reach the dialog box for these parameter groups in
one of four ways: pulldown menu, toolbar icon, keyboard shortcut, or
sidebar..
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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
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In each field, enter the values that fit your particular network.
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The IP Parameters fields are described in the table below.
IP Parameter Definitions
Field Name
Values
Description
DiffServ
Parameter
fields
DiffServ PHB (Per Hop Behavior) values
pertain to a differential prioritizing
system for IP packets as handled by
DiffServ-compatible routers. There are 64
values, each with an elaborate technical
description. These descriptions are found in
TCP/IP standards RFC2474, RFC2597, and,
for present purposes, in RFC3246, which
describes the value 34 (34 decimal; 22 hex) for
Assured Forwarding behavior (default for
Call Control PHB) and the value 46 (46
decimal; 2E hexadecimal) for Expedited
Forwarding behavior (default for Voip Media
PHB). Before using values other than these
default values of 34 and 46, consult these
standards documents and/or a qualified IP
telecommunications engineer.
To disable DiffServ, configure both fields to 0
decimal.
The next page explains DiffServ in the
context of the IP datagram.
Call Control
PHB
0 – 63
Value is used to
prioritize call setup IP
packets.
default = 34
Voip Media
PHB
0 – 63
Value is used to
prioritize the RTP/RTCP
audio IP packets.
default = 46
Frame Type
Type II, SNAP
Must be set to match
network’s frame type.
Default is Type II.
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The IP Datagram with Header, Its Type-of-Service field, & DiffServ
bits =>
0
4
8
16 19
24
31
VERS
HLEN
TYPE OF
SERVICE
TOTAL LENGTH
FLAGS
IDENTIFICATION
TIME TO LIVE PROTOCOL
SOURCE IP ADDRESS
DESTINATION IP ADDRESS
IP OPTIONS (if any)
FRAGMENT OFFSET
HEADER CHECKSUM
PADDING …
end of header
DATA
…
The TOS field consists of eight bits, of which only the first six are used. These six
bits are called the “Differentiated Service Codepoint” or DSCP bits.
The Type of Service or “TOS” field
0
1
2
3
4
5
6
7
PRECEDENCE
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 packet
based on the DSCP value. The DiffServ Parameters fields in the MultiVOIP IP
Parameters screen allow you to configure the DSCP bits to values supported by the
router. Specifically, the Voip Media PHB field relates to the prioritizing of audio
packets (RTP and RTCP packets) and the Call Control PHB field relates to the
prioritzing 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; 100010
binary – consider vis-à-vis TOS field above) for Assured Forwarding behavior. The
MultiVOIP Voip Media PHB parameter defaults to the value 46 decimal (2E hex;
101110 binary – consider vis-à-vis TOS field above). To disable DiffServ, configure
both fields to 0 decimal.
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IP Parameter Definitions (cont’d)
Field Name
Values
Description
IP Parameter fields
Enable DHCP
Y/N
Dynamic Host
Configuration Protocol is a
method for assigning IP
address and other IP
parameters to computers on
the IP network in a single
message with great
disabled by
default
flexibility. IP addresses can
be static or temporary
depending on the needs of
the computer.
IP Address
IP Mask
4-places, 0-255
4-places, 0-255
The unique LAN IP address
assigned to the MultiVOIP.
Subnetwork address that
allows for sharing of IP
addresses within a LAN.
Gateway
4-places, 0-255
The IP address of the device
that connects your
MultiVOIP to the Internet.
DNS Parameter fields
Enable DNS
Y/N
Enables Domain Name
Space/System function
where computer names
are resolved using a
worldwide distributed
database.
DNS Server IP
Address
4-places, 0-255.
IP address of specific
DNS server to be used to
resolve Internet
computer names.
FTP Parameter fields
FTP Server
Enable
Y/N
See “FTP Server
MultiVOIP unit has an
FTP Server function so
File Transfers” in that firmware and other
Operation &
Maintenance
chapter.
important operating
software files can be
transferred to the voip
via the network.
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7. Enable Web Browser GUI (Optional). After an IP address for the
MultiVOIP unit has been established, you can choose to do any further
configuration 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 do configuration work using the web browser GUI, you must first
enable it. To do so, follow the steps below.
A. Set IP address of MultiVOIP unit using the MultiVOIP
Configuration program (the Windows GUI).
B. Save Setup in Windows GUI.
C. Close Windows GUI.
D. Install Java program from MultiVOIP product CD (required 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 when prompted.
H. Use web browser GUI to configure or operate MultiVOIP unit.
The configuration screens in the web browser GUI will have the same
content as their counterparts in the Windows GUI; only the graphic
presentation will be different.
For more details on enabling the MultiVOIP web GUI, see the “Web
Browser Interface” section of the Operation & Maintenance chapter of
this manual.
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8. Set Voice/FAX Parameters. This dialog box can be reached by
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.
Accessing “Voice/FAX Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + H
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In each field, enter the values that fit your particular network.
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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 a particular channel, you can apply this entire set of
Voice/FAX parameters to another channel by using the Copy Channel
button and its dialog box. To copy a set of Voice/FAX parameters to all
channels, select “Copy to All” and click Copy.
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The Voice/FAX Parameters fields are described in the tables below.
Voice/Fax Parameter Definitions
Field Name Values
Description
Default
--
When this button is clicked, all
Voice/FAX parameters are set to their
default values.
Select
Channel
1-24 (T1)
1-30 (E1)
Channel to be configured is selected
here.
Copy
Channel
--
Copies the Voice/FAX attributes of
one channel to another channel.
Attributes can be copied to multiple
channels or all channels at once.
Voice Gain
Input Gain
--
Signal amplification (or attenuation)
in dB.
+31dB
to
–31dB
Modifies audio level entering voice
channel before it is sent over the
network to the remote VOIP. The
default & recommended value is 0 dB.
Output Gain +31dB
Modifies audio level being output to
the device attached to the voice
channel. The default and
to
–31dB
recommended value is 0 dB.
DTMF Parameters
DTMF Gain --
The DTMF Gain (Dual Tone Multi-
Frequency) controls the volume level
of the digital tones sent out for Touch-
Tone dialing.
DTMF Gain, +3dB to Default value: -4 dB. Not to be
High Tones -31dB & changed except under supervision of
“mute” MultiTech’s Technical Support.
DTMF Gain, +3dB to Default value: -7 dB. Not to be
Low Tones -31dB & changed except under supervision of
“mute” MultiTech’s Technical Support.
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Voice/Fax Parameter Definitions (cont’d)
Field Name
Values
Description
DTMF Parameters
Duration
(DTMF)
60 – 3000
ms
When DTMF: Out of Band is
selected, this setting determines
how long each DTMF digit ‘sounds’
or is held. Default = 100 ms.
DTMF
In/Out of
Band
Out of
Band, or
Inband
When DTMF Out of Band is selected
(checked), the MultiVOIP detects DTMF
tones at its input and regenerates them
at its output. When DTMF Inband is
selected, the DTMF digits are passed
through the MultiVOIP unit as they are
received.
FAX Parameters
Fax Enable
Y/N
Enables or disables fax capability
for a particular channel.
Max Baud
Rate
(Fax, bps)
2400, 4800,
7200, 9600,
12000, 14400
Set to match baud rate of fax machine
connected to channel (see Fax machine’s
user manual). Default = 14400 bps.
Fax Volume -18.5 dB
Controls output level of fax tones. To
be changed only under the direction of
Multi-Tech’s Technical Support.
Default =
-9.5 dB
to –3.5 dB
Jitter Value
(Fax)
Default =
400 ms
Defines the inter-arrival packet
deviation (in milliseconds) for the
fax transmission. A higher value
will increase the delay, allowing a
higher percentage of packets to be
reassembled. A lower value will
decrease the delay allowing fewer
packets to be reassembled.
Mode (Fax)
FRF 11;
T.38
(T.38 not
currently
sup-
FRF11 is frame-relay FAX standard
using these coders: G.711, G.728,
G.729, and G.723.1.
T.38 is an ITU-T standard for
storing and forwarding Faxes via
email using X.25 packets. It uses
T.30 fax standards and includes
special provisions to preclude FAX
timeouts during IP transmissions.
ported)
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Voice/Fax Parameter Definitions (cont’d)
Coder Parameters
Coder
Manual or Determines whether selection of
Auto-
matic
coder is manual or automatic.
When Automatic is selected, the
local and remote voice channels will
negotiate the voice coder to be used
by selecting the highest bandwidth
coder supported by both sides
without exceeding the Max
Bandwidth setting. G.723, G.729, or
G.711 are negotiated.
Selected
Coder
G.711 a/u Select from a range of coders with
law 64
kbps;
specific bandwidths. The higher the
bps rate, the more bandwidth is
used. The channel that you are
calling must have the same voice
coder selected.
G.726, @
16/24/32
/40 kbps;
G.727, @
nine bps
rates;
Default = G.723.1 @ 6.3 kbps, as
required for H.323. Here 64K of
digital voice are compressed to
6.3K, allowing several simultaneous
conversations over the same
bandwidth that would otherwise
carry only one.
G.723.1 @
5.3 kbps,
6.3 kbps;
G.729,
8kbps;
Net Coder
@
6.4, 7.2, 8,
8.8, 9.6
To make selections from the
Selected Coder drop-down list, the
Manual option must be enabled.
kbps
Max
bandwidth
(coder)
11 – 128
kbps
This drop-down list enables you to
select the maximum bandwidth
allowed for this channel. The Max
Bandwidth drop-down list is
enabled only if the Coder is set to
Automatic.
If coder selected automatically, then
enter a value for maximum
bandwidth, as directed by VOIP
administrator.
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Advanced Features
Description
Silence
Compression
Y/N
Determines whether silence
compression is enabled (checked) for
this voice channel.
With Silence Compression enabled, the
MultiVOIP will not transmit voice
packets when silence is detected,
thereby reducing the amount of
network bandwidth that is being used
by the voice channel.
Default = off.
Echo
Cancellation
Y/N
Y/N
Determines whether echo cancellation is
enabled (checked) for this voice
channel.
Echo Cancellation removes echo and
improves sound quality. Default = on.
Forward
Error
Correction
Determines whether forward error
correction is enabled (checked) for this
voice channel.
Forward Error Correction enables
some of the voice packets that were
corrupted or lost to be recovered. FEC
adds an additional 50% overhead to the
total network bandwidth consumed by
the voice channel. Default = Off
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
AutoCall/Offhook Alert
Parameters
Auto Call /
Offhook
Alert
AutoCall,
Offhook
Alert
The AutoCall option enables the local
MultiVOIP to call a remote MultiVOIP
without the user having to dial a Phone
Directory Database number. As soon as
you access the local MultiVOIP
voice/fax channel, the MultiVOIP
immediately connects to the remote
MultiVOIP identified in the Phone
Number box of this option. This is
essentially a hotline function that gives
an immediate connection between two
points.
The Offhook Alert option is not
supported in T1/E1 MultiVOIP units.
AutoCall applies on a channel-by-
channel basis. It would not be
appropriate for this function to be
applied to a channel that serves in a
pool of available channels for general
phone traffic. AutoCall requires an
entry in the Outgoing phonebook of the
local MultiVOIP and a matched setting
in the Inbound Phonebook of the
remote voip. Also, especially for the
AutoCall function,
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
AutoCall/Offhook Alert
Parameters
Offhook
Alert Timer
N/A
Not supported in T1/E1 MultiVOIP
units.
Phone
Number
N/A
Not supported in T1/E1 MultiVOIP
units.
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
Dynamic Jitter Buffer
Parameters
Dynamic
Jitter Buffer
Dynamic Jitter defines a minimum
and a maximum jitter value for
voice communications. When
receiving voice packets from a
remote MultiVOIP, varying delays
between packets may occur due to
network traffic problems. This is
called Jitter. To compensate, the
MultiVOIP uses a Dynamic Jitter
Buffer. The Jitter Buffer enables the
MultiVOIP to wait for delayed voice
packets by automatically adjusting
the length of the Jitter Buffer
between configurable minimum
and maximum values. An
Optimization Factor adjustment
controls how quickly the length of
the Jitter Buffer is increased when
jitter increases on the network. The
length of the jitter buffer directly
effects the voice delay between
MultiVOIP gateways.
The default minimum dynamic
jitter buffer of 60 milliseconds is the
minimum delay that would be
acceptable over a low jitter network.
The default maximum dynamic
jitter buffer of 300 milliseconds is
the maximum delay tolerable over a
high jitter network.
Minimum
Jitter Value
60 to 400
ms
The default minimum dynamic
jitter buffer of 60 milliseconds is the
minimum delay that would be
acceptable over a low jitter network.
Default = 60 msec
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MultiVOIP User Guide
Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
Dynamic Jitter Buffer
Parameters
Maximum
Jitter Value
60 to 400
ms
The default maximum dynamic
jitter buffer of 300 milliseconds is
the maximum delay tolerable over a
high jitter network.
Default = 300 msec
Optimizat-
ion Factor
0 to 12
The Optimization Factor
determines how quickly the length
of the Dynamic Jitter Buffer is
changed based on actual jitter
encountered on the network.
Selecting the minimum value of 0
means low voice delay is desired,
but increases the possibility of jitter-
induced voice quality problems.
Selecting the maximum value of 12
means highest voice quality under
jitter conditions is desired at the
cost of increased voice delay.
Default = 7.
Modem Relay
To place modem traffic onto the voip network (an application called “modem relay”),
use Coder G.711 mu-law at 64kbps.
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Technical Configuration (T1/E1)
Voice/Fax Parameter Definitions (cont’d) )
Field Name Values
Description
Auto Disconnect
Parameters
The Automatic Disconnection
group has four options which can
be used singly or in any
combination.
Jitter Value
1-65535
milli-
seconds
The Jitter Value defines the average
inter-arrival packet deviation (in
milliseconds) before the call is
automatically disconnected. The
default is 150 milliseconds. A higher
value means voice transmission will
be more accepting of jitter. A lower
value is less tolerant of jitter.
Inactive by default. When active,
default = 150 ms. However, value
must equal or exceed Dynamic
Minimum Jitter Value.
Call
Duration
1-65535
seconds
Call Duration defines the
maximum length of time (in
seconds) that a call remains
connected before the call is
automatically disconnected.
Inactive by default.
When active, default = 180 sec.
This may be too short for most
configurations requiring upward
adjustment.
Consecutive 1-65535
Packets Lost
Consecutive Packets Lost defines
the number of consecutive packets
that are lost after which the call is
automatically disconnected.
Inactive by default.
When active, default = 30
Network
Discon-
nection
1 to 65535 Specifies how long to wait before
seconds;
Default =
300 sec.
disconnecting the call when IP
network connectivity with the
remote site has been lost.
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Technical Configuration (T1/E1)
MultiVOIP User Guide
9. Set T1/E1/ISDN Parameters. This dialog box can be reached by
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.
Accessing “T1/E1/ISDN Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + T
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Technical Configuration (T1/E1)
In each field, enter the values that fit your particular network.
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Technical Configuration (T1/E1)
MultiVOIP User Guide
T1 Parameters. The parameters applicable to T1 and their values are
shown in the figure below. These T1 Parameter fields are described in
the tables that follow.
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Technical Configuration (T1/E1)
T1 Parameter Definitions
Field Name
Values
T1
Description
T1/E1/ISDN
North American standard.
Long-Haul
Mode
Y/N
In Long-Haul Mode, the
MultiVOIP automatically
recovers received signals as low
as –36 dB. The maximum
reachable length with 22 AWG
cable is 2000 meters. When
Long-Haul Mode is disabled,
signals as low as –10 dB can be
received.
Default: disabled.
CRC Check
Y/N
When enabled, allows
generation and checking of
CRC bits. If not enabled, all
check bits in the transmit
direction are set. Only applies
to ESF frame format.
(Cyclic
Redundancy
Check)
Default: enabled.
Frame Format of MultiVOIP
should match that used by PBX
or telco. ESF and D4 are
commonly used.
Frame Format
F4, D4, ESF,
SLC96
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Technical Configuration (T1/E1)
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T1 Parameter Definitions (cont’d)
Values Description
Field Name
CAS Protocol
E&M Immed Strt Channel Associated Signaling
E&M Wink Start
(CAS) is a method of
incorporating telephony
signaling info into a T1
E&M Wink with
dial tone
voice/data stream. In CAS, the
signaling bits (the A, B, C, and
D bits) are multiplexed into the
signal stream of each T1
FXO Ground Strt
FXO Loop Start
FXS Ground Strt
FXS Loop Start
channel. (By contrast, in
Common Channel Signaling
(CCS), one channel handles
signaling for all other channels.)
Each CAS protocol defines the
states of the signaling bits
during the various stages of a
call (IDLE, SEIZED, ANSWER,
RING-ON, RING-OFF).
The CAS protocol code allows
the VOIP to interact properly
with the PBX or central-office
switch that it serves. The need
to download CAS protocols
arises for only a small minority
of VOIP users, and only when
PBX/switch is found to be
incompatible with standard
protocols.
Match this parameter to the
setting of PBX or central-office
switch.
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Technical Configuration (T1/E1)
T1 Parameter Definitions (cont’d)
Field Name
Values
Description
Detect Flash
Hook
Y/N
This setting determines whether
or not the MultiVOIP responds
to hook-flash signals.
Detection Time 100 – 1500
milliseconds
Minimum hook-flash time that
will be interpreted as a valid
flash by the MultiVOIP.
Generation
Time
100 – 1500
In some systems, a MultiVOIP
might receive a hook-flash
signal from an upstream device
(a PBX, voip or other device)
and must replicate it to a
downstream device. This
parameter determines the
duration of the hook-flash
signal that is passed to a
downstream device.
milliseconds
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T1 Parameter Definitions (cont’d)
ISDN Parameters
Field Name
Values
Description
Enable
ISDN-PRI
Y/N
If digital connection is ISDN-
PRI type, this box should be
checked. When ISDN is
enabled, the “CAS Protocols”
field is grayed out (ISDN has its
own signaling method).
Terminal/
Network
either
“Terminal” or
“Network”
When “Terminal” is selected, it
indicates that the MultiVOIP
should emulate the subscriber
(terminal) side of the digital
connection. When “Network”
is selected, it indicates that the
MultiVOIP should emulate the
central office (network) side of
the digital connection.
Setting used for MultiVOIP
must be opposite to the setting
used in the PBX. For example,
if the PBX is set to “Terminal,”
then the MultiVOIP must be set
to “Network.”
Country
Operator
see table, later
this chapter
Country in which MultiVOIP is
operating with ISDN.
see table, later
this chapter
Indicates phone switch
manufacturer/model or refers
to telco so as to specify the
switching system in question.
ISDN is implemented
somewhat differently in
different switches.
Note on
__
[ISDN implementation options
are shown, arranged by
country, in a table below – soon
after E1 Parameter Definitions.]
Country &
Operator
options.
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T1 Parameter Definitions (cont’d)
Field Name
Values
Description
Line Build Out 0 dB, -7.5 dB,
-15 dB, -22.5 dB
To reduce the crosstalk on
received signals, a transmit
attenuator can be placed in the
data path. Transmit attenuation
is selectable. Default: O dB
Refers to length of cable
between MultiVOIP and
PBX/telco in meters. Most
common will be 0 to 40m.
Pulse Shape
Level
0 to 40 Meters
40 to 81 m
81 to 122 m
122 to 162 m
162 to 200 m
Clocking
External/Internal Set opposite to telco/PBX
setting. Example: if telco
clocking internal, set VOIP
clocking as external.
Line Coding
PCM Law
AMI / B8ZS
Match to PBX or telco.
A-Law/Mu-Law
Match to PBX or telco. “
Mu-law” is analog-to-digital
compression/expansion
standard used in North
America. “A-law” is European
standard.
Yellow Alarm
Format
Bit 2 / 1111…
Depending on the Frame
Format used, there are choices
of Yellow Alarm format, as
follows:
D4: -Bit2 = 0 in every speech
channel
-FS bit of frame 12 is forced
to one.
ESF: -Bit2 = 0 in every speech
channel
–1111111100000000 pattern
in data link channel.
Check with your PBX/telco
administrator for the correct
setting or use the default value
(1111 … ).
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E1 Parameters. The parameters applicable to E1 and their values are
shown in the figure below. These E1 Parameter fields are described in
the tables that follow.
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Technical Configuration (T1/E1)
E1 Parameter Definitions
Field Name
Values
E1
Description
T1/E1/ISDN
European standard.
Long-Haul
Mode
Y/N
In Long-Haul Mode, the
MultiVOIP automatically
recovers received signals as low
as –36 dB. The maximum
reachable length with 22 AWG
cable is 2000 meters. When
Long-Haul Mode is disabled,
signals as low as –10 dB can be
received.
Default: disabled.
CRC Check
--
Not applicable to E1.
(Cyclic
Redundancy
Check)
Frame Format of MultiVOIP
should match that used by PBX
or telco.
Frame Format
Double Frame;
MultiFrame
(with CRC4);
MultiFrame
(w/CRC4,
modified)
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E1 Parameter Definitions (cont’d)
Values Description
Field Name
CAS Protocol
E&M Immed Strt Channel Associated Signaling
E&M Wink Start
(CAS) is a method of
incorporating telephony
signaling info into an E1
E&M Wink with
dial tone
voice/data stream. In CAS, the
signaling bits (the A, B, C, and
D bits) are multiplexed into the
signal stream of each E1
FXO Ground Strt
FXO Loop Start
FXS Ground Strt
FXS Loop Start
channel. (By contrast, in
MFR2ITU
MFR2 China
MFR2 ANI
Common Channel Signaling
(CCS), one channel handles
signaling for all other channels.)
Each CAS protocol defines the
states of the signaling bits
during the various stages of a
call (IDLE, SEIZED, ANSWER,
RING-ON, RING-OFF).
The CAS protocol code allows
the VOIP to interact properly
with the PBX or central-office
switch that it serves. The need
to download CAS protocols
arises for only a small minority
of VOIP users, and only when
PBX/switch is found to be
incompatible with standard
protocols.
Match this parameter to the
setting of PBX or central-office
switch.
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E1 Parameter Definitions (cont’d)
Field Name
Values
Description
Detect Flash
Hook
Y/N
This setting determines whether
or not the MultiVOIP responds
to hook-flash signals.
Detection Time 100 – 1500
milliseconds
Minimum hook-flash time that
will be interpreted as a valid
flash by the MultiVOIP.
Generation
Time
100 – 1500
In some systems, a MultiVOIP
might receive a hook-flash
signal from an upstream device
(a PBX, voip or other device)
and must replicate it to a
downstream device. This
parameter determines the
duration of the hook-flash
signal that is passed to a
downstream device.
milliseconds
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E1 Parameter Definitions (cont’d)
ISDN Parameters
Field Name
Values
Description
Enable
ISDN-PRI
Y/N
If digital connection is ISDN-
PRI type, this box should be
checked. When ISDN is
enabled, the “CAS Protocols”
field is grayed out (ISDN has its
own signaling method).
Terminal/
Network
either
“Terminal” or
“Network”
When “Terminal” is selected, it
indicates that the MultiVOIP
should emulate the subscriber
(terminal) side of the digital
connection. When “Network”
is selected, it indicates that the
MultiVOIP should emulate the
central office (network) side of
the digital connection.
Setting used for MultiVOIP
must be opposite to the setting
used in the PBX. For example,
if the PBX is set to “Terminal,”
then the MultiVOIP must be set
to “Network.”
Country
Operator
see table, later
this chapter
Country in which MultiVOIP is
operating with ISDN.
see table, later
this chapter
Indicates phone switch
manufacturer/model or refers
to telco so as to specify the
switching system in question.
ISDN is implemented
somewhat differently in
different switches.
Note on
__
[ISDN implementation options
are shown, arranged by
country, in a table below – soon
after E1 Parameter Definitions.]
Country &
Operator
options.
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Technical Configuration (T1/E1)
E1 Parameter Definitions (cont’d)
Field Name
Values
Description
Line Build Out 0 dB, -7.5 dB,
-15 dB, -22.5 dB
To reduce the crosstalk on
received signals, a transmit
attenuator can be placed in the
data path. Transmit attenuation
is selectable. Default: O dB
Refers to length of cable
between MultiVOIP and
PBX/telco in meters. Most
common will be 0 to 40m.
Pulse Shape
Level
0 to 40 Meters
40 to 81 m
81 to 122 m
122 to 162 m
162 to 200 m
Clocking
External/Internal Set opposite to telco/PBX
setting. Example: if telco
clocking internal, set VOIP
clocking as external.
Line Coding
PCM Law
AMI / HDB3
Match to PBX or telco.
A-Law/Mu-Law
Match to PBX or telco.
“A-law” is analog-to-digital
compression/expansion
standard used in Europe.
“Mu-law” is North American
standard.
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Technical Configuration (T1/E1)
MultiVOIP User Guide
10. Set ISDN Parameters (if applicable). These parameters are acces-
sible in the T1/E1/ISDN Parameters screen. If your T1 or E1 phone line
is a Primary Rate Interface ISDN line, enable ISDN-PRI and set it for the
particular implementation of ISDN that your telco uses. The ISDN
types supported by the digital MultiVOIP units (at press time) are listed
below, organized by country.
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Technical Configuration (T1/E1)
11. Set SNMP Parameters (Remote Voip Management). This dialog
box can be reached by pulldown menu, keyboard shortcut, or
sidebar. To make the MultiVOIP controllable by a remote PC
running the MultiVoipManager software, check the “Enable SNMP
Agent” box on the SNMP Parameters screen.
Accessing “SNMP Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + M
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In each field, enter the values that fit your particular system.
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Technical Configuration (T1/E1)
The SNMP Parameter fields are described in the table below.
SNMP Parameter Definitions
Field Name
Values
Description
Enable SNMP
Agent
Y/N
Enables the SNMP code in the
firmware of the MultiVOIP. This
must be enabled for the MultiVOIP
to communicate with and be
controllable by the
MultiVoipManager software.
Default: disabled
Trap Manager Parameters
Address
4 places; n.n.n.n
n = 0-255
IP address of MultiVoipManager
PC.
Community
Name
--
A “community” is a group of VOIP
endpoints that can communicate
with each other. Often “public” is
used to designate a grouping where
all end users have access to entire
VOIP network. However, calling
permissions can be configured to
restrict access as needed.
The default port number of the
SNMP manager receiving the traps
is the standard port 162.
Port Number
162
Community
Name 1
Length = 19
characters (max.)
Case sensitive.
First community grouping.
Permissions
Read-Only,
Read/Write
If this community needs to change
MultiVOIP settings, select
Read/Write. Otherwise, select
Read-Only to view settings.
Community
Name 2
Length = 19
characters (max.)
Case sensitive.
Second community grouping
If this community needs to change
MultiVOIP settings, select
Read/Write. Otherwise, select
Read-Only to view settings.
Permissions
Read-Only,
Read/Write
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12. Set Regional Parameters (Phone Signaling Tones & Cadences and
setup for built-in Remote Configuration/Command Modem).
This dialog box can be reached by pulldown menu, keyboard
shortcut, or sidebar.
Accessing “Regional Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + R
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The Regional Parameters screen will appear. For the country selected,
the standard 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. Each MVP2410 or MVP3010
unit contains a built-in modem. This modem allows the MultiVOIP
to be configured remotely when a standard POTS line is connected to
the “Command Modem” connector on the back panel of the
MultiVOIP. In the Country Selection for Built-In Modem field
(drop-down list), select the country that best fits your situation. This
may not be the same as your selection for the Country/Region field.
The selections in the Country Selection for Built-In Modem field
entail more detailed groupings of telephony parameters than do the
Country/Region values.
In each field, enter the values that fit your particular system.
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The Regional Parameters fields are described in the table below.
“Regional Parameter” Definitions
Field Name
Values
Description
Country/
Region
USA, Japan, UK,
Custom
Name of a country or region that
uses a certain set of tone pairs for
dial tone, ring tone, busy tone,
unobtainable tone (fast busy tone),
survivability tone (tone heard
briefly, 2 seconds, after going
offhook denoting survivable mode
of VOIP unit), and re-order tone (a
tone pattern indicating the need for
the user to hang up the phone).
In some cases, the tone-pair scheme
denoted by a country name may
also be used outside of that
country. The “Custom” option
(button) assures that any tone-
pairing scheme worldwide can be
accommodated.
Type column
dial tone,
ring tone,
busy tone,
Type of telephony tone-pair for
which frequency, gain, and
cadence are being presented.
unobtainable
tone (fast busy),
re-order tone.
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“Regional Parameter” Definitions
Field Name
Values
Description
Frequency 1
frequency in
Hertz
Lower frequency of pair.
Higher frequency of pair.
Frequency 2
Gain 1
frequency in
Hertz
gain in dB
+3dB to –31dB
and “mute”
setting
Amplification factor of lower
frequency of pair.
This figure describes amplification
that the MultiVOIP
applies to outbound tones entering
the MultiVOIP at
the T1 port. Default: -16dB
Gain 2
gain in dB
+3dB to –31dB
and “mute”
setting
Amplification factor of higher
frequency of pair.
This figure describes amplification
that the MultiVOIP
applies to outbound tones entering
the MultiVOIP at
the T1 port. Default: -16dB
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“Regional Parameter” Definitions (cont’d)
Field Name
Values
Description
Cadence
n/n/n/n
On/off pattern of tone durations
used to denote phone ringing,
phone busy, connection
(msec) On/Off four integer time
values in
milli-seconds;
zero value for
dial-tone
unobtainable (fast busy), and dial
tone (continuous and described as
“0“). Default values differ for
different countries/regions.
indicates
continuous tone
Although most cadences have only
two parts (an “on” duration and
an “off” duration), some telephony
cadences have four parts. Most
cadences, then, are expressed as
two iterations of a two-part
sequence. Although this is
redundant, it is necessary to allow
for expression of 4-part cadences.
Click on the “Custom” button to
bring up the Custom Tone Pair
Settings screen. This screen allows
the user to specify tone pair
attributes that are not found in any
of the standard national/regional
telephony toning schemes.
Custom
(button)
--
MultiVOIP units operating with
the X.06 software release (and
above) include a built-in modem.
The administrator can dial into this
modem to configure the MultiVOIP
unit remotely. The country name
values in this field set telephony
parameters that allow the modem
to work in the listed country. This
value may be different than the
Country/Region value. For
example, a user may need to
choose “Europe” as the
Country
Selection for
Built-In
country name
Modem
Country/Region value but
“Denmark” as the Country-
Selection-for-Built-In-Modem
value.
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13. Set Custom Tones and Cadences (optional) . The Regional
Parameters dialog box has a secondary dialog box that allows you to
customize DTMF tone pairs to create unique ring-tones, dial tones,
busy-tones “unobtainable” tones (fast busy signal) or “re-order” tones
(telling the user that they must hang up an off-hook phone) for your
system. This screen allows the user to specify tone-pair attributes
that are not found in any of the standard national/regional telephony
toning schemes. To access this customization feature, click on the
Custom button on the Regional Parameters screen.
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The Custom Tone-Pair Settings fields are described in the table below.
Custom Tone-Pair Settings Definitions
Field Name
Values
Description
Tone Pair
dial tone
busy tone
Identifies the type of telephony
signaling tone for which
ring tone,
frequencies are being specified.
‘unobtainable’ &
re-order tones
TONE PAIR VALUES
About Defaults: US telephony
values are used as defaults on
this screen. However, since this
dialog box is provided to allow
custom tone-pair settings,
default values are essentially
irrelevant.
Frequency 1
Frequency 2
Gain 1
frequency in
Hertz
Frequency of lower tone of pair.
This outbound tone pair enters
the MultiVOIP at the T1/E1 port.
frequency in
Hertz
Frequency of higher tone of pair.
This outbound tone pair enters
the MultiVOIP at the T1/E1 port.
Amplification factor of lower
frequency of pair. This figure
describes amplification that the
MultiVOIP applies to outbound
tones entering the MultiVOIP at
the T1 port. Default = -16dB
gain in dB
+3dB to –31dB
and “mute”
setting
Gain 2
gain in dB
+3dB to –31dB
and “mute”
setting
Amplification factor of higher
frequency of pair. This figure
describes amplification that the
MultiVOIP applies to outbound
tones entering the MultiVOIP at
the T1 port. Default = -16dB
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Custom Tone-Pair Settings Definitions
Field Name
Values
Description
Cadence 1
integer time
value in
milli-seconds;
zero value for
dial-tone
On/off pattern of tone durations
used to denote phone ringing,
phone busy, connection
unobtainable tone (fast busy),
dial tone (which is continuous
and described as “0“) & re-
order tone. Cadence 1 is
indicates
continuous tone
duration of first period of tone
being “on” in the cadence of the
telephony signal (which could be
ring-tone, busy-tone,
unobtainable tone, dial tone, or
re-order tone).
Cadence 2
duration in
milliseconds
Cadence 2 is duration of first
“off” period in signaling
cadence.
Cadence 3
Cadence 4
duration in
milliseconds
Cadence 3 is duration of second
“on” period in signaling cadence.
Cadence 4 is duration of second
“off” period in the signaling
cadence, after which the 4-part
cadence pattern of the telephony
signal repeats.
duration in
milliseconds
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14. Set SMTP Parameters (Log Reports by Email). The SMTP
Parameters screen is applicable when the VOIP administrator has
chosen to receive log reports by email (this is done by selecting the
“SMTP” checkbox in the Others screen and selecting “Enable SMTP”
in the SMTP Parameters screen.). 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 will
actually be given its own email account (with Login Name and
Password) on some mail server connected to the IP network. Using this
account, the MultiVOIP will then send out email messages containing
log report information. The “Recipient” of the log report email is
ordinarily the VoIP administrator. Because the MultiVOIP cannot
receive email, a “Reply-To” address must also be set up. Ordinarily,
the “Reply-To” address is that of a technician who has access to the
mail server or MultiVOIP or both, and the VoIP administrator might
also be designated as the “Reply-To” party. The main function of the
Reply-To address is to receive error or failure messages regarding the
emailed reports.
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The SMTP Parameters screen is shown below.
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“SMTP Parameters” Definitions
Field Name
Values
Description
Enable SMTP
Y/N
In order to send log reports by
email, this box must be checked.
However, to enable SMTP
functionality, you must also select
“SMTP” in the Logs screen.
Requires
Authentication
Y/N
If this checkbox is checked, the
MultiVOIP will send Authentication
information to the SMTP server.
The authentication information
indicates whether or not the email
sender has permission to use the
SMTP server.
Login Name
Password
alpha-
numeric, per
email domain
This is the User Name for the
MultiVOIP unit’s email account.
alpha-
numeric
Login password for MultiVOIP
unit’s email account.
This mail server must be accessible
on the IP network to which the
MultiVOIP is connected.
Mail Server IP
Address
n.n.n.n
for n= 0 to
255
25 is a standard port number for
SMTP.
Port Number
25
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......
“SMTP Parameters” Definitions (cont’d)
Field Name
Values
Description
Mail Type
text or html
Mail type in which log reports will
be sent.
Subject
text
User specified. Subject line that will
appear for all emailed log reports for
this MultiVOIP unit.
User specified. This email address
functions as a source email identifier
for the MultiVOIP, which, of course,
cannot usefully receive email
messages. The Reply-To address
provides a destination for returned
messages indicating the status of
messages sent by the MultiVOIP
(esp. to indicate when log report
email was undeliverable or when an
error has occurred).
Reply-To
Address
email address
User specified. Email address at
which VOIP administrator will
receive log reports.
Recipient
Address
email address
Criteria for sending log summary by
email.
Mail Criteria
The log summary email will be sent
out either when the user-specified
number of log messages has
accumulated, or once every day or
multiple days, which ever comes first.
This is the number of log records
that must accumulate to trigger the
sending of a log-summary email.
This is the number of days that must
pass before triggering the sending of
a log-summary email.
Number of
Records
integer
integer
Number of
Days
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The SMTP Parameters dialog box has a secondary dialog box, Custom
Fields, that allows you to customize email log messages for the
MultiVOIP. The MultiVOIP software logs data about many aspects of
the call traffic going through the MultiVOIP. The Custom Fields screen
lets you pick which aspects will be included in the email log reports.
“Custom Fields” Definitions
Field
Description
Field
Description
Select All Log report to
include all fields
shown.
Channel
Number
Data channel
carrying call.
Start
Date,
Time
Call
Date and time the
phone call began.
Duration Length of call.
Voice or fax.
Mode
Total packets
received in call.
Packets
Sent
Bytes
Sent
Total packets sent
in call.
Total bytes sent in
call.
Packets
Received
Bytes
Total bytes received
in call.
Received
Packets
Lost
Packets lost in
call.
Coder
Voice Coder
/Compression Rate
used for call will be
listed in log.
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“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
Outbound Digits put out by
Prefix
Matched
When selected, the
phonebook prefix
matched in
Digits
MultiVOIP onto
the T1 or E1 line.
processing call will be
listed in log.
Call
Status
Successful or
unsuccessful.
To Details
From Details
Completing or
terminating gateway
Gateway
Number
IP Addr
Originating
gateway
IP address where
call originated.
Gatew N.
IP Addr
IP address where call
was completed or
terminated.
Identifier of site
where call was
completed or
Descript
Options
Identifier of site
where call
originated.
Descript
Options
terminated.
When selected, log
will not use/non-use
of Silence
Compression and
Forward Error
Correction by call
terminator.
When selected, log
will not use/non-
use of Silence
Compression and
Forward Error
Correction by call
originator.
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15. Set Log Reporting Method. The Logs screen lets you choose how
the VoIP administrator will receive log reports about the MultiVOIP’s
performance and the phone call traffic that is passing through it. Log
reports can 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
management program (SNMP).
Accessing “Logs” Screen
Pulldown
Icon
Shortcut
Sidebar
Ctrl + Alt + O
If you enable console messages, you can customize the types of
messages to be included/excluded in log reports by clicking on the
“Filters” button and using the Console Messages Filter Settings
screen (see subsequent page). If you use the logging function, select
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the logging option that applies to your VoIP system design. If you
intend to use a SysLog Server program for logging, click in that
Enable check box. The common SysLog logical port number is 514. If
you intend to use the MultiVOIP web browser GUI for configuration
and control of MultiVOIP units, be aware that the web browser GUI
does not support logs directly. However, when the web browser GUI
is used, log files can still be sent to the voip administrator via email
(which requires activating the SMTP logging option in this screen).
“Logs” Screen Definitions
Field Name
Values
Description
Enable
Console
Messages
Y/N
Allows MultiVOIP debugging messages
to be read via a basic tele-
communications program like
HyperTerminal ™ or similar application.
Normally, this should be disabled
because it consumers MultiVOIP pro-
cessing resources. Console messages are
meant for use by tech support personnel.
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“Logs” Screen Definitions (cont’d)
Field Name
Values
Description
Filters (button)
Click to access secondary screen on
where console messages can be
included/excluded by category and on a
per-channel basis. (See the Console
Messages Filter Settings screen on
subsequent page.)
Turn Off Logs
Logs Buttons
Y/N
Disables log reporting function.
Only one of these three log reporting
methods, GUI, SMTP, or SNMP, may be
chosen.
GUI
Y/N
Y/N
Y/N
Y/N
User must view logs at the MultiVOIP
configuration program.
SNMP
SMTP
Log messages will be delivered to the
MultiVoipManager application program.
Log messages will be sent to user-
specified email address.
SysLog Server
Enable
This box must be checked if logging is to
be done in conjunction with a SysLog
Server program. For more on SysLog
Server, see Operation & Maintenance
chapter.
IP Address
Port
n.n.n.n
for n=
0-255
IP address of computer, connected to
voip network, on which SysLog Server
program is running.
514
Logical port for SysLog Server. 514 is
commonly used.
Online
integer
Set the interval (in seconds) at which
logging information will be updated.
Statistics
Updation
Interval
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To customize console messages by category and/or by channel, click on
“Filters” and use the Console Messages Filters Settings screen.
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16. Set Supplementary Services Parameters. This dialog box can be
reached 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
brings to voip telephony functionality once only available with PSTN or
PBX telephony. Supplementary Services features can be used under
H.323 only and not under SIP.
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In each field, enter the values that fit your particular network.
Of the features implemented under Supplementary Services, three are
very closely related: Call Transfer, Call Hold, and Call Waiting. Call
Name Identification is similar but not identical to the premium PSTN
feature commonly known as Caller ID.
Call Transfer. Call Transfer allows one party to re-connect the party
with whom they have been speaking to a third party. The first party
is disconnected when the third party becomes connected. Feature is
invoked by 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 some other call management function. Invoked by
keypad sequence.
Call Waiting. Call Waiting notifies an engaged caller of an
incoming call and allows them to receive a call from a third party
while the party with whom they have been speaking is put on hold.
Invoked by keypad sequence.
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 phone stations. When the home voip is the caller, a plain
English descriptor will be sent to the remote (callee) voip identifying
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the channel over which the call is being originated (for example,
“Calling Party - Omaha Sales Office Line 2”). If that voip channel is
dedicated to a certain individual, the descriptor could say that, as
well (for example “Calling Party - Harold Smith in Omaha”). When
the home voip receives a call from any remote voip, the home voip
sends a status message back to that caller. This message confirms
that the home voip’s phone channel is either busy or ringing or that
a connection has been made (for example, “Busy Party - Omaha
Sales Office Line ”). These messages appear in the Statistics – 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
supplementary parameters for a particular channel, you can apply this
entire set of parameters to another channel by using the Copy Channel
button and its dialog box. To copy a set of Supplementary Services
parameters to all channels, select “Copy to All” and click Copy.
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The Supplementary Services fields are described in the tables below.
Supplementary Services Parameter Definitions
Field Name Values
Description
Select
Channel
1 (MVP-
130/
The channel to be configured is
selected here.
130FXS
1-2 (210);
1-4 (410);
1-8 (810)
Call
Transfer
Enable
Y/N
Select to enable the Call Transfer
function in the voip unit.
This is a “blind” transfer and the
sequence of events is as follows:
Callers A and B are having a
conversation.
Caller A wants to put B into contact
with C.
Caller A dials call transfer sequence.
Caller A hears dial tone and dials
number for caller C.
Caller A gets disconnected while
Caller B gets connected to caller C.
A brief musical jingle is played for the
caller on hold.
The numbers and/or symbols that the
caller must press on the phone keypad to
initiate a call transfer.
The call-transfer sequence can be 1 to 4
characters in length using any
Transfer
Sequence
any
phone
keypad
character
combination of digits or characters
(* or #).
The sequences for call transfer, call
hold, and call waiting can be from 1
to 4 digits in length consisting of any
combination of digits 1234567890*#.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Call Hold
Enable
Y/N
Select to enable Call Hold function in
voip unit.
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 some
other call management function.
Hold
Sequence
phone
keypad
characters
The numbers and/or symbols that the
caller must press on the phone
keypad to initiate a call hold.
The call-hold sequence can be 1 to 4
characters in length using any
combination of digits or characters
(* or #).
Call Waiting Y/N
Enable
Select to enable Call Waiting function
in voip unit.
Retrieve
Sequence
phone
keypad
The numbers and/or symbols that the
caller must press on the phone
characters, keypad to initiate retrieval of a
two
waiting call.
characters
in length
The call-waiting retrieval sequence
can be 1 to 4 characters in length
using any combination of digits or
characters
(* or #).
This is the phone keypad sequence
that a user must press to retrieve a
waiting call. Customize-able.
Sequence should be distinct from
sequence that might be used to
retrieve a waiting call via the PBX or
PSTN.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Call Name
Identification
Enable
Enables CNI function. Call Name
Identification is not the same as Caller
ID. When enabled on a given voip
unit currently being controlled by the
MultiVOIP GUI (the ‘home voip’),
Call Name Identification sends an
identifier and status information to
the administrator of the remote voip
involved in the call. The feature
operates on a channel-by-channel
basis (each channel can have a
separate identifier).
If the home voip is originating the
call, only the Calling Party field is
applicable. If the home voip is
receiving the call, then the Alerting
Party, Busy Party, and Connected
Party fields are the only applicable
fields (and any or all of these could be
enabled for a given voip channel). The
status information confirms back to
the originator that the callee (the
home voip) is either busy, or ringing,
or that the intended call has been
completed and is currently connected.
The identifier and status information
are made available to the remote voip
unit and appear in the Caller ID field
of its Statistics – Call Progress screen.
(This is how MultiVOIP units handle
CNI messages; in other voip brands,
H.450 may be implemented
differently and then the message
presentation may vary.)
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Calling
Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is originating
the call and Calling Party is selected,
then the identifier (from the Caller Id
field) will be sent to the remote voip
unit being called. The Caller Id field
gives the remote voip administrator a
plain-language identifier of the party
that is originating the call occurring
on a specific channel.
This field is applicable only when the
‘home’ voip unit is originating the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
in this example), Call Name
Identification has been enabled,
Calling Party has been enabled as an
Allowed Name Type, and “Omaha
Sales Office Voipchannel 2” has been
entered in the Caller Id field.
When channel 2 of the Omaha voip is
used to make a call to any other voip
phone station (for example, the
Denver office), the message
“Calling Party – Omaha Sales Office
Voipchannel 2” will appear in the
“Caller Id” field of the
Statistics – Call Progress screen
of the Denver voip.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Alerting
Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is receiving the
call and Alerting Party is selected,
then the identifier (from the Caller Id
field) will tell the originating remote
voip unit that the call is ringing.
This field is applicable only when the
‘home’ voip unit is receiving the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
unit in this example), Call Name
Identification has been enabled,
Alerting Party has been enabled as an
Allowed Name Type, and “Omaha
Sales Office Voipchannel 2” has been
entered in the Caller Id field of the
Supplementary Services screen.
When channel 2 of the Omaha voip
receives a call from any other voip
phone station (for example, the
Denver office), the message “Alerting
Party – Omaha Sales Office
Voipchannel 2” will be sent back and
will appear in the Caller Id field of
the Statistics – Call Progress screen of
the Denver voip. This confirms to the
Denver voip that the phone is ringing
in Omaha.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Busy Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is receiving a
call directed toward an already
engaged channel or phone station and
Busy Party is selected, then the
identifier (from the Caller Id field)
will tell the originating remote voip
unit that the channel or called party is
busy.
This field is applicable only when the
‘home’ voip unit is receiving the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
unit in this example), Call Name
Identification has been enabled, Busy
Party has been enabled as an Allowed
Name Type, and “Omaha Sales Office
Voipchannel 2” has been entered in
the Caller Id field of the
Supplementary Services screen.
When channel 2 of the Omaha voip is
busy but still receives a call attempt
from any other voip phone station
(for example, the Denver office), the
message “Busy Party – Omaha Sales
Office Voipchannel 2” will be sent
back and will appear in the Caller Id
field of the Statistics – Call Progress
screen of the Denver voip. This
confirms to the Denver voip that the
channel or phone station is busy in
Omaha.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Connected
Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is receiving a
call and Connected Party is selected,
then the identifier (from the Caller Id
field) will tell the originating remote
voip unit that the attempted call has
been completed and the connection is
made.
This field is applicable only when the
‘home’ voip unit is receiving the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
unit in this example), Call Name
Identification has been enabled,
Connected Party has been enabled as
an Allowed Name Type, and
“Omaha Sales Office Voipchannel 2”
has been entered in the Caller Id field
of the Supplementary Services
screen.
When channel 2 of the Omaha voip
completes an attempted call from any
other voip phone station (for example,
the Denver office), the message
“Connected Party – Omaha Sales
Office Voipchannel 2” will be sent
back and will appear in the Caller Id
field of the Statistics – Call Progress
screen of the Denver voip. This
confirms to the Denver voip that the
call has been completed to Omaha.
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Supplementary Services Definitions (cont’d)
Field Name Values Description
Caller ID
This is the identifier of a specific
channel of the ‘home’ voip unit. The
Caller Id field typically describes a
person, office, or location, for
example, “Harry Smith,” or “Bursar’s
Office,” or “Barnesville Factory.”
Default
--
--
When this button is clicked, all
Supplementary Service parameters
are set to their default values.
Copy
Channel
Copies the Supplementary Service
attributes of one channel to another
channel. Attributes can be copied to
multiple channels or all channels at
once.
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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 the
MultiVOIP software.
First, it is important to note that the default COM port established by
the MultiVOIP program is COM1. Do not accept the default value
until you have checked the COM port allocation on your PC. To do
this, check for COM port assignments in the system resource dialog
box(es) of your Windows operating system. If COM1 is not available,
you must change the COM port setting to COM2 or some other COM
port that you have confirmed as being available on your PC.
The default baud rate is 115,200 bps.
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18. View System Information screen and set updating interval (optional).
This dialog box can be reached by pulldown menu, keyboard shortcut,
or sidebar.
Accessing the “System Information” Screen
Pulldown
Icon
Shortcut
Sidebar
Ctrl + Alt +Y
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This screen presents vital system information at a glance. Its primary
use is in troubleshooting.
System Information Parameter Definitions
Field Name Values
Description
Boot
Version
nn.nn
Indicates the version of the code that
is used at the startup (booting) of the
voip. The boot code version is
independent of the software version.
Firmware
Version
alpha-
numeric
Indicates version of MultiVOIP
firmware.
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System Information Parameter Definitions
Field Name Values
Description
Configur-
ation
Version
nn.nn.nn. Indicates version of MultiVOIP
nn
Configuration software (which
includes screens for IP Parameters,
SNMP Parameters, SMTP Parameters,
Regional Parameters, etc.
alpha-
numeric
Phone Book
Version
numeric
Indicates the version of the inbound
and outbound phonebook portion of
the MultiVOIP software.
IFM Version numeric
Indicates the version of the firmware
running on the MultiVOIP’s Interface
Module, which is its analog telephony
hardware.
Mac
Address
alpha-
numeric
Denotes the number assigned as the
voip unit’s unique node identifier.
Up Time
days:
hours:
mm:ss
Indicates how long the voip has been
running since its last booting.
Hardware
ID
alpha-
numeric
Indicates the version of the
MultiVOIP unit’s circuit board and
components.
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The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
19. Saving the MultiVOIP Configuration. When values have been set
for all of the MultiVOIP’s various operating parameters, click on Save
Setup in the sidebar.
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20. Creating a User Default Configuration. When a “Setup” (complete
grouping of parameters) is being saved, you will be prompted about
designating that setup as a “User Default” setup. A User Default
setup may be useful as a baseline of site-specific values to which you
can easily revert. Establishing a User Default Setup is optional.
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Technical Configuration (Analog/BRI)
Configuring the Analog/BRI MultiVOIP
There are two ways in which the MultiVOIP must be configured before
operation: technical configuration and phonebook configuration.
Technical Configuration. First, the MultiVOIP must be configured to
operate with technical parameter settings that will match the
equipment with which it interfaces. 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 or
regions,
(6) its operation with a mail server on the same IP network (per SMTP
parameters) such that log reports about VoIP telephone call traffic can
be sent to 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
seven categories is what we call “technical configuration” and it is
described in this chapter.
Phonebook Configuration. The second type of configuration that is
required for the MultiVOIP pertains to the phone number dialing
sequences that it will receive and transmit when handling calls. Dialing
patterns will be affected by both the PBX/telephony equipment and the
other VOIP devices that the MultiVOIP unit interacts with. We call this
“Phonebook Configuration,” and, for analog MultiVOIP units, it is
described nominally in Chapter 9: Analog Phonebook Configuration of this
manual. But, in fact, nearly all of the descriptions and examples for
analog phonebook configuration are to be found in Chapter 7 if the
analog voip is operating under the North American telephony scheme,
or in Chapter 8 if the analog voip is operating under a European
telephony scheme. Chapter 2, the Quick Start Instructions, presents
additional examples relevant to the analog voips.
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Local/Remote Configuration. The MultiVOIP must be configured
locally at first (to establish an IP address for the MultiVOIP unit). But
changes to this initial 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 MultiVOIP configuration program is used.
Remote configuration is done through a connection between the
MultiVOIP’s Ethernet (network) port and a computer connected to the
same network. The computer could be miles or continents away from
the MultiVOIP itself. There are two ways of doing remote
configuration and operation of the MultiVOIP unit: (1) using the
MultiVoipManager SNMP program, or (2) using the MultiVOIP web
browser interface program.
MultiVoipManager. MultiVoipManager is an SNMP agent program
(Simple Network Management Protocol) that extends the capabilities of
the MultiVOIP configuration program: MultiVoipManager allows the
user to manage any number of VOIPs on a network, whereas the
MultiVOIP configuration program 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 a time.
MultiVoipManager may (but does not need to) reside on the same PC
as the MultiVOIP configuration program. The MultiVoipManager
program is on the MultiVOIP Product CD. Updates, when applicable,
may be posted at on the MultiTech FTP site. To download, go to
Web Browser Interface. The MultiVOIP web browser GUI gives access
to the same commands and configuration parameters as are available in
the MultiVOIP Windows GUI except for logging functions. When
using the web browser GUI, logging can be done by email (the SMTP
option).
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Functional Equivalence of Interfaces. The MultiVOIP configuration
program is required to do the initial configuration (that is, setting an IP
address for the MultiVOIP unit) so that the VOIP unit can communicate
with the MultiVoipManager program or with the web browser GUI.
Management of the VOIP after that point can be done from any of these
three programs since they all offer essentially the same functionality.
Functionally, either the MultiVoipManager program or the web
browser GUI can replace the MultiVOIP configuration program after
the initial configuration is complete (with minor exceptions, as noted).
WARNING: Do not attempt to interface the MultiVOIP unit with
two control programs simultaneously (that is, by
accessing the MultiVOIP configuration program via
the Command Port and either the
MultiVoipManager program or the web browser
interface via the Ethernet Port). The results of using
two programs to control a single VOIP
simultaneously would be unpredictable.
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Local Configuration
This manual primarily describes local configuration with the Windows
GUI. After IP addresses have been set locally using the Windows GUI,
most aspects of configuration (logging functions are an exception) can
be handled through the web browser GUI, as well (see the Operation and
Maintenance chapter of this manual). In most aspects of configuration,
the Windows GUI and web-browser GUI differ only graphically, not
functionally. For information on SNMP remote configuration and
management, see the MultiVoipManager documentation.
Pre-Requisites
To complete the configuration of the
MultiVOIP unit, you must know several
things about the overall system.
Before configuring your MultiVOIP Gateway unit, you must know the
values for several IP and telephone parameters that describe the IP
network system and telephony system (PBX or telco central office
equipment) with which the digital MultiVOIP will interact. If you plan
to receive log reports on phone traffic by email (SMTP), you must
arrange to have an email address assigned to the VOIP unit on the
email server on your IP network.
IP Parameters
The following parameters must be known about the network (LAN,
WAN, Internet, etc.) to which the MultiVOIP will connect:
Ask your computer network
administrator.
Info needed to operate:
all MultiVOIP models.
9
IP Network Parameters:
Record for each VOIP Site
in System
#
• IP Address
• IP Mask
• Gateway
• Domain Name Server (DNS) Info
(not implemented; for future use)
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Write down the values for these IP parameters. You will need to enter
these values in the “IP Parameters” screen in the Configuration section
of the MultiVOIP software. You must have this IP information about
every VOIP in the system.
Analog Telephony Interface Parameters
(for MVP-130/130FXS/210/410/810)
The following parameters must be known about the PBX or telco
central office equipment to which the analog MultiVOIP will connect:
Needed for:
MVP810
MVP410
Analog Phone Parameters
9
Ask phone company or
telecom manager.
MVP210
MVP130
MVP130FXS
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 a
phone, fax, or KTS (key telephone system)
• If FXO, determine if line will be an analog PBX
extension or an analog line from a telco central office
• If E&M, determine these aspects of the E&M trunk
line 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?
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ISDN-BRI Telephony Parameters
(for MVP-410ST/810ST)
The following parameters must be known about the PBX or telco
central office equipment to which the analog MultiVOIP will connect:
ISDN-BRI Phone Parameters
Needed for:
MVP810ST
MVP410ST
9
Ask phone company or
telecom manager.
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 and
which BRI ports will be terminal side.
Write down the values for these telephony parameters (whether analog
or ISDN-BRI). You will need to enter these values in the “Interface”
screen (analog) or “ISDN Parameters” screen (ISDN-BRI) in the
Configuration section of the MultiVOIP software.
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SMTP Parameters (for email call log reporting)
required if log reports of
VOIP call traffic
Optional
are to be sent by email
SMTP Parameters
Preparation Task:
To: I.T. Department
Ask Mail Server
re: email account for VOIP
administrator to set up
email account (with
password) for the
MultiVOIP unit itself.
Be sure to give a unique
identifier to each
individual MultiVOIP
unit. .
Get the IP address of the
mail server computer, as
well.
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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 must be 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 access them.
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 remote
management software is used).
11. Set Regional Parameters (Phone Signaling Tones & Cadences and
setup for built-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
Configuration program; SNMP, remotely in MultiVoipManager
program; or SMTP, via email).
15. Set Supplementary Services Parameters. The Supplementary
Services screen allows voip deployment of features that are normally
found in PBX or PSTN 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.
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19. Create a User Default Configuration (optional).
When technical configuration is complete, you will need to configure
the MultiVOIP’s inbound and outbound phonebooks. This manual has
separate chapters describing T1 Phonebook Configuration for North-
American-influenced telephony settings and E1 Phonebook
Configuration for Euro-influenced telephony settings.
Local Configuration Procedure (Detailed)
You can begin the configuration process as a continuation of the
MultiVOIP software installation. You can establish your configuration
or modify it at any time by launching the MultiVOIP program from the
Windows Start menu.
1. Check Power and Cabling. Be sure the MultiVOIP is turned on and
connected to the computer via the MultiVOIP’s Command Port (DB9
connector at computer’s COM port; RJ45 connector at MultiVOIP).
2. Start MultiVOIP Configuration Program. Launch the MultiVOIP
program from the Windows Start menu (from the folder location
determined during installation).
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3. Confirm Connection. If the MultiVOIP is set for an available COM
port and 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.)
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In the lower left corner of the screen, the connection status of the
MultiVOIP will be displayed. The messages in the lower left corner
will change as detection occurs. The message “MultiVOIP Found”
confirms that the MultiVOIP is in contact with the MultiVOIP
configuration program. Skip to step 5.
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4. Solving Common Connection Problems.
A. Fixing a COM Port Problem. If the MultiVOIP main screen appears
but is grayed out and seems inaccessible, the COM port that was
specified for its communication with the PC is unavailable and must
be changed. An error message 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
the Connection 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 are currently available, re-allocate COM port resources
in the computer’s MS Windows operating system to make one
available.)
Ctrl + G
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4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by
the computer, 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.
For instructions on MultiVOIP cable connections, see the Cabling
section of Chapter 3.
5. Configuration Parameter Groups: Getting Familiar, Learning
About Access. The first part of configuration concerns IP parameters,
Voice/FAX parameters, Telephony Interface parameters, SNMP
parameters, Regional parameters, SMTP parameters, Supplementary
Services parameters, Logs, and System Information. In the MultiVOIP
software, these seven types of parameters are grouped together under
“Configuration” and each has its own dialog box for entering values.
Generally, you can reach the dialog box for these parameter groups in
one of four ways: pulldown menu, toolbar icon, keyboard shortcut, or
sidebar.
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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
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In each field, enter the values that fit your particular network.
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The IP Parameters fields are described in the table below.
IP Parameter Definitions
Field Name
Values
Description
DiffServ
Parameter
fields
DiffServ PHB (Per Hop Behavior) values
pertain to a differential prioritizing
system for IP packets as handled by
DiffServ-compatible routers. There are 64
values, each with an elaborate technical
description. These descriptions are found in
TCP/IP standards RFC2474, RFC2597, and,
for present purposes, in RFC3246, which
describes the value 34 (34 decimal; 22 hex) for
Assured Forwarding behavior (default for
Call Control PHB) and the value 46 (46
decimal; 2E hexadecimal) for Expedited
Forwarding behavior (default for Voip Media
PHB). Before using values other than these
default values of 34 and 46, consult these
standards documents and/or a qualified IP
telecommunications engineer.
To disable DiffServ, configure both fields to 0
decimal.
The next page explains DiffServ in the
context of the IP datagram.
Call Control
PHB
0 – 63
Value is used to
prioritize call setup IP
packets.
default = 34
Voip Media
PHB
0 – 63
Value is used to
prioritize the RTP/RTCP
audio IP packets.
default = 46
Frame Type
Type II, SNAP
Must be set to match
network’s frame type.
Default is Type II.
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The IP Datagram with Header, Its Type-of-Service field, & DiffServ
bits =>
0
4
8
16 19
24
31
VERS
HLEN
TYPE OF
SERVICE
TOTAL LENGTH
FLAGS
IDENTIFICATION
TIME TO LIVE PROTOCOL
SOURCE IP ADDRESS
DESTINATION IP ADDRESS
IP OPTIONS (if any)
FRAGMENT OFFSET
HEADER CHECKSUM
PADDING …
end of header
DATA
…
The TOS field consists of eight bits, of which only the first six are used. These six
bits are called the “Differentiated Service Codepoint” or DSCP bits.
The Type of Service or “TOS” field
0
1
2
3
4
5
6
7
PRECEDENCE
D
T
R
unused
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 packet
based on the DSCP value. The DiffServ Parameters fields in the MultiVOIP IP
Parameters screen allow you to configure the DSCP bits to values supported by the
router. Specifically, the Voip Media PHB field relates to the prioritizing of audio
packets (RTP and RTCP packets) and the Call Control PHB field relates to the
prioritzing 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; 100010
binary – consider vis-à-vis TOS field above) for Assured Forwarding behavior. The
MultiVOIP Voip Media PHB parameter defaults to the value 46 decimal (2E hex;
101110 binary – consider vis-à-vis TOS field above). To disable DiffServ, configure
both fields to 0 decimal.
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IP Parameter Definitions (cont’d)
Field Name
Values
Description
IP Parameter fields
Enable DHCP
Y/N
Dynamic Host
Configuration Protocol is a
method for assigning IP
address and other IP
parameters to computers on
the IP network in a single
message with great
disabled by
default
flexibility. IP addresses can
be static or temporary
depending on the needs of
the computer.
IP Address
IP Mask
4-places, 0-255
4-places, 0-255
4-places, 0-255.
The unique LAN IP
address assigned to the
MultiVOIP.
Subnetwork address that
allows for sharing of IP
addresses within a LAN.
Gateway
The IP address of the
device that connects your
MultiVOIP to the
Internet.
Enable DNS
Enable DNS
Y/N.
Enables Domain Name
Space/System function
where computer names
are resolved using a
worldwide distributed
database.
(feature not yet
implemented; for
future use)
DNS Parameter fields
Y/N
Default = disabled
Enables Domain Name
Space/System function
where computer names
are resolved using a
worldwide distributed
database.
DNS Server IP
Address
4-places, 0-255.
IP address of specific
DNS server to be used to
resolve Internet
computer names.
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IP Parameter Definitions (cont’d)
Field Name
Values
Description
FTP Parameter fields
FTP Server Y/N
Enable
MultiVOIP unit has an
FTP Server function so
that firmware and other
Default = disabled
See “FTP Server
File Transfers” in important operating
Operation &
Maintenance
chapter.
software files can be
transferred to the voip
via the network.
TSM Routing Option Parameter
fields
Use TDM
Routing for
Intra-Gateway
calls
Y/N;
enabled by
default
Allows calls placed
between ports on the
same MultiVOIP voice
channel board to be
routed over internal
Time Division
Multiplex bus without
conversion to IP. TDM
routing effectively
eliminates the delay
introduced by IP
conversion.
If you require all calls
to be IP routed,
disable the “use TDM
Routing for Intra-
Gateway Calls”
option. Since this is
not normally required,
we generally
recommend leaving
TDM Routing enabled.
Not applicable to MVP-
130/130FXS.
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7. Enable Web Browser GUI (Optional). After an IP address for the
MultiVOIP unit has been established, you can choose to do any further
configuration 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 do configuration work using the web browser GUI, you must first
enable it. To do so, follow the steps below.
A. Set IP address of MultiVOIP unit using the MultiVOIP
Configuration program (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 when prompted.
H. Use web browser GUI to configure or operate MultiVOIP unit.
The configuration screens in the web browser GUI will have the same
content as their counterparts in the Windows GUI; only the graphic
presentation will be different.
For more details on enabling the MultiVOIP web GUI, see the “Web
Browser Interface” section of the Operation & Maintenance chapter of
this manual.
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Technical Configuration (Analog/BRI)
8. Set Voice/FAX Parameters. This dialog box can be reached by
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.
Accessing “Voice/FAX Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + H
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In each field, enter the values that fit your particular network.
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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 a particular channel, you can apply this entire set of
Voice/FAX parameters to another channel by using the Copy Channel
button and its dialog box. To copy a set of Voice/FAX parameters to all
channels, select “Copy to All” and click Copy.
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The Voice/FAX Parameters fields are described in the tables below.
Voice/Fax Parameter Definitions
Field Name Values
Description
Default
--
When this button is clicked, all
Voice/FAX parameters are set to their
default values.
Select
Channel
1-2 (210)
1-4 (410)
1-8 (810)
Channel to be configured is selected
here.
Copy
Channel
--
Copies the Voice/FAX attributes of
one channel to another channel.
Attributes can be copied to multiple
channels or all channels at once.
Not applicable to MVP130.
Voice Gain
Input Gain
--
Signal amplification (or attenuation)
in dB.
Modifies audio level entering voice
channel before it is sent over the
network to the remote VOIP. The
default & recommended value is 0 dB.
+31dB
to
–31dB
Output Gain +31dB
Modifies audio level being output to
the device attached to the voice
channel. The default and
to
–31dB
recommended value is 0 dB.
DTMF Parameters
DTMF Gain --
The DTMF Gain (Dual Tone Multi-
Frequency) controls the volume level
of the DTMF tones sent out for Touch-
Tone dialing.
DTMF Gain, +3dB to Default value: -4 dB. Not to be
High Tones
-31dB & changed except under supervision of
“mute” MultiTech’s Technical Support.
DTMF Gain, +3dB to Default value: -7 dB. Not to be
Low Tones
-31dB & changed except under supervision of
“mute” MultiTech’s Technical Support.
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Voice/Fax Parameter Definitions (cont’d)
Field Name
Values
Description
DTMF Parameters
Duration
(DTMF)
60 – 3000
ms
When DTMF: Out of Band is selected,
this setting determines how long each
DTMF digit ‘sounds’ or is held. Default
= 100 ms. Not supported in 5.02c BRI
software.
DTMF
In/Out of
Band
Out of
Band, or
Inband
When DTMF Out of Band is selected,
the MultiVOIP detects DTMF tones at
its input and regenerates them at its
output. When DTMF Inband is
selected, the DTMF digits are passed
through the MultiVOIP unit as they are
received. In 502c BRI software, “DTMF
Out of Band” can be checked or
unchecked.
FAX Parameters
Fax Enable
Y/N
Enables or disables fax capability for a
particular channel.
Max Baud
Rate
(Fax)
2400, 4800,
7200, 9600,
12000,
Set to match baud rate of fax machine
connected to channel (see Fax machine’s
user manual).
14400 bps
Default = 14400 bps.
Fax Volume -18.5 dB
Controls output level of fax tones. To
be changed only under the direction of
Multi-Tech’s Technical Support.
(Default =
-9.5 dB )
to –3.5 dB
Jitter Value
(Fax)
Default =
400 ms
Defines the inter-arrival packet
deviation (in milliseconds) for the fax
transmission. A higher value will
increase the delay, allowing a higher
percentage of packets to be
reassembled. A lower value will
decrease the delay allowing fewer
packets to be reassembled.
Mode (Fax)
FRF 11;
T.38
(T.38 not
currently
sup-
FRF11 is frame-relay FAX standard using
these coders: G.711, G.728, G.729, G.723.1.
T.38 is an ITU-T standard for storing
and forwarding FAXes via email using
X.25 packets. It uses T.30 fax standards
and includes special provisions to
preclude FAX timeouts during IP
transmissions.
ported)
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
Voice/Fax Parameter Definitions (cont’d)
Coder Parameters
Coder
Manual or Determines whether selection of
Auto-
matic
coder is manual or automatic.
When Automatic is selected, the
local and remote voice channels will
negotiate the voice coder to be used
by selecting the highest bandwidth
coder supported by both sides
without exceeding the Max
Bandwidth setting. G.723, G.729, or
G.711 are negotiated.
Selected
Coder
G.711 a/u Select from a range of coders with
law 64
kbps;
specific bandwidths. The higher the
bps rate, the more bandwidth is
used. The channel that you are
calling must have the same voice
coder selected.
G.726, @
16/24/32
/40 kbps;
G.727, @
nine bps
rates;
Default = G.723.1 @ 6.3 kbps, as
required for H.323. Here 64K of
digital voice are compressed to
6.3K, allowing several simultaneous
conversations over the same
bandwidth that would otherwise
carry only one.
G.723.1 @
5.3 kbps,
6.3 kbps;
G.729,
8kbps;
Net Coder
@
6.4, 7.2, 8,
8.8, 9.6
To make selections from the
Selected Coder drop-down list, the
Manual option must be enabled.
kbps
Max
bandwidth
(coder)
11 – 128
kbps
This drop-down list enables you to
select the maximum bandwidth
allowed for this channel. The Max
Bandwidth drop-down list is
enabled only if the Coder is set to
Automatic.
If coder is to be selected
automatically (“Auto” setting), then
enter a value for maximum
bandwidth.
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Technical Configuration (Analog/BRI)
Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Advanced Features
Description
Silence
Compression
Y/N
Determines whether silence
compression is enabled (checked) for
this voice channel.
With Silence Compression enabled, the
MultiVOIP will not transmit voice
packets when silence is detected,
thereby reducing the amount of
network bandwidth that is being used
by the voice channel.
Default = on.
Echo
Cancellation
Y/N
Y/N
Determines whether echo cancellation is
enabled (checked) for this voice
channel.
Echo Cancellation removes echo and
improves sound quality. Default = on.
Forward
Error
Correction
Determines whether forward error
correction is enabled (checked) for this
voice channel.
Forward Error Correction enables
some of the voice packets that were
corrupted or lost to be recovered. FEC
adds an additional 50% overhead to the
total network bandwidth consumed by
the voice channel.
Default = Off
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
AutoCall/Offhook Alert
Parameters
Auto Call /
Offhook
Alert
AutoCall,
Offhook
Alert
The AutoCall option enables the local
MultiVOIP to call a remote MultiVOIP
without the user having to dial a Phone
Directory Database number. As soon as
you access the local MultiVOIP
voice/fax channel, the MultiVOIP
immediately connects to the remote
MultiVOIP identified in the Phone
Number box of this option.
If the “Pass Through Enable” field is
checked in the Interface Parameters
screen, AutoCall must be used.
The Offhook Alert option applies only
to FXS channels.
The Offhook Alert option works like
this: if a phone goes offhook and yet no
number is dialed within a specific
period of time (as set in the Offhook
Alert Timer field), then that phone will
automatically dial the Alert phone
number for the voip channel. (The Alert
phone number must be set in the
Voice/Fax Parameters | Phone Number
field; if the voip system is working
without a gatekeeper unit, there must
also be a matching phone number entry
in the Outbound Phonebook.). One use
of this feature would be for emergency
use where a user goes off hook but does
not dial, possibly indicating a crisis
situation. The Offhook Alert feature
uses the Intercept Tone, as listed in the
Regional Parameters screen. This tone
will be outputted on the phone that was
taken off hook but that did not dial.
The other end of the connection will
hear audio from the “crisis” end as is it
would during a normal phone call.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
AutoCall/Offhook Alert
Parameters
Auto Call /
Offhook
Alert
AutoCall,
Offhook
Alert
(continued from previous page)
Both functions apply on a channel-by-
channel basis. It would not be
appropriate for either of these functions
to be applied to a channel that serves in
a pool of available channels for general
phone traffic. Either function requires
an entry in the Outgoing phonebook of
the local MultiVOIP and a matched
setting in the Inbound Phonebook of the
remote voip.
Generate
Local Dial
Tone
Y/N
Used for AutoCall only. If selected, dial
tone will be generated locally while the
call is being established between
gateways. The capability to generate
dial tone locally would be particularly
useful when there is a lengthy network
delay.
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Description
AutoCall/Offhook Alert
Parameters
Offhook
Alert Timer
0 – 3000
seconds
The length of time that must elapse
before the offhook alert is triggered and
a call is automatically made to the
phone number listed in the Phone
Number field.
Phone
Number
--
Phone number used for Auto Call
function or Offhook Alert Timer
function. This phone number must
correspond to an entry in the Outbound
Phonebook of the local MultiVOIP and
in the Inbound Phonebook of the
remote MultiVOIP (unless a gatekeeper
unit is used in the voip system).
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
Voice/Fax Parameter Definitions (cont’d) )
Field Name Values
Dynamic Jitter
Description
Dynamic
Jitter Buffer
Dynamic Jitter defines a minimum
and a maximum jitter value for
voice communications. When
receiving voice packets from a
remote MultiVOIP, varying delays
between packets may occur due to
network traffic problems. This is
called Jitter. To compensate, the
MultiVOIP uses a Dynamic Jitter
Buffer. The Jitter Buffer enables the
MultiVOIP to wait for delayed
voice packets by automatically
adjusting the length of the Jitter
Buffer between configurable
minimum and maximum values.
An Optimization Factor adjustment
controls how quickly the length of
the Jitter Buffer is increased when
jitter increases on the network. The
length of the jitter buffer directly
effects the voice delay between
MultiVOIP gateways.
Minimum
Jitter Value
60 to 400
ms
The minimum dynamic jitter buffer
of 60 milliseconds is the minimum
delay that would be acceptable over
a low jitter network.
Default = 150 msec
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Technical Configuration (Analog/BRI)
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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Dynamic Jitter
Description
Maximum
Jitter Value
60 to 400
ms
The maximum dynamic jitter buffer
of 400 milliseconds is the maximum
delay tolerable over a high jitter
network.
Default = 300 msec
Optimizat-
ion Factor
0 to 12
The Optimization Factor
determines how quickly the length
of the Dynamic Jitter Buffer is
changed based on actual jitter
encountered on the network.
Selecting the minimum value of 0
means low voice delay is desired,
but increases the possibility of jitter-
induced voice quality problems.
Selecting the maximum value of 12
means highest voice quality under
jitter conditions is desired at the
cost of increased voice delay.
Default = 7.
Modem Relay
To place modem traffic onto the voip network (an application called “modem relay”),
use Coder G.711 mu-law at 64kbps.
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Technical Configuration (Analog/BRI)
Voice/Fax Parameter Definitions (cont’d) )
Field Name Values
Auto Disconnect
Description
Automatic
Disconnect-
ion
--
The Automatic Disconnection
group provides four options which
can be used singly or in any
combination.
Jitter Value
1-65535
milli-
seconds
The Jitter Value defines the average
inter-arrival packet deviation (in
milliseconds) before the call is
automatically disconnected. The
default is 300 milliseconds. A higher
value means voice transmission will
be more accepting of jitter. A lower
value is less tolerant of jitter.
Inactive by default. When active,
default = 300 ms. However, value
must equal or exceed Dynamic
Minimum Jitter Value.
Call
Duration
1-65535
seconds
Call Duration defines the
maximum length of time (in
seconds) that a call remains
connected before the call is
automatically disconnected.
Inactive by default.
When active, default = 180 sec.
This may be too short for most
configurations, requiring upward
adjustment.
Consecutive 1-65535
Packets Lost
Consecutive Packets Lost defines
the number of consecutive packets
that are lost after which the call is
automatically disconnected.
Inactive by default.
When active, default = 30
Network
Discon-
nection
1 to 65535 Specifies how long to wait before
seconds;
Default =
30 sec.
disconnecting the call when IP
network connectivity with the
remote site has been lost.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
9a. (Analog VOIPs). Set Telephony Interface Parameters. This dialog
box can be reached by pulldown menu, toolbar icon, keyboard shortcut,
or sidebar.
Accessing Telephony Interface Parameters
Pulldown
Icon
Shortcut
Sidebar
Ctrl + I
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
In each field, enter the values that fit your particular network.
The kinds of parameters for which values must be chosen depend on
the type of telephony supervisory signaling or interface used (FXO,
E&M, etc.). We present here the various parameters grouped and
organized by interface type.
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Note that Interface parameters are applied on a channel-by-channel
basis. However, once you have established a set of Interface
parameters for a particular channel, you can apply this entire set of
Voice/FAX parameters to another channel by using the Copy Channel
button and its dialog box. To copy a set of Interface parameters to all
channels, select “Copy to All” and click Copy.
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Technical Configuration (Analog/BRI)
FXS Loop Start Parameters. The parameters applicable to FXS Loop
Start are shown in the figure below and described in the table that
follows.
FXS Loop Start Interface: Parameter Definitions
Field Name
Values
Description
FXS (Loop
Start)
Y/N
Enables FXS Loop Start
interface type.
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FXS Loop Start Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Dialing Options fields
Inter Digit
Timer
1 - 10 seconds
This is the length of time that
the MultiVOIP will wait
between digits. When the time
expires, the MultiVOIP will
look in the outbound
phonebook for the number
entered and place the call
accordingly.
Default = 2.
Message
Waiting
--
Not applicable to FXS Loop
Start interface
Indication
The length of time between the
outputting of DTMF digits.
Default = 100 ms.
Inter Digit
Regeneration
Time
in milliseconds
Maximum number of rings that
the MultiVOIP will issue before
giving up the attempted call.
Ring Count,
FXS
1-99
When enabled, the MultiVOIP
will interrupt loop current in
the FXS circuit to initiate a
disconnection. This tells the
device connected to the FXS
port to hang up. The Multi-
VOIP cannot drop the call; the
FXS device must go on hook.
FXS Options,
Current Loss
Y/N
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Technical Configuration (Analog/BRI)
FXS Loop Start Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Flash Hook Options fields
not applicable to FXS interface
Generation
--
Detection
Range
for Min. and Max.,
50 - 1500
For a received flash hook to be
regarded as such by the
milliseconds
MultiVOIP, its duration must
fall between the minimum and
maximum values given here.
When enabled, this parameter
creates an open audio path
through the MultiVOIP.
Pass Through
Enable
Y/N
If the Pass-Through feature is
enabled, the AutoCall feature
must be enabled for this voip
channel in the Voice/Fax
Parameters screen.
Caller ID fields
The MultiVOIP currently
supports only one
Caller ID Type Bellcore
implementation of Caller ID.
That implementation is Bellcore
type 1 with Caller ID placed
between the first and second
rings of the call.
Caller ID information is a
description of the remote
calling party received by the
called party. The description
has three parts: name of caller,
phone number of caller, and
time of call. The ‘time-of-call’
portion is always generated by
the receiving MultiVOIP unit
(on FXS channel) based on its
date and time setup.
Caller ID
enable
Y/N
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Technical Configuration (Analog/BRI)
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FXS Loop Start Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Caller ID fields
The forms of the ‘Caller Name’
and ‘Caller Phone Number’
differ depending on the IP
transmission protocol used
(H.323, SIP, or SPP) and upon
entries in the phonebook
Caller ID
enable (cont’d)
Y/N
screens of the remote (CID
generating) voip unit. The CID
Name and Number appearing
on the phone at the terminating
FXS end will come either from a
central office switch (showing a
PSTN phone number), or the
phonebook of the remote (CID
sending) voip unit.
The Caller ID feature has dependencies on both the telco central office
and the MultiVOIP phone book. See the diagram series after the FXO
Parameters section below.
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Technical Configuration (Analog/BRI)
FXO Parameters. The parameters applicable to the FXO telephony
interface type are shown in the figure below and described in the table
that follows.
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FXO Interface: Parameter Definitions*
*Not applicable to MVP130FXS
Field Name
Values
Description
Interface, FXO
Y/N
Enables FXO functionality
Dialing Options
Regeneration
Pulse, DTMF
Determines whether digits
generated and sent out will be
pulse tones or DTMF.
Inter Digit
Timer
1 to 10 seconds
This is the length of time that
the MultiVOIP will wait
between digits. When the time
expires, the MultiVOIP will
look in the phonebook for the
number entered.
Default = 2.
Message
Waiting
--
Not applicable to FXO interface.
Indication
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Technical Configuration (Analog/BRI)
FXO Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Dialing Options (cont’d)
Inter Digit
Regeneration
Time
50 to 20,000
milliseconds
The length of time between the
outputting of DTMF digits.
Default = 100 ms.
FXO Disconnect On
There are three possible criteria
for disconnection under FXO:
current loss, tone detection, and
silence detection.
Disconnection can be triggered
by more than one of the three
criteria.
Current Loss
Y/N
Disconnection to be triggered
by loss of current. That is,
when Current Loss is enabled
(“Y”), the MultiVOIP will hang
up the call when it detects a loss
of current initiated by the
attached device.
Current Loss
Detect Timer
integer values
The minimum time required for
(in milliseconds ) detecting the current loss signal
on the FXO interface. In other
words, this is the minimum
length of time the current must
be absent to validate ‘current
loss’ as a disconnection
criterion. Default = 500 ms.
Tone Detection Y/N
Disconnection to be triggered
by a tone sequence.
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Technical Configuration (Analog/BRI)
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FXO Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
FXO Disconnect On (cont’d)
Disconnect
Tone Sequence
1st tone pair
+
These are DTMF tone pairs.
Values for first tone pair are:
*, #, 0, 1-9, and A-D.
2nd tone pair
Values for second tone pair are:
none, 0, 1-9, A-D, *, and #.
The tone pairs 1-9, 0, *, and #
are the standard DTMF pairs
found on phone sets. The tone
pairs A-D are “extended
DTMF” tones, which are used
for various PBX functions.
DTMF Tone Pairs
Low Tones
697Hz
2
5
8
0
3
6
9
#
A
B
C
D
1
4
7
*
770Hz
852Hz
941Hz
High Tones 1209Hz 1336Hz 1447Hz 1633Hz
Silence
Detection
One-Way or
Two-Way
Disconnection to be triggered
by silence in one direction only
or in both directions
simultaneously.
Silence Timer
in seconds
integer value
Y/N
Duration of silence required to
trigger disconnection.
Disconnect on
Call Progress
Tone
Allows call on FXO port to be
disconnected when a PBX
issues a call-progress tone
denoting that the phone station
on the PBX that has been
involved in the call has been
hung up.
Ring Count,
FXO
1-99
Number of rings required
before the MultiVOIP answers
the incoming call.
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Technical Configuration (Analog/BRI)
FXO Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Flash Hook Options fields
Generation
50 - 1500
milliseconds
Length of flash hook that will
be generated and sent out when
the remote end initiates a flash
hook and it is regenerated
locally. Default = 600 ms.
Detection
Range
--
Not applicable to FXO.
Caller ID fields
Caller ID Type Bellcore
The MultiVOIP currently
supports only one
implementation of Caller ID.
That implementation is Bellcore
type 1 with caller ID placed
between the first and second
rings of the call.
Caller ID information is a
description of the remote
calling party received by the
called party. The description
has three parts: name of caller,
phone number of caller, and
time of call. The ‘time-of-call’
portion is always generated by
the receiving MultiVOIP unit
(on FXS channel) based on its
date and time setup. The forms
of the ‘Caller Name’ and ‘Caller
Phone Number’ differ
Caller ID
enable
Y/N
depending on the IP
transmission protocol used
(H.323, SIP, or SPP) and upon
entries in the phonebook
screens of the remote (CID
generating) voip unit. The CID
Name and Number appearing
on the phone at the terminating
FXS end will come either from a
central office switch (showing a
PSTN phone number), or the
phonebook of the remote (CID
sending) voip unit.
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Technical Configuration (Analog/BRI)
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The Caller ID feature has dependencies on both the telco central office
and the MultiVOIP phone book. See the diagram series below.
CID Flow
Call is received
here.
Call originates here
at 1:42pm, May 31.
CID
Terminating
VoIP
CID
Generating
VoIP
Central Office
with
standard telephony
Caller ID service
FXO
FXS
IP
Network
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
Clock:
5-31,
1:42pm
phone of:
Display shows:
H.323 or SPP
Melvin Jones
763-555-8794
Protocol
*
CID Number: 763-555-8794
CID Name: Melvin Jones
Time Stamp: Date: 05/31
Time:1:42pm
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
*
Figure 6-1: Voip Caller ID Case #1 – Call, through telco
central office with standard CID, enters voip system
CID Flow
Call is received
Call originates here
at 4:19pm, July 10.
here.
CID
Generating
VoIP
CID
Ch1
Terminating
VoIP
Central Office
without
FXO
FXS
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
standard telephony
Caller ID service
Ch3
Ch4
Clock:
7/10, 4:19pm
phone of:
Display shows:
CID Number: 423
H.323 Protocol
*
Wilda Jameson
763-555-4071
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 7/10
Time: 4:19pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Remove Prefix Add Prefix Forward/Addr
*
Gatekeeper RAS Param
423
748
Figure 6-2: Voip Caller ID Case #2 – Call, through telco
central office without standard CID, enters H.323 voip system
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
CID Flow
IP
Call is received
here.
Call originates here
at 5:47pm, Sept 27.
Ch1
Terminating
Generating
VoIP
Central Office
without
VoIP
FXO
FXS
Ch2
xx
x
y
y
yz
zz
z
x
x
x
y
yy
zz
zz
standard telephony
Caller ID service
J. Q. P u bl ic
J. Q. P u bl ic
Network
Ch3
Ch4
Clock:
15:26, 5-31
phone of:
Display shows:
SPP Protocol
Henry Brampton
763-555-4077
CID Number: 423
{Channel 2}
Inbound Phone Book
CID Name: Shipping Dept
Remove Prefix Add Prefix Forward/Addr
Time Stamp: Date: 0927
Time: 1747
... if “Description” field in Add/Edit
Inbound Phone Book is used
423
748
Phone Book Configuration
Anoka-Whse-VP3
Gateway Name:
OR
Add/Edit Inbound Phone Book
Use as default entry
CID Number: 423
Remove Prefix: s
Add Prefix:
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 0927
Time: 1747
Channel Number: Channel 2
Description: Shipping Dept
... if “Description” in Add/Edit
Inbound Phone Book is blank
Figure 6-3: Voip Caller ID Case #3 – Call, through telco
central office without standard CID, enters SPP voip system
CID Flow
Call is received
Call originates here
at 4:51pm, Oct 3.
here.
CID
Generating
VoIP
CID
Ch1 FXS
401
Terminating
VoIP
xxxyyyzzzz
J.Q. Public
FXS
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
402
403
phone of:
Nigel Thurston
763-555-9401
Ch3
Ch4
Clock:
10/03, 4:51pm
404
Display shows:
H.323 Protocol
*
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 10/03
Time: 4:51pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
Remove Prefix Add Prefix Forward/Addr
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
*
Gatekeeper RAS Param
423
748
Figure 6-4: Voip Caller ID Case #4 – Remote FXS call on
H.323 voip system
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
CID Flow
Call is received
Call originates here
at 6:17pm, Nov 15.
here.
CID
CID
Ch1
Generating
VoIP
Terminating
VoIP
Central Office
DID
FXS
without
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
standard telephony
Caller ID service
Ch3
Ch4
Clock:
11/15, 6:17pm
phone of:
Display shows:
H.323 Protocol
*
Edwin Smith
763-743-5873
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 11/15
Time: 6:17pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Remove Prefix Add Prefix Forward/Addr
*
Gatekeeper RAS Param
423
748
Figure 6-5: Voip Caller ID Case #5 – Call through telco central
office without standard CID enters DID channel in H.323 voip
system
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
E&M Parameters. The parameters applicable to the E&M telephony
interface type are shown in the figure below and described in the table
that follows.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
E&M Interface Parameter Definitions
*Not applicable to MVP130FXS
Field Name
Interface
Type
Values
E&M
Description
enables E&M functionality
Types 1-5.
Refers to the type of E&M
interface being used.
Mode
Signal
2-wire or 4-wire
Each E&M interface type can be
either 2-wire or 4-wire audio.
Dial Tone or
Wink
When Dial Tone is selected, no
wink is required on the E lead
or M lead in the call initiation or
setup.
When Wink is selected, a wink
is required during call setup.
Wink Timer
(in ms)
integer values,
in milliseconds
This is the length of the wink
for wink signaling.
Applicable only when Signal
parameter is set to “Wink.”
Pass Through
Enable
Y/N
When enabled (“Y”), this
feature is used to create an open
audio path for 2- or 4-wire. The
E&M leads are passed through
the voip transparently.
Applicable only for E&M
Signaling with Dial Tone.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
E&M Interface Parameter Definitions (cont’d)
Field Name
Values
Description
Dialing Options
Inter Digit integer values,
This is the length of time that
the MultiVOIP will wait
Timer
in seconds
between digits. When the time
expires, the MultiVOIP will
look in the phonebook for the
number entered. Default = 2.
Message
Waiting
Indication
Light or None
Allows MultiVOIP to pass
mode-code sequences between
Avaya Magix PBXs to turn on
and off the message-waiting
light on a PBX extension phone.
Mode codes:
*53 + PBX extension
Î turns message light on.
#53 + PBX extension
Î turns message light off.
Signals to turn message-waiting
lights on/off are not sent to
phones connected directly to
the MultiVOIP on FXS
channels, not to other non-
Avaya Magix PBX phone
stations on the voip network.
Inter Digit
Regeneration
Timer
milliseconds
The length of time between the
outputting of DTMF digits.
Default = 100 ms.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
E&M Interface Parameter Definitions (cont’d)
Field Name
Values
Description
Dialing Options (cont’d)
Disconnect on Y/N
Call Progress
Tone
Allows call on FXO port to be
disconnected when a PBX issues a
call-progress tone denoting that
the phone station on the PBX that
has been involved in the call has
been hung up.
Flash Hook Options fields
Generation
integer values, in Length of flash hook that will
milliseconds
be generated and sent out when
the remote end initiates a flash
hook and it is regenerated
locally. Default = 600 ms.
Detection
Range
for Min. and Max.,
50 1500
For a received flash hook to be
regarded as such by the
milliseconds
MultiVOIP, its duration must
fall between the minimum and
maximum values given here.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
DID Parameters. The parameters applicable to the Direct Inward Dial
(DID) telephony interface type are shown in the figure below and
described in the table that follows. The DID interface allows one phone
line to direct incoming calls to any one of several extensions without a
switchboard operator. Of course, one DID line can handle only one call
at a time. The parameters described 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 Definitions
*Not applicable to MVP130FXS
Field Name
Values
Description
Interface
DID-DPO
Enables the customer-premises
side of DID functionality
DID Options
MultiVOIP’s use of DID applies
only for incoming DID calls.
The Start Mode used by the
MultiVOIP must match that
used by the originating
telephony equipment, else DID
calls cannot be completed.
Start Modes
Immediate Start,
Wink Start,
Delay Dial
For Immediate Start, the voip
detects the off-hook condition
initiated by the telco central-
office call and becomes ready to
receive dial digits immediately.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
DID Interface Parameter Definitions (cont’d)
Field Name
Values
Description
DID Options (cont’d)
Start Modes Immediate Start,
For Wink Start, the voip detects
the off-hook condition. Then
the voip reverses battery
polarity for a specified time
(140-290 ms; a “wink”) and then
becomes ready to receive dial
digits.
Wink Start,
Delay Dial
For Delay Dial, the voip detects
detects the off-hook condition.
Then the voip reverses battery
polarity for a specified time
(reverse polarity duration has
wider acceptable range than for
Wink Start) and then becomes
ready to receive dial digits.
Wink Timer
(in ms)
integer values,
in milliseconds
This is the length of the wink
for Wink Start and Delay Dial
signaling modes..
Applicable only when Start
Mode parameter is set to “Wink
Start” or “Delay Dial.”
Dialing Options
Inter Digit
Timer
integer values,
in seconds
This is the length of time that
the MultiVOIP will wait
between digits. When the time
expires, the MultiVOIP will
look in the phonebook for the
number entered.
Default = 2.
Message
Waiting
--
Not applicable to DID-DPO
interface.
Indication
Inter-Digit
Regeneration
Timer
integer values,
in milliseconds
This parameter is applicable
when digits are dialed onto a
DID-DPO channel after the
connection has been made. The
length of time between the
outputting of DTMF digits.
Default = 100 ms.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
9b. (for ISDN-BRI MultiVOIP units). Set ISDN Parameters. This dialog
box can be reached by pulldown menu, toolbar icon, keyboard shortcut,
or sidebar.
Accessing ISDN (BRI) Parameters
Pulldown
Icon
Shortcut
Sidebar
Ctrl + T
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
In the ISDN BRI Parameters screen, select one of the BRI interfaces and
configure it for the particular implementation of ISDN that you will
use. Configure each BRI interface per the requirements of your voip
system. The MVP410ST has two ISDN-BRI interfaces and four
channels; the MVP810ST has four ISDN-BRI interfaces and eight
channels.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
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 a particular interface, you can apply this entire set of
parameters to another interface by using the Copy Interface button and
its dialog box. To copy a set of ISDN BRI parameters to all interfaces,
select “Copy to All” and click Copy.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
ISDN-BRI Parameter Definitions
Field Name
Values
Description
Select BRI
Interface
ISDNn
In this field, you will choose which
ISDN port you are configuring. The
410ST has two ISDN –BRI ports (or
“interfaces”); the 810ST has four
ISDN-BRI ports (or “interfaces”).
Each port has two channels.
for n= 1-2 (410ST)
for n=1-4 (810ST)
Layer 1 Interface
Terminal,
Network,
When “Terminal” is selected, it
indicates that the MultiVOIP should
emulate the subscriber (terminal) side
of the digital connection. When
“Network” is selected, it indicates
that the MultiVOIP should emulate
the central office (network) side of the
digital connection.
Clock Master
If connecting to a telco or PBX then
choose “Terminal.” If connecting to
an ISDN phone or terminal adapter,
then choose “Network.” Default =
Terminal.
ISDN 1 Terminal Clock Master – The
BRI VOIP uses an internal crystal as
the clock master when any one of the
ports is set for Network mode. If all
ports are set for Terminal mode, then
the first port to be activated becomes
the clock master and it gets its clock
from the network. The clock master
can only be activated on ISDN1
interface in Terminal mode. If ISDN1
is selected and any other interface is
set to Network mode, then clock
master can be enabled.
If enabled, ISDN1 is set as the clock
master for all time, whether or not it
is activated. This feature enables the
BRI VOIP to have all of its interfaces
synchronized to other network
equipment and allows the terminal
devices connected to the VOIP to be
synchronized to the network
equipment as well.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
ISDN-BRI Parameter Definitions (continued)
Field Name Values
Description
Dialing options are relevant when the
MultiVOIP provides dial tone either during
an overlap receiving mode or providing a
second dial tone. Default is 2000, which is 2
seconds. Range 250 ms to 10000 ms (1/4 sec
to 10 sec).
Dialing
Inter Digit
Options
Timer
(value in
milliseconds)
Switch Information
Country in which MultiVOIP is operating
with ISDN.
Country
see table
below
Indicates phone switch manufacturer/model
or refers to telco so as to specify the switching
system in question.
Operator
see table
below
ISDN is implemented somewhat differently
in different switches (different software stacks
are used).
PCM Law
A-Law is an analog-to-digital compression
/expansion standard used in Europe.
MU-Law is the North America standard.
Refer to the PCM-Law defaults based on
country and operator below.
A-Law or
MU-Law
TEI n
Assignment
TEI (Terminal Endpoint Identifier) is a
number to uniquely identify each device
connected to the ISDN. TEI Assignment
displays the value for each TEI assigned to
the BRI port. Depending on the layer 1
interface selection (Terminal or Network) and
the country selection, some fields are grayed
out (inactive) as they have no meaning for
this configuration. The TEI range is zero to 63
for Fixed and 64 to 126 for Automatic
assignment.
Terminal or
Network or
both
Automatic
(Point-to-
Point)
Network –
Automatic
(Point-to-
Multipoint)
Fixed (Point-
to-Point (0-
63) TEI 0 &
TEI 1 active
An interface in Network mode has the added
option of Point-to-Multipoint Automatic TEI.
This added option should be used in cases
where the interface, or BRI port, has one or
more terminal devices connected to it. This
option may also help resolve some problems
that occur when set for Point-to-Point
Automatic TEI, even with only one terminal
device connected to the interface. Point-to-
Point Automatic TEI is then to be used when
there is one network device and one terminal
device, default setting. When selecting Fixed
TEI, the connection must be Point-to-Point,
one network and one terminal device.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
ISDN-BRI Parameter Definitions (continued)
Field Name Values
Numbering Details
Description
Calling Party
Number
Type:
Support for the user to select the Calling
Party Number Type.
Unknown,
International,
National,
Net_Spf,
Subscriber,
Local,
Local is valid only for AT5 operator – local
(directory) number.
There may be cases where the default Type
of Number and/or Numbering Plan may
cause conflicts with the network or
terminal device to which the VOIP is
attached. Should this be the case, then
changing to a different Type of Number
might resolve the problem.
Abbreviated
Called Party
Number
Type:
Support for the user to select the Calling
Party Number Type.
Unknown,
International,
National,
Net_Spf,
Subscriber,
Local,
Abbreviated
Number
Plan:
Local is valid only for AT5 operator – local
(directory) number.
There may be cases where the default Type
of Number and/or Numbering Plan may
cause conflicts with the network or
terminal device to which the VOIP is
attached. Should this be the case, then
changing to a different Type of Number
might resolve the problem.
Unknown,
ISDN,
Support for the user to select the Called
Numbering Plan and Number Type.
Telephone,
Data, Telex,
National,
Private,
ISDN/telephony - CCITT E.164/E.163,
Telephony – not in CEPT, Data – CCITT
x.121, Telex – CCITT F.69, National –
Standard, Reserved for extension.
There may be cases where the default Type
of Number and/or Numbering Plan may
cause conflicts with the network or
Reserved
terminal device to which the VOIP is
attached. Should this be the case, then
changing to a different Type of Number
might resolve the problem. Typically,
setting the Called Party Type of Number to
“unknown” and the Number Plan to
“Unknown” will resolve such issues. These
problems normally occur for calls going to
the ISDN from the VOIP (numbers in the
Inbound Phone Book)
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
ISDN-BRI Parameter Definitions (continued)
Field Name Values
Description
MSN Details
MSN n
Right most
digits to be
Multiple Subscriber Numbering (MSN) – In
Euro-ISDN, and some country specific
distinguished variants, it is possible to have several ISDN
numbers for the same BRI or PRI
connection. This feature allows you to
assign different ISDN numbers to different
applications, or different physical
equipment, e.g., modem, phone
instrument, fax machine, etc. When
specifying MSN Numbers, don’t enter the
whole number, but only enough of the
right most digits so that the numbers
allocated to the line can be distinguished.
Right now, a maximum of eight MSN
Numbers is supported.
For all ISDN incoming calls, when the
dialed digits are matching with the
configured MSN number, then, if the Dial
Number corresponding to the matched
MSN number is present, then using that
Dial Number an IP outgoing call is made.
Otherwise, for the matched MSN number,
if no Dial Number is present, dial tone is
provided to the user to dial the actual
digits with which to make the IP outgoing
call.
Copies the ISDN-BRI attributes of one
interface to another interface. Attributes
can be copied to multiple interfaces or to
all interfaces at once.
“Copy
Interface”
button
MSN details are not copied due to the
nature of MSN assignment.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
Country and Operator options for the MVP-410ST/810ST voip units are
listed below.
Australia
Europe
ETSI--A-law
AUSTEL_1--A-law
ETSI--A-law
ECMA_QSIG--A-law
FT_VN6--A-law
France
FT_VN6--A-law
Hong Kong
HK_TEL
A/mu, switch depndnt
default = mu-law
Italy
ETSI--A-law
Japan
NTT--mu-law
KDD--mu-law
Korea
USA
KOREAN_OP
A/mu, switch depndnt
default = mu-law
N_ISDN1--mu-law
N_ISDN2--mu-law
ATT_5E10--mu-law
NT_DMS100--mu-law
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
10. Set SNMP Parameters (Remote Voip Management). This dialog
box can be reached by pulldown menu, keyboard shortcut, or
sidebar. To make the MultiVOIP controllable by a remote PC
running the MultiVoipManager software, check the “Enable SNMP
Agent” box on the SNMP Parameters screen.
Accessing “SNMP Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + M
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
In each field, enter the values that fit your particular system.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
The SNMP Parameter fields are described in the table below.
SNMP Parameter Definitions
Field Name
Values
Description
Enable SNMP
Agent
Y/N
Enables the SNMP code in the
firmware of the MultiVOIP. This
must be enabled for the MultiVOIP
to communicate with and be
controllable by the
MultiVoipManager software.
Default: disabled
Trap Manager Parameters
Address
4 places; n.n.n.n
n = 0-255
IP address of MultiVoipManager
PC.
Community
Name
--
A “community” is a group of VOIP
endpoints that can communicate
with each other. Often “public” is
used to designate a grouping where
all end users have access to entire
VOIP network. However, calling
permissions can be configured to
restrict access as needed.
The default port number of the
SNMP manager receiving the traps
is the standard port 162.
Port Number
162
Community
Name 1
Length = 19
characters (max.)
Case sensitive.
First community grouping.
Permissions
Read-Only,
Read/Write
If this community needs to change
MultiVOIP settings, select
Read/Write. Otherwise, select
Read-Only to view settings.
Community
Name 2
Length = 19
characters (max.)
Case sensitive.
Second community grouping
If this community needs to change
MultiVOIP settings, select
Read/Write. Otherwise, select
Read-Only to view settings.
Permissions
Read-Only,
Read/Write
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
11. Set Regional Parameters (Phone Signaling Tones & Cadences). ).
This dialog box can be reached by pulldown menu, keyboard
shortcut, or sidebar.
Accessing “Regional Parameters”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + R
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
The Regional Parameters screen will appear. For the country selected,
the standard 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. Each MVP410 and MVP810
MultiVOIP unit contains a built-in modem. This modem allows the
MultiVOIP to be configured remotely when a standard POTS line is
connected to the “Command Modem” connector on the back panel of
the MultiVOIP. In the Country Selection for Built-In Modem field
(drop-down list), select the country that best fits your situation. This
may not be the same as your selection for the Country/Region field.
The selections in the Country Selection for Built-In Modem field
entail more detailed groupings of telephony parameters than do the
Country/Region values.
In each field, enter the values that fit your particular system.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
The Regional Parameters fields are described in the table below.
“Regional Parameter” Definitions
Field Name
Country/
Region
Values
USA, Japan, UK,
Custom
Description
Name of a country or region that
uses a certain set of tone pairs for
dial tone, ring tone, busy tone,
unobtainable tone (fast busy tone),
survivability tone (tone heard
briefly, 2 seconds, after going
offhook denoting survivable mode
of VOIP unit), re-order tone (a tone
pattern indicating the need for the
user to hang up the phone), and
intercept tone (a tone that warns an
a party that has gone off hook but
has not begun dialing, within a
prescribed time, that an automatic
emergency or attendant number
will be called; the automatic call
can be used to direct an attendant’s
attention to a disabled or distressed
caller, allowing an appropriate
response to be made).
In some cases, the tone-pair scheme
denoted by a country name may
also be used outside of that
country. The “Custom” option
(button) assures that any tone-
pairing scheme worldwide can be
accommodated.
Note: Intercept tone is applicable
only when the FXS telephony
interface has been chosen in the
Interface screen and when the
AutoCall / OffHook Alert field is set
to OffHook Alert in the Voice/Fax
Parameters screen. The time
allowed for dialing before the
automatic calling process begins is
set in the Offhook Alert Timer field
of the Voice/Fax Parameters
screen.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
“Regional Parameter” Definitions
Field Name
Country/
Region
Values
USA, Japan, UK,
Custom
Description
Name of a country or region that
uses a certain set of tone pairs for
dial tone, ring tone, busy tone, and
‘unobtainable’ tone (fast busy
tone), survivability tone (tone
heard briefly, 2 seconds, after going
offhook denoting survivable mode
of voip unit) and re-order tone (a
tone pattern indicating the need for
the user to hang up the phone). In
some cases, the tone-pair scheme
denoted by a country name may
also be used outside of that
Note:
“Survivability”
tone indicates a
special type of
call-routing
redundancy &
applies to
MultiVantage
voip units only.
country. The “Custom” option
(button) assures that any tone-
pairing scheme worldwide can be
accommodated.
Type column
dial tone,
ring tone,
Type of telephony tone-pair for
which frequency, gain, and
cadence are being presented.
busy tone,
unobtainable
tone (fast busy),
survivability
tone,
re-order tone
Frequency 1
Frequency 2
Gain 1
freq. in Hertz
freq. in Hertz
Lower frequency of pair.
Higher frequency of pair.
gain in dB
+3dB to –31dB
and “mute”
setting
Amplification factor of lower
frequency of pair.
This applies to the dial, ring, busy
and ‘unobtainable’ tones that the
MultiVOIP outputs as audio to the
FXS, FXS, or E&M port. Default: -
16dB
Gain 2
gain in dB
+3dB to –31dB
and “mute”
setting
Amplification factor of higher
frequency of pair.
This applies to the dial, ring, busy,
and ‘unobtainable’ (fast busy) tones
that the MultiVOIP outputs as
audio to the FXS, FXO, or E&M
port. Default: -16dB
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
“Regional Parameter” Definitions (cont’d)
Field Name
Values
Description
Cadence
n/n/n/n
On/off pattern of tone durations
used to denote phone ringing,
phone busy, connection
(msec) On/Off four integer time
values in
milli-seconds;
zero value for
dial-tone
unobtainable (fast busy), dial tone
(“0” indicates continuous tone),
survivability, and re-order. Default
values differ for different
indicates
continuous tone
countries/regions. Although most
cadences have only two parts (an
“on” duration and an “off”
duration), some telephony
cadences have four parts. Most
cadences, then, are expressed as
two iterations of a two-part
sequence. Although this is
redundant, it is necessary to allow
for expression of 4-part cadences.
Click on the “Custom” button to
bring up the Custom Tone Pair
Settings screen. (The “Custom”
button is active only when
Custom
(button)
--
“Custom” is selected in the
Country/Region field.) This screen
allows the user to specify tone pair
attributes that are not found in any
of the standard national/regional
telephony toning schemes.
MultiVOIP units operating with
the X.06 software release (and
above) include a built-in modem.
The administrator can dial into this
modem to configure the MultiVOIP
unit remotely. The country name
values in this field set telephony
parameters that allow the modem
to work in the listed country. This
value may be different than the
Country/Region value. For
example, a user may need to
choose “Europe” as the
Country
Selection for
Built-In
Modem
(not applicable
to MVP-
130/130FXS
MVP210,
MVP410ST, or
MVP810ST)
country name
Country/Region value but
“Denmark” as the Country-
Selection-for-Built-In-Modem
value.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
12. Set Custom Tones and Cadences (optional). The Regional
Parameters dialog box has a secondary dialog box that allows you to
customize DTMF tone pairs to create unique ring-tonesdial-tones,
busy-tones or “unobtainable” tones (fast busy signal) or “re-order”
tones (telling the user that she must hang up an off-hook phone) or
“survivability” tones (an indication of call-routing redundancy) for
your system. This screen allows the user to specify tone-pair
attributes that are not found in any of the standard national/regional
telephony toning schemes. To access this customization feature, click
on the Custom button on the Regional Parameters screen. (The
“Custom” button is active only when “Custom” is selected in the
Country/Region field.)
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The Custom Tone-Pair Settings fields are described in the table below.
Custom Tone-Pair Settings Definitions
Field Name
Values
Description
Tone Pair
dial tone,
busy tone,
ring tone,
‘unobtainable’
tone,
Identifies the type of telephony
signaling tone for which
frequencies are being specified.
survivability
tone,
re-order tone
TONE PAIR VALUES
About Defaults: US telephony
values are used as defaults on
this screen. However, since this
dialog box is provided to allow
custom tone-pair settings,
default values are essentially
irrelevant.
Frequency 1
Frequency 2
Gain 1
frequency in
Hertz
Frequency of lower tone of pair.
This outbound tone pair enters
the MultiVOIP at the input port.
frequency in
Hertz
Frequency of higher tone of pair.
This outbound tone pair enters
the MultiVOIP at the input port.
Amplification factor of lower
frequency of pair. This figure
describes amplification that the
MultiVOIP applies to outbound
tones entering the MultiVOIP at
the input port. Default = -16dB
gain in dB
+3dB to –31dB
and “mute”
setting
Gain 2
gain in dB
+3dB to –31dB
and “mute”
setting
Amplification factor of higher
frequency of pair. This figure
describes amplification that the
MultiVOIP applies to outbound
tones entering the MultiVOIP at
the input port. Default = -16dB
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Technical Configuration (Analog/BRI)
Custom Tone-Pair Settings Definitions
Field Name
Values
Description
Cadence 1
integer time
value in
milli-seconds;
zero value for
dial-tone
On/off pattern of tone durations
used to denote phone ringing,
phone busy, dial tone (“0”
indicates continuous tone)
survivability and re-order.
Cadence 1 is duration of first
period of tone being “on” in the
cadence of the telephony signal
(which could be ring-tone, busy-
tone, unobtainable-tone, or dial
tone).
indicates
continuous tone
Cadence 2
duration in
milliseconds
Cadence 2 is duration of first
“off” period in signaling
cadence.
Cadence 3
Cadence 4
duration in
milliseconds
Cadence 3 is duration of second
“on” period in signaling cadence.
Cadence 4 is duration of second
“off” period in the signaling
cadence, after which the 4-part
cadence pattern of the telephony
signal repeats.
duration in
milliseconds
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
13. Set SMTP Parameters (Log Reports by Email). The SMTP
Parameters screen is applicable when the VOIP administrator has
chosen to receive log reports by email (this is done by selecting the
“SMTP” checkbox in the Others screen and selecting “Enable SMTP”
in the SMTP Parameters screen.). 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 will
actually be given its own email account (with Login Name and
Password) on some mail server connected to the IP network. Using this
account, the MultiVOIP will then send out email messages containing
log report information. The “Recipient” of the log report email is
ordinarily the VoIP administrator. Because the MultiVOIP cannot
receive email, a “Reply-To” address must also be set up. Ordinarily,
the “Reply-To” address is that of a technician who has access to the
mail server or MultiVOIP or both, and the VoIP administrator might
also be designated as the “Reply-To” party. The main function of the
Reply-To address is to receive error or failure messages regarding the
emailed reports.
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Technical Configuration (Analog/BRI)
The SMTP Parameters screen is shown below
“SMTP Parameters” Definitions
Field Name
Values
Description
Enable SMTP
Y/N
In order to send log reports by
email, this box must be checked.
However, to enable SMTP
functionality, you must also select
“SMTP” in the Logs screen.
Requires
Authentication
Y/N
If this checkbox is checked, the
MultiVOIP will send Authentication
information to the SMTP server.
The authentication information
indicates whether or not the email
sender has permission to use the
SMTP server.
Login Name
alpha-
This is the User Name for the
numeric, per
email domain
MultiVOIP unit’s email account.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
.
“SMTP Parameters” Definitions (cont’d)
Field Name
Values
Description
Password
alpha-
numeric
Login password for MultiVOIP
unit’s email account.
This is the mail server’s IP address.
This mail server must be accessible
on the IP network to which the
MultiVOIP is connected.
Mail Server IP
Address
n.n.n.n
for n= 0 to
255
25 is a standard port number for SMTP.
Port Number
Mail Type
25
text or html
Mail type in which log reports will
be sent.
Subject
text
User specified. Subject line that will
appear for all emailed log reports for
this MultiVOIP unit.
User specified. This email address
functions as a source email identifier
for the MultiVOIP, which, of course,
cannot usefully receive email
messages. The Reply-To address
provides a destination for returned
messages indicating the status of
messages sent by the MultiVOIP
(esp. to indicate when log report
email was undeliverable or when an
error has occurred).
Reply-To
Address
email address
User specified. Email address at
which VOIP administrator will
receive log reports.
Recipient
Address
email address
Criteria for sending log summary by
email.
Mail Criteria
The log summary email will be sent
out either when the user-specified
number of log messages has
accumulated, or once every day or
multiple days, which ever comes first.
This is the number of log records
that must accumulate to trigger the
sending of a log-summary email.
This is the number of days that must
pass before triggering the sending of
a log-summary email.
Number of
Records
integer
integer
Number of
Days
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
The SMTP Parameters dialog box has a secondary dialog box, Custom
Fields, that allows you to customize email log messages for the
MultiVOIP. The MultiVOIP software logs data about many aspects of
the call traffic going through the MultiVOIP. The Custom Fields screen
lets you pick which aspects will be included in the email log reports.
“Custom Fields” Definitions
Field
Description
Field
Description
Select All Log report to
include all fields
shown.
Channel
Number
Data channel
carrying call.
Start
Date,
Time
Call
Date and time the
phone call began.
Duration Length of call.
Voice or fax.
Mode
Total packets
received in call.
Packets
Sent
Bytes
Sent
Total packets sent
in call.
Total bytes sent in
call.
Packets
Received
Bytes
Total bytes received
in call.
Received
Packets
Lost
Packets lost in
call.
Coder
Voice Coder
/Compression Rate
used for call will be
listed in log.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
Outbound Digits put out by
Prefix
Matched
When selected, the
phonebook prefix
matched in
Digits
MultiVOIP onto
the phone line.
processing the call
will be listed in log.
Call
Status
Successful or
unsuccessful.
To Details
From Details
Completing or
answering gateway
Gateway
Number
IP Addr
Originating
gateway
IP address where
call originated.
Gatew N.
IP Addr
IP address where call
was completed or
answered.
Identifier of site
where call was
completed or
Descript
Options
Identifier of site
where call
originated.
Descript
Options
answered.
When selected, log
will not use/non-use
of Silence
Compression and
Forward Error
Correction by party
answering call.
When selected, log
will not use/non-
use of Silence
Compression and
Forward Error
Correction by call
originator.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
14. Set Log Reporting Method. The Logs screen lets you choose how
the VoIP administrator will receive log reports about the MultiVOIP’s
performance and the phone call traffic that is passing through it. Log
reports can 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
management program (SNMP).
Accessing “Logs” Screen
Pulldown
Icon
Shortcut
Sidebar
Ctrl + Alt + O
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
If you enable console messages, you can customize the types of
messages to be included/excluded in log reports by clicking on the
“Filters” button and using the Console Messages Filter Settings
screen (see subsequent page). If you use the logging function, select
the logging option that applies to your VoIP system design. If you
intend to use a SysLog Server program for logging, click in that
Enable check box. The common SysLog logical port number is 514. If
you intend to use the MultiVOIP web browser GUI for configuration
and control of MultiVOIP units, be aware that the web browser GUI
does not support logs directly. However, when the web browser GUI
is used, log files can still be sent to the voip administrator via email
(which requires activating the SMTP logging option in this screen).
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
“Logs” Screen Definitions
Field Name
Values
Description
Enable
Console
Messages
Y/N
Allows MultiVOIP debugging messages to be
read via a basic terminal program like
HyperTerminal ™ or equivalent. Normally,
this should be disabled because it uses
MultiVOIP processing resources. Console
messages are meant for tech support
personnel.
Filters (button)
Click to access secondary screen on where
console messages can be included/excluded
by category and on a per-channel basis. (See
the Console Messages Filter Settings screen on
subsequent page.)
Turn Off Logs
Logs Buttons
Y/N
Check to disable log-reporting function.
Only one of these three log reporting
methods, GUI, SMTP, or SNMP, may be
chosen.
GUI
Y/N
Y/N
Y/N
Y/N
User must view logs at the MultiVOIP
configuration program.
SNMP
SMTP
Log messages will be delivered to the
MultiVoipManager application program.
Log messages will be sent to user-specified
email address.
SysLog Server
Enable
This box must be checked if logging is to be
done in conjunction with a SysLog Server
program. For more on SysLog Server, see
Operation & Maintenance chapter.
IP Address
Port
n.n.n.n
for n=
0-255
IP address of computer, connected to voip
network, on which SysLog Server program is
running.
514
Logical port for SysLog Server. 514 is
commonly used.
Online Statistics
Updation
integer
Set the interval (in seconds) at which
logging information will be updated.
Interval
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Technical Configuration (Analog/BRI)
To customize console messages by category and/or by channel, click on
“Filters” and use the Console Messages Filters Settings screen.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
15. Set Supplementary Services Parameters. This dialog box can be
reached 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 brings to voip telephony functionality once only available with
PSTN or PBX telephony. Supplementary Services features can be used
under H.323 only and not under SIP.
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Technical Configuration (Analog/BRI)
In each field, enter the values that fit your particular network.
Of the features implemented under Supplementary Services, three are
very closely related: Call Transfer, Call Hold, and Call Waiting. Call
Name Identification is similar but not identical to the premium PSTN
feature commonly known as Caller ID.
Call Transfer. Call Transfer allows one party to re-connect the party
with whom they have been speaking to a third party. The first party
is disconnected when the third party becomes connected. Feature is
invoked by 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 some other call management function. Invoked by
keypad sequence.
Call Waiting. Call Waiting notifies an engaged caller of an
incoming call and allows them to receive a call from a third party
while the party with whom they have been speaking is put on hold.
Invoked by keypad sequence.
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 phone stations. When the home voip is the caller, a plain
English descriptor will be sent to the remote (callee) voip identifying
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
the channel over which the call is being originated (for example,
“Calling Party - Omaha Sales Office Line 2”). If that voip channel is
dedicated to a certain individual, the descriptor could say that, as
well (for example “Calling Party - Harold Smith in Omaha”). When
the home voip receives a call from any remote voip, the home voip
sends a status message back to that caller. This message confirms
that the home voip’s phone channel is either busy or ringing or that
a connection has been made (for example, “Busy Party - Omaha
Sales Office Line 2”). These messages appear in the Statistics – 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
supplementary parameters for a particular channel, you can apply this
entire set of parameters to another channel by using the Copy Channel
button and its dialog box. To copy a set of Supplementary Services
parameters to all channels, select “Copy to All” and click Copy.
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Technical Configuration (Analog/BRI)
The Supplementary Services fields are described in the tables below.
Supplementary Services Parameter Definitions
Field Name Values
Description
Select
Channel
1 (MVP-
130/
The channel to be configured is
selected here.
130FXS)
1-2 (210);
1-4 (410);
1-8 (810)
Call
Transfer
Enable
Y/N
Select to enable the Call Transfer
function in the voip unit.
This is a “blind” transfer and the
sequence of events is as follows:
Callers A and B are having a
conversation.
Caller A wants to put B into contact
with C.
Caller A dials call transfer sequence.
Caller A hears dial tone and dials
number for caller C.
Caller A gets disconnected while
Caller B gets connected to caller C.
A brief musical jingle is played for the
caller on hold.
The numbers and/or symbols that the
caller must press on the phone keypad to
initiate a call transfer.
The call-transfer sequence can be 1 to 4
characters in length using any
Transfer
Sequence
any
phone
keypad
character
combination of digits or characters
(* or #).
The sequences for call transfer, call
hold, and call waiting can be from 1
to 4 digits in length consisting of any
combination of digits 1234567890*#.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
Supplementary Services Definitions (cont’d)
Field Name Values Description
Call Hold
Enable
Y/N
Select to enable Call Hold function in
voip unit.
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 some
other call management function.
Hold
Sequence
phone
keypad
characters
The numbers and/or symbols that the
caller must press on the phone
keypad to initiate a call hold.
The call-hold sequence can be 1 to 4
characters in length using any
combination of digits or characters
(* or #).
Call Waiting Y/N
Enable
Select to enable Call Waiting function
in voip unit.
Retrieve
Sequence
phone
keypad
The numbers and/or symbols that the
caller must press on the phone
characters, keypad to initiate retrieval of a
two
waiting call.
characters
in length
The call-waiting retrieval sequence
can be 1 to 4 characters in length
using any combination of digits or
characters
(* or #).
This is the phone keypad sequence
that a user must press to retrieve a
waiting call. Customize-able.
Sequence should be distinct from
sequence that might be used to
retrieve a waiting call via the PBX or
PSTN.
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Technical Configuration (Analog/BRI)
Supplementary Services Definitions (cont’d)
Field Name Values
Description
Call Name
Identification
Enable
Enables CNI function. Call Name
Identification is not the same as Caller
ID. When enabled on a given voip
unit currently being controlled by the
MultiVOIP GUI (the ‘home voip’),
Call Name Identification sends an
identifier and status information to
the administrator of the remote voip
involved in the call. The feature
operates on a channel-by-channel
basis (each channel can have a
separate identifier).
If the home voip is originating the
call, only the Calling Party field is
applicable. If the home voip is
receiving the call, then the Alerting
Party, Busy Party, and Connected
Party fields are the only applicable
fields (and any or all of these could be
enabled for a given voip channel). The
status information confirms back to
the originator that the callee (the
home voip) is either busy, or ringing,
or that the intended call has been
completed and is currently connected.
The identifier and status information
are made available to the remote voip
unit and appear in the Caller ID field
of its Statistics – Call Progress screen.
(This is how MultiVOIP units handle
CNI messages; in other voip brands,
H.450 may be implemented
differently and then the message
presentation may vary.)
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Technical Configuration (Analog/BRI)
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Calling
Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is originating
the call and Calling Party is selected,
then the identifier (from the Caller Id
field) will be sent to the remote voip
unit being called. The Caller Id field
gives the remote voip administrator a
plain-language identifier of the party
that is originating the call occurring
on a specific channel.
This field is applicable only when the
‘home’ voip unit is originating the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
in this example), Call Name
Identification has been enabled,
Calling Party has been enabled as an
Allowed Name Type, and “Omaha
Sales Office Voipchannel 2” has been
entered in the Caller Id field.
When channel 2 of the Omaha voip is
used to make a call to any other voip
phone station (for example, the
Denver office), the message
“Calling Party - Omaha Sales Office
Voipchannel 2” will appear in the
“Caller Id” field of the
Statistics - Call Progress screen
of the Denver voip.
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Technical Configuration (Analog/BRI)
Supplementary Services Definitions (cont’d)
Field Name Values
Description
Alerting
Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is receiving the
call and Alerting Party is selected,
then the identifier (from the Caller Id
field) will tell the originating remote
voip unit that the call is ringing.
This field is applicable only when the
‘home’ voip unit is receiving the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
unit in this example), Call Name
Identification has been enabled,
Alerting Party has been enabled as an
Allowed Name Type, and “Omaha
Sales Office Voipchannel 2” has been
entered in the Caller Id field of the
Supplementary Services screen.
When channel 2 of the Omaha voip
receives a call from any other voip
phone station (for example, the
Denver office), the message “Alerting
Party - Omaha Sales Office
Voipchannel 2” will be sent back and
will appear in the Caller Id field of
the Statistics – Call Progress screen of
the Denver voip. This confirms to the
Denver voip that the phone is ringing
in Omaha.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Busy Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is receiving a
call directed toward an already
engaged channel or phone station and
Busy Party is selected, then the
identifier (from the Caller Id field)
will tell the originating remote voip
unit that the channel or called party is
busy.
This field is applicable only when the
‘home’ voip unit is receiving the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
unit in this example), Call Name
Identification has been enabled, Busy
Party has been enabled as an Allowed
Name Type, and “Omaha Sales Office
Voipchannel 2” has been entered in
the Caller Id field of the
Supplementary Services screen.
When channel 2 of the Omaha voip is
busy but still receives a call attempt
from any other voip phone station
(for example, the Denver office), the
message “Busy Party - Omaha Sales
Office Voipchannel 2” will be sent
back and will appear in the Caller Id
field of the Statistics – Call Progress
screen of the Denver voip. This
confirms to the Denver voip that the
channel or phone station is busy in
Omaha.
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Supplementary Services Definitions (cont’d)
Field Name Values
Description
Connected
Party,
Allowed
Name Type
(CNI)
If the ‘home’ voip unit is receiving a
call and Connected Party is selected,
then the identifier (from the Caller Id
field) will tell the originating remote
voip unit that the attempted call has
been completed and the connection is
made.
This field is applicable only when the
‘home’ voip unit is receiving the call.
Example. Suppose a voip system has
offices in both Denver and Omaha. In
the Omaha voip unit (the ‘home’ voip
unit in this example), Call Name
Identification has been enabled,
Connected Party has been enabled as
an Allowed Name Type, and
“Omaha Sales Office Voipchannel 2”
has been entered in the Caller Id field
of the Supplementary Services
screen.
When channel 2 of the Omaha voip
completes an attempted call from any
other voip phone station (for example,
the Denver office), the message
“Connect Party - Omaha Sales Office
Voipchannel 2” will be sent back and
will appear in the Caller Id field of
the Statistics – Call Progress screen of
the Denver voip. This confirms to the
Denver voip that the call has been
completed to Omaha.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
Supplementary Services Definitions (cont’d)
Field Name Values
Description
Caller ID
This is the identifier of a specific
channel of the ‘home’ voip unit. The
Caller Id field typically describes a
person, office, or location, for
example, “Harry Smith,” or “Bursar’s
Office,” or “Barnesville Factory.”
Default
--
--
When this button is clicked, all
Supplementary Service parameters
are set to their default values.
Copy
Channel
Copies the Supplementary Service
attributes of one channel to another
channel. Attributes can be copied to
multiple channels or all channels at
once.
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Technical Configuration (Analog/BRI)
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 the computer running the MultiVOIP software.
First, it is important to note that the default COM port established by
the MultiVOIP program is COM1. Do not accept the default value
until you have checked the COM port allocation on your PC. To do
this, check for COM port assignments in the system resource dialog
box(es) of your Windows operating system. If COM1 is not available,
you must change the COM port setting to COM2 or some other COM
port that you have confirmed as being available on your PC.
The default baud rate is 115,200 bps.
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Technical Configuration (Analog/BRI)
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17. View System Information screen and set updating interval (optional).
This dialog box can be reached by pulldown menu, keyboard shortcut,
or sidebar.
Accessing “System Information” Screen
Pulldown
Icon
Shortcut
Sidebar
Ctrl + Alt +Y
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This screen presents vital system information at a glance. Its primary
use is in troubleshooting.
System Information Parameter Definitions
Field Name Values
Description
Boot
Version
nn.nn
Indicates the version of the code that
is used at the startup (booting) of the
voip. The boot code version is
independent of the software version.
Firmware
Version
alpha-
numeric
Indicates version of MultiVOIP
firmware.
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MultiVOIP User Guide
System Information Parameter Definitions (cont’d)
Field Name Values Description
nn.nn.nn. Indicates version of MultiVOIP
Configur-
ation
Version
nn
alpha-
numeric
Configuration software (which
includes screens for IP Parameters,
SNMP Parameters, SMTP Parameters,
Regional Parameters, etc.
Phone Book
Version
numeric
Indicates the version of the inbound
and outbound phonebook portion of
the MultiVOIP software.
IFM Version numeric
Indicates the version of the firmware
running on the MultiVOIP’s Interface
Module, which is its analog telephony
hardware.
Mac
Address
alpha-
numeric
Denotes the number assigned as the
voip unit’s unique Ethernet address.
Up Time
days:
hours:
mm:ss
Indicates how long the voip has been
running since its last booting.
Hardware
ID
alpha-
numeric
Indicates the version of the
MultiVOIP unit’s circuit board and
components.
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MultiVOIP User Guide
Technical Configuration (Analog/BRI)
The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
18. Saving the MultiVOIP Configuration. When values have been set
for all of the MultiVOIP’s various operating parameters, click on Save
Setup in the sidebar.
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Technical Configuration (Analog/BRI)
MultiVOIP User Guide
19. Creating a User Default Configuration. When a “Setup” (complete
grouping of parameters) is being saved, you will be prompted about
designating that setup as a “User Default” setup. A User Default
setup may be useful as a baseline of site-specific values to which you
can easily revert. Establishing a User Default Setup is optional.
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T1 Phonebook Configuration
MultiVOIP User Guide
Configuring the MVP2410
MultiVOIP Phonebooks
When a VoIP serves a PBX system, it’s important that the operation of
the VoIP be transparent to the telephone end user. That is, the VoIP
should not entail the dialing of extra digits to reach users elsewhere on
the network that the VoIP serves. On the contrary, VOIP service more
commonly reduces dialed digits by allowing users (served by 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.
Furthermore, the setup of the VoIP generally should allow users to
make calls on a non-toll basis to any numbers accessible without toll by
users at all other locations on the VoIP system. Consider, for example,
a company with VOIP-equipped offices in New York, Miami, and Los
Angeles, each served by its own PBX. When the VOIP phone books are
set correctly, personnel in the Miami office should be able to make calls
without toll not only to the company’s offices in New York and Los
Angeles, but also to any number that’s local in those two cities.
To achieve transparency of the VoIP telephony system and to give full
access to all types of non-toll calls made possible by the VOIP system,
the VoIP administrator 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
dialing sequences required for a call to originate locally (typically in a
PBX in a particular facility) and reach any of its possible destinations at
remote VoIP sites, including non-toll calls completed in the PSTN at the
remote site.
The “Inbound” phonebook for a particular VoIP unit describes the
dialing sequences required for a call to originate remotely from any
other VOIP sites in the system, and to terminate on that particular
VOIP.
Briefly stated, the MultiVOIP’s Outbound phonebook lists the phone stations
it can call; its Inbound phonebook describes the dialing sequences that can be
used 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) calls originating from the Miami office and terminating in the New
York (Manhattan) office, and (2) calls originating from the Miami office
and terminating in New York City but off the company’s premises in an
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MultiVOIP User Guide
T1 PhoneBook Configuration
adjacent area code, an area code different than the company’s office but
still a local call from that office (e.g., Staten Island).
The first type of call requires an entry in the Outbound PhoneBook of
the Miami VOIP and a coordinated entry in the Inbound phonebook of
the New York VOIP. These entries would allow the Miami caller to dial
the New York office as if its phones were extensions on the Miami PBX.
The second type of call similarly requires an entry in the Outbound
PhoneBook of the Miami VOIP and a coordinated entry in the Inbound
Phonebook of the New York VOIP. However, these entries will be
longer and more complicated. Any Miami call to New York City local
numbers will be sent through the VOIP system rather than through the
regular toll public phone system (PSTN). But the phonebook entries
can be arranged so that the VOIP system is transparent to the Miami
user, such that even though that Miami user dials the New York City
local number just as they would through the public phone system, that
call will still be completed through the VOIP system.
This PhoneBook Configuration procedure is brief, but it is followed by
an example case. For many people, the example case may be easier to
grasp than the procedure steps. Configuration is not difficult, but all
phone number sequences and other information must be entered
exactly; otherwise connections will not be made.
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T1 Phonebook Configuration
MultiVOIP User Guide
Phonebook configuration screens can be accessed using icons or the
sidebar menu.
Phonebook Icons
Description
Phonebook Configuration
Inbound Phonebook
Entries List
Add Inbound Phonebook
Entry
Edit selected Inbound
Phonebook Entry
Outbound Phonebook
Entries List
Add Outbound
Phonebook Entry
Edit selected Outbound
Phonebook Entry
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T1 PhoneBook Configuration
Phonebook Sidebar Menu
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T1 Phonebook Configuration
MultiVOIP User Guide
1. Go to the PhoneBook Configuration screen (using either the sidebar
or drop-down menu).
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T1 PhoneBook Configuration
In consultation with your VOIP administrator, enter the Gateway Name
determine which protocol you will use (H.323, SIP, or SPP). Then fill in
the IP address, signaling port, and other parameters, as needed. (The
parameters needed for each protocol are different.)
The table below describes all fields in the general PhoneBook
Configuration screen.
PhoneBook Configuration Parameter Definitions
Field Name
Values
Description
Gateway
Name
Y/N
This field allows you to specify
a name for this MultiVOIP.
When placing a call, this name
is sent to the remote
MultiVOIP for display in Call
Progress listings, Logs, etc.
H.323 Parameters
Use Fast Start
Signaling Port
Y/N
Enables the H.323 Fast Start
procedure. May need to be
enabled/disabled for
compatibility with third-party
VOIP gateways.
port
Default: 1720 (H.323)
number
GateKeeper RAS Parameters
IP address of the GateKeeper.
Gatekeeper /
IP Address
n.n.n.n,
for n =
0 - 255
Signaling Port
Well-known port number for
GateKeepers.
Must match port number of
GateKeeper, 1719.
Gatekeeper
Name
alpha-
numeric
string
Optional. The name of the
GateKeeper with which this
MultiVOIP is trying to register.
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T1 Phonebook Configuration
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.
PhoneBook Configuration Parameter Definitions
(cont’d)
GateKeeper RAS Parameters
Field Name
Values
Description
RAS TTL
Value
in seconds
The H.323 Gatekeeper “Time
to Live” value. As soon as a
MultiVOIP gateway registers
with a gatekeeper (allowing
the gatekeeper to control its
call traffic) a countdown timer
begins. The RAS TTL Value is
the interval of the countdown
timer. Before the TTL
countdown expires, the
MultiVOIP gateway needs to
register with the gatekeeper in
order to maintain the
connection. If the MultiVOIP
does not register before the
TTL interval expires, the
MultiVOIP gateway’s
registration with the
gatekeeper will expire and the
gatekeeper will no longer
permit call traffic to or from
that gateway. Calls in
progress will continue to
function even if the gateway
becomes de-registered.
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T1 PhoneBook Configuration
PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values
Description
H.323 Version 4 Parameters
H.323
Multiplexing
(Mux)
Y/N
Signaling for multiple phone
calls can be carried on a single
port rather than opening a
separate signaling port for
each call. This conserves
bandwidth resources.
H.245
Values: Y/N
Tunneling
(Tun)
Description: H.245 messages are
encapsulated within the Q.931 call-signaling
channel. Among other things, the H.245
messages let the two endpoints tell each other
what their technical capabilities are and
determine who, during the call, will be the
client and who the server. Tunneling is the
process of transmitting these H.245 messages
through the Q.931 channel. The same TCP/IP
socket (or logical port) already being used for
the Call Signaling Channel is then also used
by the H.245 Control Channel. This
encapsulation reduces the number of logical
ports (sockets) needed and reduces call setup
time.
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T1 Phonebook Configuration
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PhoneBook Configuration Parameter Definitions
(cont’d)
Values Description
Field Name
H.323 Version 4 Parameters
Parallel H.245
(FS + Tun)
Values: Y/N
Description: FS (Fast Start or Fast Connect) is
a Q.931 feature of H.323v2 to hasten call
setup as well as ‘pre-opening’ the media
channel before the CONNECT message is
sent. This pre-opening is a requirement for
certain billing activities. Under Parallel
H.245 FS + Tun, this Fast Connect feature can
operate simultaneously with H.245
Tunneling (see description above).
Annex –E (AE) Values: Y/N
Description: Multiplexed UDP call signaling
transport. Annex E is helpful for high-
volume voip system endpoints. Gateways
with lesser volume can afford to use TCP to
establish calls. However, for larger volume
endpoints, the call setup times and system
resource usage under TCP can become
problematic. Annex E allows endpoints to
perform call-signaling functions under the
UDP protocol, which involves substantially
streamlined overhead. (This feature should
not be used on the public Internet because of
potential problems with security and
bandwidth usage.)
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MultiVOIP User Guide
T1 PhoneBook Configuration
PhoneBook Configuration Parameter Definitions
(cont’d)
Values Description
Field Name
SIP Proxy Parameters
Signaling Port
Port number on which the
MultiVOIP UserAgent
software module will be
waiting for any incoming SIP
requests.
Use SIP Proxy
Y/N
Allows the MultiVOIP to work
in conjunction with a proxy
server.
Proxy Domain
Name / IP
Address
n.n.n.n
where
n=0-255
Network address of the proxy
server that the voip is using.
Append SIP
Proxy Domain
Name in User
ID
Y/N
When checked, the domain
name of the SIP Proxy serving
the MultiVOIP gateway will be
included as part of the User ID
for that gateway. If
unchecked, the SIP Proxy’s IP
address will be included as
part of the User ID instead of
the SIP Proxy’s domain name.
Port Number
User Name
Logical port number for proxy
communications.
Values: alphanumeric
Description: Identifier used when proxy
server is used in network. If a proxy server is
used in a SIP voip network, all clients must
enter both a User Name and a Password
before being allowed to make a call.
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T1 Phonebook Configuration
MultiVOIP User Guide
PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values & Description
SIP Proxy Parameters
Password
Values: alphanumeric
Description: Password for proxy server
function. See “User Name” description
above.
Re-
Values: numeric (in seconds)
Registration
Time
Description: This is the timeout interval for
registration of the MultiVOIP with a SIP
proxy server. The time interval begins the
moment the MultiVOIP gateway registers
with the SIP proxy server and ends at the
time specified by the user in the Re-
Registration Time field (this field). When/if
registration lapses, call traffic routed to/from
the MultiVOIP through the SIP proxy server
will cease. However, calls in progress will
continue to function until they end.
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MultiVOIP User Guide
T1 PhoneBook Configuration
PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values
Description
Single Port Protocol (SPP)
SPP voip systems can operate
in two modes:
Mode
Direct,
Client, or
Registrar
in the direct mode, where all
voip gateways have static IP
addresses assigned to them; or
in the registrar/client mode,
where one voip gateway
serves as registrar and all
other gateways, being its
clients, point to that registrar.
The registrar assigns IP
addresses dynamically.
General Options
The UDP port on which data
transmission will occur. Each
client voip has its own port. If
two client voips are both
behind the same firewall, then
they must have different ports
assigned to them.
Port
If there are two clients and
each is behind a different
firewall, then the clients could
have different port numbers or
the same port number.
(Default port number = 10000.)
If packets are lost (as indicated
by absence of an
acknowledgment) then the
endpoint will retransmit the
lost packets after this
Re-trans-
mission
(in ms)
designated time duration has
elapsed. (Default value = 2000
milliseconds.)
Number of times the voip will
re-transmit a lost packet (if no
acknowledgment has been
received). (Default value = 3)
Max
Re-trans-
mission
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T1 Phonebook Configuration
MultiVOIP User Guide
PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values
Description
Single Port Protocol (SPP)
[continued]
Client Option fields are active
only in registrar/client mode
and only for client voip units.
This is the IP address of the
registrar voip to which this
client is assigned. (Default
value = 0.0.0.0; effectively,
there is no useful default
value.)
Client Options
Registrar IP
Address
This is the port number of the
registrar voip to which this
client is assigned. (Default port
number = 10000.)
Registrar
Port
Registrar Option fields are
active only in registrar/client
mode and only for registrar
voip units.
Registrar Options
Time-out duration before a
registrar will unregister a
client that does not send its
“I’m here” signal. Client
normally sends its “I’m here”
signal every 20 seconds.
Timeout default = 60 seconds.
Keep Alive
(in sec.)
Proxy/NAT Device
Parameters
Enables MultiVOIP (running
in SPP Registrar mode) to
operate ‘behind’ a proxy/NAT
device (NAT = Network
Address Translation).
Behind
Proxy/NAT
device
Y/N
The public IP address of the
proxy/NAT device which the
MultiVOIP is behind.
Proxy/NAT
Device
Parameters –
Public IP
n.n.n.n
where
n=0-255
Address
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MultiVOIP User Guide
T1 PhoneBook Configuration
About SPP Proxy/NAT Device Parameters
SPP Client/Registrar System
Client
Voip
IP
Public
Client
Voip
Public IP
Private IP
IP
Network
Registrar
Voip
Proxy/NAT
Device
Public IP
Client
Voip
A Proxy/NAT device is sometimes used
in a Client/Registrar SPP voip system
where the registrar voip is in a private
network but serves client voips on a
public network. The Proxy/NAT device
isolates (protects) the registrar voip
from the public network.
Public IP
Client
Voip
In such cases, you must check the
“Enable SPP Proxy/NAT device” checkbox
in the Phonebook Configuration screen
of the Registrar voip. The private registrar
voip can then function with the client voips
using the public IP address of the Proxy/NAT
device. You must enter this address in the
Public IP Address field.
Public IP
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T1 Phonebook Configuration
MultiVOIP User Guide
2. Select PhoneBook Modify and then select
Outbound Phone Book/List Entries.
Fields in the “Details” section will differ depending on the protocol
(H.323, SIP, or SPP) of the selected list entry to which the details
pertain.
Click Add.
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MultiVOIP User Guide
T1 PhoneBook Configuration
3. The Add/Edit Outbound PhoneBook screen appears.
Enter Outbound PhoneBook data for your MVP2410. Note that the
Advanced button gives access to the Alternate IP Routing feature, if
needed. Alternate IP Routing can be implemented in a secondary
screen (as described after the primary screen field definitions below).
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The fields of the Add/Edit Outbound Phone Book screen are described
in the table below.
Add/Edit Outbound Phone Book: Field Definitions
Field Name
Values
Description
Accept Any
Number
Y/N
When checked, “Any
Number” appears as the
value in the Destination
Pattern field.
The Any Number feature
works differently depending
on whether or not an external
routing device is used
(Gatekeeper for H323
protocol, Proxy for SIP
protocol, Registrar for SPP
protocol).
When no external routing
device is used. If Any
Number is selected, calls to
phone numbers not matching
a listed Destination Pattern
will be directed to the IP
Address in the Add/Edit
Outbound Phone Book
screen. “Any Number” can
be used in addition to one or
more Destination Patterns.
When external routing
device is used. If Any
Number is selected, calls to
phone numbers not matching
a listed Destination Pattern
will be directed to the
external routing device used
(Gatekeeper for H323
protocol, Proxy for SIP
protocol, Registrar for SPP
protocol). The IP Address of
the external routing device
must be set in the Phone
Book Configuration screen.
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T1 PhoneBook Configuration
Add/Edit Outbound Phone Book: Field Definitions
(cont’d)
Description
Field Name
Values
Destination
Pattern
prefixes,
area codes,
exchanges,
line
numbers,
extensions
Defines the beginning of
dialing sequences for calls
that will be connected to
another VOIP in the system.
Numbers beginning with
these sequences are diverted
from the PTSN and carried
on Internet or other IP
network.
Total Digits
as needed
This field currently disabled.
number of digits the phone
user must dial to reach
specified destination.
Remove Prefix
dialed digits portion of dialed number to
be removed before
completing call to destination
Add Prefix
IP Address
dialed digits digits to be added before
completing call to destination
n.n.n.n
for
the IP address to which the
call will be directed if it
begins with the destination
pattern given
n = 0-255
Description
alpha-
numeric
Describes the facility or
geographical location at
which the call will be
completed.
Indicates protocol to be used in
outbound transmission. Single
Port Protocol (SPP) is a non-
standard protocol designed by
Multi-Tech.
Protocol Type
SIP or H.323
or SPP
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T1 Phonebook Configuration
MultiVOIP User Guide
Add/Edit Outbound Phone Book: Field Definitions
(cont’d)
Field Name
Values
Description
H.323 fields
Use Gatekeepr Y/N
Indicates whether or not
gatekeeper is used.
The H.323 ID assigned to the
destination MultiVOIP. Only
valid if “Use Gatekeeper” is
enabled for this entry.
Gateway
H.323 ID
alpha-
numeric
Gateway
Prefix
numeric
This number becomes
registered with the
GateKeeper. Call requests
sent to the gatekeeper and
preceded by this prefix will
be routed to the VOIP
gateway.
H.323 Port
Number
1720
This parameter pertains to
Q.931, which is the H.323 call
signaling protocol for setup
and termination of calls (aka
ITU-T Recommendation
I.451). H.323 employs only
one “well-known” port (1720)
for Q.931 signaling. If Q.931
message-oriented signaling
protocol is used, 1720 must be
chosen as the H.323 Port
Number.
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T1 PhoneBook Configuration
Add/Edit Outbound Phone Book: Field Definitions
(cont’d)
Description
Field Name
SIP Fields
Use Proxy
Values
Y/N
Select if proxy server is used.
Transport
Protocol
TCP or
UDP
Voip administrator must choose
between UDP and TCP
transmission protocols. UDP is a
high-speed, low-overhead
connectionless protocol where
data is transmitted without
acknowledgment, guaranteed
delivery, or guaranteed packet
sequence integrity. TCP is slower
connection-oriented protocol
with greater overhead, but
having acknowledgment and
guarantees delivery and packet
sequence integrity.
The SIP Port Number is a
UDP logical port number.
The voip will “listen” for SIP
messages at this logical port.
If SIP is used, 5060 is the
default, standard, or “well
known” port number to be
used. If 5060 is not used,
then the port number used is
that specified in the SIP
Request URI (Universal
Resource Identifier).
SIP Port
Number
5060 or other
*See RFC3087
(“Control of
Service
Context using
SIP Request-
URI,” by the
Network
Working
Group).
Looking similar to an email
address, a SIP URL
SIP URL
sip.userphone
@
identifies a user's address.
In SIP communications, each
caller or callee is identified
by a SIP url:
hostserver,
where
“userphone”
is the
sip:user_name@host_name.
The format of a sip url is very
similar to an email address,
except that the “sip:“ prefix is
used.
telephone
number and
“hostserver”is
the domain
name or an
address on the
network
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T1 Phonebook Configuration
MultiVOIP User Guide
Add/Edit Outbound Phone Book: Field Def’ns (cont’d)
Field Name
SPP Fields
Use Registrar
Values
Description
Values: Y/N
Description: Select this checkbox to use registrar
when voip system is operating in the
“Registrar/Client” SPP mode. In this mode, one
voip (the registrar, as set in Phonebook
Configuration screen) has a static IP address and
all other voips (clients) point to the registar’s IP
address as functionally their own. However, if
your voip system overall is operating in
“Registrar/Client” mode but you want to make an
exception and use Direct mode for the destination
pattern of this particular Add/Edit Phonebook
entry, leave this checkbox unselected.
Leave this checkbox unselected if your overall voip
system is operating in the “Direct" SPP mode. In
this mode, all voips in system are peers and each
has its own static IP address.
Port Number
Values: numeric
Description: When operating in
“Registrar/Client” mode, this is the port by which
the gateway receives all SPP data and control
messages from the registrar gateway. (This ability
to receive all data and messages via one port
allows the voip to operate behind a firewall with
only one port open.)
When operating in “Direct” mode, this is the Port
by which peer voips receive data and messages.
Alternate
numeric
Phone number associated
with alternate IP routing.
When checked, this
MultiVOIP can operate with
‘first-generation’ MultiVOIP
units in the same IP network.
These include MVP-
Phone Number
Remote Device
is [legacy voip]
Y/N
110/120/200/400/800.
Advanced
Values: N/A
button
Description: Gives access to secondary screen
where an Alternate IP Route can be specified
for backup or redundancy of signal paths.
See discussion on next page. For SIP & H.323
operation only.
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T1 PhoneBook Configuration
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 carry
the 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) unit
in case one voip unit fails. The user must specify the IP address of the alternate route
for each destination pattern entry in the Outbound Phonebook.
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T1 Phonebook Configuration
MultiVOIP User Guide
Alternate Routing Field Definitions
Field
Values
Description
Name
Alternate n.n.n.n
Alternate destination for outbound data traffic
in case of excessive delay in data transmission.
IP
where
Address
n= 0-255
Round
Trip
Delay
milliseconds The Round Trip Delay is the criterion for
judging when a data pathway is considered
blocked. When the delay exceeds the
threshold specified here, the data stream will
be diverted to the alternate destination
specified as the Alternate IP Address.
The Alternate Routing function facilitates PSTN Failover protection, that is, it allows
you to re-route voip calls automatically over the PSTN if the voip system fails. The
MultiVOIP can be programmed to respond to excessive delays in the transmission of
voice packets, which the MultiVOIP interprets as a failure of the IP network. Upon
detecting 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 is
connected to the PSTN (for example, via an FXO port on the self-same MultiVOIP
could be connected to the PSTN).
4. Call completed
3. Call diverts to
via PSTN.
PSTN Line
Alt IP address in voip
accessing PSTN line.
FXO
IP
VOIP
VOIP
NETWORK
PBX
FXS
2. IP network fails.
1. Call originates.
PSTN Failover Feature. The MultiVOIP can be programmed to divert calls to the
PSTN temporarily in case the IP network fails.
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T1 PhoneBook Configuration
4. Select PhoneBook Modify and then select Inbound PhoneBook | List Entries.
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5. The Add/Edit Inbound PhoneBook screen appears.
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T1 PhoneBook Configuration
Enter Inbound PhoneBook data for your MultiVOIP. The fields of the
Add/Edit Inbound PhoneBook screen are described in the table below.
Add/Edit Inbound Phone Book: Field Definitions
Field Name
Values
Description
Accept Any
Number
Values: Y/N
Description: When checked, “Any Number”
appears as the value in the Remove Prefix
field.
The Any Number feature of the Inbound
Phone Book does not work when an external
routing device is used (Gatekeeper for H323
protocol, Proxy for SIP protocol, Registrar for
SPP protocol).
When no external routing device is used. If
Any Number is selected, calls received from
phone numbers not matching a listed Prefix
(shown in the Remove Prefix column of the
Inbound Phone Book) will be admitted into
the voip on the channel listed in the Channel
Number field. “Any Number” can be used in
addition to one or more Prefixes.
Remove Prefix
Add Prefix
dialed digits portion of dialed number to
be removed before
completing call to destination
(often a local PBX)
dialed digits digits to be added before
completing call to destination
(often a local PBX)
T1 channel number to which
the call will be assigned as it
enters the local telephony
equipment
Channel
Number
1-24, or
“Hunting”
(often a local PBX).
“Hunting” directs the call to
any available channel.
Description
--
Describes the facility or
geographical location at
which the call originated.
Call Forward Parameters
Enable Y/N
Click the check-box to enable
the call-forwarding feature.
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T1 Phonebook Configuration
MultiVOIP User Guide
Add/Edit Inbound Phone Book: Field Definitions
(cont’d)
Field Name
Values
Description
Call Forward Parameters
Unconditional. When selected,
all calls received will be
forwarded.
Busy. When selected, calls
will be forwarded when
station is busy.
Forward
Condition
Uncondit.;
Busy
No Resp.
No Response. When selected,
calls will be forwarded if
called party does not answer
after a specified number of
rings, as specified in Ring
Count field.
Forwarding can be
conditioned on both “Busy”
and “No Response.”
Forward
Destination
Phone number or IP address to which calls
will be directed.
IP address,
For H.323 calls, the Forward Destination can
phone number, be either a Phone Number or an IP Address.
port number,
etc.
For SIP calls, the Forward Destination can be
one of 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.
For SPP calls, the Forward Destination can be
one of the following:
(a) phone number, (b) IP address: port, or
(c) phone number: IP address: port.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Add/Edit Inbound Phone Book: Field Definitions
(cont’d)
Field Name
Values and Description
Ring Count
0, 1, 2, 3, etc. When “No Response” is
condition for forwarding calls, this
determines how many unanswered rings
are needed to trigger the forwarding.
Registration
Option
Parameters
In an H.323 voip system, gateways can
register with 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
with the SIP Proxy.
In an SPP voip system, gateways can register
with the SPP Registrar voip unit.
6. When your Outbound and Inbound PhoneBook entries are
completed, click on 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
the built-in Remote Configuration/Command Modem using the
MultiVOIP program. After the initial configuration is complete, all of
the MVP2410 units in the VOIP system can be configured, re-
configured, and updated from one location using the MultiVOIP web
GUI software program or the MultiVOIP program (in conjunction with
the built-in modem).
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T1 Phonebook Configuration
MultiVOIP User Guide
T1 Phonebook Examples
The following example demonstrates how Outbound and Inbound
PhoneBook entries work in a situation of multiple area codes. Consider
a company with offices in Minneapolis and Baltimore.
3 Sites, All-T1 Example
Notice first the area code situation in those two cities: Minneapolis’s
local calling area consists of multiple adjacent area codes; Baltimore’s
local calling area consists of a base area code plus an overlay area code.
Company
VOIP/PBX
5
Baltimore/
SIte
Outstate MD
Overlay
443
NW
Suburbs
St. Paul
& Suburbs
651
763
Mpls
612
Company
VOIP/PBX
SIte
...
5
SW Suburbs
952
Baltimore
410
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MultiVOIP User Guide
T1 PhoneBook Configuration
An outline of the equipment setup in both offices is shown below.
Local-Call
Area Codes:
612, 651,
952
Company HQ.
Minneapolis
North Sub.
area 763
T1
Digital
VoIP
PBX
-5174
200.2.10.3
-5173
-5172
-5171
717-5170
IP
Network
Overlay
Area Code:
443
Baltimore
ꢀ
Sales Ofc.
area 410
ꢀ
Digital
VoIP
R
o
u
t
e
r
T1
PBX
-7003
200.2.9.7
-7002
325-7001
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T1 Phonebook Configuration
MultiVOIP User Guide
The screen below shows Outbound PhoneBook entries for the VOIP
located in the company’s Baltimore facility.
The entries in the Minneapolis VOIP’s Inbound PhoneBook match the
Outbound PhoneBook entries of the Baltimore VOIP, as shown below.
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T1 PhoneBook Configuration
To call the Minneapolis/St. Paul area, a Baltimore employee must dial
eleven digits. (In this case, we are assuming that the Baltimore PBX
does not require an “8” or “9” to seize an outside phone line.)
If a Baltimore employee dials any phone number in the 612 area code,
the call will automatically be handled by the company’s voip system.
Upon receiving such a call, the Minneapolis voip will remove the digits
“1612”. But before the suburban-Minneapolis voip can complete the
call to the PSTN of the Minneapolis local calling area, it must dial “9”
(to get an outside line from the PBX) and then a comma (which denotes
a pause to get a PSTN dial tone) and then the 10-digit phone number
which includes the area code (612 for the city of Minneapolis; which is
different than the area code of the suburb where the PBX is actually
located -- 763).
A similar sequence of events occurs when the Baltimore employee calls
number in the 651 and 952 area codes because number in both of these
area codes are local calls in the Minneapolis/St. Paul area.
The simplest case is a cal from Baltimore to a phone within the
Minneapolis/St. Paul area code where the company’s voip and PBX are
located, namely 763. In that case, that local voip removes 1763 and
dials 9 to direct 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 to Minneapolis employees will stay within the
suburban Minneapolis PBX and will not reach or be carried on the local
PSTN.
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T1 Phonebook Configuration
MultiVOIP User Guide
Similarly, the Inbound PhoneBook for the Baltimore VOIP (shown first
below) generally matches the Outbound PhoneBook of the Minneapolis
VOIP (shown second below).
Notice the extended prefix to be removed: 14103257. This entry allows
Minneapolis users to contact Baltimore co-workers as though they were
in the Minneapolis 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. A one-second delay is commonly used to allow a second dial
tone to be generated for calls going outside of the facility’s PBX system.
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MultiVOIP User Guide
T1 PhoneBook Configuration
The Outbound PhoneBook for the Minneapolis VOIP is shown below.
The third destination pattern, “7” facilitates reception of co-worker calls
using local-appearing-extensions only. In this case, the “Add Prefix”
field value for this phonebook entry would be “1410325” .
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T1 Phonebook Configuration
MultiVOIP User Guide
Configuring Mixed Digital/Analog VOIP Systems
The MVP2410 digital MultiVOIP unit is compatible with analog VOIPs.
In many cases, digital and analog VOIP units will appear in the same
telephony/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 same VOIP 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 MVP200 units).
EXAMPLE:
Site D:
Digital & Analog VOIPs
Pierre, SD
in Same System
Area Code 615
PSTN
PBX
200.2.9.9
Digital
VoIP
MVP2410
T1
Other extensions
x3101 - x3199
Router
615-492-3100
Site E:
Site A:
Cheyenne, WY
Area Code 307
Bismarck, ND
Area Code 701
200.2.9.6
Series #1 Analog MultiVOIP
(Server/Client Phonebook)
MVP200
Series #2 Analog MultiVOIP
MVP210
FXS
Unit
FXS
CH1
#200
CH1
421
201
200.2.9.7
Client
IP
Network
Site F:
Lincoln, NE
Area Code 402
Site B:
Rochester, MN
Area Code 507
200.2.9.5
FXO
Series #1 Analog MultiVOIP
(Server/Client Phonebook)
PSTN
Series #2 Analog MultiVOIP
MVP410
Port #4
102
MVP200
CH2
FXS
FXO
Unit
#100
CH1
FXS Port
FXS Ports
CO Ports
717-5000
200.2.9.8
Host
(Holds phonebook for both
Series #1 analog VOIPs.)
CO Port
Key
System
Other extensions
x7401 - x7429
PSTN
402-263-7400
507-717-5662
Site C:
Suburban Rochester
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MultiVOIP User Guide
T1 PhoneBook Configuration
The Series I analog VOIP phone book resides in the “Host” VOIP unit at
Site B. 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
digital MultiVOIP requires its own inbound and outbound
phonebooks, as well.
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T1 Phonebook Configuration
MultiVOIP User Guide
These seven phone books are shown below.
Phone Book for Series I Analog VOIP Host Unit (Site B)
VOIP Dir #
-OR-
IP Address Channel Comments
Destination
Pattern
102
101
200.2.9.8
200.2.9.8
2
1
Site B, FXS channel.
Site B, FXO
channel.
421
201
200.2.9.6
200.2.9.7
0
1
Site E FXS channel.
Site A, FXS
channel.
1615
xxx
xxxx
200.2.9.9
200.2.9.9
200.2.9.5
200.2.9.5
0
Gives remote voip
(Note 2.) users access to local
PSTN of Site D
(Pierre, SD, area
code 615).
3xxx
0
0
0
Allows remote voip
users to call all PBX
extensions at Site D
(Pierre, SD) using
only four digits.
(Note 1.)
1402
Gives remote voip
users access to local
PSTN of Site F
(Lincoln, NE; area
code 402).
140226374
(Note 1)
(Note 3)
Gives remote voip
users access to key
phone system
extensions at Site F
(Lincoln).
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T1 PhoneBook Configuration
Note 1. The “x” is a wildcard character.
Note 2. By specifying “Channel 0,” we instruct the
MVP2400/2410 to choose 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 (140226374)
actually directs calls to 402-263-7430 through
402-263-7499 into the key system, as well.
This means that such calls, which belong on the
PSTN, cannot be completed. In some cases, this
might be inconsequential because an entire
exchange (fully used or not) might have been
reserved for the company or it might be
unnecessary to reach those numbers. However, to
specify only the 30 lines actually used by the key
system, the destination pattern 140226374 would
have to be replaced by three other destination
patterns, namely 1402263740, 1402263741, and
1402263742. In this way, calls to 402-263-7430
through 402-263-7499 would be properly directed
to the PSTN. In the Site D outbound phonebook,
the 30 lines are defined exactly, that is, without
making any adjacent phone numbers unreachable
through the voip system.
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T1 Phonebook Configuration
MultiVOIP User Guide
Outbound Phone Book for MVP2410 Digital VOIP
(Site D)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls to
Site A (Bismarck).
1507
1507
101#
Note 3.
200.2.9.8 To originate calls
to Rochester local
PSTN using the
FXO channel
(channel #1) of the
Site B VOIP.
102
200.2.9.8 To originate calls
to phone
connected to FXS
port (channel #2)
of the Site B VOIP.
200.2.9.6 Calls to Site E
(Cheyenne).
421
1402
200.2.9.5 Calls to Lincoln
area local PSTN
(via FXO channel,
CH4, of the Site F
VOIP).
1402
263
740
200.2.9.5 Calls to extensions
(thirty) of key
system at Site F
(Lincoln). Human
operator or auto-
attendant is
needed to
complete these
calls.
1402
263
741
1402
263
742
200.2.9.5
200.2.9.5
Note 3. The pound sign (“#”) is a delimiter separating the
VOIP number from the standard telephony phone number.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Inbound Phonebook for MVP2410 Digital VOIP (Site D)
Remove Add
Channel
Number
0
Comment
Prefix
Prefix
1615
9,
Allows phone users at remote
voip sites to call non-toll
numbers within the Site D area
code (615; Pierre, SD) over the
VOIP network.
Note 4.
Note 5.
1615
49231
31
0
Allows voip calls directly to
employees at Site D (at
extensions x3101 to x3199).
Note 4. “9” gives PBX station users access to outside line.
Note 5. The comma represents a one-second pause, the
time required for the user to receive a dial tone on
the outside line (PSTN). The comma is only
allowed in the Inbound phonebook.
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T1 Phonebook Configuration
MultiVOIP User Guide
Outbound Phone Book for MVP410 Analog VOIP
(Site F)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A
(Bismarck).
1507
1507
101#
Note 3.
200.2.9.8 To originate calls
to any PSTN
phone in
Rochester area
using the FXO
channel (channel
#1) of the Site B
VOIP.
102
200.2.9.8 To originate calls
to phone
connected to FXS
port (channel #2)
of the Site B VOIP
(Rochester).
421
200.2.9.6 Calls to Site E
(Cheyenne).
1615
200.2.9.9 Calls to Pierre area
PSTN via Site D
PBX.
31
1615
492
200.2.9.9 Calls to Pierre PBX
extensions with
four digits.
Note 3. The pound sign (“#”) is a delimiter separating the
VOIP number from the standard telephony phone number.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Inbound Phonebook for MVP410 Analog VOIP (Site F)
Remove Add
Channel
Number
4
Comment
Prefix
Prefix
1402
Access to Lincoln local PSTN by
users at remote VOIP locations
via FXO port at Site F.
1402
263740
1402
263741
1402
740
741
742
0
0
0
Gives remote voip users access
to extension of key phone
system at Site F (Lincoln).
Because call is completed at key
system, abbreviated dialing (4
digits) is not workable. Human
operator or auto-attendant is
needed to complete these
calls.
263742
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T1 Phonebook Configuration
MultiVOIP User Guide
Outbound Phone Book for MVP210 Analog VOIP
(Site E)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A.
1507
1507
101#
Note 3.
200.2.9.8 To originate calls
to any PSTN
phone in
Rochester area
using the FXO
channel (channel
#1) of the Site B
VOIP.
102
200.2.9.8 To originate calls
to phone
connected to FXS
port (channel #2)
of the Site B VOIP.
200.2.9.5 Calls to Lincoln
area PSTN (via
1402
FXO channel,
CH4, of the Site F
VOIP).
7
1402
263
200.2.9.5 Calls to Lincoln
key extensions
with four digits.
200.2.9.9 Calls to Pierre area
PSTN via Site D
PBX.
200.2.9.9 Calls to Pierre PBX
extensions with
four digits.
1615
31
1615
492
Note 3. The pound sign (“#”) is a delimiter separating the
VOIP number from the standard telephony phone number.
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T1 PhoneBook Configuration
Inbound Phonebook for MVP210 Analog VOIP (Site E)
Remove Add
Channel
Number
1
Comment
Prefix
Prefix
421
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.
Site A calling Site C, Method 2
1. Dial 101#7175662
2. Await completion. Talk.
Note: Some analog VOIP gateways will allow
completion 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.
4. Await completion. Talk.
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T1 Phonebook Configuration
MultiVOIP User Guide
Site D calling Site C
1. Dial 9,15077175662.
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 PBX units can be programmed to identify the
destination patterns of all calls to be directed to the VOIP.
3. PBX at Site D is programmed to divert all calls made to the 507
area code and exchange 717 into the VOIP network. (It would
also be possible to divert all calls to all phones in area code 507
into the VOIP network, 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 number from the digits that the analog
VOIP must dial onto its local PSTN to complete the call. The
digits “101#7175662” are forwarded to the Site B 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 MVP2410: 101#7175662. The analog
VOIP, seeing the “101” prefix, uses its own channel #1 (an FXO
port) to connect the call to the PSTN. Then the analog VOIP
dials its local phone number 7175662 to complete the call.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Site D calling Site F
A 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 of the call, would be 1402263742. The call would be directed to
the Lincoln voip’s IP address, 200.2.9.5.
(Generally on such a call, the caller would have to dial an initial “9.” But
typically the 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 facilitate
completion of the call would be
1402263742
1402
for calls within the office at Lincoln
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
voip
at IP address 200.2.9.5.
4. The dialed string matches an inbound phonebook entry at the
Lincoln voip, namely 1402263742.
5. The Lincoln voip rings one of the three FXS ports connected to the
Lincoln
key 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
the
extension to which they wish to be connected).
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T1 Phonebook Configuration
MultiVOIP User Guide
Site F calling Site D
A voip call from a Lincoln key extension to extension 3117 on the PBX in Pierre, South
Dakota.
A. The required entry in the Lincoln Outbound Phonebook to facilitate
origination of the 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
completion of the call would be 1615492.
1. Caller at Lincoln picks up phone receiver, presses button on key
phone set. This button has been assigned to a particular voip
channel (any one of the 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
dialing string, 16154923117, to the Pierre voip
at IP address 200.2.9.9.
5. The Pierre voip matches the called digits 16154923117 to its
Inbound Phonebook entry “1615492” .
6. The Pierre PBX dials extension 3117 in the office at Pierre.
Variations in PBX Characteristics
The exact dialing strings needed in the Outbound and Inbound
Phonebooks of the MVP2410 will depend on the capabilities of the PBX.
Some PBXs require trunk access codes (like an “8” or “9” to access an
outside line or to access the VOIP network). Other PBXs can
automatically distinguish between intra-PBX calls, PSTN calls, and
VOIP calls.
Some PBX units can also insert digits automatically when they receive
certain dialing strings from a phone station. For example, a PBX may
be programmable to insert automatically the three-digit VOIP identifier
strings into calls to be directed to analog VOIPs.
The MVP2410 offers complete flexibility for inter-operation with PBX
units so that a coherent dialing scheme can be established to connect a
company’s multiple sites together in a way that is convenient and
intuitive for phone users. When working together with modern PBX
units, the presence of the MVP2410 can be completely transparent to
phone users within the company.
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E1 Phonebook Configuration
MultiVOIP User Guide
MVP3010 Inbound and Outbound MultiVOIP
Phonebooks
Important
Definition:
The MultiVOIP’s Outbound phonebook
lists the phone stations it can call;
its Inbound phonebook describes the
dialing sequences that can be used to
call that MultiVOIP and how those calls
will be directed.
When a VOIP serves a PBX system, the operation of the VOIP should be
transparent to the telephone end user and savings in long-distance
calling charges should be enjoyed. Use of the VOIP should not require
the dialing of extra digits to reach users elsewhere on the VOIP
network. On the contrary, VOIP service more commonly reduces
dialed digits by allowing users (served by 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, the VOIP system
should be configured to maximize savings in long-distance calling
charges. To achieve both of these objectives, ease of use and maximized
savings, the VOIP phonebooks must be set correctly.
NOTE: VOIPs are commonly used for
another reason, as well: VOIPs
allow an organization to
integrate phone and data traffic
onto a single network. Typically
these are private networks.
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E1 PhoneBook Configuration
Free Calls: One VOIP Site to Another
The most direct use of the VOIP system is making calls between the
offices where the VOIPs are located. Consider, for example, the Wren
Clothing Company. This company has VOIP-equipped offices in
London, Paris, and Amsterdam, each served by its own PBX. VOIP
calls between the three offices completely avoid international long-
distance charges. These calls are free. The phonebooks can be set up to
allow all Wren Clothing employees to contact each other using 3-, 4-, or
5-digit numbers, as though they were all in the same building.
United Kingdom
Wren Clothing Co.
5 Wren Clothing Co.
VOIP/PBX Site
VOIP/PBX Site
5London
Amsterdam
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Paris
Free VOIP Calls
France
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E1 Phonebook Configuration
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Local Rate Calls: Within Local Calling Area of Remote
VOIP
In the second use of the VOIP system, the local calling area of each
VOIP location becomes accessible to all of the VOIP system’s users. As
a result, international calls can be made at local calling rates. For
example, suppose that Wren Clothing buys its zippers from The
Bluebird Zipper Company in the western part of metropolitan London.
In that case, Wren Clothing personnel in both Paris and Amsterdam
could call the Bluebird Zipper Company without paying international
long-distance rates. Only London local phone rates would be charged.
This applies to calls completed anywhere in London’s local calling area
(which includes both Inner London and Outer London). Generally,
local calling rates apply only within a single area code, and, for all calls
outside that area code, national rates apply. There are, however, some
European cases where local calling rates extend beyond a single area
code. Local rates between 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 this is rare.)
United Kingdom
Wren Clothing Co.
5Wren Clothing Co.
VOIP/PBX Site
Bluebird Zipper Co.
VOIP/PBX Site
5London
London
Amsterdam
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Calls at London local rates
Paris
Local Calling Area
France
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MultiVOIP User Guide
E1 PhoneBook Configuration
Similarly, the VOIP system allows Wren Clothing employees in London
and Amsterdam to call anywhere in Paris at local rates; it allows Wren
Clothing employees in Paris and London to call anywhere in
Amsterdam at local rates.
United Kingdom
Wren Clothing Co.
VOIP/PBX Site
Amsterdam
Wren Clothing Co.
VOIP/PBX Site
5London
5
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Paris
Calls at Amsterdam local rates
Calls at Paris local rates
Local Calling Areas
France
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E1 Phonebook Configuration
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National Rate Calls: Within Nation of Remote VOIP Site
In the third use of the VOIP system, the national calling area of each
VOIP location becomes accessible to all of the VOIP system’s users. As
a result, international calls can be made at national calling rates. Again,
significant savings are possible. For example, suppose that the Wren
Clothing Company buys its buttons from the Chickadee Button
Company in the Dutch city of Rotterdam. In that case, Wren Clothing
personnel in both London and Paris could call the Chickadee Button
Company without paying international long-distance rates; only Dutch
national calling rates would be charged. This applies to calls completed
anywhere in The Netherlands.
United Kingdom
The
Wren Clothing Co.
VOIP/PBX Site
Netherlands
5London
Wren Clothing Co.
5
VOIP/PBX Site
Amsterdam
Chickadee Button Co.
Rotterdam
Wren Clothing Co.5
VOIP/PBX Site
Paris
Calls at Dutch
National Rates
France
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MultiVOIP User Guide
E1 PhoneBook Configuration
Similarly, the VOIP system allows Wren Clothing employees in London
and Amsterdam to call anywhere in France at French national rates; it
allows Wren Clothing employees in Paris and Amsterdam to call
anywhere in the United Kingdom at its national rates.
United Kingdom
Wren Clothing Co.
VOIP/PBX Site
5 Wren Clothing Co.
London
VOIP/PBX Site
Amsterdam
5
The
Netherlands
Wren Clothing Co.
VOIP/PBX Site
Paris
5
Calls at French
National Rates
Calls at UK
National Rates
France
Inbound versus Outbound Phonebooks
To make the VOIP system transparent to phone users and to allow all
possible free 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
dialing sequences required for a call to originate locally (typically in a
PBX in a particular facility) and reach any of its possible destinations at
remote VOIP sites, including calls terminating at points beyond the
remote VOIP site.
The “Inbound” phonebook for a particular VOIP unit describes the
dialing sequences required for a call to originate remotely from any
other VOIP sites in the system, and to terminate on that particular
VOIP.
Briefly stated, the MultiVOIP’s Outbound phonebook lists the phone stations
it can call; its Inbound phonebook lists the dialing sequences that can be used
to call that MultiVOIP. (Of course, the phone numbers are not literally
“listed” individually.) The phone stations that can originate or
complete calls over the VOIP system are described by numerical rules
called “destination patterns.” These destination patterns generally
consist of country codes, area codes or city codes, and local phone
exchange numbers.
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E1 Phonebook Configuration
MultiVOIP User Guide
In order for any VOIP phone call to be made, there must be both an
Inbound Phonebook entry and an Outbound Phonebook entry that
describe the end-to-end connection. The phone station originating the
call must be connected to the VOIP system. The Outbound Phonebook
for that VOIP unit must have a destination pattern entry that includes
the ‘called’ phone (that is, the phone completing the call). The Inbound
Phonebook of the VOIP where the call is completed must have a
destination pattern entry that includes the digit sequence dialed by the
originating phone station.
The PhoneBook Configuration procedure below is brief, but it is
followed by an example case. For many people, the example case may
be easier to grasp than the procedure steps. Configuration is not
difficult, but all phone number sequences, destination patterns, and
other information must be entered exactly; otherwise connections will
not be made.
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E1 PhoneBook Configuration
Phonebook configuration screens can be accessed using icons or the
sidebar menu.
Phonebook Icons
Description
Phonebook Configuration
Inbound Phonebook
Entries List
Add Inbound Phonebook
Entry
Edit selected Inbound
Phonebook Entry
Outbound Phonebook
Entries List
Add Outbound
Phonebook Entry
Edit selected Outbound
Phonebook Entry
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In consultation with your VOIP administrator, enter the Gateway
Name determine which protocol you will use (H.323, SIP, or SPP).
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E1 Phonebook Configuration
MultiVOIP User Guide
Then fill in the IP address, signaling port, and other parameters, as
needed. (The parameters needed for each protocol are different.)
The table below describes all fields in the general PhoneBook
Configuration screen.
PhoneBook Configuration Parameter Definitions
Field Name
Values
Description
Gateway
Name
Y/N
This field allows you to specify
a name for this MultiVOIP.
When placing a call, this name
is sent to the remote
MultiVOIP for display in Call
Progress listings, Logs, etc.
H.323 Parameters
Use Fast Start
Signaling Port
Y/N
Enables the H.323 Fast Start
procedure. May need to be
enabled/disabled for
compatibility with third-party
VOIP gateways.
port
Default: 1720 (H.323)
number
GateKeeper RAS Parameters
IP address of the GateKeeper.
Gatekeeper /
IP Address
n.n.n.n,
for n =
0 - 255
Signaling Port
1 - 64000
Well-known port number for
GateKeepers.
Must match port number of
GateKeeper, 1719.
Gatekeeper
Name
alpha-
numeric
string
Optional. The name of the
GateKeeper with which this
MultiVOIP is trying to register.
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E1 PhoneBook Configuration
.
PhoneBook Configuration Parameter Definitions
(cont’d)
GateKeeper RAS Parameters
Field Name
Values
Description
RAS TTL
Value
in seconds
The H.323 Gatekeeper “Time
to Live” value. As soon as a
MultiVOIP gateway registers
with a gatekeeper (allowing
the gatekeeper to control its
call traffic) a countdown timer
begins. The RAS TTL Value is
the interval of the countdown
timer. Before the TTL
countdown expires, the
MultiVOIP gateway needs to
register with the gatekeeper in
order to maintain the
connection. If the MultiVOIP
does not register before the
TTL interval expires, the
MultiVOIP gateway’s
registration with the
gatekeeper will expire and the
gatekeeper will no longer
permit call traffic to or from
that gateway. Calls in
progress will continue to
function even if the gateway
becomes de-registered.
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E1 Phonebook Configuration
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PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values
Description
H.323 Version 4 Parameters
H.323
Multiplexing
(Mux)
Y/N
Signaling for multiple phone
calls can be carried on a single
port rather than opening a
separate signaling port for
each call. This conserves
bandwidth resources.
H.245
Values: Y/N
Tunneling
(Tun)
Description: H.245 messages are
encapsulated within the Q.931 call-signaling
channel. Among other things, the H.245
messages let the two endpoints tell each other
what their technical capabilities are and
determine who, during the call, will be the
client and who the server. Tunneling is the
process of transmitting these H.245 messages
through the Q.931 channel. The same TCP/IP
socket (or logical port) already being used for
the Call Signaling Channel is then also used
by the H.245 Control Channel. This
encapsulation reduces the number of logical
ports (sockets) needed and reduces call setup
time.
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E1 PhoneBook Configuration
PhoneBook Configuration Parameter Definitions
(cont’d)
Values Description
Field Name
H.323 Version 4 Parameters
Parallel H.245
(FS + Tun)
Values: Y/N
Description: FS (Fast Start or Fast Connect) is
a Q.931 feature of H.323v2 to hasten call
setup as well as ‘pre-opening’ the media
channel before the CONNECT message is
sent. This pre-opening is a requirement for
certain billing activities. Under Parallel
H.245 FS + Tun, this Fast Connect feature can
operate simultaneously with H.245
Tunneling (see description above).
Annex –E (AE) Values: Y/N
Description: Multiplexed UDP call signaling
transport. Annex E is helpful for high-
volume voip system endpoints. Gateways
with lesser volume can afford to use TCP to
establish calls. However, for larger volume
endpoints, the call setup times and system
resource usage under TCP can become
problematic. Annex E allows endpoints to
perform call-signaling functions under the
UDP protocol, which involves substantially
streamlined overhead. (This feature should
not be used on the public Internet because of
potential problems with security and
bandwidth usage.)
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PhoneBook Configuration Parameter Definitions
(cont’d)
Values Description
Field Name
SIP Proxy Parameters
Signaling Port
1 - 64000
Port number on which the
MultiVOIP UserAgent
software module will be
waiting for any incoming SIP
requests.
Use SIP Proxy
Y/N
Allows the MultiVOIP to work
in conjunction with a proxy
server.
Proxy Domain
Name / IP
Address
n.n.n.n
where
n=0-255
Network address of the proxy
server that the voip is using.
Append SIP
Proxy Domain
Name in User
ID
Y/N
When checked, the domain
name of the SIP Proxy serving
the MultiVOIP gateway will be
included as part of the User ID
for that gateway. If
unchecked, the SIP Proxy’s IP
address will be included as
part of the User ID instead of
the SIP Proxy’s domain name.
Port Number
User Name
numeric
Logical port number for proxy
communications.
Values: alphanumeric
Description: Identifier used when proxy
server is used in network. If a proxy server is
used in a SIP voip network, all clients must
enter both a User Name and a Password
before being allowed to make a call.
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E1 PhoneBook Configuration
PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values & Description
SIP Proxy Parameters
Password
Values: alphanumeric
Description: Password for proxy server
function. See “User Name” description
above.
Re-
Values: numeric (in seconds)
Registration
Time
Description: This is the timeout interval for
registration of the MultiVOIP with a SIP
proxy server. The time interval begins the
moment the MultiVOIP gateway registers
with the SIP proxy server and ends at the
time specified by the user in the Re-
Registration Time field (this field). When/if
registration lapses, call traffic routed to/from
the MultiVOIP through the SIP proxy server
will cease. However, calls in progress will
continue to function until they end.
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E1 Phonebook Configuration
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PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values
Description
Single Port Protocol (SPP)
SPP voip systems can operate
in two modes:
Mode
Direct,
Client, or
Registrar
in the direct mode, where all
voip gateways have static IP
addresses assigned to them; or
in the registrar/client mode,
where one voip gateway
serves as registrar and all
other gateways, being its
clients, point to that registrar.
The registrar assigns IP
addresses dynamically.
General Options
The UDP port on which data
transmission will occur. Each
client voip has its own port. If
two client voips are both
behind the same firewall, then
they must have different ports
assigned to them.
Port
If there are two clients and
each is behind a different
firewall, then the clients could
have different port numbers or
the same port number.
(Default port number = 10000.)
If packets are lost (as indicated
by absence of an
acknowledgment) then the
endpoint will retransmit the
lost packets after this
Re-trans-
mission
(in ms)
designated time duration has
elapsed. (Default value = 2000
milliseconds.)
Number of times the voip will
re-transmit a lost packet (if no
acknowledgment has been
received). (Default value = 3)
Max
Re-trans-
mission
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E1 PhoneBook Configuration
PhoneBook Configuration Parameter Definitions
(cont’d)
Field Name
Values
Description
Single Port Protocol (SPP)
[continued]
Client Option fields are active
only in registrar/client mode
and only for client voip units.
This is the IP address of the
registrar voip to which this
client is assigned. (Default
value = 0.0.0.0; effectively,
there is no useful default
value.)
This is the port number of the
registrar voip to which this
client is assigned. (Default port
number = 10000.)
Client Options
Registrar IP
Address
n.n.n.n
1-64000
Registrar
Port
Registrar Option fields are
active only in registrar/client
mode and only for registrar
voip units.
Registrar Options
Time-out duration before a
registrar will unregister a
client that does not send its
“I’m here” signal. Timeout
default = 60 seconds.
Keep Alive
(in sec.)
30 - 300
Proxy/NAT Device
Parameters
Enables MultiVOIP (running
in SPP Registrar mode) to
operate ‘behind’ a proxy/NAT
device (NAT = Network
Address Translation).
Behind
Proxy/NAT
device
Y/N
The public IP address of the
proxy/NAT device which the
MultiVOIP is behind.
Proxy/NAT
Device
Parameters –
Public IP
n.n.n.n
where
n=0-255
Address
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E1 Phonebook Configuration
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2. Select PhoneBook Modify and then select
Outbound Phone Book/List Entries.
Click Add.
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3. The Add/Edit Outbound PhoneBook screen appears.
Enter Outbound PhoneBook data for your MVP3010. Note that the
Advanced button gives access to the Alternate IP Routing feature, if
needed. Alternate IP Routing can be implemented in a secondary
screen (as described after the primary screen field definitions below).
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The fields of the Add/Edit Outbound Phone Book screen are described
in the table below.
Add/Edit Outbound Phone Book: Field Definitions
Field Name
Values
Description
Accept Any
Number
Y/N
When checked, “Any
Number” appears as the
value in the Destination
Pattern field.
The Any Number feature
works differently depending
on whether or not an external
routing device is used
(Gatekeeper for H323
protocol, Proxy for SIP
protocol, Registrar for SPP
protocol).
When no external routing
device is used. If Any
Number is selected, calls to
phone numbers not matching
a listed Destination Pattern
will be directed to the IP
Address in the Add/Edit
Outbound Phone Book
screen. “Any Number” can
be used in addition to one or
more Destination Patterns.
When external routing
device is used. If Any
Number is selected, calls to
phone numbers not matching
a listed Destination Pattern
will be directed to the
external routing device used
(Gatekeeper for H323
protocol, Proxy for SIP
protocol, Registrar for SPP
protocol). The IP Address of
the external routing device
must be set in the Phone
Book Configuration screen.
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E1 PhoneBook Configuration
Add/Edit Outbound Phone Book: Field Definitions
Field Name
Values
Description
Destination
Pattern
prefixes,
area codes,
exchanges,
line
numbers,
extensions
Defines the beginning of
dialing sequences for calls
that will be connected to
another VOIP in the system.
Numbers beginning with
these sequences are diverted
from the PTSN and carried
on Internet or other IP
network.
Total Digits
as needed
number of digits the phone
user must dial to reach
specified destination
Remove Prefix
dialed digits portion of dialed number to
be removed before
completing call to destination
Add Prefix
IP Address
dialed digits digits to be added before
completing call to destination
n.n.n.n
the IP address to which the
call will be directed if it
begins with the destination
pattern given
for = 0-255
Description
alpha-
numeric
Describes the facility or
geographical location at
which the call will be
completed.
Indicates protocol to be used in
outbound transmission.
Protocol Type
SIP, H.323,
or SPP
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E1 Phonebook Configuration
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Add/Edit Outbound Phone Book: Field Definitions
(cont’d)
Field Name
H.323 fields
Values
Description
Indicates whether or not
gatekeeper is used.
Use Gatekeepr Y/N
The H.323 ID assigned to the
destination MultiVOIP. Only
valid if “Use Gatekeeper” is
enabled for this entry.
Gateway H.323 alpha-
ID
numeric
numeric
Gateway
Prefix
This number becomes
registered with the
GateKeeper. Call requests
sent to the gatekeeper and
preceded by this prefix will
be routed to the VOIP
gateway.
H.323 Port
Number
1720
This parameter pertains to
Q.931, which is the H.323 call
signaling protocol for setup
and termination of calls (aka
ITU-T Recommendation
I.451). H.323 employs only
one “well-known” port (1720)
for Q.931 signaling. If Q.931
message-oriented signaling
protocol is used, the port
number 1720 must be chosen.
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E1 PhoneBook Configuration
Add/Edit Outbound Phone Book: Field Definitions
(cont’d)
Description
Field Name
SIP Fields
Use Proxy
Values
Y/N
Select if proxy server is used.
Transport
Protocol
TCP or
UDP
Voip administrator must choose
between UDP and TCP
transmission protocols. UDP is a
high-speed, low-overhead
connectionless protocol where
data is transmitted without
acknowledgment, guaranteed
delivery, or guaranteed packet
sequence integrity. TCP is slower
connection-oriented protocol
with greater overhead, but
having acknowledgment and
guarantees delivery and packet
sequence integrity.
The SIP Port Number is a
UDP logical port number.
The voip will “listen” for SIP
messages at this logical port.
If SIP is used, 5060 is the
default, standard, or “well
known” port number to be
used. If 5060 is not used,
then the port number used is
that specified in the SIP
Request URI (Universal
Resource Identifier).
SIP Port
Number
5060 or other
*See RFC3087
(“Control of
Service
Context using
SIP Request-
URI,” by the
Network
Working
Group).
Looking similar to an email
address, a SIP URL
SIP URL
sip.userphone
@
identifies a user's address.
In SIP communications, each
caller or callee is identified
by a SIP url:
hostserver,
where
“userphone”
is the
sip:user_name@host_name.
The format of a sip url is very
similar to an email address,
except that the “sip:“ prefix is
used.
telephone
number and
“hostserver”
is the domain
name or an
address on the
network
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E1 Phonebook Configuration
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Add/Edit Outbound Phone Book: Field Def’ns (cont’d)
Field Name
SPP Fields
Use Registrar
Values
Description
Values: Y/N
Description: Select this checkbox to use registrar
when voip system is operating in the
“Registrar/Client” SPP mode. In this mode, one
voip (the registrar, as set in Phonebook
Configuration screen) has a static IP address and
all other voips (clients) point to the registar’s IP
address as functionally their own. However, if
your voip system overall is operating in
“Registrar/Client” mode but you want to make an
exception and use Direct mode for the destination
pattern of this particular Add/Edit Phonebook
entry, leave this checkbox unselected.
Leave this checkbox unselected if your overall voip
system is operating in the “Direct" SPP mode. In
this mode, all voips in system are peers and each
has its own static IP address.
Port Number
Values: numeric
Description: When operating in
“Registrar/Client” mode, this is the port by which
the gateway receives all SPP data and control
messages from the registrar gateway. (This ability
to receive all data and messages via one port
allows the voip to operate behind a firewall with
only one port open.)
When operating in “Direct” mode, this is the Port
by which peer voips receive data and messages.
Alternate
Phone Number
numeric
Phone number associated
with alternate IP routing.
Remote
Device is …
Y/N
Check when system includes
1st-generation MultiVOIPs to
allow inter-operation. These
include MVP-
110/120/200/400/800
MultiVOIP units.
Advanced
Values: N/A
button
Description: Gives access to secondary screen
where an Alternate IP Route can be specified
for backup or redundancy of signal paths.
See discussion on next page. For SIP & H.323
operation only.
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E1 PhoneBook Configuration
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 carry
the 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) unit
in case one voip unit fails. The user must specify the IP address of the alternate route
for each destination pattern entry in the Outbound Phonebook.
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E1 Phonebook Configuration
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Alternate Routing Field Definitions
Field
Values
Description
Name
Alternate n.n.n.n
Alternate destination for outbound data traffic
in case of excessive delay in data transmission.
IP
where
Address
n= 0-255
Round
Trip
Delay
milliseconds The Round Trip Delay is the criterion for
judging when a data pathway is considered
blocked. When the delay exceeds the
threshold specified here, the data stream will
be diverted to the alternate destination
specified as the Alternate IP Address.
4. Select PhoneBook Modify and then select Inbound PhoneBook/List
Entries.
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5. The Add/Edit Inbound PhoneBook screen appears.
Enter Inbound PhoneBook data for your MVP3010. The fields of the
Add/Edit Inbound PhoneBook screen are described in the table below.
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E1 Phonebook Configuration
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Add/Edit Inbound Phone Book: Field Definitions
Field
Values
Description
Name
Accept
Any
Y/N
When checked, “Any Number” appears as the
value in the Remove Prefix field.
Number
The Any Number feature of the Inbound
Phone Book does not work when an external
routing device is used (Gatekeeper for H323
protocol, Proxy for SIP protocol, Registrar for
SPP protocol).
When no external routing device is used. If
Any Number is selected, calls received from
phone numbers not matching a listed Prefix
(shown in the Remove Prefix column of the
Inbound Phone Book) will be admitted into
the voip on the channel listed in the Channel
Number field. “Any Number” can be used in
addition to one or more Prefixes.
Remove
Prefix
dialed digits portion of dialed number to be removed
before completing call to destination
(often a local PBX)
Add
dialed digits digits to be added before completing call to
Prefix
destination
(often a local PBX)
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MultiVOIP User Guide
E1 PhoneBook Configuration
Add/Edit Inbound Phone Book: Field Definitions
(cont’d)
Field Name
Values
Description
E1 channel number to which
the call will be assigned as it
enters the local telephony
equipment
Channel
Number
1-30, or
“Hunting”
(often a local PBX).
“Hunting” directs the call to
any available channel.
Description
--
Describes the facility or
geographical location at
which the call originated.
Call Forward Parameters
Enable
Y/N
Click the check-box to enable
the call-forwarding feature.
Unconditional. When selected,
all calls received will be
forwarded.
Busy. When selected, calls
will be forwarded when
station is busy.
Forward
Condition
Uncondit.;
Busy
No Resp.
No Response. When selected,
calls will be forwarded if
called party does not answer
after a specified number of
rings, as specified in Ring
Count field.
Forwarding can be
conditioned on both “Busy”
and “No Response.”
385
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E1 Phonebook Configuration
MultiVOIP User Guide
Add/Edit Inbound Phone Book: Field Definitions
(cont’d)
Description
Field Name
Values
Forward
Destination
Phone number or IP address to which calls
will be directed.
IP address,
For H.323 calls, the Forward Destination can
phone number, be either a Phone Number of an IP Address.
port number,
etc.
For SIP calls, the Forward Destination can be
one of 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.
For SPP calls, the Forward Destination can be
one of the following:
(a) phone number, (b) IP address: port, or
(c) phone number: IP address: port.
Ring Count
integer
When No Response is
condition for forwarding
calls, this determines how
many unanswered rings
are needed to trigger the
forwarding.
Registration
Option
Parameters
In an H.323 voip system, gateways can
register with 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
with the SIP Proxy.
In an SPP voip system, gateways can register
with the SPP Registrar voip unit.
6. When your Outbound and Inbound PhoneBook entries are
completed, click on Save Setup in the sidebar menu to save your
configuration.
You can change your configuration at any time as needed for your
system.
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MultiVOIP User Guide
E1 PhoneBook Configuration
Remember that the initial MVP3010 setup must be done locally or via
the built-in Remote Configuration/Command Modem using the
MultiVOIP program. However, after the initial configuration is
complete, all of the MVP3010 units in the VOIP system can be
configured, re-configured, and updated from one location using the
MultiVOIP web GUI software program or the MultiVOIP program (in
conjunction with the built-in modem).
E1 Phonebook Examples
To demonstrate how Outbound and Inbound PhoneBook entries work
in an international VOIP system, we will re-visit our previous example
in greater detail. It’s an international company with offices in London,
Paris, and Amsterdam. In each office, a MVP3010 has been connected
to the PBX system.
3 Sites, All-E1 Example
The VOIP system will have the following features:
1. Employees in all cities will be able
to call each other over the VOIP
system using 4-digit extensions.
2. Calls to Outer London and Inner
London, greater Amsterdam, and
greater Paris will be accessible to all
company offices as local calls.
3. Vendors in Guildford, Lyon, and
Rotterdam can be contacted as
national calls 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.
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E1 Phonebook Configuration
MultiVOIP User Guide
France Country Code: 33
Lille
Paris: Area 01
Reims
Rouen
Nantes
Strasbourg
Lyon
Bordeaux
Toulouse
Marseille
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MultiVOIP User Guide
E1 PhoneBook Configuration
The Netherlands
Country Code: 31
050
Groningen
058
Leeuwarden
Texel 0222
Den Helder 0223
038 Zwolle
0299 Purmerend
Beverwijk 0251
Haarlem 023
020 Amsterdam
Aalsmeer0297
053
Enschede
0294 Weesp
070
The Hague
026
Arnhem
010
Rotterdam
0118
Middelburg
040
Eindhoven
043
Maastricht
389
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E1 Phonebook Configuration
MultiVOIP User Guide
An outline of the equipment setup in these three offices is shown
below.
Wren Clothing Co.
London Office
Country Code: +44
Area Code: 0208
E1
Digital
VoIP
PBX
-5174
200.2.10.3
-5173
-5172
IP
Network
-5171
979-5170
Wren Clothing Co.
Paris Office
Country Code: +33
Area Code: 01
R
o
E1
PBX
u
t
Digital
VoIP
e
r
-29 83
Digital
VoIP
200.2.9.7
Wren Clothing Co.
Amsterdam Office
Country Code: +31
Area/City Code: 020
-29 82
200.2.8.5
E1
74 71 29 81
PBX
-4804
-4803
-4802
-4801
688-4800
390
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MultiVOIP User Guide
E1 PhoneBook Configuration
The screen below shows Outbound PhoneBook entries for the VOIP located in the
company’s London facility
The Inbound PhoneBook for the London VOIP is shown below.
NOTE: Commas are allowed in the Inbound Phonebook, but not in the
Outbound Phonebook. Commas denote a brief pause for a dial
tone, allowing time for the PBX to get an outside line.
391
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E1 Phonebook Configuration
MultiVOIP User Guide
The screen below shows Outbound PhoneBook entries for the VOIP
located in the company’s Paris facility.
The Inbound PhoneBook for the Paris VOIP is shown below.
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MultiVOIP User Guide
E1 PhoneBook Configuration
The screen below shows Outbound PhoneBook entries for the VOIP in
the company’s Amsterdam facility.
The Inbound PhoneBook for the Amsterdam VOIP is shown below.
393
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E1 Phonebook Configuration
MultiVOIP User Guide
Configuring Digital & Analog VOIPs in Same System
The MVP3010 digital MultiVOIP unit is compatible with analog VOIPs.
In many cases, digital and analog VOIP units will appear in the same
telephony/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 same VOIP 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 MVP200 units).
EXAMPLE:
Digital & Analog VOIPs
in Same System
Site D:
Inner London, UK
Area Code 0207
PSTN
PBX
200.2.9.9
Digital
VoIP
MVP3010
E1
Other extensions
x8301 - x8399
Router
020-7398-8300
Site E:
Carlisle, UK
Area Code 0122 8
Site A:
Birmingham, W. Midlands, UK
Area Code 0121
200.2.9.6
Series #1 Analog MultiVOIP
(Server/Client Phonebook)
MVP200
Series #2 Analog MultiVOIP
MVP210
FXS
Unit
FXS
CH1
#200
CH1
421
201
200.2.9.7
Client
IP
Network
Site F:
Site B:
Tavistock, UK
Area Code 0182
Reading, Berkshire, UK
Area Code 0118
200.2.9.5
FXO
Series #1 Analog MultiVOIP
(Server/Client Phonebook)
PSTN
Series #2 Analog MultiVOIP
MVP410
Port #4
102
MVP200
CH2
FXS
FXO
Unit
#100
CH1
FXS Port
FXS Ports
CO Ports
943-6161
200.2.9.8
Host
(Holds phonebook for both
Series #1 analog VOIPs.)
CO Port
Key
System
Other extensions
x7401 - x7429
PSTN
263-7400
118-943-5632
Site C:
Reading Area Residential
394
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MultiVOIP User Guide
E1 PhoneBook Configuration
The Series I analog VOIP phone book resides in the “Host” VOIP unit at
Site B. 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
digital MultiVOIP requires its own inbound and outbound
phonebooks, as well.
These seven phone books are shown below.
Phone Book for Analog VOIP Host Unit (Site B)
VOIP Dir #
-OR-
IP Address Channel Comments
Destination
Pattern
102
200.2.9.8
200.2.9.8
200.2.9.7
200.2.9.6
200.2.9.5
2
1
1
0
0
Site B, FXS channel.
(Reading, UK)
101
201
421
Site B, FXO channel.
(Reading, UK)
Site A, FXS channel.
(Birmingham)
Site E, FXS channel.
(Carlisle, UK)
018226374
Note 3.
Gives remote voip users
access to key phone
system extensions at
Tavistock office (Site F).
The key system might be
arranged either so that
calls go through a human
operator or through an
auto-attendant (which
prompts user to dial the
desired extension).
0182
3xx
200.2.9.5
200.2.9.9
4
0
Gives remote voip users
access to Tavistock PSTN
via FXO port (#4) at Site
F.
Allows remote voip users
(Note 1.) to call all PBX extensions
at Site D (Inner London)
using only three digits.
395
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E1 Phonebook Configuration
MultiVOIP User Guide
Phone Book for Analog VOIP Host Unit (Site B)
(continued)
VOIP Dir #
-OR-
IP Address Channel Comments
Destination
Pattern
0207
xxx
xxxx
200.2.9.9
200.2.9.9
0
Gives remote voip users
(Note 2.) access to phone numbers
in 0207 area code (Inner
London) in which Site D
is located.
0208
xxx
xxxx
0
Gives remote voip users
(Note 2.) access to phone numbers
in 0208 area code (Outer
London) for which calls
are local from Site D
(Inner London).
Note 1. The “x” is a wildcard character.
Note 2. By specifying “Channel 0,” we instruct the MVP3010 to
choose 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 to 402-263-7430 through
402-263-7499 into the key system, as well.
This means that such calls, which belong on the PSTN, cannot be
completed. In some cases, this might be inconsequential because
an entire exchange (fully used or not) might have been reserved
for the company or it might be unnecessary to reach those
numbers. However, to specify only the 30 lines actually used by
the key system, the destination pattern 018226374 would have to
be replaced by three other destination patterns, namely
0182263740, 0182263741, and 0182263742. In this way, calls to
0182-263-7430 through 0182-263-7499 would be properly directed
to the PSTN. In the Site D outbound phonebook, the 30 lines are
defined exactly, that is, without making any adjacent phone
numbers unreachable through the voip system.
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MultiVOIP User Guide
E1 PhoneBook Configuration
The Outbound PhoneBook of the MVP3010 is shown below.
Outbound Phone Book for MVP3010 Digital VOIP (Site D)
Destin.
Pattern
Remov
e
Add
Prefix
IP
Comment
Address
Prefix
201
200.2.9.7 To originate calls to Site A
(Birmingham).
901189 901189 101#
200.2.9.8 To originate calls to any
PSTN phone in Reading
area using the FXO channel
(channel #1) of the Site B
VOIP (Reading, UK).
Note 3.
421
90182
--
--
200.2.9.6 Calls to Site E (Carlisle).
Calls to Tavistock local
PSTN (Site F) could be
arranged by operator or
possibly by auto-attendant.
200.2.9.5 Calls to extensions of key
phone system at Tavistock
office.
90182
263
740
90182
263
741
9
9
9
--
--
--
200.2.9.5
90182
263
200.2.9.5
742
102
200.2.9.8 To originate calls to phone
connected to FXS port
(channel #2) of the Site B
VOIP (Reading).
Note 3. The pound sign (“#”) is a delimiter separating the VOIP
number from the standard telephony phone number.
397
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E1 Phonebook Configuration
MultiVOIP User Guide
The Inbound PhoneBook of the MVP3010 is shown below.
Inbound Phone Book for MVP3010 Digital VOIP (Site D)
Remove
Prefix
Add
Prefix
Channel
Number
Comments
0207
9,7
Note 4.
Note 5.
0
Allows phone users at remote voip sites
to call local numbers (those within the
Site D area code, 0207, Inner London)
over the VOIP network.
0208
9,8
0
0
Allows phone users at remote voip sites
to call local numbers (those in Outer
London) over the VOIP network.
Allows phone users at remote voip sites
to call extensions of the Site D PBX
using three digits, beginning with “3” .
Note 4.
Note 5.
3
0207
39883
Note 4. “9” gives PBX station users access to outside line.
Note 5. The comma represents a one-second pause, the time
required for the user to receive a dial tone on the outside line
(PSTN). Commas can be used in the Inbound Phonebook, but not
in the Outbound Phonebook.
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MultiVOIP User Guide
E1 PhoneBook Configuration
Outbound Phone Book for MVP410 Analog VOIP
(Site F)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A
(Birmingham).
200.2.9.8 To originate calls
to any PSTN
01189
0118
101#
Note 3.
phone in Reading
area using the
FXO channel
(channel #1) of the
Site B VOIP.
102
200.2.9.8 To originate calls
to phone
connected to FXS
port (channel #2)
of the Site B VOIP
(Reading).
421
200.2.9.6 Calls to Site E
(Carlisle).
0207
200.2.9.9 Calls to Inner
London area
PSTN via Site D
PBX.
0208
3
200.2.9.9 Calls to Inner
London area
PSTN via Site D
PBX.
200.2.9.9 Calls to Inner
London PBX
--
0207
398
8
extensions with
three digits.
Note 3. The pound sign (“#”) is a delimiter separating the
VOIP number from the standard telephony phone number.
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E1 Phonebook Configuration
MultiVOIP User Guide
Inbound Phonebook for MVP410 Analog VOIP (Site F)
Remove Add
Channel
Number
4
Comment
Prefix
Prefix
01822
2
Calls to Tavistock local
PSTN through FXO port
(Port #4) at Site F.
0182
263
740
0182
263
741
0182
263
742
740.
741.
742
0
0
0
Gives remote voip users, access
to extensions of key phone
system atTavistock office.
Because call is completed at key
system, abbreviated dialing (3-
digits) is not workable.
Human operator or auto-
attendant is needed to
complete these calls.
400
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MultiVOIP User Guide
E1 PhoneBook Configuration
Outbound Phone Book for MVP210 Analog VOIP
(Site E)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A
(Birmingham).
01189
0118
101#
Note 3.
200.2.9.8 To originate calls
to any PSTN
phone in Reading
area using the
FXO channel
(channel #1) of the
Site B VOIP.
102
200.2.9.8 To originate calls
to phone
connected to FXS
port (channel #2)
of the Site B VOIP
(Reading).
01822
01822
--
200.2.9.5 Calls to Tavistock
area PSTN (via
FXO channel of
the Site F VOIP).
200.2.9.5 Calls to Tavistock
key system
0182
26374
operator or auto-
attendant.
0207
8
0207
200.2.9.9 Calls to London
area PSTN via Site
D PBX.
200.2.9.9 Calls to London
PBX extensions
0207
398
with four digits.
Note 3. The pound sign (“#”) is a delimiter separating the
VOIP number from the standard telephony phone number.
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E1 Phonebook Configuration
MultiVOIP User Guide
Inbound Phonebook for MVP210 Analog VOIP (Site E)
Remove Add
Channel
Number
1
Comment
Prefix
Prefix
421
Call Completion Summaries
Site A calling Site C, Method 1
Dial 101.
Hear dial tone from Site B.
Dial 9435632.
Await completion. Talk.
Site A calling Site C, Method 2
Dial 101#9435632
Await completion. Talk.
Note: Some analog VOIP gateways will allow completion by
Method 2. Others will not.
Site C calling Site A
1.
2.
3.
4.
Dial 9436161.
Hear dial tone from Site B VOIP.
Dial 201.
Await completion. Talk.
402
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MultiVOIP User Guide
E1 PhoneBook Configuration
Site D calling Site C
1. 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 PBX units can be programmed to identify the
destination patterns of all calls to be directed to the VOIP.
3. PBX at Site D is programmed to divert all calls made to the 118
area code and exchange 943 into the VOIP network. (It would
also be possible to divert all calls to all phones in area code 118
into the VOIP network, 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 number from the digits that the analog VOIP
must dial onto its local PSTN to complete the call. The digits
“101#9435632” are forwarded to the Site B 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) to connect the call to the PSTN. Then the analog VOIP
dials its local phone number 9435632 to complete the call.
NOTE: In the case of Reading, Berkshire,,
England, both “1189” and “1183” are
considered local area codes. This is, in a
sense however, a matter of terminology.
It simply means that numbers of the
form 9xx-xxxx and
3xx-xxxx are both local calls for users at
other sites in the VOIP network.
403
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E1 Phonebook Configuration
MultiVOIP User Guide
Site D calling Site F
A voip call from Inner London PBX to extension 7424 on the key telephone system in
Tavistock, UK.
A. The required entry in the London Outbound Phonebook to facilitate
origination of the call, would be 90182263742. The call would be directed to
the Tavistock voip’s IP address, 200.2.9.5. (Generally on such a call, the caller
would have to dial an initial “9”. But typically the PBX would not pass the
initial “9” dialed 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 Tavistock Inbound Phonebook to facilitate
completion of the call would be
0182263742
01822
for calls within the office at Tavistock
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 the
Tavistock voip
at IP address 200.2.9.5.
4. The dialed string matches an inbound phonebook entry at the
Tavistock voip, namely 0182263742.
5. The Tavistock voip rings one of the three FXS ports connected to
the Tavistock
key 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
the
extension to which they wish to be connected).
404
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MultiVOIP User Guide
E1 PhoneBook Configuration
Site F calling Site D
A voip call from a Tavistock key extension to extension 3117 on the PBX in Inner
London.
A. The required entry in the Tavistock Outbound Phonebook to facilitate
origination of the call, would be “3”. The string 02073988 is added, preceding
the “3”. The call would 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
facilitate completion of the call would be 020739883.
1. The caller in Tavistock picks up the phone receiver, presses a
button on the key phone set. This button has been assigned to a
particular voip channel.
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 voip
at IP address 200.2.9.9.
5. The Inner-London voip matches the called digits 02073983117to its
Inbound Phonebook entry “020739883, ” which it removes. Then it
adds back the “3” as a prefix.
6. The Inner-London PBX dials extension 3117 in the office in Inner
London.
Variations in PBX Characteristics
The exact dialing strings needed in the Outbound and Inbound
Phonebooks of the MVP3010 will depend on the capabilities of the PBX.
Some PBXs require trunk access codes (like an “8” or “9” to access an
outside line or to access the VOIP network). Other PBXs can
automatically distinguish between intra-PBX calls, PSTN calls, and
VOIP calls.
Some PBX units can also insert digits automatically when they receive
certain dialing strings from a phone station. For example, a PBX may
be programmable to insert automatically the three-digit VOIP identifier
strings into calls to be directed to analog VOIPs.
The MVP3010 offers complete flexibility for inter-operation with PBX
units so that a coherent dialing scheme can be established to connect a
company’s multiple sites together in a way that is convenient and
intuitive for phone users. When working together with modern PBX
units, the presence of the MVP3010 can be completely transparent to
phone users within the company.
405
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E1 Phonebook Configuration
MultiVOIP User Guide
International Telephony Numbering Plan Resources
Due 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 have been established; new service categories have been
established (for example, to accommodate GSM, personal numbering,
corporate numbering, etc.). Below we list several web sites that present
up-to-date information on the telephony numbering plans used around
the world. While we find these to be generally good resources, we
would note that URLs may change or become nonfunctional, and we
cannot guarantee the quality of information on these sites.
URL
Description
/wtng
The World Telephone
Numbering Guide
presents excellent
international
numbering info that
is both broad and
detailed. This
includes info on re-
numbering plans
carried out
worldwide in recent
years to
accommodate new
technologies.
/number.htm
UK numbering plan
from the Office of
Telecommunications,
the UK telephony
authority.
The International
Telecommunications
Union is an excellent
source and authority
on international
telecom regulations
and standards.
National and
international number
plans are listed on
this site.
406
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MultiVOIP User Guide
E1 PhoneBook Configuration
URL
Description
Guide to
international use of
modems.
National and
international
numbering plans
based on direct input
from regulators
worldwide. Includes
lists of telecom
carriers per country.
European
Telecommunications
Office. Primarily
concerned with
mobile/wireless
radiotelephony,
GSM, etc.
European Telephony
Numbering Space.
Resources for pan-
European telephony
services, standards,
etc. Part of ETO site.
http://www.regtp.de/en/reg_tele/start List of European
telecom regulatory
agencies by country
(from German
telecom authority).
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Operation & Maintenance
Phonebooks for Series II analog MultiVOIP units (MVP130,
MVP130FXS, MVP210, MVP410, and MVP810) and BRI MultiVOIP
units (MVP410ST/810ST) are, in principle, configured the same as
phonebooks for digital MultiVOIP products that would operate in the
same environment (under either North American or European
telephony standards, T1 or E1).
Therefore, if you are operating an analog MultiVOIP unit in a North
American telephony environment, you will find useful phonebook
instructions and examples in Chapter 7: T1 Phonebook Configuration. If
you are operating an analog MultiVOIP unit in a European telephony
environment, you will find useful phonebook instructions and
examples in Chapter 8: E1 Phonebook Configuration.
Most of the examples in Chapters 7 and 8 describe systems containing
both digital and analog MultiVOIP units.
You will also find useful information in Chapter 2: Quick Start Guide.
See especially these sections:
Phonebook Starter Configuration
Phonebook Tips
Phonebook Example (One Common Situation)
Chapter 2 also contains a “Phonebook Worksheet” section. You may
want to print out several worksheet copies. Paper copies can be very
helpful in comparing phonebooks at multiple sites at a glance. This will
assist you in making the phonebooks clear and consistent and will
reduce ‘surfing’ between screens on the configuration program.
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Operation & Maintenance
Operation and Maintenance
Although most Operation and Maintenance functions of the software
are in the Statistics group of screens, an important summary appears in
the System Information of the Configuration screen group.
System Information screen
This screen presents vital system information at a glance. Its primary
use is in troubleshooting. This screen is accessible via the
Configuration pulldown menu, the Configuration sidebar menu, or by
the keyboard shortcut Ctrl + Alt + Y.
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System Information Parameter Definitions
Field Name Values Description
Boot
Version
nn.nn
alpha-
numeric
Indicates the version of the code that
is used at the startup (booting) of the
voip. The boot code version is
independent of the software version.
Firmware
Version
nn.nn.nn Indicates the version of the
alpha-
MultiVOIP firmware.
numeric
Configur-
ation
Version
nn.nn.
nn.nn
alpha-
numeric
Indicates the version of the
MultiVOIP configuration software.
Phone Book
Version
nn.nn
alpha-
numeric
Indicates the version of the
MultiVOIP phone book being used.
IFM Version nn
alpha-
Indicates version of the IFM module,
the device that performs the
transformation between telephony
signals and IP signals.
numeric
Mac
Address
numeric
Denotes the number assigned as the
voip unit’s unique Ethernet address.
Up Time
days:
hours:
mm:ss
Indicates how long the voip has been
running since its last booting.
Hardware
ID
alpha-
numeric
Indicates version of the MultiVOIP
circuit board assembly being used.
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The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
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MultiVOIP User Guide
Statistics Screens
Ongoing operation of the MultiVOIP, whether it is in a
MultiVOIP/PBX setting or MultiVOIP/telco-office setting, can be
monitored for performance using the Statistics functions of the
MultiVOIP software.
About Call Progress
Accessing Call-Progress Statistics
Channel Icons (Main Screen Lower Left)
Channel icons are green when data
traffic is present, red when idle.
In the web GUI, call progress details can be viewed by
clicking on an icon (one for each channel) arranged
similarly on the web-browser screen.
Pulldown
Icon
Shortcut
Sidebar
Alt + A
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The Call Progress Details Screen
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Call Progress Details: Field Definitions
Field Name
Values
Description
Channel
1-n
Number of data channel or time
slot on which the call is carried.
This is the channel for which call-
progress details are being viewed.
Call Details
Duration
Mode
Hours:
Minutes:
Seconds
The length of the call in hours,
minutes, and seconds (hh:mm:ss).
Indicates whether the call being
described was a voice call or a
FAX call.
Voice or FAX
Voice Coder
Packets Sent
G.723, G.729,
G.711, etc.
The voice coder being used on
this call.
integer value
integer value
integer value
integer value
integer value
0-9, #, *
The number of data packets sent
over the IP network in the course
of this call.
Packets Rcvd
Bytes Sent
The number of data packets
received over the IP network in
the course of this call.
The number of bytes of data sent
over the IP network in the course
of this call.
Bytes Rcvd
Packets Lost
The number of bytes of data
received over the IP network in
the course of this call.
The number of voice packets from
this call that were lost after being
received from the IP network.
Outbound
Digits
The digits transmitted by the
MultiVOIP to the PBX/telco for
this call.
Prefix
Matched
Displays the dialed digits that
were matched to a phonebook
entry.
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Call Progress Details: Field Definitions (cont’d)
From – To Details
Description
Gateway
Name
alphanumeric
string
Identifier for the VOIP gateway
that handled this call.
IP Address
x.x.x.x,
IP address from which the call
was received.
where x has a
range of 0 to
255
Options
SC, FEC
Displays VOIP transmission
options in use on the current call.
These may include Forward Error
Correction or Silence
Compression.
Silence
SC
“SC” stands for Silence
Compression
Compression. With Silence
Compression enabled, the
MultiVOIP will not transmit voice
packets when silence is detected,
thereby reducing the amount of
network bandwidth that is being
used by the voice channel.
Forward Error FEC
Correction
“FEC” stands for Forward Error
Correction. Forward Error
Correction enables some of the
voice packets that were corrupted
or lost to be recovered. FEC adds
an additional 50% overhead to the
total network bandwidth
consumed by the voice channel.
Default = Off
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Call Progress Details: Field Definitions (cont’d)
Field Name Values Description
Supplementary Services
Status
Call on Hold
alphanumeric
Describes held call by its IP
address source, location/gateway
identifier, and hold duration.
Location/gateway identifiers
comes from Gateway Name field
in Phone Book Configuration
screen of remote voip.
Call Waiting
alphanumeric
Describes waiting call by its IP
address source, location/gateway
identifier, and hold duration.
Location/gateway identifiers
comes from Gateway Name field
in Phone Book Configuration
screen of remote voip.
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Call Progress Details: Field Definitions (cont’d)
Field Name Values Description
Supplementary Services
Status
Caller ID
There are four
values:
“Calling Party
+ identifier”;
“Alerting
This field shows the identifier and
status of a remote voip (which has
Call Name Identification enabled)
with which this voip unit is
currently engaged in some voip
transmission. The status of the
engagement (Connected, Alerting,
Busy, or Calling) is followed by
the identifier of a specific channel
of a remote voip unit. This
Party +
identifier”;
“Busy Party
+ identifier”;
and
identifier comes from the “Caller
Id” field in the Supplementary
Services screen of the remote
voip unit.
“Connected
Party +
identifier”
Status
hangup, active Shows condition of current call.
Tun, FS + Tun, Displays the H.323 version 4
Call Control
Status
AE, Mux
features in use for the selected
call. These include tunneling
(Tun), Fast Start with tunneling
(FS + Tun), Annex E multiplexed
UDP call signaling transport (AE),
and Q.931 Multiplexing (Mux).
See Phonebook Configuration
Parameters (in T1 or E1 chapters)
for more on H.323v4 features.
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The Logs Screen
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Logs Screen Details: Field Definitions
Field Name
Values
Description
Log # column
1 or higher
All calls are assigned an event
number in chronological order,
with the most recent call having
the highest event number.
Start Date,Time
column
dd:mm:yyyy
hh:mm:ss
The starting time of the call (event).
The date is presented as a day
expression of one or two digits, a
month expression of one or two
digits, and a four-digit year. This is
followed by a time-of-day expression
presented as a two-digit hour, a two-
digit minute, and a two-digit seconds
value. (statistics, logs) field
This describes how long the call
(event) lasted in hours, minutes, and
seconds.
Duration column hh:mm:ss
Status column
Mode column
success or
failure
Displays the status of the call, i.e.,
whether the call was completed
successfully or not.
Indicates whether the (event) being
described was a voice call or a FAX
call.
voice or FAX
From column
To column
gateway name
gateway name
Displays the name of the voice
gateway that originates the call.
Displays the name of the voice
gateway that completes the call.
Special Buttons
Previous
Next
--
Displays log entry before
currently selected one.
Displays log entry after currently
selected one.
--
First
Last
Delete File
--
--
--
Displays first log entry
Displays last log entry.
Deletes selected log file.
Call Details
Packets sent
integer value
The number of data packets sent
over the IP network in the course
of this call.
Bytes sent
integer value
The number of bytes of data sent
over the IP network in the course
of this call.
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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
from this call that were lost
after being received from the IP
network.
Voice coder
G.723, G.729,
G.711, etc.
The voice coder being used on
this call.
Packets received
integer value
The number of data packets
received over the IP network in
the course of this call.
Bytes received
integer value
0-9, #, *
The number of bytes of data
received over the IP network in
the course of this call.
The digits transmitted by the
MultiVOIP to the PBX/telco for
this call.
Outbound digits
FROM Details
Gateway Name
alphanumeric
string
x.x.x.x,
where x has a
range of 0 to 255
FEC, SC
Identifier for the VOIP gateway
that originated this call.
IP address of the VOIP gateway
from which the call was
received.
Displays VOIP transmission
options used by the VOIP
gateway originating the call.
These may include Forward
Error Correction or Silence
Compression.
IP Address
Options
TO Details
Gateway Name
IP Address
Options
alphanumeric
string
Identifier for the VOIP gateway
that completed (terminated)
this call.
IP address of the VOIP gateway
at which the call was completed
(terminated).
Displays VOIP transmission
options used by the VOIP
gateway terminating the call.
These may include Forward
Error Correction or Silence
Compression.
x.x.x.x,
where x has a
range of 0 to 255
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Logs Screen Details: Field Definitions (cont’d)
Supplementary Services Info
Call Transferred
To
phone number
string
Number of party called in
transfer.
Call Forwarded
To
phone number
string
Number of party called in
forwarding.
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IP Statistics: Field Definitions
Field
Values
Description
Name
UDP versus TCP. (User Datagram
Protocol versus Transmission Control
Protocol). UDP provides
unguaranteed, connectionless
transmission of data across an IP
network. By contrast, TCP provides
reliable, connection-oriented
transmission of data.
Both TCP and UDP split data into
packets called “datagrams.” However,
TCP includes extra headers in the
datagram to enable retransmission of
lost packets and reassembly of packets
into their correct order if they arrive out
of order. UDP does not provide this.
Lost UDP packets are unretrievable;
that is, out-of-order UDP packets
cannot be reconstituted in their proper
order..
Despite these obvious disadvantages,
UDP packets can be transmitted much
faster than TCP packets -- as much as
three times faster. In certain
applications, like audio and video data
transmission, the need for high speed
outweighs the need for verified data
integrity. Sound or pictures often
remain intelligible despite a certain
amount of lost or disordered data
packets (which appear as static).
Clears packet tallies from memory.
“Clear”
button
--
Total Packets
Transmit integer
Sum of data packets of all types.
Total number of packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
ted
value
within the MultiVOIP software.
Total number of packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
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IP Statistics: Field Definitions (cont’d)
Values Description
Field
Name
Total Packets
(cont’d)
Sum of data packets of all types.
Received integer
Total number of error-laden packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
UDP Packets
User Datagram Protocol packets.
Transmit integer
Number of UDP packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of UDP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden UDP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
TCP Packets
Transmission Control Protocol packets.
Transmit integer
Number of TCP packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of TCP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden TCP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
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IP Statistics: Field Definitions (cont’d)
RTP Packets
Voice signals are transmitted in
Realtime Transport Protocol packets.
RTP packets are a type or subset of
UDP packets.
Transmit integer
Number of RTP packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of RTP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden RTP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
RTCP Packets
Realtime Transport Control Protocol
packets convey control information to
assist in the transmission of RTP (voice)
packets. RTCP packets are a type or
subset of UDP packets.
Transmit integer
Number of RTCP packets transmitted
by this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of RTCP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden RTCP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
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About Link Management
The Link Management screen is essentially an automated utility for
pinging endpoints on your voip network. This utility generates pings
of variable sizes at variable intervals and records the response to the
pings.
Accessing Link Management
Pulldown
none
Shortcut/Icon
Sidebar
none/none
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Link Management screen Field Definitions
Field Name Values Description
Monitor Link fields
IP Address to
Ping
a.b.c.d
0-255
This is the IP address of the target
endpoint to be pinged.
No. of Pings
1-999
This field determines how many
pings will be generated by the
Start Now command.
Response
Timeout
500 – 5000
milliseconds
The duration after which a ping
will be considered to have failed.
Ping Size in
Bytes
32 – 128 bytes
This field determines how long or
large the ping will be.
Timer Interval 0 or 30 – 6000
between Pings minutes
This field determines how long of
a wait there is between one ping
and the next.
Start Now
command
button
--
--
Initiates pinging.
Clear
command
button
Erases ping parameters in
Monitor Link field group and
restores default values.
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Link Management screen Field Definitions (cont’d)
Field Name Values Description
Link Status Parameters
These fields summarize the results
of pinging.
IP Address
column
a.b.c.d
0-255
Target of ping.
No. of Pings
Sent
as listed
as listed
as listed,
Number of pings sent to target
endpoint.
No. of Pings
Received
Number of pings received by
target endpoint.
Round Trip
Delay
(Min/Max/
Avg)
Displays how long it took from
in milliseconds time ping was sent to time ping
response was received.
Last Error
as listed
Indicates when last data error
occurred.
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About Packetization Time
You can use the Packetization Time screen to specify definite
packetization rates for coders selected in the Voice/FAX Parameters
screen (in the “Coder Options” group of fields). The Packetization
Time screen is accessible under the “Advanced” options entry in the
sidebar list of the main voip software screen. In dealing with RTP
parameters, the Packetization Time screen is closely related to both
Voice/FAX Parameters and to IP Statistics. It is located in the
“Advanced” group for ease of use.
Accessing Packetization Time
Pulldown
Shortcut/Icon
Sidebar
none/none
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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 Increments
Coder Types
Range (in Kbps);
{default value}
5-120
30-120
10-120
20-120
Increments (in Kbps)
G711, G726, G727
G723
G729
{5}
5
{30}
{10}
{20}
30
10
20
Netcoder
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Once the packetization rate has been set for one channel, it can be copied into other
channels.
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About T1/E1 and BRI Statistics
Accessing 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
and MVP810ST
.
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T1 Statistics Screen
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T1 Statistics: Field Definitions
Field
Values
Description
Name
Red Alarm
Integer tally of
alarms
counted since
last reset.
The alarm condition declared when a
device receives no signal or cannot
synchronize to the signal being received. A
Red Alarm is generated if the incoming
data stream has no transitions for 176
consecutive pulse positions.
Blue Alarm
Tally since last
reset.
Alarm signal consisting of all 1’s (including
framing bit positions) which indicates
disconnection or failure of attached equipment.
Loss of
Frame
Tally since last Loss of data frame synchronization.
reset.
Alignment
Excessive
Zeroes
Tally since last Displayed value will increment if
reset.
consecutive zeroes beyond a set threshold
are detected. I.e., tally increments if more
than 7 consecutive zeroes in the received
data stream are detected under B8ZS line
coding, or if 15 consecutive zeroes are
detected under AMI line coding.
Signaling has been frozen at the most
recent values due to loss of frame
alignment, loss of multiframe
Status
Freeze
Signaling
Active
alignment or due to a receive slip.
Line loopback deactivation signal has
been detected in the receive bit stream.
Line
Loopback
Deactivation
Signal
A short exists between the transmit pair for
at least 32 consecutive pulses.
Transmit
Line Short
For use by MTS Technical Support
personnel.
Transmit
Data
Overflow
The frequency of the transmit clock is less
than the frequency of the transmit system
interface working clock. A frame is
repeated.
Transmit
Slip Positive
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T1 Statistics: Field Definitions (cont’d)
Values Description
Field
Name
Yellow
Alarm
Tally since last The alarm signal sent by a remote T1/E1
reset.
device to indicate that it sees no receive
signal or cannot synchronize on the
receive signal.
[To be supplied.]
Frame
Search
Restart Flag
Loss of
Tally since last In D4 or ESF mode, displayed value will
MultiFrame
Alignment
reset.
increment if multiframe alignment has
been lost or if loss of frame alignment has
been detected.
Transmit
Slip
Tally since last Slip in transmitted data stream. Slips
reset.
indicate a clocking mismatch (or lack of
synchronization) between T1/E1 devices.
When slips occur, data may be lost or
repeated.
Pulse
Density
Violation
The pulse density of the received data
stream is below the requirement defined
by ANSI T1.403 or more than 15
consecutive zeros are detected.
Line
The line loopback activation signal
has been detected in the received bit
stream.
Loopback
Activation
Signal
Transmit
Line Open
At least 32 consecutive zeros were
transmitted.
Transmit
Data
For use by MTS Technical Support
Personnel.
Underrun
Transmit
Slip
Negative
The frequency of the transmit clock is
greater than the frequency of the
transmit system interface working
clock. A frame is skipped.
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T1 Statistics: Field Definitions (cont’d)
Field
Values
Description
Name
Two successive pulses of the same
polarity have been received and these
pulses are not part of zero substitution.
On an AMI-encoded line, this represents a
line error. On a B8ZS line, this may
represent the substitution for a string of 8
zeroes.
Bipolar
Violation
Integer tally of
violation count
since last reset.
Receive Slip
Tally since last A receive slip (positive or negative) has
reset.
occurred. Slips indicate a clocking
mismatch (or lack of synchronization)
between T1/E1 devices. When slips occur,
data may be lost or repeated.
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E1 Statistics Screen
E1 Statistics: Field Definitions
Field
Values
Description
Name
Red Alarm
Integer tally of
alarms
counted since
last reset.
The alarm condition declared when a
device receives no signal or cannot
synchronize to the signal being received. A
Red Alarm is generated if the incoming
data stream has no transitions for 176
consecutive pulse positions.
Blue Alarm
Tally since last Alarm signal consisting of all 1’s (including
reset.
framing bit positions) which indicates
disconnection or failure of attached
equipment.
Loss of
Frame
Tally since last Loss of data frame synchronization.
reset.
Alignment
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E1 Statistics: Field Definitions (cont’d)
Values Description
Field
Name
Detected alarm indication signal in
timeslot 16 according to ITU-T G.775.
Indicates the incoming time slot 16
contains less than 4 zeros in each of
two consecutive time slot 16
Receive
Timeslot 16
Alarm
Indication
Signal
multiframe periods.
Transmit
Line Short
A short exists between the transmit
pair for at least 32 consecutive pulses.
Transmit
Data
For use by MTS personnel.
Overflow
Transmit
Slip Positive
The frequency of the transmit clock is
less than the frequency of the transmit
system interface working clock. A
frame is repeated.
Yellow
Alarm
Tally since last The alarm signal sent by a remote T1/E1
reset.
device to indicate that it sees no receive
signal or cannot synchronize on the
receive signal.
Status
Signaling has been frozen at the most
recent values due to loss of frame
alignment, loss of multiframe alignment
or due to a receive slip.
Freeze
Signaling
Active
Loss of
Tally since last In D4 or ESF mode, displayed value will
MultiFrame
Alignment
reset.
increment if multiframe alignment has
been lost or if loss of frame alignment has
been detected.
Receive
Timeslot 16
Loss of
The time slot 16 data stream contains all
zeros for at least 16 contiguously received
time slots.
Signal
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E1 Statistics: Field Definitions (cont’d)
Field
Values
Description
Name
Receive
The framing pattern '0000' in 2
Timeslot 16
Loss of
MultiFrame
Alignment
consecutive CAS multiframes were not
found or in all time slot 16 of the previous
multiframe all bits were reset.
At least 32 consecutive zeroes were
transmitted.
Transmit
Line Open
For use by MTS Technical Support
Personnel.
Transmit
Data
Underrun
The frequency of the transmit clock is
greater than the frequency of the transmit
system interface working clock. A frame
is skipped.
Transmit
Slip
Negative
Bipolar Violation (or BPV) refers to two
successive pulses of the same polarity on
the E1 line. On an AMI-encoded line, this
represents a line error. On a B8ZS line,
this may represent the substitution for a
string of 8 zeroes.
Bipolar
Violation
Integer tally of
violation count
since last reset.
Excessive
Zeroes
Tally since last Displayed value will increment if
reset.
consecutive zeroes beyond a set threshold
are detected. I.e., tally increments if more
than 7 consecutive zeroes in the received
data stream are detected under B8ZS line
coding, or if 15 consecutive zeroes are
detected under AMI line coding.
Transmit
Slip
Tally since last Slip in transmitted data stream. Slips
reset.
indicate a clocking mismatch (or lack of
synchronization) between T1/E1 devices.
When slips occur, data may be lost or
repeated.
Receive Slip
Tally since last
reset.
Slip in received data stream. Slips indicate a
clocking mismatch (or lack of synchronization)
between T1/E1 devices. When slips occur, data
may be lost or repeated.
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ISDN BRI Statistics Screen
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ISDN BRI Statistics: Field Definitions
Field
Values
Description
Name
Select BRI
Interface
ISDNn
In this field, you can choose the ISDN
port for which you want to view the
status. The 410ST has two ISDN –BRI
ports (or “interfaces”); the 810ST has
four ISDN-BRI ports (or “interfaces”).
Each interface has two channels.
For n=1-2 (410ST)
For n-1-4 (810ST)
Layer 1 Interface
Status
inactive (F1),
sensing (F2),
Shows the current Layer 1 status of the
ISDN connection. Each status
description (inactive, sensing, etc.)
corresponds to a particular “state” label
(F1-F8 and G1-G4).
deactivated (F3),
awaiting signal (F4),
identifying input (F5),
synchronized (F6),
activated (F7),
lost framing (F8),
deactive (G1),
pending
activation (G2),
active (G3),
pending
deactivation (G4)
State
F1-F8 (for Terminal
mode ports),
Shows the I.430 state name for Layer 1.
An “F” state name indicates this port is
in Terminal mode (F1-F8), as set in the
ISDN BRI Parameters screen.
G1-G4 (for Network
mode ports)
A “G” state name indicates that this
port is in Network mode (G1-G4), as set
in the ISDN BRI Parameters screen.
Shows the number of lost-framing
events on the ISDN physical layer.
Loss Of
Framing
integer
integer
Loss of
Sync
Shows the number of lost-
synchronization events on the ISDN
physical layer.
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ISDN BRI Statistics: Field Definitions
(continued)
Field
Values
Description
Name
Switch Information:
TEI Assignment
TEI 0
through
TEI 7
0-63 (point-to-point
assignments)
Displays the value for each TEI
assigned to the BRI port. The TEI
(Terminal Endpoint Identifier) uniquely
identifies each device connected to the
ISDN physical layer.
64-126 (automatic
assignments)
Switch Information:
D-Channel Information
Shows the number of packets
transmitted on the channel. When the
value exceeds 4294967295 packets, it
will reset to zero and continue
counting.
Tx Packets
0 to 4294967295
Shows the number of packets received
on the channel. When the value exceeds
4294967295 packets, it will reset to zero
and continue counting.
Rx Packets
0 to 4294967295
Switch Information:
SPID 0
A SPID (Service Profile Identifier) is
assigned by the ISDN provider and
pertains to one channel of the BRI
interface (port), in this case channel
0. The SPID identifies an ISDN
terminal uniquely. The SPID
associates a set of services (features)
with the terminal. (In Terminal
mode the provider is a telco or PBX.
In Network mode MultiVOIP is the
provider.) A SPID is only used
when the “Country” field is set to
“USA” in the ISDN BRI Parameters
screen.
(SPID 0
number)
numeric,
3 to 20 digits
Status
Not Checked,
Correct,
Indicates whether SPID0 is correct,
incorrect, or not being checked.
Incorrect
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ISDN BRI Statistics: Field Definitions
(continued)
Field
Values
Description
Name
Switch Information:
SPID 1
SPID for channel 1 of the BRI
interface. Otherwise, same as
SPID0 description above.
(SPID 1
number)
numeric
Indicates whether SPID1 is correct,
incorrect, or not being checked.
Status
Not Checked,
Correct,
Incorrect
“Clear”
button
Clears (sets to zero) all
ISDN BRI Statistics fields
with numeric tally values (these are
Loss of Framing, Loss of Sync, Tx
Packets, Rx Packets).
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About Registered Gateway Details
The Registered Gateway Details screen presents a real-time display of
the special operating parameters of the Single Port Protocol (SPP).
These are configured in the PhoneBook Configuration screen and in
the Add/Edit Outbound PhoneBook screen.
Accessing Registered Gateway Details
Pulldown
Icon
Shortcut
Sidebar
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Registered Gateway Details: Field Definitions
Field
Values
Description
Name
Column Headings
Description
alphanumeric
This is a descriptor for a particular voip
gateway unit. This descriptor should
generally identify the physical location of
the unit (e.g., city, building, etc.) and
perhaps even its location in an equipment
rack.
IP Address
Port
n.n.n.n,
The RAS address for the gateway.
for n = 0-255
Port by which the gateway exchanges
H.225 RAS messages with the gatekeeper. .
Register
Duration
The time remaining in seconds before the
TimeToLive timer expires. If the gateway
fails to reregister within this time, the
endpoint is unregistered.
The current status of the gateway, either
registered or unregistered.
Status
No. of
Entries
The number of gateways currently
registered to the Registrar. This includes all
SPP clients registered and the Registrar
itself.
Details
Count of
Registered
Numbers
If a registered gateway is selected (by
clicking on it in the screen), The "Count of
Registered Numbers" will indicate the
number of registered phone numbers for the
selected gateway. When a client registers, all
of its inbound phonebook's phone numbers
become registered.
Lists all of the registered phone numbers for
the selected gateway.
List of
Registered
Numbers
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MultiVoip Program Menu Items
After the MultiVoip program is installed on the PC, it can be
launched from the Programs group of the Windows Start menu ( Start
| Programs | MultiVOIP ____ | … ). In this section, we describe the
software functions available on this menu.
Several basic software functions are accessible from the MultiVoip
software menu, as shown below.
MultiVOIP Program Menu
Menu Selection
Description
Configuration
Select this to enter the Configuration
program where values for IP,
telephony, and other parameters are
set.
Configuration Port Setup
Date and Time Setup
Select this to access the COM Port
Setup screen of the MultiVOIP
Configuration program.
Select this for access to set
calendar/clock used for data logging.
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MultiVOIP Program Menu (cont’d)
Description
Menu Selection
Download Factory Defaults Select this to return the configuration
parameters to the original factory
values.
Download Firmware
Select this to download new versions
of firmware as enhancements become
available.
Download IFM Firmware
Select this to download new versions
of IFM firmware as enhancements
become available. The Interface
Module (IFM) is the telephony
interface for analog MultiVOIP units
(MVP130, MVP130FXS, MVP210,
MVP410, MVP810). There is one IFM
for each channel of the MultiVOIP
unit. For each channel, the IFM
handles the analog signals to and from
the attached telephone, PBX or CO
line.
Download User Defaults
Set Password
To be used after a full set of parameter
values, values specified by the user,
have been saved (using Save Setup).
This command loads the saved user
defaults into the MultiVOIP.
Select this to create a password for
access to the MultiVOIP software
programs (Program group commands,
Windows GUI, web browser GUI, &
FTP server). Only the FTP Server
function requires a password for access.
The FTP Server function also requires
that a username be established along
with the password.
Uninstall
Select this to uninstall the MultiVOIP
software (most, but not all components
are removed from computer when this
command is invoked).
Upgrade Software
Loads firmware (including H.323
stack) and factory default settings from
the controller PC to the MultiVOIP
unit.
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“Downloading” here refers to transferring program files from the PC to
the nonvolatile “flash” memory of the MultiVOIP. Such transfers are
made via the PC’s serial port. This can be understood as a “download”
from the perspective of the MultiVOIP unit.
When new versions of the MultiVoip software become available, they
will be posted on MultiTech’s web or FTP sites. Although transferring
updated program files from the MultiTech web/FTP site to the user’s
PC can generally be considered a download (from the perspective of
the PC), this type of download cannot be initiated from the MultiVoip
software’s Program menu command set.
Generally, updated firmware must be downloaded from the MultiTech
web/FTP site to the PC before it can be loaded from the PC to the
MultiVOIP.
Configuration Option
The “Configuration” option in the MultiVOIP Program menu launches
the MultiVOIP Configuration software program.
Configuration Port Setup
The Configuration Port Setup option in the MultiVOIP Program menu
brings up the COM Port Setup screen of the MultiVOIP configuration
software.
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Date and Time Setup
The dialog box below allows you to set the time and date indicators of
the MultiVOIP system.
Obtaining Updated Firmware
Generally, updated firmware must be downloaded from the MultiTech
web/FTP site to the user’s PC before it can be downloaded from that
PC to the MultiVOIP.
Note that the structure of the MultiTech web/FTP site may change
without notice. However, firmware updates can generally be found
using standard web techniques. For example, you can access updated
firmware by doing a search or by clicking on Support.
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If you conduct a search, for example, on the word “MultiVoip,” you
will be directed to a list of firmware that can be downloaded.
If you choose Support, you can select “MultiVoip” in the Product
Support menu and then click on Firmware to find MultiVOIP
resources.
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Once the updated firmware has been located, it can be downloaded
from the web/ftp site using normal PC/Windows procedures. While
the next 3 screens below pertain to the MVP3010, similar screens will
appear for any MultiVOIP model described in this manual.
MVP3000x.EXE from ftp.multitech.com
Saving:
MVP3000x.EXE from ftp.multitech.com
Estimated time left: Not known (Opened so far 781 KB)
Download to:
Transfer rate:
C:\VoipSystem\MVP3000\...\MVP301f.EXE
260 KB/sec
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Generally, the firmware file will be a self-extracting compressed file
(with .zip extension), which must be expanded (decompressed, or
“unzipped”) on the user’s PC in a user-specified directory.
C:\Acme-Inc\MVP3000-firm
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Implementing a Software Upgrade
MultiVOIP software can be upgraded locally using a single command
at the MultiVOIP Windows GUI, namely Upgrade Software. This
command downloads firmware (including the H.323 stack), and
factory default settings from the controller PC to the MultiVOIP unit.
When using the MultiVOIP Windows GUI, firmware and factory
default settings can also be transferred from controller PC to MultiVOIP
piecemeal using separate commands.
When using the MultiVOIP web browser GUI to control/configure the
voip remotely, upgrading of software must be done on a piecemeal
basis using the FTP Server function of the MultiVOIP unit.
When performing a piecemeal software upgrade (whether from the
Windows GUI or web browser GUI), follow these steps in order:
1. Identify Current Firmware Version
2. Download Firmware
3. Download Factory Defaults
When upgrading firmware, the software commands “Download
Firmware,” and “Download Factory Defaults” must be implemented in
order, else the upgrade is incomplete.
Identifying Current Firmware Version
Before implementing a MultiVOIP firmware upgrade, be sure to verify
the firmware version currently loaded on it. The firmware version
appears in the MultiVoip Program menu. Go to Start | Programs |
MultiVOIP ____ x.xx. The final expression, x.xx, is the firmware
version number. In the illustration below, the firmware version is
4.00a, made for the E1 MultiVOIP (MVP3010).
When a new firmware version is installed, the MultiVOIP software can
be upgraded in one step using the Upgrade Software command, or
piecemeal using the Download Firmware command and the
Download Factory Defaults command.
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Download Firmware transfers the firmware (including the H.323
protocol stack) in the PC’s MultiVOIP directory into the nonvolatile
flash memory of the MultiVOIP.
Download Factory Defaults sets all configuration parameters to the
standard default values that are loaded at the MultiTech factory.
Upgrade Software implements both the Download Firmware
command and the Download Factory Defaults command.
Downloading Firmware
1. 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.
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3. If a password has been established, the Password Verification screen
will appear.
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
the file transfer process.
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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.
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Downloading Factory Defaults
1. The MultiVoip Configuration program must be off when invoking
the Download Factory Defaults 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 Factory Defaults.
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3. If a password has been established, the Password Verification screen
will appear.
Type in the password and click OK.
4. The MVP____- Firmware screen appears saying “MultiVOIP [model
number] 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
the file transfer process.
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5. After the PC gets a response from the MultiVOIP, the Dialog – IP
Parameters screen will appear.
The user should verify that the correct IP parameter values are listed
on the screen and revise them if necessary. Then click OK.
6. Progress bars will appear at the bottom of the screen during the data
transfer.
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.
7. The Download Factory Defaults procedure is complete.
Downloading IFM Firmware (Analog Voips only)
The Interface Module (IFM) is the telephony interface for analog
MultiVOIP units (MVP130, MVP130FXS, MVP210, MVP410, MVP810).
There is one IFM for each channel of the MultiVOIP unit. For each
channel, the IFM handles the analog signals to and from the attached
telephone, PBX or CO line. The IFM communicates with the main
processor indicating the status of the telephone line. For example, it
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might indicate that a phone is off hook (FXS) or that an incoming ring is
present (FXO). The IFM receives operating instructions from the voip’s
main processor. For example, the IFM might be instructed to ring the
phone (FXS) or seize the line (FXO). The IFM contains a codec
(coder/decoder) to convert the incoming audio to a PCM stream (pulse
code modulation) which it sends to the DSP (digital signal processor).
The IFM’s codec also converts outgoing PCM to audio.
The firmware of the IFMs will change from time to time and you may
need to upgrade the firmware on your MultiVOIP unit. To do so,
follow these instructions.
1. In the System Information screen of the MultiVOIP Configuration
software, check the version number of the IFM firmware already
installed on the MultiVOIP unit. Write down the version number.
2. Exit the Configuration software program. The MultiVoip
Configuration program must be off when invoking the Download
IFM Firmware command. If it is on, the command will not work.
3.To invoke the Download IFM Firmware command, go to Start |
Programs | MVP____ x.xx | Download IFM Firmware.
4. A warning window will appear: “Downloading IFM Firmware will
reboot the MultiVOIP. Do you want to continue?” Click OK.
4. The “Boot” LED on the front panel of the MultiVOIP will come on.
5. The software will search for an IFM firmware file to use to upgrade
the system. If the file found represents firmware newer than that
already installed on the MultiVOIP (or if you want to overwrite the
same version of firmware) click Open.
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6. The IFM Firmware Download screen will appear. Select “Copy to
All IFMs” and click OK. (Only in very special circumstances would
different IFMs in the same voip be loaded with different IFM
firmware.)
7. The main MultiVOIP Configuration screen will appear. Progress
bars can be seen at the bottom of the screen while files are being
copied.
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8. Then a completion screen entitled IFM Test will appear.
Click OK.
9. The MultiVOIP will reboot itself. When the reboot is complete, the
MultiVOIP Configuration screen will close.
10. The IFM firmware downloading process is complete.
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Setting and Downloading User Defaults
The Download User Defaults command allows you to maintain a known
working configuration that is specific to your VOIP system. You can then
experiment with alterations or improvements to the configurations confident
that a working configuration can be restored if necessary.
1. Before you can invoke the Download User Defaults command, you
must first save a set of configuration parameters by using the Save
Setup command in the sidebar menu of the MultiVOIP software.
2. Before the setup configuration is saved, you will be prompted to save
the setup 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 [model
number] 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 the file transfer
process.
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4. Progress bars will appear during the file transfer process.
5. When the file transfer process is complete, the Dialog-- IP
Parameters screen will appear.
6. Set the IP values per your particular VOIP system. Click OK.
Progress bars will appear as the MultiVOIP reboots itself.
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Setting a Password (Windows GUI)
After a user name has been designated and a password has been set,
that password is required to gain access to any functionality of the
MultiVOIP software. Only one user name and password can be
assigned to a voip unit. The user name will be required when
communicating with the MultiVOIP via the web browser GUI.
NOTE: Record your user name and password in a safe place. If
the password is lost, forgotten, or unretrievable, the user
must contact MultiTech Tech Support in order to resume
use of the MultiVOIP unit.
1. The MultiVoip configuration program must be off when invoking
the Set Password command. If it is on, the command will not work.
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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.
4. The Password screen will appear. If you intend to use the FTP Server
function that is built into the MultiVOIP, enter a user name. (A User
Name is not needed to access the local Windows GUI, the web
browser GUI, or the commands in the Program group.) Type your
password in the Password field of the Password screen. Type this
same password again in the Confirm Password field to verify the
password you have chosen.
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NOTE: Be sure to write down your password in a convenient but
secure place. If the password is forgotten, contact
MultiTech Technical 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-
boot itself and, in so doing, its BOOT LED will light up.
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6. After the password has been set, the user will be required to enter the
password to gain access to the web browser GUI and any part of the
MultiVOIP software listed in the Program group menu. User Name
and Password are both needed for access to the FTP Server residing in
the MultiVOIP.
When MultiVOIP program asks for password at launch of program, the
program will simply shut down if CANCEL is selected.
The MultiVOIP program will produce an error message if an invalid
password is entered.
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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 software functions (Windows GUI, web browser GUI, FTP
server, and all Program menu commands, e.g., Upgrade Software –
only the FTP Server function requires a User Name in addition to the
password). After a password has been set, that password is required to
access the MultiVOIP web browser GUI.
NOTE: Record your user name and password in a safe place. If
the password is lost, forgotten, or unretrievable, the user
must contact MultiTech Tech Support in order to resume
use of the MultiVOIP web browser GUI.
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Un-Installing the MultiVOIP Software
1. To un-install the MultiVOIP configuration software, go to Start |
Programs and locate the MultiVOIP entry. Select Uninstall MVP____
vx.xx (versions may vary).
2. Two confirmation screens will appear. Click Yes and OK when you
are certain you want to continue with the uninstallation process.
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3. A special warning message similar to that shown below may appear
for the MultiVOIP software’s “.bin” file. Click Yes.
An option that you selected requires that files be installed to your system,
or files be uninstalled from your system, or both. A read-only file,
C:\ProgramFiles\MVP3000\v4.00a\mvpt1.bin was found while
performing the needed file operations on your system.
To perform the file operation, click the Yes button;
otherwise, click No.
4. A completion screen will appear.
Click Finish.
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Upgrading Software
As noted earlier (see the section Implementing a Software Upgrade above),
the Upgrade Software command transfers, from the controller PC to the
MultiVOIP unit, firmware (including the H.323 stack) and factory
default configuration settings. As such, Upgrade Software implements
the functions of both Download Firmware and Download Factory
Defaults in a single command.
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FTP Server File Transfers (“Downloads”)
With the 4.03/6.03 software release, MultiTech has built an FTP server
into the MultiVOIP unit. Therefore, file transfers from the controller PC
to the voip unit 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
this context. File transfers from a client to a server are typically
considered “uploads.” File transfers from a large repository of data to
machines with less data capacity are considered “downloads.” In this
case, these metaphors are contradictory: the FTP server is actually
housed in the MultiVOIP unit, and the 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 the transfer of files from the PC
to the voip “downloads.” (Be aware that some FTP client programs
may use the opposite terminology, i.e., they may refer to the file
transfer as an “upload “)
You can download firmware, CAS telephony protocols, default
configuration parameters, and phonebook data for the MultiVOIP unit
with this FTP functionality. These downloads are done over a network,
not by a local serial port connection. Consequently, voips at distant
locations can be updated from a central control point.
The phonebook downloading feature greatly reduces the data-entry
required to establish 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 the first few voip units have
been compiled, phonebooks for additional voips become much simpler:
you copy the common material by downloading and then do data entry
for the few phonebook items that are unique to that particular voip unit
or voip site.
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Operation & Maintenance
To transfer files using the FTP server functionality in the MultiVOIP,
follow these directions.
1. Establish Network Connection and IP Addresses. Both the
controller PC and the MultiVOIP unit(s) must be connected to the same
IP network. An IP address 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. (When connection is made via a local serial connection
between the PC and the voip unit, no user name is needed.)
As shown above, the username and password can be set in the web
GUI as well as in the Windows GUI.
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3. Install FTP Client Program or Use Substitute. You should install an
FTP client program on the controller PC. FTP file transfers can be done
using a web browser (e.g., Netscape or Internet Explorer) in conjunction
with a local Windows browser a (e.g., Windows Explorer), but this
approach is somewhat clumsy (it requires use of two application
programs rather than one) and it limits downloading to only one VOIP
unit at a time. With an FTP client program, multiple voips can receive
FTP file transmissions in response to a single command (the transfers
may occur serially however).
Although MultiTech does not provide an FTP client program with the
MultiVOIP software or endorse any particular FTP client program, we
remind our readers that adequate FTP programs are readily available
under retail, shareware and freeware licenses. (Read and observe any
End-User License Agreement carefully.) Two examples of this are the
“WSFTP” client and the “SmartFTP” client, with the former having an
essentially text-based interface and 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
on the “FTP Server: Enable” box.
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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 file to be transferred will have “Ftp” as the part of its filename
just before the suffix (or extension). So, for example, the file
“mvpt1Ftp.bin” can be transferred to update the bin file (firmware)
residing in the MultiVOIP. Similarly, the file “fxo_loopFtp.cas” could
be transferred to enable use of the FXO Loop Start telephony interface
in one of the analog voip units and the file “r2_brazilFtp.cas” could be
transferred to enable a particular telephony protocol used in Brazil.
File Type
File Names
Description
firmware
“bin” file
mvpt1Ftp.bin
This is the MultiVOIP
firmware file. Only one
file of this type will be
in the directory.
factory defaults
fdefFtp.cnf
This file contains
factory default settings
for user-changeable
configuration
parameters. Only one
file of this type will be
in the directory.
CAS file
fxo_loopFtp.cas,
em_winkFtp.cas, for Channel Associated
These telephony files are
r2_brazilFtp.cas
r2_chinaFtp.cas
Signaling. The directory
contains many CAS files,
some labeled for specific
functionality, others for
countries or regions where
certain attributes are
standard.
H323 PDL file
This file is specific to
the particular version of
the H.323 standard
being used. This file
rarely needs to be
updated.
inbound
phonebook
InPhBk.tmr
This file updates the
inbound phonebook in
the MultiVOIP unit.
outbound
phonebook
OutPhBk.tmr
This file updates the
outbound phonebook in
the MultiVOIP unit.
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Operation and Maintenance
MultiVOIP User Guide
6. Contact MultiVOIP FTP Server. You must make contact with the
FTP Server in the voip using either a web browser or FTP client
program. Enter the IP address of the MultiVOIP’s FTP Server. If you
are using a browser, the address must be preceded by “ftp://”
(otherwise you’ll reach the web GUI within the MultiVOIP unit).
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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 to be done with a web browser (see first screen below) or
with an FTP client program (see second screen below).
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MultiVOIP User Guide
8. Invoke Download. Downloading can be done with a web browser
or with an FTP client program.
8A. Download with Web Browser.
8A1. In the local Windows browser, locate the directory
holding the MultiVOIP program files. The default
location will be C:\Program Files \Multi-Tech Systems
\MultiVOIP xxxx yyyy (where x and y represent
MultiVOIP model numbers and software version
numbers).
8A2. Drag-and-drop files from the local Windows browser (e.g.,
Windows Explorer) to the web browser.
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You may be asked to confirm the overwriting of files on the MultiVOIP.
Do so.
File transfer between PC and voip will look like transfer within voip
directories.
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8B. Download with FTP Client Program.
8B1. In the local directory browser of the FTP client program,
locate the directory holding the MultiVOIP program files.
The default location will be C:\Program Files \Multi-Tech
Systems \MultiVOIP xxxx yyyy (where x and y represent
MultiVOIP model numbers and software version
numbers).
8B2. In the FTP client program window, drag-and-drop files
from the local browser pane to the pane for the MultiVOIP
FTP server. FTP client GUI operations vary. In some
cases, you can choose between immediate and queued
transfer. In some cases, there may be automated
capabilities to transfer to multiple destinations with a
single command.
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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
the MultiVOIP.
10. Log Out of FTP Session. Whether the file transfer was done with a
web browser or with an FTP client program, you must log out of the
FTP session before opening the MultiVOIP Windows GUI.
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MultiVOIP User Guide
Web Browser Interface
Beginning with the 4.03/6.03 software release, you can control the
MultiVOIP unit with a graphic user interface (GUI) based on the
common web browser platform. Qualifying browsers are
InternetExplorer6 and Netscape6.
MultiVOIP Web Browser GUI Overview
Function
Remote configuration and control
of MultiVOIP units.
Configuration
Prerequisite
Local Windows GUI must be used
to assign IP address to MultiVOIP.
Browser Version
Requirement
Internet Explorer 6.0 or higher; or
Netscape 6.0 or higher
Java Requirement
Video Usability
Java Runtime Environment
version 1.4.0_01 or higher
(this application program is
included with MultiVOIP)
large video monitor recommended
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Operation & Maintenance
The initial configuration step of assigning the voip unit an IP address
must still be done locally using the Windows GUI. However, all
additional configuration can be done via the web GUI.
The content and organization of the web GUI is directly parallel to the
Windows GUI. For each screen in the Windows GUI, there is a
corresponding screen 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
menus whereas the web GUI does not.
The web GUI, however, cannot perform logging in the same direct
mode done in the Windows GUI. However, when the web GUI is used,
logging can be done by email (SMTP).
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MultiVOIP User Guide
The graphic layout of the web GUI is also somewhat larger-scale than
that of the Windows GUI. For that reason, it’s helpful to use as large of
a video monitor as possible.
The primary advantage of the web GUI is remote access for control and
configuration. The controller PC and the MultiVOIP unit itself must
both be connected to the same IP network and their IP addresses must
be known.
In order to use the web GUI, you must also install a Java application
program on the controller PC. This Java program is included on the
MultiVOIP product CD. ). Java is needed to support drop-down menus
and multiple windows in the web GUI.
To install the Java program, go to the Java directory on the MultiVOIP
product CD. Double-click on the EXE file to begin the installation.
Follow the instructions on the Install Shield screens.
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During the installation, you must specify which browser you’ll use in
the Select Browsers screen.
When installation is complete, the Java program becomes accessible in
your Start | Programs menu (Java resources are readily available via
the web). However, the Java program runs automatically in the
background as a plug-in supporting the MultiVOIP web GUI. No overt
user actions are required.
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MultiVOIP User Guide
After the Java program has been installed, you can access the
MultiVOIP using the web browser GUI. Close the MultiVOIP
Windows GUI. Start the web browser. Enter the IP address of the
MultiVOIP unit. Enter a password when prompted. (A password is
needed here only if password has been set for the local Windows GUI
or for the MultiVOIP’s FTP Server function. See “Setting a Password --
Web Browser GUI” earlier in this chapter.) The web browser GUI
offers essentially the same control over the voip as can be achieved
using the Windows GUI. As noted earlier, logging functions cannot be
handled via the web GUI. And, because network communications will
be slower than direct communications over a serial PC cable, command
execution will be somewhat slower over the web browser GUI than
with the Windows GUI.
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MultiVOIP User Guide
Operation & Maintenance
SysLog Server Functions
Beginning with the 4.03/6.03 software release, we have built SysLog
server functionality into the software of the MultiVOIP units. SysLog is
a de facto standard for logging events in network communication
systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement
this functionality, you will need a SysLog client program (sometimes
referred to as a “daemon”). SysLog client programs, both paid and
freeware, can be obtained from Kiwi Enterprises, among other firms.
Read the End-User License Agreement carefully and observe license
essentially give you a means of structuring 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 with MultiVOIP units. Kiwi’s brief description of their SysLog
program is as follows:
“Kiwi Syslog Daemon is a freeware Syslog
Daemon for the Windows platform. It
receives, logs, displays and forwards Syslog
messages from hosts such as routers,
switches, Unix hosts and any other syslog
enabled device. There are many customizable
options available.”
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MultiVOIP User Guide
Before a SysLog client program is used, the SysLog functionality must
be enabled 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’)
logical port, 514.
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Configuring the SysLog Client Program. Configure the SysLog client
program for your own needs. In various SysLog client programs, you
can define where log messages will be saved/archived, opt for
interaction with an SNMP system (like MultiVoipManager), set the
content and format of log messages, determine disk space allocation
limits for log messages, and establish a hierarchy for the seriousness of
messages (normal, alert, critical, emergency, etc.). A sample
presentation of SysLog info in the Kiwi daemon is shown below.
SysLog programs will vary in features and presentation.
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MultiVOIP User Guide
Regulatory Information
EMC, Safety, and R&TTE Directive Compliance
The CE mark is affixed to this product to confirm compliance with the
following European Community Directives:
Council Directive 89/336/EEC of 3 May 1989 on the approximation of the
laws of Member States relating to electromagnetic compatibility,
and
Council Directive 73/23/EEC of 19 February 1973 on the harmonization of
the laws of Member States relating to electrical equipment designed for use
within certain voltage limits,
and
Council Directive 1999/5/EC of 9 March 1999 on radio equipment and
telecommunications terminal equipment and the mutual recognition of their
conformity.
FCC Declaration
NOTE: This equipment has been tested and found to comply with the
limits for a Class A digital device, pursuant to Part 15 of the FCC Rules.
These limits are designed to provide reasonable protection against
harmful interference when the equipment is operated in a commercial
environment. This equipment generates, uses and can radiate radio
frequency energy, and if not installed and used in accordance with the
instructions, may cause harmful interference to radio communications.
Operation of this equipment in a residential area is likely to cause
harmful interference in which case the user will be required to correct
the interference at his own expense.
This device complies with Part 15 of the FCC rules.
Operation is subject to the following two conditions:
(1) This device may not cause harmful interference.
(2) This device must accept any interference that may cause
undesired operation.
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MultiVOIP User Guide
Warning: Changes or modifications to this unit not expressly approved
by the party responsible for compliance could void the user’s authority
to operate the equipment.
Industry Canada
This Class A digital apparatus meets all requirements of the Canadian
Interference-Causing Equipment Regulations.
Cet appareil numérique de la classe A
respecte toutes les exigences du
Reglement Canadien sur le matériel brouilleur.
FCC Part 68 Telecom
1. This equipment complies with part 68 of the Federal
Communications Commission Rules. On the outside surface of this
equipment is a label that contains, among other information, the FCC
registration number. This information must be provided to the
telephone company.
2. As indicated below, the suitable jack (Universal Service Order Code
connecting arrangement) for this equipment is shown. If applicable,
the facility interface codes (FIC) and service order codes (SOC) are
shown.
3. An FCC compliant telephone cord and modular plug is provided
with this equipment. This equipment is designed to be connected to
the telephone network or premises wiring using a compatible
modular jack that is Part 68 compliant. See installation instructions
for details.
4. If this equipment causes harm to the telephone network, the
telephone company will notify you in advance that temporary
discontinuance of service may be required. If advance notice is not
practical, the telephone company will 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. If this happens, the telephone company will provide
advance notice to allow you to make necessary modifications to
maintain uninterrupted service.
6. If trouble is experienced with this equipment (the model of which is
indicated below), please contact Multi-Tech Systems, Inc. at the
address shown below for details of how to have repairs made. If the
equipment is causing harm to the network, the telephone company
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Regulatory Information
may request you to remove 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 and warranty.
8. Manufacturer:
Trade name:
Multi-Tech Systems, Inc.
MultiVOIP
MVP-2410/3010/810/410/210/130/
130FXS
Model number:
FCC registration number:
Modular jack (USOC):
Service center in USA:
US: AU7DDNAN46050
RJ-48C
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, MN 55112
Tel: (763) 785-3500
FAX: (763) 785-9874
Canadian Limitations Notice
Notice: The Industry Canada label identifies certified equipment. This
certification means that the equipment meets certain
telecommunications network protective, operational and safety
requirements. The Department does not guarantee the equipment will
operate to the user’s satisfaction.
Before installing this equipment, users should ensure that it is
permissible to be connected to the facilities of the local
telecommunications company. The equipment must also be installed
using an acceptable method of connection. The customer should be
aware that compliance with the above conditions may not prevent
degradation of service in some situations.
Repairs to certified equipment should be made by an authorized
Canadian maintenance facility designated by the supplier. Any repairs
or alterations made by the user to this equipment, or equipment
malfunctions, may give the telecommunications company cause to
request the user to disconnect the equipment.
Users should ensure for their own protection that the electrical ground
connections of the power utility, telephone lines and internal metallic
water pipe system, if present, are connected together. This precaution
may be particularly important in rural areas.
Caution: Users should not attempt to make such connections
themselves, but should contact the appropriate electric inspection
authority, or electrician, as appropriate.
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MultiVOIP User Guide
T1/E1 Expansion Cards
Installation
Both the MVP2410 and the MVP3010 use the same mechanical chassis.
This chassis accommodates a second MultiVOIP circuit card or
motherboard module. The add-on module for the MVP2410 is the
MVP24-48 product; the add-on module for the MVP3010 is the MVP30-
60 product.
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 of the 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
the MVP2410/3010 unit. This power cable has a two-pin “molex”
connector. When the rear cover plate has been removed, the cable is
accessible from the rear at the right side of the expansion slot. Locate
this connector within the MVP2410/3010. See Figure A-2.
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T1/E1 Expansion Cards
MultiVOIP User Guide
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 or
MVP30-60 card into the grooves of the expansion slot. Push it in far
enough to allow connection of the power cable to the receptacle on the
vertical plate of the 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.
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T1/E1 Expansion Cards
Operation
The MVP2410/3010 front panel has two sets of identical LEDs. In the
MVP2410/3010 without an expansion card, only the left-hand set of
LEDs is functional. However, when the MultiVOIP unit has been
upgraded with an MVP24-48 or MVP30-60 expansion card, the right-
hand set of LEDs will also become active.
Remember that the expansion card must be configured as though it
were simply another complete MultiVOIP unit: it requires its own
T1/E1 line; it requires its own connection to a computer running the
MultiVOIP configuration software. All of the procedures and
operations that apply to the original motherboard of the
MVP2410/3010 will also apply to the expansion card. See applicable
User Guide chapters for details.
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Cable Pinouts
Appendix B: Cable Pinouts
Command Cable
RJ-45 Connector
End-to-End Pin Info
5-
'%)
3,112
3,112
1 2 3 4 5 6 7 8
&/($5726(1'
7R'7(
7R&RPPDQG
3RUW&RQQHFWRU
75$160,7'$7$
'HYLFH
HJ3&
5(&(,9('$7$
6,*1$/*5281'
RJ-45 connector plugs into Command Port of
MultiVOIP.
DB-9 connector plugs into serial port of command
PC (which runs MultiVOIP configuration
software).
Ethernet Connector
The functions of the individual conductors of the MultiVOIP’s Ethernet port are
shown on a pin-by-pin basis below.
RJ-45 Ethernet Connector
Pin Circuit Signal Name
1
2
3
6
TD+ Data Transmit Positive
TD- Data Transmit Negative
RD+ Data Receive Positive
RD- Data Receive Negative
1 2 3 4 5 6 7 8
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Cable Pinouts
MultiVOIP User Guide
T1/E1 Connector
T1/E1 Connector
1
2
Receive Pair (from line)
Transmit Pair (to line)
1 2 3 4 5 6 7 8
}
}
4
5
Voice/Fax Channel Connectors
1 2 3 4 5 6 7 8
1 2 3 4
Pin Functions (E&M Interface)
Pin
1
Descr
M
Function
Input
2
E
Output
3
T1
R
4-Wire Output
4
4-Wire Input, 2-Wire Input
4-Wire Input, 2-Wire Input
4-Wire Output
5
T
6
R1
SG
SB
7
Signal Ground (Output)
Signal Battery (Output)
8
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Cable Pinouts
Pin Functions (FXS/FXO Interface)
FXS Pin
Description
N/C
FXO Pin
Description
2
3
4
5
2
3
4
5
N/C
Tip
Ring
Tip
Ring
N/C
N/C
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Cable Pinouts
MultiVOIP User Guide
ISDN BRI RJ-45 Pinout Information
The S/T interface uses an 8-conductor modular cable terminated with
an 8-pin RJ-45 plug. An 8-pin RJ-45 jack located on the terminal is used
to connect the terminal to the DSL (Digital Subscriber Loops) using this
modular cable.
The table below shows the Pin Number, Terminal Pin Signal Name and
Network Pin Signal name for the S/T interface.
Pin
1
TE Signal
Not used
Not used
Tx+
NT Signal
Not used
Not used
Rx+
Pin
1
2
2
3
3
4
Rx-
Tx-
4
5
Rx+
Tx+
5
6
Tx-
Rx-
6
7
Not used
Not used
Not used
Not used
7
8
8
1 2 3 4 5 6 7 8
TE=Terminal Equipment
NT=Network
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Cable Pinouts
ISDN Interfaces: “ST” and “U”
The MVP410ST and MVP810ST are ISDN-BRI voip units that use an
S/T outlet interface. You will need an NT1 device to connect these units
to any network equipment that has the “U” ISDN interface. In the UK,
and in many European countries, the telco supplies an NT1 device for
ISDN-BRI service.
An ISDN Basic Rate (BRI) U-Loop consists of two conductors from the
telco central office to the customer premises. The equipment on both
sides of the U-loop accommodates the extensive length of the U-loop
and the noisy environment in which it may operate. At the customer
premises, the U-loop is terminated by an NT1 (network termination 1 )
device. An NT1 device makes an end-user’s 4-wire terminal equipment
compatible with the telco’s 2-wire twisted pair ISDN-BRI line.
The NT1 drives an S/T bus. The S/T bus is usually made up of 4 wires,
but in some 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
the terminal. (“Terminal” can mean any sort of end-user ISDN device:
data terminals, telephones, FAX machines, voip units, etc.)
Point T refers to the connection between the NT1 device and customer
supplied equipment. Terminals can connect directly to the NT1 device
at point T, or there may be a PBX (private branch exchange, i.e., a
customer-owned telephone exchange). The figure below shows “S” and
“T” connection points in an ISDN network.
Point “S”
4-8 Wires
Te lco
Central
Office
Point “T”
4-8 Wires
NT2
(PBX)
NT1
Point “U”
2 Wires
Terminal
Point “S”
Point “S”
Terminal
Terminal
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Cable Pinouts
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) web
site (www.iana.org).
“The Well Known Ports are assigned by
the IANA and on most systems can only
be used by system (or root) processes or
by programs executed by privileged
users. Ports are used in the TCP
[RFC793] to name the ends of logical
connections which carry long term
conversations. For the purpose of
providing services to unknown callers, a
service contact port is defined. This list
specifies the port used by the server
process as its contact port. The contact
port is sometimes called the "well-
known port". To the extent possible,
these same port assignments are used
with the UDP [RFC768]. The range for
assigned ports managed by the IANA is
0-1023.”
Well-known port numbers especially pertinent to MultiVOIP operation are listed
below.
Port Number Assignment List
Well-Known Port Numbers
Function
telnet
Port Number
23
tftp
69
snmp
snmp tray
gatekeeper registration
H.323
SIP
161
162
1719
1720
5060
514
SysLog
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MultiVOIP User Guide
8-Channel Analog Expansion Card
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.
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
with standoffs here
*
(2 places).
Add standoffs hereꢁ
(2 places).
(C)
Attach upgrade card
(screws into standoffs
-- 4 places).
ꢁ
ꢁ
(B)
Mate 60-pin
connectors.
Figure D-1. Installation Summary
Procedure in Detail
1. Power down and unplug the MVP410 unit.
2. Using a Phillips driver, remove the blank cover plate at the rear of the
MVP410 chassis. Save the screws.
screws on blank cover plate (2)
Figure D-2: Removing screws from blank cover plate
511
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8-Channel Analog Expansion Card
MultiVOIP User Guide
3. Using a Phillips driver, remove the three screws that secure the main circuit
board and back panel assembly to the chassis.
NOTE:
Follow standard ESD
precautions to protect the
circuit board from static
electricity 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 the
power 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.
512
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8-Channel Analog Expansion Card
8. On the phone-jack side of the circuit card, three screws attach the circuit
card 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 edge
of 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 with
board) and attach a standoff to each location).
Standoff locations (2) at LED edge
of board (top view).
Standoff/nut attachment
(rear bottom view)
Figure D-6: Standoffs at LED edge of board (top view)
513
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8-Channel Analog Expansion Card
MultiVOIP User Guide
11. Locate the male 60-pin vertical connector near the LED edge of the main
circuit card. Check that pins are straight and evenly spaced. If not, then
correct for straightness and spacing. Locate the 60-pin female connector
on the upgrade circuit card.
12. Set the upgrade circuit card on top of the main circuit card. Align the
upgrade card’s 4 pairs of phone-jacks with the 4 pairs of holes in the
backplane 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.
*
*
These screws (4 places)
*
attach upgrade card
to main card.
*
*
60-pin connectors
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 and
rear edges of the upgrade card. Attach the upgrade card to the main card
using 4 Phillips screws. The upgrade card should now be firmly attached
to 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.
514
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MultiVOIP User Guide
Index
INDEX
accessing logs screen
(
analog........................................286
T1/E1 ........................................185
accessing Regional Parameters
Alternate Phone Number field, SPP
E1.............................................. 380
Alternate Phone Number, SPP
analog........................................272
T1/E1 ........................................170
accessing Registered Gateway Details
(Statistics) screen ......................448
accessing Registered Gateway
Details screen..................447, 448
accessing RTP Parameters screen .432
accessing SMTP parameters
analog........................................280
T1/E1 ........................................178
accessing SNMP parameters
analog........................................269
T1/E1 ........................................167
accessing Supplementary Services
screen
T1.............................................. 328
A
abbreviated dialing, inter-office
E1.............................................. 357
T1.............................................. 309
Accept Any Number (inbound)
E1.............................................. 384
T1.............................................. 333
Accept Any Number (outbound) field
E1.............................................. 376
T1.............................................. 324
access codes, PBX .......................... 66
access codes, types
PBX ............................................ 71
PSTN .......................................... 71
special......................................... 71
access digits, PBX 66, See phonebook
digits, types used
analog........................................290
T1/E1 ........................................189
accessing System Information screen
analog........................................302
T1/E1 ........................................201
accessing T1 Statistics screen....435
accessing T1/E1/ISDN Parameters
screen ........................................152
accessing Voice/FAX Parameters
screen ................................140, 227
access to network
analog........................................ 271
T1/E1 ........................................ 169
access to remote PSTN
E1................................................ 18
T1................................................ 11
accessing Statistics, Logs screen
.................................................. 420
accessing Call Progress (Statistics)
screen........................................ 414
accessing configuration parameter
groups
dd Inbound Phonebook Entry icons
E1..............................................363
T1..............................................310
A
Add Outbound Phonebook Entry icon
E1..............................................363
T1..............................................310
Add Prefix (inbound) field
analog........................................ 219
T1/E1 ........................................ 133
accessing interface parameters...... 240
accessing IP Parameters screen
analog........................................ 220
T1/E1 ........................................ 134
accessing IP Statistics screen........ 425
accessing Logs (Statistics) screen
.................................................. 420
E1..............................................384
T1..............................................333
Add Prefix (outbound) field
E1..............................................377
T1..............................................325
Add/Edit Inbound Phonebook field
definitions
E1.............................. 384, 385, 386
T1.............................. 333, 334, 335
515
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Index
MultiVOIP User Guide
Add/Edit Inbound Phonebook screen
E1.............................................. 384
T1.............................................. 333
Add/Edit Inbound Phonebook screen
fields (E1)
Accept Any Number................. 384
Add Prefix................................. 384
Channel Number....................... 385
Description (callee location)..... 385
Enable (Call Forwarding) ......... 385
Forward Condition.................... 385
Forward Destination ................. 386
Registration Option Parameters 386
Remove Prefix .......................... 384
Ring Count................................ 386
Add/Edit Inbound Phonebook screen
fields (T1)
Accept Any Number................. 333
Add Prefix................................. 333
Channel Number....................... 333
Description (callee location)..... 333
Enable (Call Forwarding) ......... 333
Forward Condition.................... 334
Forward Destination ................. 334
Registration Option Parameters 335
Remove Prefix .......................... 333
Ring Count................................ 335
Add/Edit Outbound Phonebook field
definitions
Total Digits ...............................377
Transport Protocol (SIP) ...........379
Use Gatekeeper .................378, 380
Use Proxy (SIP) ........................379
Add/Edit Outbound Phonebook fields
(T1)
Accept Any Number .................324
Add Prefix.................................325
Advanced button .......................327
Description................................325
Destination Pattern....................325
Gateway H.323 ID ....................326
Gateway Prefix..........................326
IP Address.................................325
Protocol Type............................325
Q.931 Port Number...................326
Remove Prefix ..........................325
SIP Port Number.......................327
SIP URL....................................327
Total Digits ...............................325
Transport Protocol (SIP) ...........327
Use Gatekeeper .................326, 328
Use Proxy (SIP) ........................327
Add/Edit Outbound Phonebook screen
E1..............................................375
T1..............................................323
Add/Edit Outbound Phonebook SPP
Fields
E1..............................................380
T1..............................................328
add-on module (analog, 4-to-8
channel), installation.................511
add-on module (T1/E1)
E1.............. 376, 377, 378, 379, 380
T1.............. 324, 325, 326, 327, 328
Add/Edit Outbound Phonebook fields
(E1)
Accept Any Number................. 376
Add Prefix................................. 377
Advanced button....................... 379
Description................................ 377
destination pattern..................... 377
Gateway H.323 ID.................... 378
Gateway Prefix ......................... 378
H.323 Port Number................... 378
IP Address................................. 377
Protocol Type............................ 377
Remote Device is [legacy]
operation ...................................501
add-on module (T1/E1), installation
..................................................499
Address (SNMP) field
analog........................................271
T1/E1 ........................................169
Advanced button, Outbound
Phonebook
E1..............................................380
T1..............................................328
Advanced Features field group
analog........................................233
T1/E1 ........................................146
airflow.............................................91
Alerting Party
MultiVOIP............................ 380
Remove Prefix .......................... 377
SIP Port Number....................... 379
SIP URL ................................... 379
516
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Index
Supplementary Services (analog)
.............................. 297, 298, 299
Supplementary Services (T1/E1)
.............................. 196, 197, 198
Allowed Name Type (analog)
Alerting Party............ 297, 298, 299
Calling Party............................. 296
Allowed Name Type (T1/E1)
Alerting Party............ 196, 197, 198
Calling Party............................. 195
Allowed Name Types, Call Name ID
(analog)
Alerting Party............................ 297
Busy Party................................. 298
Calling Party............................. 296
Connected Party........................ 299
Allowed Name Types, Call Name ID
(T1/E1)
Alerting Party............................ 196
Busy Party................................. 197
Calling Party............................. 195
Connected Party........................ 198
Alternate IP Address field
E1.............................................. 382
T1.............................................. 330
Alternate IP Routing
analog telephony interface parameters
..................................................211
Annex E field
E1..............................................369
T1..............................................316
Append SIP Proxy Domain Name in
User ID (proxy server)
E1..............................................370
T1..............................................317
area codes........................................70
Auto Disconnect field group
analog........................................239
T1/E1 ........................................151
AutoCall
analog/BRI................................234
T1/E1 ........................................147
AutoCall (Voice/Fax Params - analog)
and Pass Through Enable (FXS
Loop Start) ................................234
AutoCall/Offhook Alert field
analog/BRI........................234, 235
T1/E1 ........................................147
Automatic Disconnection field
analog........................................239
B
E1.............................................. 375
T1.............................................. 323
Alternate Phone Number, SPP
(Add/Edit Outbound Phonebook)
E1.............................................. 380
T1.............................................. 328
Alternate Routing
bandwidth
coder (analog) ...........................232
coder (T1/E1)............................145
battery caution.................................85
baud rate, default (MultiVOIP
ware connection)
soft
T1/E1.....................................200
PSTN failover feature, and ....... 330
Alternate Routing field definitions
E1.............................................. 382
T1.............................................. 330
Alternate Routing field definitions
(E1)
Alternate IP Address................. 382
Round Trip Delay ..................... 382
Alternate Routing field definitions
(T1)
Alternate IP Address................. 330
Round Trip Delay ..................... 330
analog phonebook......................... 409
using T1 & E1 examples for..... 409
analog phonebook examples......... 207
analog....................................301
baud rate, fax
analog........................................231
T1/E1 ........................................144
baud rate, setting
analog........................................301
T1/E1 ........................................200
Behind Proxy/NAT device
E1..............................................373
T1..............................................320
Bipolar Violation (E1 stats) field ..442
Bipolar Violation (T1 stats) field ..439
Blue Alarm (E1 stats) field ...........440
Blue Alarm (T1 stats) field ...........437
Boot LED
517
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Index
MultiVOIP User Guide
analog models............................. 32
Bytes Sent (call progress) field .....416
Bytes Sent (SMTP logs) field
analog........................................283
T1/E1 ........................................182
Bytes sent (statistics, logs) field....422
analog, MVP130......................... 33
BRI models................................. 39
MVP-210 .................................. 107
MVP-410/810 ............................. 99
MVP-410ST/810ST.................. 103
on MVP-2410/3010 .................... 94
Boot Version
C
cable length, maximum span
System Info............................... 412
System Info (ANALOG/BRI)... 303
System Info (T1/E1) ................. 202
booting time
analog.......................................... 32
analog, MVP130......................... 33
BRI ............................................. 39
E1................................................ 24
T1................................................ 17
box contents
verifying...................................... 86
BRI connector pinout.................... 506
BRI interface types
ST and U................................... 507
built-in modem
setup in Regional Parameters
screen...................................... 61
built-in modem (analog)
setup in Regional Parameters
screen............................ 214, 273
built-in modem (T1/E1)
setup in Regional Parameters
screen.................... 128, 170, 171
busy & no-response (E1)
E1..............................................161
T1..............................................155
cabling diagram, quick
analog models ..... 52, 53, 55, 56, 57
BRI models .................................54
E1 models....................................52
MVP130......................................56
MVP130FXS...............................57
MVP210......................................55
MVP2410....................................52
MVP3010....................................52
MVP410......................................53
MVP-410ST/810ST ....................54
MVP810......................................53
T1 models....................................52
cabling problem, fixing
analog models ...........................219
T1/E1 models............................133
cabling procedure
MVP130....................................108
MVP210....................................104
MVP2410....................................93
MVP3010....................................93
MVP410......................................95
forwarding, dual conditions ...... 385
busy & no-response (T1)
forwarding, dual conditions ...... 334
busy tone, custom
410ST..............................100
MVP-
MVP810......................................95
MVP-810ST..............................100
Cadence 1 (custom) field
analog........................................ 278
T1/E1 ................................ 175, 176
busy-tones
analog........................................279
T1/E1 ........................................177
Cadence 2 (custom) field
analog........................................ 277
T1/E1 ........................................ 175
Bytes Received (call progress) field
.................................................. 416
Bytes Received (SMTP logs) field
analog........................................ 283
T1/E1 ........................................ 182
Bytes received (statistics, logs) field
.................................................. 423
analog........................................279
T1/E1 ........................................177
Cadence 3 (custom) field
analog........................................279
T1/E1 ........................................177
Cadence 4 (custom) field
analog........................................279
T1/E1 ........................................177
Cadence field
518
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Index
analog........................................ 276
T1/E1 ........................................ 174
cadences, custom
T1.E1 ................................ 177, 279
T1/E1 ........................................ 175
cadences, signaling
analog........................................ 272
T1/E1 ........................................ 170
Call Control PHB field
analog........................................ 222
T1/E1 ........................................ 136
Call Control Status
Call Progress Details (statistics)
field....................................... 419
Call Control Status (call progress)
field........................................... 419
Call Duration field
analog........................................ 239
T1/E1 ........................................ 151
Call Forward Parameters (inbound
phonebook)
E1.............................................. 385
T1.............................................. 334
Call Forwarded To
logs (statistics) field.................. 424
Call Hold
ANALOG ................................... 30
BRI ............................................. 38
E1................................................ 23
T1................................................ 16
Call Hold (analog) ........................ 291
Call Hold (T1/E1)......................... 190
Call Hold Enable
analog........................................ 294
T1/E1 ........................................ 193
Call Mode (SMTP logs) field
analog........................................ 283
T1/E1 ........................................ 182
Call Name Identification
ANALOG ................................... 30
BRI ............................................. 38
E1................................................ 23
T1................................................ 16
Call Name Identification (analog)
Alerting Party............ 297, 298, 299
Calling Party............................. 296
Call Name Identification (T1/E1)
Alerting Party............ 196, 197, 198
Calling Party .............................195
Call Name Identification (analog).291
Call Name Identification (T1/E1) .190
Call On Hold
Call Progress Details (statistics)
field ...............................416, 418
Call on Hold (call progress) field..418
Call Progress (Statistics) ...............414
Call Progress Details (statistics)
screen field
Call On Hold.........................416
Call Waiting..........................416
Caller ID................................416
Call On Hold.........................418
Call Waiting..........................418
Caller ID................................419
Call Progress Details (statistics)
screen fields
Channel .................................416
Duration ................................416
Mode .....................................416
Voice Coder ..........................416
Packets Sent ..........................416
Packets Received...................416
Bytes Sent .............................416
Bytes Received......................416
Packets Lost ..........................416
Outbound Digits....................416
Prefix Matched......................416
Gateway Name......................417
IP Address.............................417
Options..................................417
Silence Compression.............417
Forward Error Correction......417
Status.....................................419
Call Control Status................419
Call Status (SMTP logs) field
analog........................................284
T1/E1 ........................................183
Call Transfer
ANALOG....................................30
BRI..............................................38
E1................................................23
T1................................................16
Call Transfer (analog)...................291
Call Transfer (T1/E1)....................190
Call Transfer Enable
analog........................................293
519
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Index
MultiVOIP User Guide
T1/E1 ........................................ 192
Canadian Limitations Notice
Call Transfer music jingle during hold
analog........................................ 293
T1/E1 ........................................ 192
Call Transferred To
logs (statistics) field.................. 424
Call Waiting
ANALOG ................................... 30
BRI ............................................. 38
Call Progress Details (statistics)
field............................... 416, 418
E1................................................ 23
T1................................................ 16
Call Waiting (analog) ................... 291
Call Waiting (call progress) field.. 418
Call Waiting (T1/E1) .................... 190
Call Waiting Enable
analog........................................ 294
T1/E1 ........................................ 193
Caller ID
Call Progress Details (statistics)
field............................... 416, 419
Caller ID (analog)......................... 291
Caller ID (call progress) field ....... 419
Caller ID (Supplementary Services)
field
analog........................................ 300
T1/E1 ........................................ 199
Caller ID (T1/E1).......................... 191
Caller ID enable
FXO .......................................... 251
FXS Loop Start................. 245, 246
Caller ID examples ....... 252, 253, 254
Caller ID fields
(regulatory) ...............................497
CAS Protocol field
E1..............................................162
T1..............................................156
CAS vs. CCS
T1......................................156, 162
CCS vs. CAS
T1......................................156, 162
CD
MultiVOIP ..................................44
Channel (call progress) field.........416
channel capacity................................9
analog..........................................26
BRI..............................................34
E1................................................18
T1................................................11
Channel Number (inbound) field
E1..............................................385
T1..............................................333
Channel Number (SMTP logs) field
analog........................................283
T1/E1 ........................................182
channel tracing on/off (logging)
analog........................................289
T1/E1 ........................................188
city codes ........................................70
Clear (button), ISDN BRI Statistics
screen ........................................446
Clear (IP Statistics) button ............426
Clear command (Link Management)
button ........................................430
Client Options fields
373
E1..............................................
FXO .......................................... 251
Caller ID Type
T1..............................................320
Clocking field
FXO .......................................... 251
FXS Loop Start......................... 245
Caller Name Identification Enable
analog........................................ 295
T1/E1 ........................................ 194
calling area codes............................ 70
Calling Party
Supplementary Services (analog)
.............................................. 296
Supplementary Services (T1/E1)
.............................................. 195
Canadian Class A requirements.... 496
E1..............................................165
T1..............................................159
coder (analog)
bandwidth, max.........................232
G.711.........................................232
G.723.1......................................232
G.726.........................................232
G.727.........................................232
G.729.........................................232
Net Coder..................................232
Coder (SMTP logs) field
analog........................................283
520
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Index
T1/E1 ........................................ 182
coder (T1/E1)
and Regional Parameters screen
(analog) .........................214, 273
and Regional Parameters screen
(T1/E1).................. 128, 170, 171
Command Modem
setup for ......................................61
Command Modem (analog)
setup for ............................214, 273
Command Modem (T1/E1)
setup for .................... 128, 170, 171
command PC
COM port assignment (detailed)116
COM port requirement................51
demands upon .............................51
non-dedicated use........................51
operating system .........................51
settings ........................................51
specifications...............................51
Command PC
bandwidth, max......................... 145
G.711 ........................................ 145
G.723.1 ..................................... 145
G.726 ........................................ 145
G.727 ........................................ 145
G.729 ........................................ 145
Net Coder.................................. 145
Coder field
analog........................................ 232
T1/E1 ........................................ 145
coder options
packetization rates and.............. 432
Coder Parameters field group
analog........................................ 232
T1/E1 ........................................ 145
coder types (voice/fax, RTP
packetization)
T1/E1 ........................................ 433
COM port
on command PC........................ 116
COM port (analog models)
COM port requirement................40
non-dedicated use of ...................40
operating system .........................40
community (voip) defined
conflict, resolving ..................... 218
error message............................ 218
COM port (T1/E1 models)
conflict, resolving ..................... 132
error message............................ 132
COM port allocation
analog........................................271
T1/E1 ........................................169
Community Name 1 (SNMP) field
analog........................................271
T1/E1 ........................................169
compatibility, Fast Start
analog........................................ 301
T1/E1 ........................................ 200
COM port assignments
analog........................................ 301
T1/E1 ........................................ 200
COM port conflict
error message............................ 116
COM Port Setup screen ................ 116
COM Port Setup screen (analog
models) ..................................... 218
COM Port Setup screen (T1/E1
models) ..................................... 132
comma
E1..............................................366
T1..............................................313
compatibility, H.450 with H.323, not
with SIP
analog..................................27, 290
BRI..............................................35
E1................................................19
T1................................................12
T1/E1 ........................................189
compression standard
E1..............................................165
T1..............................................159
compression, silence
meaning/use in phonebook ......... 72
comma use
and second dial tone.................... 72
command cable pinout.................. 503
command modem
analog........................................233
T1/E1 ........................................146
Compression, Silence (SMTP logs)
analog........................................284
T1/E1 ........................................183
computer requirements....................40
and Regional Parameters screen . 61
521
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Index
MultiVOIP User Guide
configuration of voip (analog)
local versus remote ................... 208
configuration of voip (T1/E1)
local versus remote ........... 122, 123
Configuration option description
(MultiVOIP program menu) ..... 449
Configuration Parameter Groups,
accessing
analog........................................ 219
T1/E1 ........................................ 133
Configuration Port Setup option
description (MultiVOIP program
menu)........................................ 449
configuration procedure, local
detailed, analog......................... 215
detailed, T1/E1.......................... 129
summary, analog....................... 214
summary, T1/E1 ....................... 128
Configuration Version
confirmation of............................81
confirming with remote voip.50, 63
pinging and..................................82
connectivity test ..............................78
Consecutive Packets Lost field
analog........................................239
T1/E1 ........................................151
Console Message Settings, Filters for
analog........................................289
T1/E1 ........................................188
console messages .......... 61, 78, 80, 81
console messages, enabling
analog........................................287
T1/E1 ........................................186
console parameters tracked
analog........................................289
T1/E1 ........................................188
coordinated phonebook entries
E1..............................................362
T1..............................................309
Copy Channel command (Interface
Parameters)
System Info (ANALOG/BRI)... 304
System Info (T1/E1) ................. 203
configuration, local
analog/BRI................................ 210
T1/E1 ........................................ 124
configuration, phonebook
analog........................................242
Copy Channel command (Voice/Fax
Parameters)
E1.............................................. 362
starter .......................................... 63
T1.............................................. 309
configuration, saving
analog........................................ 305
T1/E1 ........................................ 204
user ........................................... 466
configuration, starter
phone/IP...................................... 59
configuration, user default
analog........................................ 306
T1/E1 ........................................ 205
Configuring MultiVOIP phonebooks,
general
E1.............................................. 356
T1.............................................. 308
confirming connectivity.................. 81
conflicts
COM port.................................. 116
Connection Problems, Solving
analog........................................ 218
T1/E1 ........................................ 132
connectivity
analog........................................229
T1/E1 ........................................142
Copy Channel field
analog........................................230
Copy Channel field (Voice/Fax
Parameters)
T1/E1 ........................................143
Copy Channel, Supplementary
Services command
analog........................................292
T1/E1 ........................................191
Copy Channel, Supplementary
Services field
analog........................................300
T1/E1 ........................................199
Copy Interface command
BRI............................................263
Count of Registered Numbers field
(Registered Gateway Details) ...448
country
ISDN type and...........................166
switch type and ISDN ...............166
Country (ISDN) field
522
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Index
E1/ISDN ................................... 164
country codes.................................. 70
Country definitions
ISDN-BRI................................. 268
Country field
ISDN-BRI................................. 265
Country field (ISDN)
T1/ISDN ................................... 158
Country Selection for Built-In Modem
field
analog (except MVP130).......... 276
T1/E1 ........................................ 174
Country/Region (tone schemes) field
analog................................ 274, 275
T1/E1 ........................................ 172
CRC and ESF frame format (T1).. 155
CRC Check field
Options......................................284
Description (callee)...................284
Description (caller) ...................284
Duration ....................................283
From Gateway Number.............284
From IP Address .......................284
Outbound Digits........................284
Packets Lost ..............................283
Packets Received.......................283
Packets Sent ..............................283
Prefix Matched..........................284
Select All...................................283
Start Date, Time........................283
To Gateway Number.................284
To IP Address ...........................284
Custom Fields, SMTP log email
(T1/E1)
T1.............................................. 155
Creating a User Default Configuration
analog........................................ 306
T1/E1 ........................................ 205
Current Loss (FXO disconnect
criteria) field ............................. 249
Current Loss Detect Timer (FXO)
field........................................... 249
Current Loss field
FXS Loop Start......................... 244
Custom (tones, Regional)field
analog........................................ 276
T1/E1 ........................................ 174
custom cadences
analog........................................ 279
T1/E1 ........................................ 177
custom DTMF
analog........................................ 278
T1/E1 ................................ 175, 176
Custom Fields (SMTP) definitions
analog................................ 283, 284
T1/E1 ................................ 182, 183
Custom Fields, SMTP log email
(analog)
Bytes Received..........................182
Bytes Sent .................................182
Call Mode..................................182
Call Status.................................183
Channel Number .......................182
Coder.........................................182
Options......................................183
Options......................................183
Description (callee)...................183
Description (caller) ...................183
Duration ....................................182
From Gateway Number.............183
From IP Address .......................183
Outbound Digits........................183
Packets Lost ..............................182
Packets Received.......................182
Packets Sent ..............................182
Prefix Matched..........................183
Select All...................................182
Start Date, Time........................182
To Gateway Number.................183
To IP Address ...........................183
Custom Tone-Pair Settings (analog)
fields
Bytes Received ......................... 283
Bytes Sent................................. 283
Call Mode ................................. 283
Call Status................................. 284
Channel Number....................... 283
Coder ........................................ 283
Options...................................... 284
Cadence 1..................................279
Cadence 2..................................279
Cadence 3..................................279
Cadence 4..................................279
Custom Tone-Pair Settings (T1/E1)
fields
Cadence 1..................................177
523
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MultiVOIP User Guide
Cadence 2 ................................. 177
Default (Supplementary Services)
field
analog........................................300
T1/E1 ........................................199
Default (Voice/FAX) field
analog........................................230
T1/E1 ........................................143
default baud rate (MultiVOIP
software connection)
analog........................................301
T1/E1 ........................................200
default configuration, user
analog........................................306
T1/E1 ........................................205
default values, software.................460
delay, packets
analog........................................237
T1/E1 ........................................149
delay, versus voice quality
analog........................................238
T1/E1 ........................................150
Delete File button
Logs (Statistics) screen .............422
Description (callee location)
E1..............................................385
T1..............................................333
Description (callee, outbound
phonebook)
E1..............................................377
T1..............................................325
Description field (Registered Gateway
Details)......................................
Description, From Details (SMTP
logs) field
analog........................................284
T1/E1 ........................................183
Description, To Details (SMTP logs)
field
analog........................................284
T1/E1 ........................................183
Destination Pattern (outbound) field
E1..............................................377
T1..............................................325
destination patterns
Cadence 3 ................................. 177
Cadence 4 ................................. 177
Custom Tone-Pair Settings definitions
analog................................ 278, 279
T1/E1 ................................ 176, 177
Custom Tone-Pair Settings fields
(analog)
Frequency 1 .............................. 278
Frequency 2 .............................. 278
Gain 1 ....................................... 278
Gain 2 ....................................... 278
Tone Pair................................... 278
Custom Tone-Pair Settings fields
(T1/E1)
Frequency 1 .............................. 176
Frequency 2 .............................. 176
Gain 1 ....................................... 176
Gain 2 ....................................... 176
Tone Pair................................... 176
custom tones, setting
T1/E1 ........................................ 175
customized log email
analog................................ 283, 284
T1/E1 ................................ 182, 183
D
D Channel Information fields (ISDN
BRI Statistics)........................... 445
data capacity ..................................... 9
analog.......................................... 26
BRI ............................................. 34
E1................................................ 18
T1................................................ 11
data compression
analog.......................................... 27
BRI ............................................. 35
E1................................................ 19
T1................................................ 12
Date & Time Setup (program menu
option), command..................... 452
Date and Time Setup option
448
description (MultiVOIP program
menu)........................................ 449
debugging messages
digits used ...................................70
tips about.....................................70
destination patterns, discussion
E1..............................................361
analog........................................ 288
T1/E1 ........................................ 186
524
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Index
T1.............................................. 308
Detection Flash Hook field
E1.............................................. 163
T1.............................................. 157
Detection Range, Flash Hook Options
field
E&M......................................... 258
FXO .......................................... 251
FXS Loop Start......................... 245
Detection Time
E1.............................................. 163
T1.............................................. 157
Detection Time field
E1.............................................. 163
T1.............................................. 157
dial tone, custom
analog........................................ 278
T1/E1 ................................ 175, 176
dial tone, second
and comma use ........................... 72
pausing for .................................. 72
Dialing Options (E&M) fields...... 257
Dialing Options (FXO) fields248, 249
Dialing Options field
ISDN-BRI................................. 265
dialing patterns
digits used................................... 70
inbound/outbound matching ....... 72
tips about..................................... 70
dial-tones
analog........................................ 277
T1/E1 ........................................ 175
DID interface (MVP210)
uses of....................................... 107
DID interface (MVP-410/810)
uses of......................................... 98
DID Interface Parameter definitions
.................................................. 260
DID Interface Parameter fields
Message Waiting Indication ..... 260
DID Interface Parameters ............. 259
DID jumper
MVP210.................................... 104
MVP-410/810 ............................. 96
DID lines (MVP210)
polarity sensitivity and.............. 107
DID lines (MVP-410/810)
DID-DPO Interface Parameter
definitions .................................259
DID-DPO Interface Parameter fields
Inter Digit Timer (dialing) ........260
Start Modes...............................259
Wink Timer...............................259
DID-DPO Parameter fields
Inter-Digit Regeneration Timer
(dialing).................................260
DID-DPO vs. DID-DPT................259
DiffServ and IP datagram
analog........................................223
T1/E1 ........................................137
DiffServ PHB (Per Hop Behavior)
value
analog........................................222
T1/E1 ........................................136
digits in phonebook
specialized codes.........................71
types............................................70
dimensions
analog models .............................43
E1 models....................................42
Disconnect on Call Progress Tone
(E&M) field ..............................258
Disconnect on Call Progress Tone
(FXO) field................................250
Disconnect Tone Sequence (FXO)
field ...........................................250
disconnection criteria, FXO ..249, 250
DNS Server IP Address
T1/E1 ................................138, 224
Download Factory Defaults (program
menu option) , command...........460
Download Factory Defaults option
description (MultiVOIP program
menu) ........................................450
Download Firmware (program menu
option), command .............456, 457
Download Firmware option
description (MultiVOIP program
menu) ........................................450
Download IFM Firmware (program
menu option) , command...462, 463
Download IFM Firmware option
description (MultiVOIP program
menu) ........................................450
polarity sensitivity and................ 98
525
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MultiVOIP User Guide
Download User Defaults (program
menu option) , command .......... 466
Download User Defaults option
description (MultiVOIP program
menu)........................................ 450
downloading firmware, machine
perspective........................ 451, 476
downloading IFM firmware.......... 462
downloading user defaults ............ 466
downloads vs. uploads (FTP)........ 476
dropping digits, in phonebook ........ 71
DTMF
extended.................................... 250
standard..................................... 250
DTMF frequency chart ................. 250
DTMF Gain (High Tones) field
analog........................................ 230
T1/E1 ........................................ 143
DTMF Gain (Low Tones) field
analog........................................ 230
T1/E1 ........................................ 143
DTMF Gain field
analog........................................ 230
T1/E1 ........................................ 143
DTMF In/Out of Band field
analog........................................ 231
T1/E1 ........................................ 144
DTMF inband
analog........................................ 231
T1/E1 ........................................ 144
DTMF out of band
analog........................................ 231
T1/E1 ........................................ 144
DTMF Parameters
T1/E1 ........................................149
Dynamic Jitter field group
analog........................................237
T1/E1 ........................................149
Dynamic Jitter fields
analog........................................238
T1/E1 ........................................150
E
E&M interface (MVP210)
matching telco trunk line...........106
uses of .......................................106
E&M interface (MVP-410/810)
matching telco trunk line.............98
.........................................98
uses of
E&M Interface Parameter fields
Detection Range (flash hook)....258
Disconnect on Call Progress Tone
..............................................258
Flash Hook................................258
Inter Digit Timer (dialing) ........257
Interface ....................................256
Message Waiting Indication......257
Pass Through.............................256
Regeneration (dialing)...............257
Signal ........................................256
Type ..........................................256
Wink Timer...............................256
E&M Parameter definitions .256, 257,
258
E&M Parameters...........................255
E1 Parameter definitions......161, 162,
163, 165
Clocking....................................165
Line Build-Out..........................165
Line Coding ..............................165
PCM Law..................................165
Pulse Shape Level.....................165
E1 Parameter fields
T1/E1 ........................................ 143
DTMF, custom tone pairs
analog........................................ 278
T1/E1 ................................ 175, 176
Duration (call progress) field........ 416
Duration (DTMF) field
analog........................................ 231
T1/E1 ........................................ 144
Duration (SMTP logs) field
analog........................................ 283
T1/E1 ........................................ 182
Duration (statistics, logs) field...... 422
Dynamic Jitter Buffer field
CAS Protocol ........................162
CRC Check ...............................162
Detect Flash Hook.....................163
Frame Format............................162
Long-Haul Mode.......................162
E1 Parameters screen ....................160
E1 Statistics field definitions440, 441,
442
analog........................................ 237
E1 Statistics fields
526
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Index
Bipolar Variation ...................... 442
Blue Alarm ............................... 440
Excessive Zeroes ...................... 442
Loss of Frame Alignment ......... 440
Loss of MultiFrame Alignment 441
Receive Slip.............................. 442
Receive Timeslot 16 Alarm
quick............................................50
T1/E1 ................................127, 178
email log reports
analog........................................280
quick............................................61
recipient.......................................61
reply-to address...........................61
subject line ..................................61
T1/E1 ........................................178
email logs, illustration
analog........................................285
T1/E1 ........................................184
EMC, Safety, R&TTE Directive
Compliance ...............................495
emergency phone numbers
caution about...............................72
Enable (Call Fwdg)
E1..............................................385
T1..............................................333
Enable Call Hold
analog........................................294
T1/E1 ........................................193
Enable Call Transfer
analog........................................293
T1/E1 ........................................192
Enable Call Waiting
analog........................................294
T1/E1 ........................................193
Enable Caller Name Identification
analog........................................295
T1/E1 ........................................194
Enable Console Messages field
analog........................................288
T1/E1 ........................................186
Enable DNS field
analog........................................224
T1/E1 ................................138, 224
Enable ISDN-PRI field
E1/ISDN....................................164
T1/ISDN....................................158
Enable SMTP field
Indication Signal................... 441
Receive Timeslot 16 Loss of
MultiFrame Alignment ......... 442
Receive Timeslot 16 Loss of Signal
.............................................. 441
Red Alarm................................. 440
Status Freeze Signalling Active 441
Transmit Data Overflow........... 441
Transmit Data Underrun........... 442
Transmit Line Open.................. 442
Transmit Line Short.................. 441
Transmit Slip ............................ 442
Transmit Slip Negative............. 442
Transmit Slip Positive............... 441
Yellow Alarm ........................... 441
E1 telephony parameters............... 126
E1/ISDN Parameter definitions .... 164
E1/ISDN Parameter fields
Country..................................... 164
Enable ISDN-PRI ..................... 164
Operator.................................... 164
Terminal Network..................... 164
Echo Cancellation field
analog........................................ 233
T1/E1 ........................................ 146
echo, removing
analog........................................ 233
T1/E1 ........................................ 146
Edit selected Inbound Phonebook
Entry icon
E1.............................................. 363
T1.............................................. 310
Edit selected Outbound Phonebook
Entry icon
E1.............................................. 363
T1.............................................. 310
email account for voip unit
analog........................................ 281
T1/E1 ........................................ 180
email address for voip
analog........................................281
T1/E1 ........................................180
Enable SNMP Agent.............167, 269
Enable SNMP Agent field
analog........................................271
T1/E1 ........................................169
enabling SMTP
analog................................ 213, 280
527
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MultiVOIP User Guide
analog........................................ 280
T1/E1 ........................................ 178
enabling web browser GUI
analog.................................. 59, 226
T1/E1 ........................................ 139
Error Correction (SMTP logs)
analog........................................ 284
T1/E1 ........................................ 183
error correction, forward
analog........................................ 233
T1/E1 ........................................ 146
error message
COM port conflict..................... 116
COM port conflict (analog models)
.............................................. 218
error message (analog models)
MultiVOIP Not Found.............. 219
Phone Database Not Read......... 219
error message (T1/E1 models)
T1 models....................................12
failover (PSTN) feature.................330
FAQ for MultiVOIPs ......................10
fast busy (unobtainable) tones
analog................................175, 277
Fast Connect... See Fast Start, See Fast
Start
E1..............................................369
T1..............................................316
Fast Start compatibility
E1..............................................366
T1..............................................313
Fast Start plus H.245 Tunneling field
E1..............................................369
T1..............................................316
fax baud rate, default
analog........................................231
T1/E1 ........................................144
Fax Enable field
MultiVOIP Not Found.............. 133
Phone Database Not Read......... 133
ESF and CRC frame format (T1).. 155
ethernet cable pinout..................... 503
Ethernet interface
analog.......................................... 26
BRI ............................................. 34
European Community Directives.. 495
Excessive Zeroes (E1 stats) field.. 442
Excessive Zeroes (T1 stats) field.. 437
exchanges, phone
dedicated..................................... 71
institutional ................................. 71
local ............................................ 71
non-local..................................... 71
organizational ............................. 71
expansion card (analog, 4-to-8
channel) installation.................. 511
expansion card (T1/E1) installation
.................................................. 499
expansion card (T1/E1)operation.. 501
analog........................................231
T1/E1 ........................................144
fax machine
connecting to analog voip
(MVP130) .............................108
connecting to analog voip
(MVP130FXS)......................109
connecting to analog voip
(MVP210) .....................106, 107
connecting to analog voip (MVP-
410/810)............................97, 98
FAX Parameters
analog........................................231
T1/E1 ........................................144
fax tones, output level
analog........................................231
T1/E1 ........................................144
Fax Volume field
analog........................................231
T1/E1 ........................................144
FCC Declaration ...........................495
FCC Part 68 Telecom rules...........496
FCC registration number...............497
FCC rules, Part 15.........................495
FDX LED
F
factory default software settings... 460
factory defaults, downloading....... 460
failover (PSTN)
analog models .............................32
analog, MVP130 .........................33
BRI models .................................39
E1................................................24
analog models............................. 27
BRI models................................. 35
E1 models ................................... 19
528
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Index
T1................................................ 17
Filters (Console Message Settings)
analog........................................ 289
T1/E1 ........................................ 188
Filters button (Console Message
Settings)
analog........................................ 288
T1/E1 ........................................ 187
firmware upgrade, implementing.. 456
Firmware Version (System Info)
ANALOG/BRI.......................... 303
T1/E1 ........................................ 202
firmware version, identifying ....... 456
firmware, downloading................. 457
firmware, obtaining updated......... 452
Flash Hook Options fields
analog........................................231
T1/E1 ........................................144
Frame Search Restart Flag (T1 stats)
field ...........................................438
Frame Type field
analog........................................222
T1/E1 ........................................136
free calls
E1..............................................357
T1..............................................308
frequencies, touch tone .................250
Frequency 1 (custom tone) field
analog........................................278
T1/E1 ........................................176
Frequency 1 (tone pair scheme)
analog........................................275
T1/E1 ........................................173
Frequency 2 (custom tone) field
analog........................................278
T1/E1 ........................................176
Frequency 2 (tone pair scheme)
analog........................................275
T1/E1 ........................................173
frequency, power
analog models .............................43
E1 models....................................42
FRF11
analog........................................231
T1/E1 ........................................144
From (gateway, statistics, logs) field
..................................................422
front panel
analog models .............................32
BRI models .................................39
E1................................................24
MVP2400....................................16
MVP2410....................................16
MVP3010....................................24
T1................................................16
FTP client program .......................476
FTP client program, obtaining ......478
FTP client programs
E&M......................................... 258
FXO .......................................... 251
forgotten password................ 468, 472
Forward Address/Number
T1.............................................. 334
Forward Condition (Call Fwdg)
E1.............................................. 385
T1.............................................. 334
Forward Destination (Inbound PhBk)
E1.............................................. 386
Forward Error Correction (call
progress) field........................... 417
Forward Error Correction (SMTP
logs)
analog........................................ 284
T1/E1 ........................................ 183
Forward Error Correction field
analog........................................ 233
T1/E1 ........................................ 146
forward on busy
T1...................................... 334, 385
Forward upon No Response
E1.............................................. 385
T1.............................................. 334
forwarding, dual conditions (E1)
busy & no-response .................. 385
forwarding, dual conditions (T1)
busy & no-response .................. 334
Frame Format field
E1.............................................. 161
T1.............................................. 155
frame relay, and fax
graphic vs. textual orientation...485
FTP file transfers
using FTP client program..........478
using web browser ....................478
FTP Server Enable field
analog........................................225
529
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MultiVOIP User Guide
T1/E1 ........................................ 138
FXO Parameter fields
FTP Server function
Caller ID enable ........................251
Caller ID Type ..........................251
FXO Current Detect Timer .......249
Tone Detection..........................249
FXO Parameters............................247
FXS interface(MVP130)
as added feature ........................ 476
enabling .................................... 478
FTP Server, contacting ................. 480
FTP Server, invoking
download/transfer
using FTP client program ......... 484
using web browser .................... 482
FTP Server, logging in.................. 481
FTP Server, logging out................ 485
FTP transfers
file types ........................... 476, 479
phonebooks............................... 476
server location........................... 476
function tracing on/off (logging)
analog........................................ 289
T1/E1 ........................................ 188
FXO Disconnect On fields.... 249, 250
FXO disconnection criteria........... 249
FXO disconnection, triggering of 249,
250
uses of .......................................108
FXS interface(MVP130FXS)
uses of .......................................109
FXS interface(MVP210)
uses of .......................................106
FXS interface(MVP-410/810)
uses of .........................................97
FXS Loop Start Interface parameter
definitions .................................243
FXS Loop Start Interface Parameter
fields
Caller ID enable ................245, 246
Caller ID Type ..........................245
Current Loss..............................244
Detection Range (flash hook)....245
Inter Digit Regeneration Timer.244
Inter Digit Timer.......................244
Message Waiting Indication......244
Pass Through Enable.................245
Ring Count................................244
FXS Loop Start Parameter fields
Inter Digit Timer.......................243
Message Waiting Light .............243
FXS Loop Start Parameters...........243
FXS/FXO connector
FXO interface (MVP130)
uses of....................................... 108
FXO Interface Parameter definitions
.......................................... 248, 249
FXO Interface Parameter Definitions
.................................................. 250
FXO Interface Parameter fields
Current Loss ............................. 249
Current Loss Detect Timer ....... 249
Detection Range (flash hook) ... 251
Disconnect on Call Progress Tone
.............................................. 250
Disconnect Tone Sequence....... 250
Flash Hook................................ 251
Inter Digit Regeneration Timer. 249
Inter Digit Timer (dialing) ........ 248
Message Waiting Indication ..... 248
Regeneration (dialing) .............. 248
Ring Count................................ 250
Silence Detection...................... 250
Silence Timer............................ 250
Tone Detection.......................... 249
FXO interface(MVP210)
MVP130....................................108
MVP210....................................106
MVP-410/810 .............................97
G
G711 coders (RTP packetization,
voice/fax)
T1/E1 ........................................433
G723 coders (RTP packetization,
voice/fax)
T1/E1 ........................................433
G726 coders (RTP packetization,
voice/fax)
uses of....................................... 106
FXO interface(MVP-410/810)
uses of......................................... 97
T1/E1 ........................................433
G727 coders (RTP packetization,
voice/fax)
530
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Index
T1/E1 ........................................ 433
G729 coders (RTP packetization,
voice/fax)
T1/E1 ........................................ 433
Gain 1 (custom tone) field
analog........................................ 278
T1/E1 ........................................ 176
Gain 1 (tone pair scheme)
analog........................................ 275
T1/E1 ........................................ 173
Gain 2 (custom tone) field
analog........................................ 278
T1/E1 ........................................ 176
Gain 2 (tone pair scheme)
analog........................................ 275
T1/E1 ........................................ 173
gatekeeper interaction
analog models............................. 27
BRI models................................. 35
E1 models ............................. 19, 20
T1 models ............................. 12, 13
Gatekeeper IP Address (Gatekeeper
RAS) field
Gateway Number, From Details
(SMTP logs) field
analog........................................284
T1/E1 ........................................183
Gateway Number, To Details (SMTP
logs) field
analog........................................284
T1/E1 ........................................183
Gateway Prefix (outbound
phonebook) field
E1..............................................378
T1..............................................326
General Options fields
E1..............................................372
T1..............................................319
Generate Local Dial Tone
(Voice/FAX – AutoCall/Offhook
Alert) field
analog/BRI................................235
Generation Flash-Hook Options field
E&M .........................................258
FXO ..........................................251
Generation Time
E1.............................................. 366
T1.............................................. 313
Gatekeeper Name (Gatekeeper RAS)
field
E1..............................................163
T1..............................................157
Generation Time field
E1..............................................163
T1..............................................157
grounding
E1.............................................. 366
T1.............................................. 313
GateKeeper RAS Parameters
E1...................................... 366, 367
T1...................................... 313, 314
Gateway (IP Parameters) field
analog........................................ 224
T1/E1 ........................................ 138
Gateway H.323 ID (Outbound
Phonebook) field
T1...................................... 326, 378
Gateway Name (call progress) field
.................................................. 417
Gateway Name (callee, statistics,
logs) field.................................. 423
Gateway Name (caller, statistics, logs)
field........................................... 423
Gateway Name field
in rack installations .....................91
MVP210....................................107
MVP410ST ...............................103
...............................103
MVP810ST
grounding screw, diagrams
(MVP-2410/3010).......................52
(MVP-410/810)...........................53
(MVP-410ST/810ST)..................54
GUI (log reporting type) button
analog........................................288
T1/E1 ........................................187
H
H.245 Tunneling field
E1..............................................368
T1..............................................315
H.323
E1.............................................. 366
T1.............................................. 313
compatibility (analog models).....27
compatibility (BRI models).........35
531
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MultiVOIP User Guide
compatibility (E1 models) .......... 19
compatibility (T1 models) .......... 12
H.323 Annex E field
E1.............................................. 369
T1.............................................. 316
H.323 coder
Hold Sequence (T1/E1).................190
hold, caller on (analog)
musical jingle for ......................293
hold, caller on (T1/E1)
musical jingle for ......................192
hookup
analog........................................ 232
T1/E1 ........................................ 145
H.323 fields (Outbound Phonebook)
E1.............................................. 378
T1.............................................. 326
H.323 Multiplexing field
E1.............................................. 368
T1.............................................. 315
H.323 Parameters
MVP130......................................56
MVP130FXS...............................57
MVP210......................................55
MVP2410....................................52
MVP3010....................................52
MVP410......................................53
MVP-410ST/810ST ....................54
MVP810......................................53
HyperTerminal program
E1.............................................. 366
T1.............................................. 313
H.323 Port Number (outbound
phonebook) field
E1.............................................. 378
H.323 version 4 features
analog.......................................... 27
BRI ............................................. 35
E1................................................ 19
T1................................................ 12
H.323 Version 4 Parameters
E1 ............................................. 369
T1 ............................................. 316
H.450 features, incompatible with SIP
analog.................................. 27, 290
BRI ............................................. 35
E1................................................ 19
T1................................................ 12
T1/E1 ........................................ 189
H.450 functionality
and connectivity testing ..............79
I
IANA ............................................509
icon
variable version...................10, 113
icons, phonebook
E1..............................................363
T1..............................................310
identifying current firmware version
..................................................456
IFM (interface module, analog voips
only) description .......................462
IFM firmware, downloading.462, 463
IFM Version
System Info (ANALOG/BRI)...304
System Info (T1/E1)..................203
implementing firmware upgrade...456
in band, DTMF
analog........................................231
T1/E1 ........................................144
inbound phonebook
logs for...................................... 424
H.450 standard
ANALOG ................................... 30
BRI ............................................. 38
E1................................................ 23
T1................................................ 16
Hardware ID
example.......................................73
Inbound Phonebook Entries List icon
E1..............................................363
T1..............................................310
nbound Phonebook entries, list
I
System Info (ANALOG/BRI)... 304
System Info (T1/E1) ................. 203
Hold Sequence
E1..............................................382
T1..............................................331
inbound phonebook example
analog........................................ 294
T1/E1 ........................................ 193
Hold Sequence (analog)................ 291
quick............................................67
inbound vs. outbound phonebooks
E1..............................................361
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Index
T1.............................................. 308
Industry Canada requirements ...... 496
info sources
voip email account(analog models)
..............................................213
voip email account(T1/E1 models)
..............................................127
installation preparations (optional)
log reports by email.....................50
voip email account ......................50
installation, mechanical
analog models .............................26
BRI models .................................34
E1 models....................................18
T1 models....................................11
installation, quick
log reports by email.....................50
voip email account ......................50
installing Java vis-a-vis web GUI .488
integrated phone/data networks.....356
Inter Digit Regeneration Time (FXO)
field ...........................................249
Inter Digit Regeneration Time (FXS
Loop Start) field........................244
Inter Digit Regeneration Timer
(E&M) field ..............................257
Inter Digit Timer (dialing) field
DID-DPO..................................260
E&M .........................................257
FXO ..........................................248
FXS Loop Start .........................244
Intercept Tone (Regional Params) and
Offhook Alert (Voice/Fax Params)
..................................................274
Intercept Tone and required Interface
& Voice/Fax settings
analog........................................274
Interface field (DID-DPO)............259
Interface field (E&M) ...................256
interface parameters, accessing.....240
interface parameters, setting..........240
interface types, BRI
ST and U ...................................507
interfaces
BRI telephony .............................54
inter-office dialing
E1..............................................357
T1..............................................309
inter-operation (analog)
with T1/E1 voips.........................25
inter-operation (BRI)
analog telephony details...... 48, 211
BRI telephony details.................. 49
E1 details .................................... 48
E1 telephony details.................. 126
IP details ..................................... 47
IP details (analog system)......... 210
IP details (T1/E1 system).......... 124
ISDN-BRI telephony details..... 212
SMTP details .............................. 50
T1 details .................................... 47
T1 telephony details.................. 125
voip email account...................... 50
info sources (analog models)
SMTP details ............................ 213
voip email account.................... 213
info sources (T1/E1 models)
SMTP details ............................ 127
voip email account.................... 127
Input Gain field
analog........................................ 230
T1/E1 ........................................ 143
installation
airflow......................................... 91
analog prerequisites .......... 210, 211
BRI prerequisites ........................ 49
E1 prerequisites .................. 48, 126
expansion card (analog, 4-to-8
channel) ................................ 511
expansion card (T1/E1)............. 499
full summary............................... 46
in a nutshell................................. 44
in rack ......................................... 90
IP prerequisites ........................... 47
ISDN-BRI prerequisites............ 212
log reports by email (analog
models) ................................. 213
log reports by email (T1/E1
models) ................................. 127
software (detailed) .................... 111
T1 prerequisites .................. 47, 125
T1/E1 prerequisites................... 124
upgrade card (analog, 4-to-8
channel) ................................ 511
upgrade card (T1/E1)................ 499
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with T1/E1/BRI voips................. 34
Gateway ....................................224
IP Address.................................224
IP Mask.....................................224
IP Parameter fields (T1/E1)
Enable DHCP............................138
IP Parameters screen, accessing
analog........................................220
T1/E1 ........................................134
IP startup configuration...................59
IP Statistics field definitions .425, 427
IP Statistics fields
inter-operation with phone system
analog models............................. 26
BRI models................................. 34
E1 models ................................... 18
T1 models ................................... 11
IP Address (call progress) field..... 417
IP Address (callee, statistics, logs)
field........................................... 423
IP Address (caller, statistics, logs)
field........................................... 423
IP Address (outbound phonebook)
E1.............................................. 377
T1.............................................. 325
IP Address field
analog........................................ 224
T1/E1 ........................................ 138
IP Address field (Registered Gateway
Details)...................................... 448
IP Address to Ping (Link
Management) field.................... 430
IP Address, From Details (SMTP
logs) field
Clear..........................................425
Received (RTCP Packets).........428
Received (RTP Packets)............428
Received (TCP Packets)............427
Received (Total Packets) ..........425
Received (UDP Packets)...........427
Received with errors (RTCP
Packets).................................428
Received with errors (RTP Packets)
..............................................428
Received with errors (TCP Packets)
..............................................427
Received with errors (Total
analog........................................ 284
T1/E1 ........................................ 183
IP address, SysLog Server
Packets).................................427
Received with errors (UDP
analog........................................ 288
T1/E1 ........................................ 187
IP Address, To Details (SMTP logs)
field
analog........................................ 284
T1/E1 ........................................ 183
IP datagram and DiffServ
analog........................................ 223
T1/E1 ........................................ 137
IP Mask field
analog........................................ 224
T1/E1 ........................................ 138
IP parameter definitions
Packets).................................427
Transmitted (RTCP Packets).....428
Transmitted (RTP Packets) .......428
Transmitted (TCP Packets) .......427
Transmitted (Total Packets)......425
Transmitted (UDP Packets).......427
IP Statistics function .....................425
ISDN BRI Interface screen fields
Status, Layer 1 Interface ...........444
Status, SPID0 ............................445
Status, SPID1 ............................446
ISDN BRI Parameters
TEI n Assignment .....................265
ISDN BRI Parameters fields
analog........................................ 222
T1/E1 ........................................ 136
IP Parameter fields (analog)
Country .....................................265
Dialing Options.........................265
Inter Digit Timer.......................265
Layer 1 Interface.......................264
Operator ....................................265
Select BRI Interface..................264
Switch Information ...................265
ISDN BRI Statistics screen fields
Diffserv Parameters .................. 222
DNS Server IP Address ............ 224
Enable DHCP............................ 224
Enable DNS .............................. 224
Frame Type............................... 222
FTP Server Enable.................... 224
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Index
Clear (button)............................ 446
D Channel Information (field
T1..............................................320
key system
group).................................... 445
Layer 1 Interface (field group).. 444
Loss of Framing........................ 444
Loss of Sync ............................. 444
Rx Packets ................................ 445
Select BRI Interface.................. 444
SPID0........................................ 445
SPID1........................................ 446
State .......................................... 444
Switch Information (field group)
.............................................. 445
Tx Packets................................. 445
ISDN parameters, setting.............. 166
ISDN-BRI operating modes
MVP-410ST/810ST)................. 102
ISDN-BRI Parameter definitions.. 264
ISDN-BRI telephony interfaces
uses of....................................... 102
ISDN-BRI telephony parameters.. 212
ISDN-PRI
connecting to analog voip
(MVP130) .............................108
connecting to analog voip
(MVP210) .............................106
connecting to analog voip (MVP-
410/810)..................................97
Knowledge Base (online, for
MultiVOIPs) ...............................10
L
lab voip network
use in setup..................................72
Last button
Logs (Statistics) screen .............422
Last Error (Link Management) field
..................................................431
Layer 1 Interface
ISDN-BRI .................................264
Layer 1 Interface fields (ISDN BRI
Statistics)...................................444
LED definitions
analog models .............................32
BRI models .................................39
E1................................................24
MVP2400....................................16
MVP2410....................................17
MVP3010....................................24
T1................................................17
LED definitions (analog models)
FDX ............................................32
LNK ............................................32
LED definitions (analog)
Boot.............................................32
Ethernet.......................................32
Power ..........................................32
RCV (channel)...........................32
RSG.............................................32
XMT (channel) ..........................32
XSG.............................................32
LED definitions (analog, E1)
types supported ......................... 166
ISDN-PRI implementations.......... 166
J
Java
installing ................................... 488
web GUI and............................. 488
jitter buffer
analog........................................ 237
T1/E1 ........................................ 149
Jitter Value (Fax) field
analog........................................ 231
T1/E1 ........................................ 144
Jitter Value field
analog........................................ 239
T1/E1 ........................................ 151
jitter, dynamic
analog........................................ 237
T1/E1 ........................................ 149
jumper, DID
MVP210.................................... 104
MVP-410/810 ............................. 96
LNK ............................................24
LED definitions (analog, MVP130)
Boot.............................................33
Ethernet.......................................33
FDX ............................................33
LNK ............................................33
K
Keep Alive field
E1.............................................. 373
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MultiVOIP User Guide
Power.......................................... 33
LED sets (T1/E1), left and right....501
LED types
analog models .............................30
BRI models .................................38
lifting
precaution about..........................85
limitations notice (regulatory),
Canadian ...................................497
Line Build Out field
E1..............................................165
T1..............................................159
Line Coding field
E1..............................................165
T1..............................................159
Line Loopback Activation Signal (T1
stats) field..................................438
Line Loopback Deactivation Signal
(T1 stats) field...........................437
Link Management (Statistics) fields
Clear command button..............430
IP Address column....................431
IP Address to Ping.....................430
Last Error ..................................431
No. of Pings ..............................430
No. of Pings Received...............431
No. of Pings Sent ......................431
Ping Size in Bytes .....................430
Response Timeout.....................430
Round Trip Delay......................431
Start Now command button ......430
Timer Interval between Pings ...430
Link Management (Statistics) screen
field definitions .................430, 431
IP Address (ping target .................431
Link Status fields
RCV (channel) .......................... 33
RSG ............................................ 33
XMT (channel).......................... 33
XSG ............................................ 33
LED definitions (analog, T1)
LNK............................................ 17
LED definitions (BRI models)
FDX ............................................ 39
LNK............................................ 39
LED definitions (BRI)
Boot ............................................ 39
Ethernet....................................... 39
Power.......................................... 39
RCV (channel) .......................... 39
XMT (channel).......................... 39
LED definitions (E1)
Boot ............................................ 24
E1................................................ 24
FDX ............................................ 24
IC ................................................ 24
LC............................................... 24
LS ............................................... 24
ONL............................................ 24
Power.......................................... 24
PRI.............................................. 24
LED definitions (T1)
Boot ............................................ 17
FDX ............................................ 17
IC ................................................ 17
LC............................................... 17
LS ............................................... 17
ONL............................................ 17
Power.......................................... 17
PRI.............................................. 17
LED indicators
Link Management (Statistics)
screen ....................................431
List of Registered Numbers field
(Registered Gateway Details) ...448
lithium battery caution ....................85
LNK LED
analog models .............................32
analog, MVP130 .........................33
BRI models .................................39
E1................................................24
T1................................................17
loading of weight in rack ................91
local configuration
E1................................................ 23
T1................................................ 16
LED indicators (analog)
channel operation........................ 30
general operation ........................ 30
LED indicators (BRI)
channel operation........................ 38
general operation ........................ 38
LED indicators, active
analog.......................................... 30
E1................................................ 23
T1................................................ 16
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Index
analog/BRI................................ 210
T1/E1 ........................................ 124
local configuration procedure
T1/E1 ........................................180
Logs (Statistics) fields
Bytes received...........................423
Bytes Sent .................................422
Call Forwarded to......................424
Call Transferred to ....................424
CT Ph# ......................................424
Duration ....................................422
From (gateway).........................422
Gateway Name (callee).............423
Gateway Name (caller) .............423
H.450 functionality ...................424
IP Address (callee)....................423
IP Address (caller) ....................423
Log #.........................................422
Mode.........................................422
Options (caller) .........................423
Options callee............................423
Outbound digits.........................423
Packets Lost ..............................423
Packets received........................423
Packets Sent ..............................422
Start Date, Time........................422
Status.........................................422
Supplementary Services info.....424
To (gateway) .............................422
Voice coder...............................423
Logs (Statistics) function ...........420
Logs (Statistics) screen
detailed, analog......................... 215
detailed, T1/E1.......................... 129
summary, analog....................... 214
summary, T1/E1 ....................... 128
local exchange numbers.................. 71
local voip configuration (analog).. 208
local voip configuration (T1/E1) .. 122
local Windows GUI vs. web GUI
comparison................................ 487
local-rate access (E1)
to remote PSTN .......................... 18
local-rate calls to remote voip sites
E1.............................................. 358
Log # (statistics, logs) field........... 422
log report email, customizing
analog................................ 283, 284
T1/E1 ................................ 182, 183
log report email, triggering
analog ...................................... 282
T1/E1....................................... 181
log reporting method, setting
analog........................................ 286
T1/E1 ........................................ 185
log reports
analog models........................... 213
T1/E1 models............................ 127
log reports & SMTP
analog........................................ 280
T1/E1 ........................................ 178
log reports and SMTP
quick ........................................... 61
log reports by email
Delete File button......................422
First button................................422
Last button ................................422
Next button................................422
Previous button .........................422
logs and web browser GUI
analog........................................287
T1/E1 ........................................186
logs by email, illustration
analog........................................285
T1/E1 ........................................184
Logs screen definitions
analog........................................287
T1/E1 ........................................186
Logs screen field definitions
analog........................................288
T1/E1 ........................................187
Logs screen parameters (analog)
Enable Console Messages.........288
analog........................................ 280
quick ........................................... 61
T1/E1 ........................................ 178
log reports, quick ............................ 50
logging options
analog........................................ 287
T1/E1 ........................................ 186
logging update interval
analog........................................ 287
T1/E1 ........................................ 186
logging, web GUI and................... 487
Login Name (SMTP) field
analog........................................ 281
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MultiVOIP User Guide
Filters........................................ 288
lost password.........................468, 472
GUI........................................... 288
IP Address (SysLog Server) ..... 288
Online Statistics Updation Interval
.............................................. 288
Port (SysLog Server) ................ 288
SMTP........................................ 288
SNMP ....................................... 288
SysLog Server Enable............... 288
Turn Off Logs........................... 288
Logs screen parameters (T1/E1)
Console Message Settings ........ 187
Enable Console Messages......... 186
Filters........................................ 187
GUI........................................... 187
IP Address (SysLog Server) ..... 187
Online Statistics Updation Interval
.............................................. 187
Port (SysLog Server) ................ 187
SMTP........................................ 187
SNMP ....................................... 187
SysLog Server Enable............... 187
Turn Off Logs........................... 187
logs screen, accessing
M
Mac Address
System Info...............................412
System Info (ANALOG/BRI)...304
System Info (T1/E1)..................203
mail criteria (SMTP), records
analog........................................282
T1/E1 ........................................181
Mail Server IP Address (SMTP) field
analog........................................282
T1/E1 ........................................180
Mail Type (SMTP logs) field
analog........................................282
T1/E1 ........................................181
mains frequency
analog models .............................43
E1 models....................................42
management (E1 models)
local.............................................20
remote (SNMP)...........................20
remote (web browser GUI) .........20
management of voips, remote
analog........................................269
T1/E1 ........................................167
Max bandwidth (coder)
analog........................................ 286
T1/E1 ........................................ 185
long distance call savings
T1.............................................. 308
long-distance call savings
E1.............................................. 356
Long-Haul Mode field
analog........................................232
T1/E1 ........................................145
Max Baud Rate field
analog........................................231
E1.............................................. 161
T1.............................................. 155
Loss of Frame Alignment (E1 stats)
field........................................... 440
Loss of Frame Alignment (T1 stats)
field........................................... 437
Loss Of Framing field (ISDN BRI
Statistics, Layer 1 Interface) ..... 444
Loss of MultiFrame Alignment (E1
stats) field ................................. 441
Loss of MultiFrame Alignment (T1
stats) field ................................. 438
Loss of Sync field (ISDN BRI
T1/E1 ........................................144
ax Retransmission (SPP, General
Options) field
E1..............................................372
T1..............................................319
maximum cable span
E1..............................................161
T1..............................................155
Maximum Jitter Value field
analog........................................238
T1/E1 ........................................150
Message Waiting Indication (DID-
DPO) .........................................260
Message Waiting Indication (E&M)
and DID.....................................257
Message Waiting Indication field
DID-DPO..................................260
M
Parameters, Layer 1 Interface).. 444
lost packets, consecutive
analog........................................ 239
T1/E1 ........................................ 151
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Index
E&M......................................... 257
FXO .......................................... 248
FXS Loop Start......................... 244
Minimum Jitter Value field
analog........................................ 237
T1/E1 ........................................ 149
Mode (call progress) field............. 416
Mode (Fax) field
analog........................................ 231
T1/E1 ........................................ 144
Mode (SPP) field
E1.............................................. 372
T1.............................................. 319
Mode (statistics, logs) field........... 422
model descriptions
mounting in rack .............................90
procedure for...............................92
safety.....................................85, 91
mounting options ..............................9
Multiplexed UDP field
E1..............................................369
T1..............................................316
MultiVOIP configuration software .58
E1 models....................................20
T1 models....................................13
MultiVOIP FAQ (on MTS web site)
....................................................10
MultiVOIP Program Menu items..449
MultiVOIP Program Menu options
Configuration ............................449
Configuration Port Setup ..........449
Date & Time Setup ...................449
Download Factory Defaults ......450
Download Firmware .................450
Download IFM Firmware .........450
Set Password .............................450
Uninstall....................................450
Upgrade Software .....................450
MultiVOIP program menu, option
descriptions .......................449, 450
MultiVOIP software
E1................................................ 18
modem relay
analog........................................ 238
T1/E1 ........................................ 150
modem traffic on voip network
analog........................................ 238
T1/E1 ........................................ 150
modem, command
and Regional Parameters Country
Selection ................................. 61
modem, command (analog)
and Regional Parameters Country
Selection ....................... 214, 273
modem, command (T1/E1)
and Regional Parameters Country
Selection ............... 128, 170, 171
modem, remote
installing....................................111
location of files .........................114
program icon location ...............115
uninstalling........................118, 473
MultiVOIP software (analog)
moving around in ......................219
MultiVOIP software (T1/E1)
configuration/command
setup for...................................... 61
modem, remote
configuration/command (analog)
setup for ............................ 214, 273
modem, remote
moving around in ......................133
MultiVoipManager..........................10
analog........................................208
T1/E1 ........................................123
MultiVoipManager software
configuration/command (T1/E1)
setup for.................... 128, 170, 171
Monitor Link fields
Link Management (Statistics)
screen.................................... 430
mounting
E1 models....................................20
T1 models....................................13
musical jingle during call transfer
analog........................................293
T1/E1 ........................................192
MVP130
analog models............................. 26
BRI models................................. 34
E1 models ................................... 18
T1 models ................................... 11
cabling procedure......................108
Introduction.................................25
unpacking....................................89
MVP210
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MultiVOIP User Guide
grounding.................................. 107
MVP210x
T1/ISDN....................................158
No. of Entries field (Registered
Gateway Details).......................448
No. of Pings (Link Management) field
..................................................430
No. of Pings Received (Link
Management) field....................431
No. of Pings Sent (Link Management)
field ...........................................431
no-response & busy(E1)
cabling procedure...................... 104
unpacking.................................... 88
MVP2410
cabling procedure........................ 93
unpacking.................................... 86
MVP-2410
remote configuration modem...... 94
MVP3010
cabling procedure........................ 93
remote configuration modem...... 94
unpacking.................................... 86
MVP410
forwarding, dual conditions ......385
no-response & busy(T1)
forwarding, dual conditions ......334
NT1 device
cabling procedure........................ 95
remote configuration modem...... 99
MVP410ST
when required for MVP410ST..101
when required for MVP810ST..101
NT1 device, use of
grounding.................................. 103
MVP-410ST
cabling procedure...................... 100
MVP410x
unpacking.................................... 87
MVP810
BRI voip units.....................49, 101
Number of Days (email log criteria)
analog........................................282
T1/E1 ........................................181
Number of Records (email log
criteria)
cabling procedure........................ 95
remote configuration modem...... 99
MVP810ST
analog........................................282
T1/E1 ........................................181
numbering plan resources .............406
grounding.................................. 103
MVP-810ST
O
cabling procedure...................... 100
MVP810x
obtaining updated firmware ..........452
Offhook alert.................................234
Offhook Alert (Voice/Fax Params)
and Intercept Tone (Regional
Params) .....................................234
Offhook Alert Timer (Voice/FAX --
AutoCall/Offhook Alert) field
analog/BRI................................236
T1/E1 ........................................148
official phone numbers
caution about...............................72
Online Statistics Updation Interval
field (Logs)
analog........................................288
T1/E1 ........................................187
operating system ..........................40
operating temperature .....................91
operating voltage
unpacking.................................... 87
N
national-rate calls to foreign voip sites
E1.............................................. 360
Netcoder coders (RTP packetization,
voice/fax)
T1/E1 ........................................ 433
network access
analog........................................ 271
T1/E1 ........................................ 169
Network Disconnection field
analog........................................ 239
T1/E1 ........................................ 151
network/terminal settings, voip and
PBX
E1/ISDN ................................... 164
ISDN-BRI................................. 264
analog models .............................43
T1 models....................................42
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Index
operation
T1/E1 ........................................143
output level, fax tones
analog........................................231
T1/E1 ........................................144
outside line, access to................71, 73
expansion card (T1/E1)............. 501
Operator (ISDN) field
E1/ISDN ................................... 164
T1/ISDN ................................... 158
Operator definitions
P
ISDN-BRI................................. 268
Operator field
packet priority and DiffServ
ISDN-BRI................................. 265
Optimization Factor field
analog........................................223
T1/E1 ........................................137
packetization (RTP), ranges &
increments
T1/E1 ........................................433
packetization rates
analog........................................ 238
T1/E1 ........................................ 150
Options (call progress) field ......... 417
Options (callee, statistics, logs) field
.................................................. 423
Options, From Details (SMTP logs)
field
analog........................................ 284
T1/E1 ........................................ 183
Options, To Details (SMTP logs) field
analog........................................ 284
T1/E1 ........................................ 183
out of band, DTMF
analog........................................ 231
T1/E1 ........................................ 144
Outbound Digits (call progress) field
.................................................. 416
Outbound Digits (SMTP logs) field
analog........................................ 284
T1/E1 ........................................ 183
Outbound digits (statistics, logs) field
.................................................. 423
outbound phonebook
coder options and ......................432
Packets Lost (call progress) field ..416
Packets Lost (SMTP logs) field
analog........................................283
T1/E1 ........................................182
Packets lost (statistics, logs) field .423
Packets Received (call progress) field
..................................................416
Packets Received (SMTP logs) field
analog........................................283
T1/E1 ........................................182
Packets received (statistics, logs) field
..................................................423
Packets Sent (call progress) field ..416
Packets Sent (SMTP logs) field
analog........................................283
T1/E1 ........................................182
ets sent (statistics, logs) field.422
Pack
packets, consecutive lost
example....................................... 73
Outbound Phonebook Entries List
icon
analog........................................239
T1/E1 ........................................151
Parallel H.245 field
E1.............................................. 363
T1.............................................. 310
Outbound Phonebook entries, list
E1.............................................. 374
T1.............................................. 322
outbound phonebook example
quick ........................................... 63
outbound vs. inbound phonebooks
E1.............................................. 361
T1.............................................. 308
Output Gain field
E1..............................................369
T1..............................................316
parameters tracked by console
analog........................................289
T1/E1 ........................................188
Pass Through (E&M) field............256
Pass Through Enable (FXS Loop Start
interface) and AutoCall (Voice/Fax
Params) .....................................245
Password (proxy server) field
E1..............................................371
T1..............................................318
analog........................................ 230
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MultiVOIP User Guide
Password (SMTP) field
Phone Signaling Tones & Cadences
analog........................................272
T1/E1 ........................................170
phone startup configuration.............59
phone switch types
analog........................................ 282
T1/E1 ........................................ 180
password, lost/forgotten........ 468, 472
password, setting........................... 468
web browser GUI...................... 472
patents ............................................. 2
patterns, destination
tips about..................................... 70
PBX characteristics, variations in
E1.............................................. 405
T1.............................................. 354
PBX interaction
ISDN implementations in..........166
phone/IP details
importance of writing down........46
importance of writing down
(analog) .................................210
importance of writing down
(T1/E1)..................................124
phonebook
FTP remote file transfers...........476
phonebook configuration
starter ..........................................63
phonebook configuration (analog)207,
409
analog models............................. 26
BRI models................................. 34
E1 models ................................... 18
T1 models ................................... 11
PC, command
phonebook configuration (remote) 476
phonebook configuration (T1/E1).122
Phonebook Configuration icon
E1..............................................363
T1..............................................310
Phonebook Configuration Parameter
definitions
E1...... 366, 367, 368, 369, 370, 371
T1...... 313, 314, 315, 316, 317, 318
Phonebook Configuration procedure
T1..............................................309
COM port assignment (detailed)116
COM port requirement................ 51
demands upon............................. 51
non-dedicated use ....................... 51
operating system ......................... 51
settings........................................ 51
specifications .............................. 51
PCM Law field
E1.............................................. 165
T1.............................................. 159
Permissions (SNMP) field
nebook Configuration Procedure
Pho
analog........................................ 271
T1/E1 ........................................ 169
personnel requirement
for rack installation..................... 91
to lift during installation ............. 92
to lift unit during installation ...... 85
Phone Book Version
System Info (ANALOG/BRI)... 304
System Info (T1/E1) ................. 203
phone exchanges
dedicated..................................... 71
institutional ................................. 71
local ............................................ 71
non-local..................................... 71
organizational ............................. 71
Phone Number (Voice/FAX –
AutoCall/Offhook Alert) field
analog/BRI................................ 236
T1/E1 ........................................ 148
E1..............................................362
Phonebook Configuration screen
E1..............................................364
T1..............................................309
Phonebook Configuration screen (E1)
Mode (SPP Protocol) ................372
Phonebook Configuration screen (T1)
Mode (SPP Protocol) ................319
Phonebook configuration screen fields
(E1)
Password (proxy server)............371
Re-Registration Time (proxy
server) ...................................371
Phonebook Configuration screen
fields (E1)
Annex E (H.323, UDP
multiplexing).........................369
Client Options...........................372
542
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Index
Gatekeeper IP Address.............. 366
Gatekeeper Name...................... 366
Gateway Name.......................... 366
General Options........................ 372
H.245 Tunneling....................... 368
H.323 Multiplexing................... 368
Keep Alive................................ 372
Max Retransmission (SPP, General
Options) ................................ 372
Parallel H.245 (Tunneling with Fast
Start) ..................................... 369
Port (SPP, General Options) ..... 372
Proxy Domain Name / IP Address
.............................................. 370
RAS TTL Value........................ 367
Register with GateKeeper......... 366
Registrar IP Address................. 372
Registrar Options...................... 372
Registrar Port............................ 372
Retransmission (SPP, General
Parallel H.245 (Tunneling with Fast
Start)......................................316
Port (SPP, General Options) .....319
Proxy Domain Name / IP Address
..............................................317
RAS TTL Value........................314
Register with GateKeeper .........313
Registrar IP Address .................319
Registrar Options ......................319
Registrar Port ............................319
Retransmission (SPP, General
Options).................................319
Signaling Number (proxy server)
..............................................317
Signaling Port............................313
Signaling Port (Gatekeeper)......313
Use Fast Start............................313
Use SIP Proxy...........................317
User Name (proxy server).........317
phonebook destination patterns.......70
phonebook dialing patterns .............70
phonebook digits
dropping......................................71
leading.........................................71
non-PSTN type............................71
specialized codes.........................71
types used....................................70
phonebook entries, coordinating
E1..............................................362
T1..............................................309
phonebook examples
analog........................................207
mixed digital/analog....................73
phonebook icons
E1..............................................363
T1..............................................310
phonebook objectives &
considerations
E1..............................................361
phonebook sidebar menu
Options) ................................ 372
Signaling Number (proxy server)
.............................................. 370
Signaling Port ........................... 366
Signaling Port (Gatekeeper)...... 366
Use Fast Start............................ 366
Use SIP Proxy........................... 370
User Name (proxy server)......... 370
Phonebook configuration screen fields
(T1)
Password (proxy server) ........... 318
Re-Registration Time (proxy
server) ................................... 318
Phonebook Configuration screen
fields (T1)
Annex E (H.323, UDP
multiplexing)......................... 316
Client Options........................... 319
Gatekeeper IP Address.............. 313
Gatekeeper Name...................... 313
Gateway Name.......................... 313
General Options........................ 319
H.245 Tunneling....................... 315
H.323 Multiplexing................... 315
Keep Alive................................ 319
Max Retransmission (SPP, General
Options) ................................ 319
E1..............................................364
T1..............................................311
phonebook tips................................70
phonebook worksheet ...............76, 77
phonebook, analog voips...............409
phonebook, inbound
example.......................................73
example, quick ............................67
543
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MultiVOIP User Guide
phonebook, outbound
T1..............................................328
power consumption
analog models .............................43
E1 models....................................42
power frequency
analog models .............................43
E1 models....................................42
Power LED
example....................................... 73
example, quick............................ 63
phonebooks, inbound vs. outbound
E1.............................................. 361
T1.............................................. 308
phonebooks, objectives &
considerations
T1.............................................. 308
Phonebooks, objectives &
analog models .......................32, 33
BRI models .................................39
powering of ISDN-BRI phones
MVP-410ST/810ST ..................102
Prefix Matched (call progress) field
..................................................416
Prefix Matched (SMTP logs) field
analog........................................284
T1/E1 ........................................183
prerequisites
for technical configuration (analog)
..............................................210
for technical configuration (T1/E1)
..............................................124
prerequisites for installation
considerations
E1.............................................. 356
phonebooks, sample........................ 75
Ping Size in Bytes (Link
Management) field.................... 430
pinging and connectivity................. 82
pinout
BRI connector........................... 506
command cable ......................... 503
ethernet cable............................ 503
T1/E1 connector........................ 504
Voice/FAX connector............... 504
polarity sensitivity
DID lines and (MVP210).......... 107
DID lines and (MVP-410/810) ... 98
Port (SPP, General Options) field
E1.............................................. 372
T1.............................................. 319
Port field (Registered Gateway
Details)...................................... 448
Port field, SysLog Server
analog........................................ 288
T1/E1 ........................................ 187
Port Number (Gatekeeper RAS) field
E1.............................................. 366
T1.............................................. 313
Port Number (proxy server) field
E1.............................................. 370
T1.............................................. 317
Port Number (SMTP) field
analog........................................ 282
T1/E1 ........................................ 180
port number (SNMP) field
analog........................................ 271
T1/E1 ........................................ 169
Port Number field, SPP (Outbound
Phonebook)
BRI info ......................................49
E1 info.........................................48
IP info .........................................47
T1 info.........................................47
PRI
ISDN implementations..............166
product CD......................................44
use in software installation..58, 111
Product CD
E1 models....................................20
T1 models....................................13
product family.............................9, 10
product groups ..................................8
Program Menu items.....................449
Protocol Type (outbound phonebook)
E1..............................................377
T1..............................................325
Proxy Domain Name / IP Address
field
E1..............................................370
T1..............................................317
Proxy/NAT Device Parameters –
Public IP Address
E1..............................................373
T1..............................................320
E1.............................................. 380
544
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Index
PSTN failover feature
Received (RTCP Packets, IP Stats)
field ...........................................428
Received (RTP Packets, IP Stats) field
..................................................428
Received (TCP Packets, IP Stats) field
..................................................427
Received (Total Packets, IP Stats)
field ...........................................426
Received (UDP Packets, IP Stats)
field ...........................................427
Received with Errors (RTCP Packets,
IP Stats) field.............................428
Received with Errors (RTP Packets,
IP Stats) field.............................428
Received with Errors (TCP Packets,
IP Stats) field.............................427
Received with Errors (Total Packets,
IP Stats) field.............................427
Received with Errors (UDP Packets,
IP Stats) field.............................427
Recipient Address (email logs) field
T1/E1 ........................................181
Recipient Address (email logs)field
analog........................................282
recovering voice packets
Alternate Routing, and.............. 330
analog models............................. 27
BRI models................................. 35
E1 models ................................... 19
T1 models ................................... 12
Pulse Density Violation (T1 stats)
field........................................... 438
Pulse Shape Level field
E1.............................................. 165
T1.............................................. 159
Q
Q.931 Port Number (outbound
phonebook) field
T1.............................................. 326
quality-of-service
analog.......................................... 27
BRI ............................................. 35
E1................................................ 19
T1................................................ 12
R
rack mounting
grounding.................................... 91
safety..................................... 85, 91
rack mounting instructions.............. 90
rack mounting procedure ................ 92
rack, equipment
weight capacity of....................... 91
rack-mountable voip models........... 85
RAS TTL Value (Gatekeeper RAS)
field
3
analog........................................23
T1/E1 ........................................146
Red Alarm (E1 stats) field ............440
Red Alarm (T1 stats) field ............437
Regeneration (dialing, FXO) field 248
Regional Parameter definitions
analog........................ 274, 275, 276
T1/E1 ........................ 172, 173, 174
Regional Parameter fields (analog)
Cadence.....................................276
Country/Region (tone schemes) 274
Custom (tones)..........................276
Frequency 1...............................275
Frequency 2...............................275
Gain 1........................................275
Gain 2........................................275
Pulse Generation Ratio..............276
type (of tone).............................275
Regional Parameter fields (T1/E1)
Cadence.....................................174
Country/Region (tone schemes) 172
Custom (tones)..........................174
Frequency 1...............................173
E1.............................................. 367
T1.............................................. 314
RCV (channel) LED
analog models .......................... 32
analog, MVP130 ....................... 33
BRI models................................ 39
Receive Slip (E1 Stats) field......... 442
Receive Slip (T1 Stats) field......... 439
Receive Timeslot 16 Alarm Indication
Signal (E1 stats) field................ 441
Receive Timeslot 16 Loss of
MultiFrame Alignment (E1 stats)
field........................................... 442
Receive Timeslot 16 Loss of Signal
(E1 stats) field........................... 441
545
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MultiVOIP User Guide
Frequency 2 .............................. 173
MVP810......................................99
Remote Configuration/Command
Modem
setup for ......................................61
Remote Configuration/Command
Modem (analog)
Gain 1 ....................................... 173
Gain 2 ....................................... 173
type (of tone)............................. 172
Regional Parameters fields (analog,
except MVP130)
Country Selection for Built-In
Modem.................................. 276
Regional Parameters fields (T1/E1)
Country Selection for Built-In
Modem.................................. 174
regional parameters, setting
analog........................................ 272
T1/E1 ........................................ 170
Register Duration field (Registered
Gateway Details) ...................... 448
Registered Gateway Details
(Statistics) screen, accessing..... 448
Registered Gateway Details
‘Statistics’ function......... 447, 448
Registered Gateway Details screen448
Registered Gateway Details screen
fields
Description................................ 448
IP Address................................. 448
No. of Entries............................ 448
Port ........................................... 448
Register Duration...................... 448
Status ........................................ 448
Registered Gateway Details screen
fields: ........................................ 448
Registrar IP Address field
E1.............................................. 373
T1.............................................. 320
Registrar Options fields
E1.............................................. 373
T1.............................................. 320
Registrar Port field
E1.............................................. 373
T1.............................................. 320
Registration Option Parameters
(Inbound Phone Book)
E1.............................................. 386
T1.............................................. 335
remote configuration modem
setup for ............................214, 273
Remote Configuration/Command
Modem (T1/E1)
setup for .................... 128, 170, 171
remote control/configuration
web GUI and.............................488
Remote Device is [legacy voip]
(Outbound Phonebook)
T1..............................................328
Remote Device is [legacy] MultiVOIP
checkbox ...................................380
remote phonebook configuration ..476
remote voip
using to confirm configuration...50,
63
remote voip configuration (analog)
..................................................208
remote voip configuration (T1/E1)122
Remote Voip Management
analog........................................269
T1/E1 ........................................167
Remove Prefix (inbound) field
E1..............................................384
T1..............................................333
Remove Prefix (outbound) field
E1..............................................377
325
T1..............................................
re-order tone, custom
T1/E1 ........................................175
Reply-To Address (email logs) field
T1/E1 ........................................181
Reply-To Address (email logs)field
analog........................................282
Requires Authentication (SMTP) field
analog........................................281
T1/E1 ........................................180
Re-Registration Time (proxy server)
E1..............................................371
T1..............................................318
Resolutions (MultiVOIP
MVP-2410 .................................. 94
MVP3010.................................... 94
MVP410...................................... 99
troubleshooting) ..........................10
546
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Response Timeout (Link
sample phonebooks.........................75
Save Setup command
analog........................................305
T1/E1 ........................................204
saving configuration
analog........................................305
T1/E1 ........................................204
user............................................466
Saving the MultiVOIP Configuration
analog........................................305
T1/E1 ........................................204
savings on toll calls
E1..............................................356
T1..............................................308
scale-ability
Management) field.................... 430
Retransmission (SPP, General
Options) field
E1.............................................. 372
T1.............................................. 319
Retrieve Sequence
analog........................................ 294
T1/E1 ........................................ 193
Retrieve Sequence (analog) .......... 291
Retrieve Sequence (T1/E1)........... 190
RFC768......................................... 509
RFC793......................................... 509
ring cadences, custom
analog........................................ 279
T1/E1 ................................ 175, 177
Ring Count (FXO) field................ 250
Ring Count field
E1................................................18
T1................................................11
second dial tone
FXS Loop Start......................... 244
Ring Count forwarding condition
E1.............................................. 386
T1.............................................. 335
ring tone, custom
analog........................................ 278
T1/E1 ................................ 175, 176
ring-tones
and comma use............................72
Select All (SMTP logs) field
analog........................................283
T1/E1 ........................................182
Select BRI Interface field..............444
Select BRI Interface ISDN-BRI field
BRI............................................264
Select Channel field
analog........................................ 277
T1/E1 ........................................ 175
Round Trip Delay (Link
Management) field.................... 431
Round Trip Delay field
E1.............................................. 382
T1.............................................. 330
RSG LED
analog models .......................... 32
analog, MVP130 ....................... 33
RTP packetization, ranges &
increments................................. 433
RTP Parameters screen................. 433
Rx Packets field (ISDN BRI
analog........................................230
T1/E1 ........................................143
Select Channel, Supplementary
Services field
analog........................................293
T1/E1 ........................................192
Selected Coder field
analog........................................232
T1/E1 ........................................145
Set Baud Rate
analog........................................301
T1/E1 ........................................200
Set Custom Tones & Cadences
T1/E1 ........................................175
Set ISDN Parameters ....................166
Set Log Reporting Method
Statistics, D-Channel Information)
.................................................. 445
analog........................................286
T1/E1 ........................................185
Set Password (program menu option) ,
command...................................468
Set Password (web browser GUI) ,
command...................................472
S
Safety Recommendations for Rack
Installations................................. 91
safety warnings............................... 85
Safety Warnings Telecom.......... 85
547
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Set Password option description
(MultiVOIP program menu) ..... 450
Set Regional Parameters
analog........................................ 272
T1/E1 ........................................ 170
Set SMTP Parameters
signaling types
analog telephony (MVP130).....108
analog telephony (MVP130FXS)
..............................................109
analog telephony (MVP210)....106,
107
analog........................................ 280
T1/E1 ........................................ 178
Set SNMP Parameters
analog........................................ 269
T1/E1 ........................................ 167
Set Supplementary Services
analog telephony (MVP-410/810)
..........................................97, 98
Silence Compression (call progress)
field ...........................................417
Silence Compression (SMTP logs)
analog........................................284
T1/E1 ........................................183
Silence Compression field
analog........................................233
T1/E1 ........................................146
Silence Detection (FXO) field ......250
Silence Timer (FXO) field ............250
simulated voip network
Parameters
analog........................................ 290
T1/E1 ........................................ 189
Set T1/E1/ISDN Parameters ......... 152
Set Telephony Interface Parameters
.................................................. 240
Set Voice/FAX Parameters
analog........................................ 227
T1/E1 ........................................ 140
setting IP parameters
use in startup ...............................72
Single-Port Protocol, general
description
analog........................................ 220
T1/E1 ........................................ 134
setting password............................ 468
web browser GUI...................... 472
setting RTP Parameters................. 433
setting user defaults ...................... 466
setup, saving
analog........................................ 305
T1/E1 ........................................ 204
user ........................................... 466
setup, saving user values............... 466
Signal (type, E&M) field .............. 256
signaling cadences
analog........................................ 272
T1/E1 ........................................ 170
signaling parameters (analog
telephony) ................................. 240
Signaling Port (proxy server) field
E1.............................................. 370
T1.............................................. 317
Signaling Port field
analog..........................................27
BRI..............................................35
E1................................................19
T1................................................12
SIP
compatibility
analog models .........................27
BRI models .............................35
E1 models................................19
T1 models................................12
SIP Fields (Outbound Phonebook)
E1..............................................379
T1..............................................327
SIP incompatibility with H.450
Supplementary Services
analog..................................27, 290
BRI..............................................35
E1................................................19
T1................................................12
T1/E1 ........................................189
SIP Port Number field
E1.............................................. 366
T1.............................................. 313
signaling tones
E1..............................................379
T1..............................................327
SIP port number, standard
analog........................................ 272
T1/E1 ........................................ 170
E1..............................................379
T1..............................................327
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SIP Proxy Parameters
T1/E1 ........................................178
SMTP port, standard
analog.......................................282
T1/E1 .......................................180
SMTP prerequisites
analog models ...........................213
quick............................................50
T1/E1 models............................127
SMTP, enabling
analog........................................280
T1/E1 ........................................178
SNMP (log reporting type) button
analog........................................288
T1/E1 ........................................187
SNMP agent program
E1.............................................. 370
T1.............................................. 317
SIP URL field
E1.............................................. 379
T1.............................................. 327
SMTP
quick setup.................................. 61
SMTP (log reporting type) button
analog........................................ 288
T1/E1 ........................................ 187
SMTP logs by email, illustration
analog........................................ 285
T1/E1 ........................................ 184
SMTP Parameters definitions
analog........................................ 282
T1/E1 ........................................ 179
SMTP Parameters fields (analog)
Enable SMTP............................ 281
Login Name .............................. 281
Mail Server IP Address............. 282
Mail Type ................................. 282
Number of Days........................ 282
Number of Records................... 282
Password................................... 282
Port Number ............................. 282
Recipient Address..................... 282
Reply-To Address..................... 282
Requires Authentication ........... 281
Subject ...................................... 282
SMTP Parameters fields (T1/E1)
Enable SMTP............................ 179
Login Name .............................. 179
Mail Server IP Address............. 179
Mail Type ................................. 181
Number of Days........................ 181
Number of Records................... 181
Password................................... 179
Port Number ............................. 179
Recipient Address..................... 181
Reply-To Address..................... 181
Requires Authentication ........... 179
Subject ...................................... 181
SMTP parameters, accessing
analog........................................208
T1/E1 ........................................123
SNMP agent, enabling
analog........................................269
T1/E1 ........................................167
SNMP Parameter Definitions
T1/E1 ........................................169
SNMP Parameter fields (analog)
Address .....................................271
Community Name (2) ...............271
Community Name 1..................271
Enable SNMP Agent.................271
Permissions (1)..........................271
Permissions (2)..........................271
Port Number..............................271
SNMP Parameter fields (T1/E1)
Address .....................................169
Community Name (2) ...............169
munity Name 1..................169
Com
Enable SNMP Agent.................169
Permissions (1)..........................169
Permissions (2)..........................169
Port Number..............................169
SNMP Parameters, setting
analog........................................269
T1/E1 ........................................167
software
control .........................................58
uninstalling (detailed) ...............118
updates (analog)........................208
updates (T1/E1).........................123
software (MultiVOIP)
analog........................................ 280
T1/E1 ........................................ 178
SMTP parameters,setting
analog........................................ 280
uninstalling................................473
549
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MultiVOIP User Guide
software configuration
ST interface (ISDN-BRI)
summary ................................... 111
software installation
description.................................507
Start Date, Time (SMTP logs) field
analog........................................283
T1/E1 ........................................182
Start Date,Time (statistics, logs) field
..................................................422
Start Modes (DID-DPO) field259, 260
Start Now command (Link
detailed...................................... 111
quick ........................................... 58
software loading............................ 111
software loading, quick................... 58
software version numbers............. 113
software, MultiVOIP (analog)
screen-surfing in ....................... 219
software, MultiVOIP (T1/E1)
Management) button .................430
starter configuration
moving around in...................... 133
screen-surfing in ....................... 133
software, MultiVOIP(analog)
moving around in...................... 219
software, on command PC.............. 58
Solving Common Connection
inbound phonebook.....................67
outbound phonebook...................63
phone/IP......................................59
startup tasks.....................................46
State field (ISDN BRI Statistics,
Layer 1 Interface)......................444
Options (caller...............................423
Status (call progress) field.............419
Status (statistics, logs) field ..........422
Status field (ISDN BRI Statistics,
Layer 1 Interface)......................444
Status field (ISDN BRI Statistics,
SPID0) ......................................445
Status field (ISDN BRI Statistics,
SPID1) ......................................446
Status field (Registered Gateway
Details)......................................448
Status Freeze Signalling Active (E1
stats) field..................................441
Status Freeze Signalling Active (T1
stats) field..................................437
Subject (email logs) field
Problems
analog........................................ 218
T1/E1 ........................................ 132
sound quality, improving
analog........................................ 233
T1/E1 ........................................ 146
specialized codes, in dialing ........... 71
specifications
E1 models ................................... 42
T1 models ................................... 41
SPID0 field (ISDN BRI Statistics,
Switch Information).................. 445
SPID1 field (ISDN BRI Statistics,
Switch Information).................. 446
SPP Fields (Outbound Phonebook)
E1.............................................. 380
T1.............................................. 328
SPP Fields (Phonebook Configuration
screen)
analog........................................282
T1/E1 ........................................181
supervisory signaling (analog)......241
supervisory signaling parameters
(analog telephony).....................240
supervisory signaling types
E1.............................................. 372
T1.............................................. 319
SPP, general description
analog.......................................... 27
BRI ............................................. 35
E1................................................ 19
T1................................................ 12
SPP, strengths & compatibilities of
analog.......................................... 27
BRI ............................................. 35
E1................................................ 19
T1................................................ 12
MVP130....................................108
MVP130FXS.............................109
MVP210............................106, 107
MVP-410/810 .......................97, 98
Supplementary (Telephony) Services
ANALOG....................................30
BRI..............................................38
E1................................................23
T1................................................16
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Supplementary Services (analog)
Alerting Party............ 297, 298, 299
Call Hold................................... 291
Call Hold Enable....................... 294
Call Name Identification........... 291
Call Transfer............................. 291
Call Transfer Enable................. 293
Call Waiting.............................. 291
Call Waiting Enable.................. 294
Caller Name Identification Enable
.............................................. 295
Calling Party............................. 296
Enable Call Hold....................... 294
Enable Call Transfer................. 293
Enable Call Waiting.................. 294
Enable Caller Name Identification
.............................................. 295
Hold Sequence.......................... 294
Retrieve Sequence .................... 294
Select Channel .......................... 293
Transfer Sequence .................... 293
Supplementary Services (T1/E1)
Alerting Party............ 196, 197, 198
Call Hold................................... 190
Call Hold Enable....................... 193
Call Name Identification........... 191
Call Transfer............................. 190
Call Transfer Enable................. 192
Call Waiting.............................. 190
Call Waiting Enable.................. 193
Caller Name Identification Enable
.............................................. 194
Calling Party............................. 195
Enable Call Hold....................... 193
Enable Call Transfer................. 192
Enable Call Waiting.................. 193
Enable Caller Name Identification
.............................................. 194
Hold Sequence.......................... 193
Retrieve Sequence .................... 193
Select Channel .......................... 192
Transfer Sequence .................... 192
Supplementary Services Info
Supplementary Services Parameter
buttons (T1/E1)
Copy Channel............................199
Default ......................................199
Supplementary Services Parameter
Definitions
analog293, 294, 295, 296, 297, 298,
299, 300
T1/E1 192, 193, 194, 195, 196, 197,
198, 199
Supplementary Services Parameter
fields (analog)
Call Transfer Enable .................293
Call Waiting Enable..................294
Hold Sequence ..........................294
Retrieve Sequence.....................294
Transfer Sequence.....................293
Supplementary Services Parameter
fields (analog)
Select Channel ..........................293
Supplementary Services Parameter
fields (analog)
Call Hold Enable.......................294
Supplementary Services Parameter
fields (analog)
Call Name Identification Enable295
Supplementary Services Parameter
fields (analog)
Calling Party .............................296
Supplementary Services Parameter
fields (analog)
Allowed Name Types................296
Supplementary Services Parameter
fields (analog)
Alerting Party............................297
Supplementary Services Parameter
fields (analog)
Allowed Name Types................297
Supplementary Services Parameter
fields (analog)
Busy Party.................................298
Supplementary Services Parameter
fields (analog)
logs for...................................... 424
Supplementary Services Parameter
buttons (analog)
Allowed Name Types................298
Supplementary Services Parameter
fields (analog)
Copy Channel ........................... 300
Default ...................................... 300
Connected Party........................299
551
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MultiVOIP User Guide
Supplementary Services Parameter
fields (analog)
Allowed Name Types ............... 299
Supplementary Services Parameter
fields (analog)
Caller ID ................................... 300
Supplementary Services Parameter
fields (T1/E1)
Call Transfer Enable................. 192
Call Waiting Enable.................. 193
Hold Sequence.......................... 193
Retrieve Sequence .................... 193
Transfer Sequence .................... 192
Supplementary Services Parameter
fields (T1/E1)
Caller ID....................................199
Supplementary Services Parameters
screen, accessing
analog........................................290
T1/E1 ........................................189
Supplementary Services parameters,
setting
analog........................................290
T1/E1 ........................................189
Supplementary Services, incompatible
with SIP
analog..................................27, 290
BRI..............................................35
E1................................................19
T1................................................12
T1/E1 ........................................189
Switch Information fields (ISDN BRI
Statistics)...................................445
switch types (phone) and ISDN-PRI
..................................................166
SysLog client
Select Channel .......................... 192
Supplementary Services Parameter
fields (T1/E1)
Call Hold Enable....................... 193
Supplementary Services Parameter
fields (T1/E1)
Call Name Identification Enable194
Supplementary Services Parameter
fields (T1/E1)
Calling Party............................. 195
Supplementary Services Parameter
fields (T1/E1)
Allowed Name Types ............... 195
Supplementary Services Parameter
fields (T1/E1)
Alerting Party............................ 196
Supplementary Services Parameter
fields (T1/E1)
Allowed Name Types ............... 196
Supplementary Services Parameter
fields (T1/E1)
Busy Party................................. 197
Supplementary Services Parameter
fields (T1/E1)
ANALOG....................................29
BRI..............................................37
E1................................................22
T1................................................15
SysLog client programs
availability.................................491
features & presentation types....493
SysLog functionality
ANALOG....................................29
BRI..............................................37
E1................................................22
T1................................................15
SysLog server
ANALOG....................................29
BRI..............................................37
E1................................................22
T1................................................15
SysLog Server Enable field
Allowed Name Types ............... 197
Supplementary Services Parameter
fields (T1/E1)
analog........................................288
T1/E1 ........................................187
SysLog Server function
Connected Party........................ 198
Supplementary Services Parameter
fields (T1/E1)
Allowed Name Types ............... 198
Supplementary Services Parameter
fields (T1/E1)
as added feature.........................491
capabilities of............................493
enabling.....................................492
location of .................................491
SysLog Server IP Address field
analog........................................288
552
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Index
T1/E1 ........................................ 187
SysLog Server, enabling
analog........................................ 287
T1/E1 ........................................ 186
System Information screen
for op & maint .......................... 411
System Information screen, accessing
analog........................................ 302
T1/E1 ........................................ 201
System Information update interval,
setting
T1/E1 connector pinout.................504
T1/E1 Statistics function...........435
T1/E1/ISDN field
E1..............................................161
T1..............................................155
T1/E1/ISDN Parameters screen,
accessing ...................................152
T1/E1/ISDN parameters, setting...152
T1/ISDN Parameter definitions.....158
T1/ISDN Parameter fields
Country .....................................158
Enable ISDN-PRI......................158
Operator ....................................158
Terminal Network.....................158
table-top voip models......................85
TCP/UDP compared
analog........................................ 302
for op & maint .......................... 413
T1/E1 ........................................ 201
T
T1 model descriptions..................... 11
T1 Parameter definitions...... 155, 156,
157, 159
Clocking.................................... 159
Line Build-Out.......................... 159
Line Coding .............................. 159
PCM Law.................................. 159
Pulse Shape Level..................... 159
Yellow Alarm Format............... 159
T1 Parameter fields
CAS Protocol........................ 156
CRC Check............................... 155
Detect Flash Hook .................... 157
Frame Format............................ 155
Long-Haul Mode ...................... 155
T1/E1/ISDN.............................. 155
T1 Parameters screen.................... 154
T1 Statistics field definitions 438, 439
T1 Statistics fields
E1..............................................379
IP Statistics context...................426
T1..............................................327
technical configuration
startup..........................................59
technical configuration (analog)
prerequisites to..........................210
summary....................................207
echnical configuration (T1/E1)
t
prerequisites to..........................124
summary....................................122
technical configuration procedure
detailed, analog .........................215
detailed, T1/E1..........................129
summary, analog.......................214
summary, T1/E1........................128
TEI Assignment fields (ISDN BRI
Statistics, Switch Information)..445
TEI n Assignment
Bipolar Violation ...................... 439
Frame Search Restart Flag........ 438
Line Loopback Activation Signal
.............................................. 438
Loss of MultiFrame Alignment 438
Pulse Density Violation ............ 438
Receive Slip.............................. 439
Transmit Data Underrun........... 438
Transmit Line Open.................. 438
Transmit Slip ............................ 438
Transmit Slip Negative............. 438
Yellow Alarm ........................... 438
T1 telephony parameters............... 125
ISDN-BRI .................................265
TEIn fields (ISDN BRI Statistics,
Switch Information)..................445
telco authorities and ISDN............166
telecom safety warnings.............85
telephony interface parameters,
setting........................................240
telephony interfaces
uses of ... 97, 98, 106, 107, 108, 109
telephony interfaces, BRI................54
telephony signaling cadences
analog........................................272
T1/E1 ........................................170
553
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Index
MultiVOIP User Guide
telephony signaling tones
Total Digits (outbound) field
analog........................................ 272
T1/E1 ........................................ 170
telephony startup configuration ...... 59
telephony toning schemes
E1..............................................377
T1..............................................325
touch tone frequencies ..................250
trace on/off (logging)
analog........................................ 277
T1/E1 ........................................ 175
temperature
analog........................................289
T1/E1 ........................................188
Transfer Sequence
operating ..................................... 91
terminal mode (ISDN-BRI) & D-
channel support
MVP-410ST/810ST.................. 101
Terminal Network field
E1/ISDN ................................... 164
T1/ISDN ................................... 158
terminal/network settings, voip and
PBX
E1/ISDN ................................... 164
ISDN-BRI................................. 264
T1/ISDN ................................... 158
timeout interval
analog........................................293
T1/E1 ........................................192
Transfer Sequence (analog)...........291
Transfer Sequence (T1/E1) ...........190
Transmit Data Overflow (E1 stats)
field ...........................................441
Transmit Data Overflow (T1 stats)
field ...........................................437
Transmit Data Underrun (E1 stats)
field ...........................................442
Transmit Data Underrun (T1 stats)
field ...........................................438
Transmit Line Open (E1 stats) field
..................................................442
Transmit Line Open (T1 stats) field
..................................................438
Transmit Line Short (E1 stats) field
..................................................441
Transmit Line Short (T1 stats) field
..................................................437
Transmit Slip (E1 stats) field ........442
Transmit Slip (T1 stats) field ........438
Transmit Slip Negative (E1 stats) field
..................................................442
Transmit Slip Negative (T1 stats) field
..................................................438
Transmit Slip Positive (E1 stats) field
..................................................441
Transmit Slip Positive (T1 stats) field
..................................................437
Transmitted (RTCP Packets, IP Stats)
field ...........................................428
Transmitted (RTP Packets, IP Stats)
field ...........................................428
Transmitted (TCP Packets, IP Stats)
field ...........................................427
Transmitted (Total Packets, IP Stats)
field ...........................................426
Transmitted (UDP Packets, IP Stats)
field ...........................................427
voips under H.323 gatekeeper.. 314,
367
voips under SIP proxy server... 318,
371
Timer Interval between Pings (Link
Management) field.................... 430
tips, phonebook............................... 70
To (gateway, statistics, logs) field 422
toll-call savings
E1.............................................. 356
T1.............................................. 308
toll-free access (T1)
to remote PSTN .......................... 11
within voip network.................... 11
toll-free access (within voip network)
E1................................................ 18
T1................................................ 11
Tone Detection (FXO disconnect
criteria) field ............................. 249
Tone Pair (custom) field
analog........................................ 278
T1/E1 ........................................ 176
tone pairs, custom
T1/E1 ........................................ 175
tones, signaling
analog........................................ 272
T1/E1 ........................................ 170
554
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Index
Transport Protocol (SIP) field
unobtainable tones
E1.............................................. 379
T1.............................................. 327
trap manager parameters (SNMP)
T1/E1 ........................................ 169
triggering log report email
analog ...................................... 282
T1/E1....................................... 181
troubleshooting ............................... 82
Troubleshooting Resolutions for
MultiVOIPs ................................ 10
Turn Off Logs field
analog........................................ 288
T1/E1 ........................................ 187
Tx Packets field (ISDN BRI Statistics,
D-Channel Information) ........... 445
Type (E&M type) field ................. 256
Type (of tone) field
analog................................175, 277
unpacking
MVP130......................................89
MVP210x....................................88
MVP2410....................................86
MVP3010....................................86
MVP410x....................................87
MVP810x....................................87
Up Time
System Info...............................412
System Info (ANALOG/BRI)...304
System Info (T1/E1)..................203
update interval (logging)
analog........................................287
T1/E1 ........................................186
updated firmware, obtaining .........452
upgrade
E1................................................18
T1................................................11
upgrade card (analog, 4-to-8 channel)
installation.................................511
upgrade card (T1/E1) installation..499
Upgrade Software option description
MultiVOIP program menu........450
upgrade, firmware.........................456
uploads vs. downloads (FTP)........476
Use Fast Start (Q.931) field
analog........................................ 275
T1/E1 ........................................ 172
Type-of-Service IP header field &
DiffServ
analog........................................ 223
T1/E1 ........................................ 137
U
U interface (ISDN-BRI)
description ................................ 507
UDP multiplexed (H.323 Annex E)
field
E1.............................................. 369
T1.............................................. 316
UDP/TCP compared
E1.............................................. 379
IP Statistics context................... 426
T1.............................................. 327
unconditional forwarding
E1.............................................. 385
T1.............................................. 334
Uninstall (program menu option) ,
command .................................. 473
Uninstall option description
(MultiVOIP program menu) ..... 450
uninstalling MultiVOIP software. 118,
473
E1..............................................366
T1..............................................313
Use Gatekeeper (Outbound
Phonebook) field
E1..............................................378
T1..............................................326
Use Proxy (SIP) field
E1..............................................379
T1..............................................327
Use Registrar field (Outbound
nebook)
Pho
E1..............................................380
T1..............................................328
Use SIP Proxy field
E1..............................................370
T1..............................................317
user default configuration, creating
analog........................................306
T1/E1 ........................................205
user defaults, downloading ...........466
unobtainable tone, custom
analog........................................ 278
T1/E1 ................................ 175, 176
555
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MultiVOIP User Guide
user defaults, setting ..................... 466
user name
T1/E1 ................................150, 151
Voice/FAX Parameter Definitions
analog........ 230, 231, 232, 233, 237
T1/E1 ........ 143, 144, 145, 146, 149
Voice/FAX Parameter fields (analog)
Automatic Disconnection..........239
Call Duration.............................239
Consecutive Packets Lost..........239
Copy Channel............................230
Default ......................................230
DTMF Gain...............................230
DTMF Gain (High Tones) ........230
DTMF Gain (Low Tones).........230
DTMF In/Out of Band ..............230
Duration (DTMF)......................230
Dynamic Jitter Buffer................237
Echo Cancellation .....................233
Fax Enable ................................231
Fax Volume...............................231
Forward Error Correction..........233
Input Gain .................................230
Jitter Value................................239
Jitter Value (Fax) ......................231
Max Baud Rate (Fax)................231
Maximum Jitter Value ..............238
Minimum Jitter Value...............237
Mode (Fax)................................231
Network Disconnection.............239
Optimization Factor ..................238
Output Gain...............................230
Select Channel ..........................230
Silence Compression.................233
Voice Gain ................................230
Voice/FAX Parameter fields
Windows GUI........................... 468
User Name (proxy server) field
E1.............................................. 370
T1.............................................. 317
user values (software), saving....... 466
V
variations in PBX characteristics
E1.............................................. 405
T1.............................................. 354
version numbers.............................. 10
version numbers (software) .......... 113
version, firmware.......................... 456
Voice Coder (call progress) field.. 416
Voice coder (statistics, logs) field. 423
VOICE connector
MVP130FXS ............................ 108
voice delay
analog................................ 237, 238
T1/E1 ................................ 149, 150
Voice Gain field
analog........................................ 230
T1/E1 ........................................ 143
voice packets (analog)
recovering lost/corrupted .......... 233
voice packets (T1/E1)
recovering lost/corrupted .......... 146
voice packets, consecutive lost
analog........................................ 239
T1/E1 ........................................ 151
voice packets, delayed
analog................................ 237, 238
T1/E1 ................................ 149, 150
voice packets, re-assembling
(analog/BRI)
AutoCall/Offhook Alert ....234, 235
AutoCall/Offhook Alert fields .234,
235
analog........................................ 231
voice packets, re-assembly
Generate Local Dial Tone.........235
Offhook Alert Timer .................236
Phone Number (Auto Call/Offhook
Alert).....................................236
Voice/FAX Parameter fields (T1/E1)
AutoCall/Offhook Alert ............147
AutoCall/Offhook Alert fields ..147
Automatic Disconnection..........151
Call Duration.............................151
Consecutive Packets Lost..........151
T1/E1 ........................................ 144
voice quality, improving
analog........................................ 233
T1/E1 ........................................ 146
voice quality, versus delay
analog........................................ 238
T1/E1 ........................................ 150
Voice/FAX connector pinout........ 504
Voice/FAX Parameter definitions
analog................................ 238, 239
556
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Index
Copy Channel ........................... 143
Default ...................................... 143
DTMF Gain .............................. 143
DTMF Gain (High Tones)........ 143
DTMF Gain (Low Tones)......... 143
DTMF In/Out of Band.............. 143
Duration (DTMF) ..................... 143
Dynamic Jitter Buffer ............... 149
Echo Cancellation..................... 146
Fax Enable ................................ 144
Fax Volume .............................. 144
Forward Error Correction ......... 146
Input Gain................................. 143
Jitter Value................................ 151
Jitter Value (Fax) ...................... 144
Max Baud Rate ......................... 144
Maximum Jitter Value .............. 150
Minimum Jitter Value............... 149
Mode (Fax) ............................... 144
Network Disconnection ............ 151
Offhook Alert Timer................. 148
Optimization Factor.................. 150
Output Gain .............................. 143
Phone Number .......................... 148
Phone Number (AutoCall/Offhook
Alert)..................................... 148
Select Channel .......................... 143
Silence Compression ................ 146
Voice Gain................................ 143
Voice/FAX Parameters screen,
types used....................................70
voip email account
analog........................................281
T1/E1 ........................................180
voip management, remote
analog........................................269
T1/E1 ........................................167
Voip Media PHB field
analog........................................222
T1/E1 ........................................136
voip network, lab/simulated
use in startup ...............................72
voip software
host PC..................................40, 51
voip software (analog)
host PC......................................208
voip software (T1/E1)
host PC......................................123
voip system example, conceptual (E1)
calls to remote PSTN ................358
foreign calls, national rates .......360
voip site to voip site ..................357
voip system example, digital &
analog, with phonebook details
E1..............................................394
T1..............................................342
voip system example, digital only,
with phonebook details
E1..............................................387
T1..............................................336
voip(E1)
accessing
analog........................................ 227
T1/E1 ........................................ 140
Voice/FAX parameters, setting
analog........................................ 227
T1/E1 ........................................ 140
Voip Caller ID Case #1 –telco
basic functions of ........................19
voip(T1)
basic functions of ........................12
voltage, operating
analog models .............................43
E1 models....................................42
standard CID enters voip system
.................................................. 252
Voip Caller ID Case #2 – H.323 voip
system, no telco CID................. 252
Voip Caller ID Case #3 –SPP....... 253
Voip Caller ID Case #4 – Remote
FXS call on H.323 voip system 253
Voip Caller ID Case #5 –DID channel
in H.323 voip system ................ 254
voip dialing digits
W
warnings, safety ..............................85
web browser GUI and logs
analog........................................287
T1/E1 ........................................186
web browser GUI, enabling
analog..................................59, 226
T1/E1 ........................................139
web browser interface
non-PSTN type ........................... 71
557
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Index
MultiVOIP User Guide
browser version requirement ... 486,
489
T1..............................................326
well-known port, Q.931 params,
H.323
E1..............................................366
T1..............................................313
well-known port, SIP
E1..............................................379
T1..............................................327
well-known port, SNMP
analog........................................271
T1/E1 ........................................169
Windows GUI vs. web GUI
BRI..............................................37
wink signaling (DID-DPO)...........260
wink signaling (E&M) ..................256
Wink Timer (DID-DPO) field.......260
Wink Timer (E&M) field..............256
worksheet
general ...................................... 486
Java requirement....................... 486
prerequisite local assigning of IP
address .................................. 487
video useability......................... 486
web GUI
Java and .................................... 488
remote control/configuration and
.............................................. 488
web GUI vs. local Windows GUI
comparison................................ 487
web GUI vs. Windows GUI
BRI ............................................. 36
web GUI, logging and................... 487
weight
analog models............................. 43
E1 models ................................... 42
T1 models ................................... 41
weight loading
phonebook.............................76, 77
X
in rack ......................................... 91
weight of unit
XMT (channel) LED
analog models...........................32
analog, MVP130........................33
BRI models ................................39
XSG LED
lifting precaution......................... 85
personnel requirement................. 85
Well Known Ports......................... 509
well-known port number, SMTP
analog ...................................... 282
T1/E1....................................... 180
well-known port, gatekeeper
analog models...........................32
analog, MVP130 .........................33
Y
registration
Yellow Alarm (E1 stats) field .......441
Yellow Alarm (T1 stats) field .......438
Yellow Alarm Format field (T1)...159
E1.............................................. 366
T1.............................................. 313
well-known port, H.323 params
E1.............................................. 378
558
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S000249K
559
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