Voice / Fax over IP Networks
User Guide for Voice/IP Gateways
Digital Models (T1, E1, ISDN-PRI):
MVP-2410/3010
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CONTENTS
CHAPTER 1: OVERVIEW.......................................................................................6
ABOUT THIS MANUAL...............................................................................................7
INTRODUCTION TO TI MULTIVOIPS (MVP2410 & MVP24-48) ...............................9
T1 Front Panel LEDs..........................................................................................15
INTRODUCTION TO EI MULTIVOIPS (MVP3010 & MVP30-60) ............................17
E1 Front Panel LEDs .........................................................................................23
E1 LED Descriptions..........................................................................................24
SPECIFICATIONS ......................................................................................................25
Specs for Digital T1 MultiVOIP Units................................................................25
Specs for Digital E1 MultiVOIP Units................................................................26
INSTALLATION AT A GLANCE ..................................................................................27
RELATED DOCUMENTATION....................................................................................27
CHAPTER 2: QUICK START INSTRUCTIONS.................................................28
CHAPTER 3: MECHANICAL INSTALLATION AND CABLING...................30
INTRODUCTION........................................................................................................31
SAFETY WARNINGS .................................................................................................31
Lithium Battery Caution .....................................................................................31
Safety Warnings Telecom....................................................................................31
UNPACKING YOUR MULTIVOIP..............................................................................32
Unpacking the MVP2410/3010...........................................................................32
RACK MOUNTING INSTRUCTIONS ............................................................................33
Safety Recommendations for Rack Installations.................................................34
19-Inch Rack Enclosure Mounting Procedure....................................................35
CABLING .................................................................................................................36
Cabling Procedure..............................................................................................36
CHAPTER 4: SOFTWARE INSTALLATION .....................................................38
INTRODUCTION........................................................................................................39
LOADING MULTIVOIP SOFTWARE ONTO THE PC....................................................39
UN-INSTALLING THE MULTIVOIP CONFIGURATION SOFTWARE.............................46
CHAPTER 5: TECHNICAL CONFIGURATION................................................49
CONFIGURING THE MULTIVOIP..............................................................................50
LOCAL CONFIGURATION..........................................................................................53
Pre-Requisites.....................................................................................................53
IP Parameters................................................................................................................54
T1 Telephony Parameters (for MVP2410) ...................................................................55
E1 Telephony Parameters (for MVP3010) ...................................................................56
SMTP Parameters (for email call log reporting)...........................................................57
Config Info CheckList..................................................................................................58
Local Configuration Procedure (Summary) .......................................................59
Local Configuration Procedure (Detailed).........................................................60
Modem Relay ......................................................................................................87
3
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Contents
MultiVOIP User Guide
CHAPTER 6: T1 PHONEBOOK CONFIGURATION ......................................170
T1 VERSUS E1 TELEPHONY ENVIRONMENTS.........................................................171
CONFIGURING T1 (NAM) TELEPHONY MULTIVOIP PHONEBOOKS......................171
T1 PHONEBOOK EXAMPLES...................................................................................189
3 Sites, All-T1 Example.....................................................................................189
Configuring Mixed Digital/Analog VOIP Systems ...........................................195
Call Completion Summaries .............................................................................204
Variations in PBX Characteristics....................................................................207
CHAPTER 7: E1 PHONEBOOK CONFIGURATION ......................................208
E1 VERSUS T1 TELEPHONY ENVIRONMENTS.........................................................209
E1-STANDARD INBOUND AND OUTBOUND MULTIVOIP PHONEBOOKS.................209
Free Calls: One VOIP Site to Another.............................................................210
Local Rate Calls: Within Local Calling Area of Remote VOIP.......................211
National Rate Calls: Within Nation of Remote VOIP Site...............................213
Inbound versus Outbound Phonebooks.............................................................214
PHONEBOOK CONFIGURATION PROCEDURE...........................................................218
E1 PHONEBOOK EXAMPLES...................................................................................231
3 Sites, All-E1 Example ....................................................................................231
Configuring Digital & Analog VOIPs in Same System.....................................238
Call Completion Summaries.......................................................................................246
Variations in PBX Characteristics....................................................................249
International Telephony Numbering Plan Resources.......................................250
CHAPTER 8: OPERATION AND MAINTENANCE ........................................252
OPERATION AND MAINTENANCE ...........................................................................253
System Information screen................................................................................253
Statistics Screens ..............................................................................................256
About Call Progress..........................................................................................256
About Logs........................................................................................................264
About IP Statistics.............................................................................................271
About Link Management...................................................................................276
About Registered Gateway Details ...................................................................287
About Alternate Server Statistics ......................................................................290
About Packetization Time .................................................................................294
MULTIVOIP PROGRAM MENU ITEMS .....................................................................297
Configuration Option........................................................................................299
Configuration Port Setup..................................................................................299
Date and Time Setup.........................................................................................300
Obtaining Updated Firmware...........................................................................300
Implementing a Software Upgrade ...................................................................304
Identifying Current Firmware Version .......................................................................304
Downloading Firmware..............................................................................................305
Downloading CAS Protocol .......................................................................................308
Downloading Factory Defaults...................................................................................310
Setting and Downloading User Defaults ..........................................................313
Setting a Password (Windows GUI) .................................................................316
Setting a Password (Web Browser GUI) ..........................................................320
4
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MultiVOIP User Guide
ContentsVOIP
Un-Installing the MultiVOIP Software .............................................................321
Upgrading Software..........................................................................................323
FTP SERVER FILE TRANSFERS (“DOWNLOADS”)...................................................324
WEB BROWSER INTERFACE ...................................................................................334
SYSLOG SERVER FUNCTIONS ................................................................................340
CHAPTER 9 WARRANTY, SERVICE, AND TECH SUPPORT.....................343
LIMITED WARRANTY.............................................................................................344
REPAIR PROCEDURES FOR U.S. AND CANADIAN CUSTOMERS ...............................344
TECHNICAL SUPPORT ............................................................................................346
Contacting Technical Support ..........................................................................346
CHAPTER 10: REGULATORY INFORMATION ............................................347
EMC, Safety, and R&TTE Directive Compliance.............................................348
FCC DECLARATION...............................................................................................348
Industry Canada ...............................................................................................349
FCC Part 68 Telecom.......................................................................................349
Canadian Limitations Notice............................................................................350
WEEE Statement...............................................................................................351
APPENDIX A: CABLE PINOUTS......................................................................352
APPENDIX A: CABLE PINOUTS..............................................................................353
Command Cable ...............................................................................................353
Ethernet Connector...........................................................................................353
T1/E1 Connector...............................................................................................354
Voice/Fax Channel Connectors ........................................................................354
ISDN BRI RJ-45 Pinout Information ................................................................356
ISDN Interfaces: “ST” and “U” .....................................................................357
APPENDIX B: TCP/UDP PORT ASSIGNMENTS............................................358
WELL KNOWN PORT NUMBERS.............................................................................359
PORT NUMBER ASSIGNMENT LIST.........................................................................359
APPENDIX C: INSTALLATION INSTRUCTIONS FOR MVP428 UPGRADE
CARD.......................................................................................................................360
INSTALLATION INSTRUCTIONS FOR MVP428 UPGRADE CARD..............................361
INDEX .....................................................................................................................366
5
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Chapter 1: Overview
6
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MultiVOIP User Guide
Overview
About This Manual
This manual is about Voice-over-IP products made by Multi-Tech
Systems, Inc. It describes three analog MultiVOIP units,
models MVP810, MVP410, and MVP210.
These MultiVOIP units can inter-operate with other contemporary
analog MultiVOIP units (MVP130 & MVP130FXS), with contemporary
BRI MultiVOIP units (MVP410ST & MVP810ST), with contemporary
digital T1/E1/ISDN-PRI MultiVOIP units (MVP2410 and MVP3010),
and with the earlier generation of MultiVOIP products (MVP200,
MVP400, MVP800, MVP120, etc.)
The table below (on next page) describes the vital characteristics of the
various models described in this manual.
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. 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).
7
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Overview
MultiVOIP User Guide
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.
8
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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 1-1. 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.
9
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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.
10
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MultiVOIP User Guide
Overview
RADIUS Support. Inter-operation with a RADIUS server allows for
call accounting (especially for billing) on a voip system. The MultiVOIP
supports inter-operation with RADIUS servers for the RADIUS
accounting function (but not the RADIUS authentication function).
STUN Support. The STUN protocol (Simple Traversal of UDP through
NATs (Network Address Translation)) assists with the packet routing
functions of devices behind NAT firewalls or routers. The MultiVOIP
supports inter-operation with STUN servers and NATs (SIP based
environment only).
Gatekeeper. T1 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, 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.
11
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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).
12
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MultiVOIP User Guide
Overview
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.
13
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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.
See www.kiwisyslog.com. SysLog client programs 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 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.”
14
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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.
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 1-2: 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.
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MultiVOIP User Guide
MVP2410 Front Panel LED Definitions
LED NAME
Power
DESCRIPTION
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
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MultiVOIP User Guide
Overview
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 1-3. 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.
17
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Overview
MultiVOIP User Guide
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.
RADIUS Support. Inter-operation with a RADIUS server allows for
call accounting (especially for billing) on a voip system. The MultiVOIP
18
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MultiVOIP User Guide
Overview
supports inter-operation with RADIUS servers for the RADIUS
accounting function (but not the RADIUS authentication function).
STUN Support. The STUN protocol (Simple Traversal of UDP through
NATs (Network Address Translation)) assists with the packet routing
functions of devices behind NAT firewalls or routers. The MultiVOIP
supports inter-operation with STUN servers and NATs (SIP based
environment only).
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.
19
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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).
20
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MultiVOIP User Guide
Overview
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.
See www.kiwisyslog.com. SysLog client programs 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 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.”
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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 1-4: 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.
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Overview
MultiVOIP User Guide
E1 LED Descriptions
MVP3010 Front Panel LED Definitions
DESCRIPTION
LED NAME
Power
Indicates presence of power.
Boot
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.
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
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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
Mains
Frequencies
100-240 VAC
1.2 - 0.6 A
50/60 Hz
100-240 VAC
1.2 - 0.6 A
50/60 Hz
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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Overview
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
Mains
Frequencies
100-240 VAC
1.2 - 0.6 A
50/60 Hz
100-240 VAC
1.2 - 0.6 A
50/60 Hz
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
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:
www.adobe.com/prodindex/acrobat/readstep.html
This MultiVOIP User Guide is also available on Multi-Tech’s Web site
at:
http://www.multitech.com
Viewing and printing a user guide from the Web also requires that you
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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Chapter 2: Quick Start Instructions
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MultiVOIP User Guide
Quick Start Instructions
The Quick Start Guide is a separate manual with streamlined
instructions to get the MultiVOIP up and running quickly. These start-
up instructions include assistance on setting up the MultiVOIP’s
Inbound and Outbound Phonebooks. These sections of the Quick Start
Guide may be particularly useful for phonebook configuration:
Phonebook Starter Configuration
Phonebook Tips
Phonebook Example (One Common Situation)
The Quick Start Guide 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.
A printed Quick Start Guide is shipped with the MultiVOIP and an
electronic copy is included on the Product CD.
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Mechanical Installation & Cabling
MultiVOIP User Guide
Chapter 3: Mechanical Installation
and Cabling
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MultiVOIP User Guide
Mechanical Installation & Cabling
Introduction
When the MVP2410 or MVP3010 unit is 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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MultiVOIP User Guide
UnpackingYour MultiVOIP
When unpacking your MultiVOIP, check to see that all of the items
shown are included in the box. 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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MultiVOIP User Guide
Mechanical Installation & Cabling
Rack Mounting Instructions
The MultiVOIPs can be mounted in an industry-standard EIA 19-inch
rack enclosure, as shown in Figure 3-2.
Figure 3-2: Rack-Mounting
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MultiVOIP User Guide
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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MultiVOIP User Guide
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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 3-3, and then secure unit to rack rails by the
brackets, as shown in Figure 3-4. 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 3-3: Bracket Attachment for Rack Mounting
Figure 3-4: Attaching MultiVOIP to Rack Rail
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Mechanical Installation & Cabling
MultiVOIP User Guide
Cabling
Cabling Procedure
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 3-5 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 3-5.
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 3-5. 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 3-5.
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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MultiVOIP User Guide
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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 3-6.
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
TRUNK
10 BASET
RS232
DIGITAL VOICE
ETHERNET COMMAND
COMMAND
MODEM
Grounding Screw
Telco POTS Line
Figure 3-6. 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 3-6) 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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Technical Configuration (T1/E1)
MultiVOIP User Guide
Chapter 4: Software Installation
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MultiVOIP User Guide
Mechanical Installation & Cabling
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), 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 Chapter 6 for North American
(T1) telephony standards and in Chapter 7 for European (E1) telephony
standards.
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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Technical Configuration (T1/E1)
MultiVOIP User Guide
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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Technical Configuration (T1/E1)
MultiVOIP User Guide
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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Technical Configuration (T1/E1)
MultiVOIP User Guide
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 chapter of this
manual.
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Technical Configuration (T1/E1)
MultiVOIP User Guide
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.
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
concerning the MultiVOIP software’s “.bin” file. Click Yes.
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Technical Configuration (T1/E1)
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4. A completion screen will appear.
Click Finish.
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Chapter 5:Technical Configuration
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Technical Configuration
MultiVOIP User Guide
Configuring the 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 in Chapter 6. The Quick Start Guide presents additional
information on phonebook setup.
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.
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MultiVOIP User Guide
Technical Configuration
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
ftp://ftp.multitech.com/MultiVoip/.
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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MultiVOIP User Guide
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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Technical Configuration
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. A summary of this configuration
information appears on page 58 (“Config Info CheckList”).
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MultiVOIP User Guide
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.
Ê
IP Network Parameters:
Record for each VOIP Site
in System
#
• IP Address
• IP Mask
• Gateway
• Domain Name Server (DNS) Info
• If SIP protocol is used, determine whether or not
802.1p Packet Prioritization will be used.
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.
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MultiVOIP User Guide
Technical Configuration
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:
Ê
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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Technical Configuration
MultiVOIP User Guide
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:
Ê
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 Ethernet/IP Parameters.
7. Set up web browser GUI (optional).
8. Set Voice/Fax Parameters.
9. Set T1/E1 Parameters.
10. Set ISDN Parameters (if applicable).
11. Set Call Signaling parameters. The choice of H.323, SIP, or SPP is
made in the Outbound Phonebook, but details are configured in the
Call Signaling Parameters screen.
12. Set SNMP Parameters (applicable if MultiVoipManager remote
management software is used).
13. 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).
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17. Set NAT Traversal (STUN) parameters. Optional. Applicable only
under SIP Call Signaling when the UDP transport protocol is used.
18. Set RADIUS parameters. Optional. Used only if system interfaces
with RADIUS server for billing or other accounting functions.
19. Set Baud Rate (of COM port connection to ‘Command’ PC).
20. View System Info screen and set updating interval (optional).
21. Save the MultiVOIP configuration.
22. 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 Ethernet/IP Parameters. This dialog box can be reached by
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.
Accessing “Ethernet/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 Ethernet/IP Parameters fields are described in the tables and text
passages below. Note that both DiffServ parameters (Call Control PHB
and VoIP Media PHB) must be set to zero if you enable Packet
Prioritization (802.1p). Nonzero DiffServ values negate the
prioritization scheme.
Ethernet/IP Parameter Definitions (cont’d)
Field Name
Values
Description
Ethernet Parameters
Packet
Prioritization
(802.1p)
Y/N
Select to activate
prioritization under 802.1p
protocol (described below).
.
Frame Type
802.1p
Type II, SNAP
Must be set to match
network’s frame type.
Default is Type II.
A draft standard of the IEEE about data traffic
prioritization on Ethernet networks. The 802.1p
draft is an extension of the 802.1D bridging
standard. 802.1D determines how prioritization
will operate within a MAC-layer bridge for any
kind of media. The 802.1Q draft for virtual local-
area-networks (VLANs) addresses the issue of
prioritization for Ethernet networks in particular.
802.1p enacts this Quality-of-Service feature
using 3 bits. This 3-bit code allows data switches to
reorder packets based on priority level. The
descriptors for the 8 priority levels are given below.
802.1p PRIORITY LEVELS
LOWEST PRIORITY
1 – Background: Bulk transfers and other
activities permitted on the network,
but should not affect the use of
network by other users and
applications.
2 – Spare: An unused (spare) value of the
user priority.
0 – Best Effort (default): Normal priority for
ordinary LAN traffic.
3 – Excellent Effort: The best effort type of
service that an information services
organization would deliver to its most
important customers.
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Ethernet/IP Parameter Definitions (cont’d)
Field Name Values Description
Ethernet Parameters
802.1p
(continued)
4 – Controlled Load: Important business
applications subject to some form of
“Admission Control”, such as
preplanning of Network requirement,
characterized by bandwidth
reservation per flow.
5 – Video: Traffic characterized by
delay < 100 ms.
6 – Voice: Traffic characterized by
delay < 10 ms.
7 - Network Control: Traffic urgently
needed to maintain and support
network infrastructure.
HIGHEST PRIORITY
Call Control
Priority
0-7, where 0 is
lowest priority
Sets the priority for
signaling packets.
VoIP Media
Priority
0-7, where 0 is
lowest priority
Sets the priority for media
packets.
Others
(Priorities)
0-7, where 0 is
lowest priority
Sets the priority for SMTP,
DNS, DHCP, and other
packet types.
VLAN ID
1 - 4094
The 802.1Q IEEE standard
allows virtual LANs to be
defined within a network.
This field identifies each
virtual LAN by number.
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Ethernet/IP Parameter Definitions (cont’d)
Field Name Values Description
IP Parameter fields
Gateway
Name
alphanumeric
Descriptor of current voip
unit to distinguish it from
other units in system.
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.
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Ethernet/IP Parameter Definitions (cont’d)
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
default = 34
.
Value is used to
prioritize call setup IP
packets.
Voip Media
PHB
0 – 63
default = 46
n
Value is used to
prioritize the RTP/RTCP
audio IP packets.
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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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Ethernet/IP Parameter Definitions (cont’d)
Field Name Values Description
FTP Parameter fields
FTP Server
Enable
Y/N
Default = disabled
See “FTP Server
MultiVOIP unit has an
FTP Server function so
that firmware and other
File Transfers” in important operating
Operation &
Maintenance
chapter.
software files can be
transferred to the voip
via the network.
DNS Parameter fields
Enable DNS
Y/N
Default = disabled
Enables Domain Name
Space/System function
where computer names
are resolved using a
worldwide distributed
database.
Enable SRV
Y/N
Enables ‘service record’
function. Service record
is a category of data in
the Internet Domain
Name System specifying
information on available
servers for a specific
protocol and domain, as
defined in RFC 2782.
Newer internet protocols
like SIP, STUN, H.323,
POP3, and XMPP may
require SRV support
from clients. Client
implementations of older
protocols, like LDAP and
SMTP, may have been
enhanced in some
settings to support SRV.
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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About Service Records
An SRV record holds the following information:
•
•
•
•
•
•
•
•
•
Service: the symbolic name of the desired service.
Protocol: this is usually either TCP or UDP.
Domain name: the domain for which this record is valid.
TTL: standard DNS time to live field.
Class: standard DNS class field (this is always IN).
Priority: the priority of the target host.
Weight: A relative weight for records with the same priority.
Port: the TCP or UDP port on which the service is to be found.
Target: the hostname of the machine providing the service.
An example SRV record might look like this:
_sip._tcp.example.com 86400 IN SRV 0 5 5060 sipserver.example.com.
This expression denotes a server named sipserver.example.com. This server listens on
TCP port 5060 for SIP protocol connections. The priority given here is 0, and the
weight is 5.
TDM 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.
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7. Set up the 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 set
it up. 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. Set browser to allow pop-ups. The MultiVOIP Web GUI makes
extensive use of pop-up windows to access screens and
commands.
I. 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
1-2 (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.
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.
Out of Band RFC 2833, RFC2833 method. Uses an RTP
Mode
SIP Info
mode defined in RFC 2833 to
transmit the DTMF digits.
SIP Info method. Generates dual
tone multi frequency (DTMF) tones
on the telephony call leg. The SIP
INFO message is sent along the
signaling path of the call.
You must set this parameter per the
capabilities of the remote endpoint
with which the voip will
communicate. The RFC2833
method is the more common of the
two methods.
FAX Parameters
Fax Enable
Y/N
Enables or disables fax capability for a
particular channel.
Modem
Relay
Enable
Y/N
When enabled, modem traffic can be
carried on voip system. When disabled,
modem traffic will bypass the voip
system (Modem Bypass mode).
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.
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Voice/Fax Parameter Definitions (cont’d)
Field Name
Valuee
Description
FAX Parameters
(cont’d)
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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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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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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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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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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Voice/Fax Parameter Definitions (cont’d)
Field Name Values
Dynamic Jitter
Maximum 60 to 400
Description
The maximum dynamic jitter buffer
of 400 milliseconds is the maximum
delay tolerable over a high jitter
network.
Jitter Value
ms
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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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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9. Set T1/E1/ISDN Parameters. This dialog box can be reached by
pulldown menu, keyboard shortcut, or sidebar.
Accessing “T1/E1/ISDN Parameters”
Pulldown
Icon
--
Shortcut
Sidebar
Ctrl + T
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In each field, enter the values that fit your particular network.
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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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T1 Parameter Definitions
Field Name
Values
Description
T1/E1/ISDN
T1
North American digital
telephony 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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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.
If a user has an old MultiVOIP
unit (with a firmware version
lower than 4.08), and wants to
upgrade to 4.08, the latest CAS
file (4.08) should also be
downloaded into that
MultiVOIP unit. The new CAS
file ensures proper operation
between the MultiVOIP and a
PBX.
Match this parameter to the
setting of PBX or central-office
switch.
FXS Options –
No Response
Timer
1 – 65535
(in seconds)
Length of time before call
connection attempt is
abandoned. Applicable only
when FXS CAS protocol is
selected.
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T1 Parameter Definitions
Field Name
Values
Description
FXS Ground Start Supervision
Parameters
Answer Delay
(Enable)
Y/N
When this option is selected, the
FXS interface sends the
connection notice to the calling
party only when the Answer
Delay Timer expires. The
connection notice is sent
regardless of whether or not the
called extension has gone
offhook.
Answer Delay
Timer
numeric
(in seconds)
When Answer Delay is enabled,
this value determines when the
FXS interface sends the
connection notice.
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T1 Parameter Definitions (cont’d)
Values Description
Field Name
FXS Ground Start Supervision
Parameters
Tone Detection Y/N
(Enable)
After a specified tone (chosen
from the Available Tones list)
coming from the PBX is
stopped, the FXS interface will
send the ‘connect’ signal to the
calling party.
Available
Busy Tone, Dial
List from which tones can be
Tones (List)
Tone, Reorder Tone chosen to signal call answer.
Survivability Dial
Tone, Unobtainable
Tone
Answer Tones
(List)
Busy Tone, Dial
Currently chosen call-answer
Tone, Reorder Tone supervision tone.
Survivability Dial
Tone, Unobtainable
Tone
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.”
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T1 Parameter Definitions (cont’d)
Values Description
Field Name
ISDN Parameters
Country
see table, later
this chapter
Country in which MultiVOIP is
operating with ISDN.
Operator
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.
Numbering Details Parameters
Calling Party
Number Type
unknown,
national,
Calling party type is part of
calling party Number
international,
Information element that is sent
network specific, on ISDN line. The Calling party
subscriber,
number information element
identifies the origin of a call.
abbreviated,
as received from
network
Called Party
Number Type
unknown,
national,
international,
Called Party Number Type and
Called Party Number Plan are
part of Calling Party Number
network specific, Information element that is sent
subscriber,
on ISDN line. The Called party
number information element
identifies destination of a call.
abbreviated,
as received from
network
Called Party
Number Plan
unknown,
ISDN telephony,
data,
The call dialing plan under
which the called party operates.
telex,
national standard,
private,
as received from
network
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T1 Parameter Definitions (cont’d)
Values Description
Field Name
General T1/E1/ISDN Parameters
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
Caller ID Parameters
Caller ID
Y/N
Turns Caller ID feature on (if
Enable
checked) and off (if unchecked).
Calling
Number Prefix
(Caller ID)
0-9, *, #
A DTMF symbol used to mark the
beginning of the calling party
number for use with Caller ID.
Maximum length: 4 characters.
Calling
Number Suffix
(Caller ID)
0-9, *, #
Y/N
A DTMF symbol used to mark
the end of the calling party
number for use with Caller ID.
Maximum length: 4 characters.
Detect Flash
Hook
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
milliseconds
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.
Clocking
External/Internal Set opposite to telco/PBX setting.
Example: if telco clocking internal,
set VOIP clocking as external.
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T1 Parameter Definitions (cont’d)
Field Name
Line Coding
PCM Law
Values
Description
AMI / B8ZS
A-Law/Mu-Law
Match to PBX or telco.
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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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.
FXS Options –
No Response
Timer
1 – 65535
(in seconds)
Length of time before call
connection attempt is
abandoned. Applicable only
when FXS is selected as CAS
protocol.
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E1 Parameter Definitions
Field Name
Values
Description
FXS Ground Start Supervision
Parameters
Answer Delay
(Enable)
Y/N
When this option is selected, the
FXS interface sends the
connection notice to the calling
party only when the Answer
Delay Timer expires. The
connection notice is sent
regardless of whether or not the
called extension has gone
offhook.
Answer Delay
Timer
numeric
(in seconds)
When Answer Delay is enabled,
this value determines when the
FXS interface sends the
connection notice.
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E1 Parameter Definitions (cont’d)
Values Description
Field Name
FXS Ground Start Supervision
Parameters
Tone Detection Y/N
(Enable)
After a specified tone (chosen
from the Available Tones list)
coming from the PBX is
stopped, the FXS interface will
send the ‘connect’ signal to the
calling party.
Available
Busy Tone, Dial
List from which tones can be
Tones (List)
Tone, Reorder Tone chosen to signal call answer.
Survivability Dial
Tone, Unobtainable
Tone
Answer Tones
(List)
Busy Tone, Dial
Currently chosen call-answer
Tone, Reorder Tone supervision tone.
Survivability Dial
Tone, Unobtainable
Tone
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.”
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E1 Parameter Definitions (cont’d)
Values Description
Field Name
ISDN Parameters
Country
see table, later
this chapter
Country in which MultiVOIP is
operating with ISDN.
Operator
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.
Numbering Details Parameters
Calling Party
Number Type
unknown,
national,
Calling party type is part of
calling party Number
international,
Information element that is sent
network specific, on ISDN line. The Calling party
subscriber,
number information element
identifies the origin of a call.
abbreviated,
as received from
network
Called Party
Number Type
unknown,
national,
international,
Called Party Number Type and
Called Party Number Plan are
part of Calling Party Number
network specific, Information element that is sent
subscriber,
on ISDN line. The Called party
number information element
identifies destination of a call.
abbreviated,
as received from
network
Called Party
Number Plan
unknown,
ISDN telephony,
data,
The call dialing plan under
which the called party operates.
telex,
national standard,
private,
as received from
network
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E1 Parameter Definitions (cont’d)
Values Description
Field Name
General E1/E1/ISDN Parameters
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
Caller ID Parameters
Caller ID
Y/N
Turns Caller ID feature on (if
Enable
checked) and off (if unchecked).
Calling
Number Prefix
(Caller ID)
0-9, *, #
A DTMF symbol used to mark the
beginning of the calling party
number for use with Caller ID.
Maximum length: 4 characters.
Calling
Number Suffix
(Caller ID)
0-9, *, #
Y/N
A DTMF symbol used to mark
the end of the calling party
number for use with Caller ID.
Maximum length: 4 characters.
Detect Flash
Hook
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
Clocking
External/Internal Set opposite to telco/PBX
setting. Example: if telco
clocking internal, set VOIP
clocking as external.
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E1 Parameter Definitions (cont’d)
Field Name
Line Coding
PCM Law
Values
Description
AMI / B8ZS
A-Law/Mu-Law
Match to PBX or telco.
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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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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11. Set Call Signaling Parameters. This dialog box leads to 3 others,
one for each of the call-signaling types supported (H.323, SIP, and
SPP). These dialog boxes can be reached by pulldown menu,
keyboard shortcut, or a sidebar menu.
Accessing “Call Signaling Parameters”
Pulldown
Shortcut
Sidebar
Alt + C
Accessing the Signaling Protocols
Protocol
H.323
SIP
Ctrl + Alt + 3
Ctrl + Alt + Shft + P
Ctrl + Alt + Shft + P
SPP
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The tables below describes all fields in the general H.323 Call Signaling
screen.
H.323 Call Signaling Parameter Definitions
Field Name
Values
Description
Use Fast Start
Y/N
Enables the H.323 Fast Start
procedure. May need to be
enabled/disabled for
compatibility with third-party
VOIP gateways.
Signaling Port
port
Default: 1720 (H.323)
number
Register with
Gatekeeper
Y/N
Check this field to have traffic
on current voip gateway
controlled by a gatekeeper.
Allow
Y/N
When selected, incoming calls
are accepted only if those calls
come through the gatekeeper.
Incoming Calls
Through
Gatekeeper
Only
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H.323 Call Signaling Parameter Defns (cont’d)
Field Name
Values
Description
GateKeeper RAS Parameters
This is the preferred gatekeeper
for controlling the traffic of the
current voip.
Primary GK
(Gatekeeper)
--
A first and a second alternate
gatekeeper can be specified for
use by the current voip for
situations where the Primary GK
is busy or otherwise unavailable.
IP address of the GateKeeper.
Alternate GK
(Gatekeepers)
1 and 2
--
Gatekeeper /
IP Address
n.n.n.n,
for n =
0 - 255
RAS Port
1719
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.
A primary gatekeeper and two
alternate units are listed.
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Technical Configuration
.
H.323 Call Signaling Parameter Defns (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.
Gatekeeper
Discovery
Polling
integer
60 - 300
The interval between the voip
gateway’s successive attempts
to connect to and be governed
by a higher level gatekeeper.
The Primary GK is the highest
level gatekeeper. Alternate
GK1 is second; Alternate GK2
is the lowest order gatekeeper.
Interval
Use Online
Alternate
Gatekeeper
List
When selected, voip will seek an alternate
gatekeeper (when none of the 3 gatekeepers
shown on this screen are available) from a
list. The list will reside on the Primary
gatekeeper or one of the Alternate
(Y/N)
gatekeepers. The gatekeeper holding the list
would download that list onto the voip
gateways within the system.
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H.323 Call Signaling 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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H.323 Call Signaling 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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The tables below describes all fields in the general SIP Call Signaling screen.
SIP Call Signaling Parameter Definitions
Field Name
Values
Description
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.
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SIP Call Signaling Parameter Definitions (cont’d)
Field Name Values Description
SIP Proxy Parameters
Allow
Y/N
When selected, incoming calls
are accepted only if those calls
come through the gatekeeper.
Incoming Calls
Through SIP
Proxy Only
This is the preferred SIP proxy
server for controlling the traffic of
the current voip.
A first and a second alternate SIP
proxy server can be specified for
use by the current voip for
situations where the Primary
proxy server is busy or otherwise
unavailable.
Primary Proxy
--
--
Alternate
Proxy 1 and 2
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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SIP Call Signaling 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.
Proxy Polling
Interval
integer
60 - 300
The interval between the voip
gateway’s successive attempts
to connect to and be governed
by a higher level SIP proxy
server. The Primary Proxy is
the highest level gatekeeper.
Alternate Proxy 1 is second;
Alternate Proxy 2 is the lowest
order SIP proxy server.
TTL Value
in seconds
The SIP proxy “Time to Live” value. As soon as a
MultiVOIP gateway registers with a SIP proxy
server (allowing the proxy server to control its call
traffic) a countdown timer begins. The 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 proxy
server will expire and the proxy server 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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The tables below describes all fields in the general SPP Call Signaling screen.
SPP Call Signaling 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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SPP Call Signaling 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.
Client Options
This is the preferred SPP registrar
gateway for controlling the traffic
of the current voip.
A first and a second alternate SPP
Registrar gateway can be
specified for use by the current
voip for situations where the
Primary Registrar gateway is busy
or otherwise unavailable.
Primary
Registrar
--
--
Alternate
Registrar 1 and
2
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.)
Registrar IP
Address
n.n.n.n
This is the port number of the
registrar voip to which this
client is assigned. (Default port
number = 10000.)
Registrar
Port
10000 or
other
Polling
Interval
integer
60 - 300
The interval between the voip
gateway’s successive attempts to
connect to and be governed by a
higher level SPP registrar gateway.
The Primary Registrar is the
highest level registrar gateway.
Alternate Registrar 1 is second;
Alternate Registrar 2 is the lowest
order SPP registrar gateway.
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.
Keep Alive
(in sec.)
30 – 300
(seconds)
Timeout default = 60 seconds.
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SPP Call Signaling Parameter Definitions
(cont’d)
Values Description
Field Name
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
An example of a NAT-equipped SPP network is shown below.
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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Technical Configuration
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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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).
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), ring tone, and
other, more specialized tones.
Remote Configuration/Command Modem. Each MVP2410 and
MVP3010 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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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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“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.
Advisory
screen
This message screen appears whenever the
Country field is changed. It informs the
operator that, upon change of the Country
field value, all User Defined Tones will be
deleted.
Standard Tones fields
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),
survivability
tone,
re-order tone
Frequency 1
Frequency 2
freq. in Hertz
freq. in Hertz
Lower frequency of pair.
Higher frequency of pair.
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“Regional Parameter” Definitions (cont’d)
Field Name Values Description
Standard Tones fields (cont’d)
Gain 1
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
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.
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“Regional Parameter” Definitions (cont’d)
Field Name
Values
Description
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
Country
Selection for
Built-In
country name
Modem
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/Region value
but “Denmark” as the Country-
(not applicable
to MVP-
130/130FXS
MVP210,
MVP410ST, or
MVP810ST)
Selection-for-Built-In-Modem value.
User Defined Tones fields
Name of supervisory tone pair.
Cannot be same as name of any
standard tone pair.
Type column
alphanumeric
name specified
by user
Frequency 1
freq. in Hertz
freq. in Hertz
gain in dB
Lower frequency of pair.
Higher frequency of pair.
Amplification factor of lower
frequency of pair.
Frequency 2
Gain 1
+3dB to –31dB
and “mute” setting This applies to any supervisory 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
Amplification factor of higher
frequency of pair.
and “mute” setting This applies to any supervisory tones
that the MultiVOIP outputs as audio to
the FXS, FXO, or E&M port. Default:
-
16dB
Cadence
n/n/n/n
On/off pattern of tone durations used
to denote supervisory tones specified
by user. Supervisory tones relate to
answering and disconnection of calls.
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
(msec) On/Off four integer time
values in
milli-seconds;
zero value for
dial-tone
indicates
continuous tone
redundant, it is necessary to allow for
expression of 4-part cadences.
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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 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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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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MultiVOIP User Guide
Technical Configuration
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
“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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.
“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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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,
Date and time the
phone call began.
Time
Duration Length of call.
Call
Voice or fax.
Mode
Packets
Received
Total packets
received in call.
Packets
Sent
Total packets sent
in call.
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“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
Bytes
Sent
Bytes
Received
Total bytes sent in
call.
Total bytes received
in call.
Packets
Lost
Coder
Packets lost in
call.
Voice Coder
/Compression Rate
used for call will be
listed in log.
Outbound
Digits
Received
Prefix
Matched
The DTMF dialing
digits received by
this gateway from
the remote
When selected, the
phonebook prefix
matched in
processing the call
will be listed in log.
gateway
presuming that
DTMF is set to
"Out of Band."
Successful or
Call
Status
Call Type
Indicates the Call
Signaling protocol
used for the call
unsuccessful.
(H.323, SIP, or SPP).
DTMF
Capability
Call
Direction
Indicates call’s
originating party.
Indicates whether the
DTMF dialing digits
are carried "Inband"
or "Out of Band." The
corresponding field
values differ for the 3
different voip
protocols.
For H.323, this field
can display "Out of
Band" or "Inband".
For SIP it can display
either "Out of Band
RFC2833" or "Out of
Band SIP INFO" to
indicate the out-of-
band condition or
"Inband" to indicate
the in-band condition.
For SPP it can
display "Out of Band
RFC2833" or
"Inband".
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“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
Outbound
Digits Sent
Server
Details
The dialing digits
sent by this gateway
to the remote
gateway presuming
that DTMF is set to
"Out of Band."
The IP address of
the traffic control
server (if any)
being used
(whether an H.323
gatekeeper, a SIP
proxy, or an SPP
registrar gateway)
will be displayed
here if the call is
handled through
that server.
Disconnect
Reason
Indicates whether the call was disconnected simply
because the desired conversation was done or some
other irregular cause occasioned disconnection (e.g., a
technical error or failure). Values are "Normal" and
"Local" disconnection.
To Details
From Details
Completing or
answering gateway
IP address where call
was completed or
answered.
Gateway
Number
IP Addr
Originating
gateway
IP address where
call originated.
Gatew N.
IP Addr
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 Silence
Compression and
Forward Error
Correction by party
answering call.
When selected, log
will not 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/Traces” Screen
Pulldown
Icon
Shortcut
Sidebar
Ctrl + Alt + L
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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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“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
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. Even though the H.450 standard
refers only to H.323, Supplementary Services are still applicable to the
SIP and SPP voip protocols.
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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 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
The Supplementary Services fields are described in the tables below.
Supplementary Services Parameter Definitions
Field Name Values
Description
Select
Channel
1-24
(2410);
1-30
The channel to be configured is
selected here.
(3010)
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
“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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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
17. Set NAT Traversal parameters. NAT (Network Address
Translation) parameters are applicable only when the MultiVOIP is
operating in SIP mode. The use of STUN (Simple Traversal of UDP
NATs) servers to aid networks with NAT devices is described in RFC
3489.
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Accessing “NAT Traversal” Parameters
Pulldown Icon
Shortcut
Sidebar
Ctrl + Alt + Sft
+ VH
Descriptions for NAT Traversal screen fields are presented in the
table below.
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NAT Traversal Definitions (cont’d)
Field Name Values
Description
Enable
(STUN)
Y/N
Enables STUN client functionality in
the MultiVOIP.
STUN (Simple Traversal of UDP
through NATs (Network Address
Translation)) is a protocol that allows
a server to assist client gateways
behind a NAT firewall or router with
their packet routing.
Name/IP
(Server)
n.n.n.n
0 - 255
IP address of the STUN server.
Port (Server; numeric; The data port (TDM time slot) at
NAT/STUN default=
which STUN info will be transmitted
and received.
)
3478
Keep Alive
(Timers;
NAT/STUN seconds)
)
60 – 3600 The interval at which the STUN client
(in
sends indicator (“Keep Alive”)
packets to the STUN server to
determine whether or not the STUN
server is available.
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18. Set RADIUS parameters. In general, RADIUS is concerned with
authentication, authorization, and accounting. The MultiVOIP
supports the accounting and authentication functions. The
accounting function is sell suited for billing of voip telephony
services. In the Attributes secondary screen (accessed by clicking on
Select Attributes), the voip administrator can select the parameters to
be tallied by the RADIUS server.
Accessing “RADIUS” Parameters
Pulldown
Icon
--
Shortcut
Sidebar
Ctrl + Alt + U
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The fields of the RADIUS screen are described in the table below.
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RADIUS Screen Field Definitions
Field Name Values
Description
Enable
Accounting
Y/N
When checked, the MultiVOIP will
access the accounting functionality of
the
Server
Address
n.n.n.n
0 – 255
IP address of the RADIUS server that
handles accounting (billing) for the
current MultiVOIP unit.
Accounting
Port
numeric; TDM time slot at which RADIUS
1 - 65535
accounting information will be
transmitted and received.
Retrans-
mission
Interval
If the MultiVOIP sends out a packet to
the RADIUS server and doesn't
receive a response in the retransmit
interval, it will retransmit that packet
again and wait the retransmit interval
again for a response. How many
times it does this is determined by the
setting in the Number of
Number of
Re-transmis-
sions
0 - 255
Retransmissions field.
Shared
Secret
alpha-
numeric
Client encryption key for the current
voip unit.
Select
Attributes
(button)
--
Gives access to RADIUS Attributes
screen. On Attributes screen, one can
specify the parameters to be tallied by
the RADIUS server for accounting
(usually billing) purposes.
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The RADIUS Parameters dialog box has a secondary dialog box,
Custom Fields, that allows you to customize accounting information
sent to the RADIUS server by 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 accounting reports sent to the RADIUS server.
“Custom Fields” Definitions
Field
Description
Field
Description
Select All Log report to
include all fields
shown.
Channel
Number
Data channel
carrying call.
Start
Date,
Date and time the
phone call began.
Time
Duration Length of call.
Call
Voice or fax.
Mode
Packets
Received
Total packets
received in call.
Packets
Sent
Total packets sent
in call.
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“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
Bytes
Received
Bytes
Sent
Total bytes sent in
call.
Total bytes received
in call.
Packets
Lost
Packets lost in
call.
Coder
Voice Coder
/Compression Rate
used for call will be
listed in log.
Outbound The DTMF dialing Prefix
When selected, the
phonebook prefix
matched in
processing the call
will be listed in log.
Digits
Sent
digits received by
this gateway from
the remote
Matched
gateway
presuming that
DTMF is set to
"Out of Band."
Successful or
unsuccessful.
The IP address (etc.) of the traffic control server (if any)
being used (whether an H.323 gatekeeper, a SIP proxy,
or an SPP registrar gateway) will be displayed here if
the call is handled through that server. The Options
field refers to non-mandatory server features that might
be activated. For example, with H.323, various H.323
Version 4 options might be listed (Multiplexing,
Tunneling, etc.).
Call
Status
Server
Details
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Technical Configuration
“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
To Details
Completing or
answering gateway
IP address where call
was completed or
answered.
From Details
Originating
Gateway
Number
IP Addr
Gatew N.
IP Addr
gateway
IP address where
call originated.
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 Silence
Compression and
Forward Error
Correction by party
answering call.
When selected, log
will not use
Silence
Compression and
Forward Error
Correction by call
originator.
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19. 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
20. 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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System Information Parameter Definitions (cont’d)
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 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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The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
21. 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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22. 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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Chapter 6:T1 Phonebook
Configuration
(North American Telephony Standards)
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MultiVOIP User Guide
T1 PhoneBook Configuration
T1 Versus E1 Telephony Environments
We present separate chapters for the MVP2410 MultiVOIP (this
chapter) and the MVP3010 MultiVOIP (Chapter 7) because the
respective telephony environments in which they operate have
different standards and conventions. The MVP2410 is designed to
operate under North American or T1 standards; the MVP3010 is
designed to operate under European or E1 standards. The
configuration of the phonebook is the same in either case. However,
differences in the telephony environment give rise to different
examples in each case. Series II analog MultiVOIP units (MVP130,
MVP130FXS, MVP210, MVP410, and MVP810) can be operated in
either the T1 or E1 environments. The examples in this chapter show
these analog voip units being used in the same system as the MVP2410
digital MultiVOIP.
Configuring T1 (NAM) Telephony
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
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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
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
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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Phonebook Pulldown Menu
Inbound Phonebook Shortcut
Outbound Phonebook
Shortcut
Alt + I
Alt + O
Phonebook Sidebar Menu
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1. 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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2. The Add/Edit Outbound PhoneBook screen appears.
Enter Outbound PhoneBook data for your MultiVOIP unit. 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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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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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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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 RFC 3087
(“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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Add/Edit Outbound Phone Book: Field Def’ns (cont’d)
Field Name
SPP Fields
Values
Description
Use Registrar
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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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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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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3. Select Inbound PhoneBook | List Entries.
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4. The Add/Edit Inbound PhoneBook screen appears.
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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
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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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.
5. 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 MultiVOIP 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 MultiVOIP 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
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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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
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
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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
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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T1 Phonebook Configuration
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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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Configuring Mixed Digital/Analog VOIP Systems
Analog MultiVOIP units, like the MVP-210/410/810 are compatible
with digital MultiVOIP units like the MVP2410. 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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T1 Phonebook Configuration
MultiVOIP User Guide
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
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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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
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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T1 Phonebook Configuration
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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
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
102
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.
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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T1 Phonebook Configuration
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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
263742
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.
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T1 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.
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
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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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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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T1 Phonebook Configuration
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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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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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Chapter 7: E1 Phonebook
Configuration
(European Telephony Standards)
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E1 PhoneBook Configuration
E1 Versus T1 Telephony Environments
We present separate chapters for the MVP3010 MultiVOIP (this
chapter) and the MVP2410 MultiVOIP (Chapter 6) because the
respective telephony environments in which they operate have
different standards and conventions. The MVP3010 is designed to
operate under European or E1 standards; the MVP2410 is designed to
operate under North American or T1 standards. The configuration of
the phonebook is the same in either case. However, differences in the
telephony environment give rise to different examples in each case.
Series II analog MultiVOIP units (MVP130, MVP130FXS, MVP210,
MVP410, and MVP810) can be operated in either the T1 or E1
environments. The examples in this chapter show these analog voip
units being used in the same system as the MVP3010 digital MultiVOIP.
E1-Standard 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
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E1 Phonebook Configuration
MultiVOIP User Guide
onto a single network. Typically
these are private networks.
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
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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E1 Phonebook Configuration
MultiVOIP User Guide
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
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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E1 Phonebook Configuration
MultiVOIP User Guide
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
Wren Clothing Co.
London
VOIP/PBX Site
5
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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
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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MultiVOIP User Guide
E1 PhoneBook Configuration
Phonebook Pulldown Menu
Inbound Phonebook Shortcut
Outbound Phonebook
Shortcut
Alt + I
Alt + O
Phonebook Sidebar Menu
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E1 Phonebook Configuration
MultiVOIP User Guide
Phonebook Configuration Procedure
1. Select Outbound Phone Book/List Entries.
Click Add.
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MultiVOIP User Guide
E1 PhoneBook Configuration
2. The Add/Edit Outbound PhoneBook screen appears.
Enter Outbound PhoneBook data for your MultiVOIP unit. 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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
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
MultiVOIP User Guide
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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MultiVOIP User Guide
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
MultiVOIP User Guide
Add/Edit Outbound Phone Book: Field Def’ns (cont’d)
Field Name
SPP Fields
Values
Description
Use Registrar
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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MultiVOIP User Guide
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
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.
3. Select Inbound PhoneBook/List Entries.
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MultiVOIP User Guide
E1 PhoneBook Configuration
4. The Add/Edit Inbound PhoneBook screen appears.
Enter Inbound PhoneBook data for your MultiVOIP unit. The fields of
the Add/Edit Inbound PhoneBook screen are described in the table
below.
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E1 Phonebook Configuration
MultiVOIP User Guide
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.”
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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.
5. 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 MultiVOIP 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 MultiVOIP 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 MultiVOIPs (MVP-
210/410/810) 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
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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
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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.
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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.
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E1 Phonebook Configuration
MultiVOIP User Guide
Configuring Digital & Analog VOIPs in Same System
Analog MultiVOIP units, like the MVP-210/410/810 are compatible
with digital MultiVOIP units like the MVP3010. 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
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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.
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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
90182
9
9
9
--
--
--
200.2.9.5
200.2.9.5
263
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.
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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).
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).
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.
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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
0207
--
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
operator or auto-
attendant.
200.2.9.9 Calls to London
area PSTN via Site
D PBX.
0182
26374
0207
8
0207
398
200.2.9.9 Calls to London
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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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
1.
2.
3.
4.
Dial 101.
Hear dial tone from Site B.
Dial 9435632.
Await completion. Talk.
Site A calling Site C, Method 2
5.
6.
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.
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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.
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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).
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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.
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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
http://phonebooth.interocitor.net
/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.
http://www.oftel.gov.uk/numbers
/number.htm
UK numbering plan
from the Office of
Telecommunications,
the UK telephony
authority.
http://www.itu.int/home/index.html
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.
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MultiVOIP User Guide
E1 PhoneBook Configuration
URL
Description
http://kropla.com/phones.htm
Guide to
international use of
modems.
http://www.numberplan.org/
National and
international
numbering plans
based on direct input
from regulators
worldwide. Includes
lists of telecom
carriers per country.
http://www.eto.dk/
European
Telecommunications
Office. Primarily
concerned with
mobile/wireless
radiotelephony,
GSM, etc.
http://www.eto.dk/ETNS.htm
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
/fs_05.html
telecom regulatory
agencies by country
(from German
telecom authority).
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Chapter 8: Operation and
Maintenance
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MultiVOIP User Guide
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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Operation and Maintenance
MultiVOIP User Guide
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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MultiVOIP User Guide
Operation & Maintenance
The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
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Operation and Maintenance
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
Ctrl +
Alt + A
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MultiVOIP User Guide
Operation & Maintenance
The Call Progress Details Screen
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Operation and Maintenance
MultiVOIP User Guide
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
IP Call Type
G.723, G.729,
G.711, etc.
The voice coder being used on
this call.
H.323, SIP, or
SPP
Indicates the Call Signaling
protocol used for the call (H.323,
SIP, or SPP).
IP Call
Direction
incoming,
outgoing
Indicates whether the call in
question is an incoming call or an
outgoing call.
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MultiVOIP User Guide
Operation & Maintenance
Call Progress Details: Field Definitions
Field Name Values Description
Packet Details
Packets Sent
integer value
integer value
integer value
integer value
integer value
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.
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Operation and Maintenance
MultiVOIP User Guide
Call Progress Details: Field Definitions (cont’d)
From – To Details
Description
Gateway
Name (from)
alphanumeric
string
Identifier for the VOIP gateway
that handled the origination of
this call.
IP Address
(from)
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.
Gateway
Name (to)
alphanumeric
string
Identifier for the VOIP gateway
that handled the completion of
this call.
IP Address
(to)
x.x.x.x,
IP address to which the call was
sent.
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.
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Operation & Maintenance
Call Progress Details: Field Definitions (cont’d)
DTMF/Other Details
Field Name Values
Description
Prefix
Matched
specified
dialing digits
Displays the dialed digits that
were matched to a phonebook
entry.
Outbound
Digits Sent
0-9, #, *
The digits transmitted by the
MultiVOIP to the PBX/telco for
this call.
Outbound
Digits
Received
0-9, #, *
Of the digits transmitted by the
MultiVOIP to the PBX/telco for
this call, these are the digits that
were confirmed as being received.
Server Details n.n.n.n
(for n=0-255)
The IP address (etc.) of the traffic
control server (if any) being used
(whether an H.323 gatekeeper, a
SIP proxy, or an SPP registrar
gateway) will be displayed here if
the call is handled through that
server.
and/or other
server IP-
related
descriptions
DTMF
Capability
inband,
out of band
Indicates whether the DTMF
dialing digits are carried "Inband"
or "Out of Band." The
corresponding field values differ
for the 3 different voip protocols.
Expressions
differ slightly
for different
Call Signaling
protocols
For H.323, this field can display
"Out of Band" or "Inband". For SIP
it can display either "Out of Band
RFC2833" or "Out of Band SIP
INFO" to indicate the out-of-band
condition or "Inband" to indicate
the in-band condition. For SPP it
can display "Out of Band
(H.323, SIP, or
SPP).
RFC2833" or "Inband".
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MultiVOIP User Guide
Call Progress Details: Field Definitions (cont’d)
Field Name Values Description
Supplementary Services
Status
Call on Hold
Call Waiting
Caller ID
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.
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.
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”
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Operation & Maintenance
Call Progress Details: Field Definitions (cont’d)
Field Name Values Description
Call Status fields
Call Status
hangup, active Shows condition of current call.
Call Control
Status
Tun, FS + Tun, Displays the H.323 version 4
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.
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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Operation and Maintenance
MultiVOIP User Guide
About Logs
Accessing “Statistics: Logs”
Pulldown
Icon
Shortcut
Sidebar
Ctrl + O
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MultiVOIP User Guide
Operation & Maintenance
The Logs Screen
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Operation and Maintenance
MultiVOIP User Guide
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
Type
H.323, SIP, or SPP
Indicates the Call Signaling protocol
used for the call (H.323, SIP, or SPP).
Displays the status of the call, i.e.,
whether the call was completed
successfully or not.
Status column
success or
failure
IP Direction
incoming,
outgoing
Indicates whether the call is
"incoming" or "outgoing" with
respect to the gateway.
Mode column
voice or FAX
Indicates whether the (event) being
described was a voice call or a FAX
call.
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.
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Logs Screen Details: Field Definitions (cont’d)
Field Name
Values
Description
Call Details
Voice coder
G.723, G.729,
G.711, etc.
The voice coder being used on
this call.
Disconnect
Reason
Values are
"Normal" and
"Local"
Indicates whether the call was
disconnected simply because the
desired conversation was done
or some other irregular cause
occasioned disconnection (e.g., a
technical error or failure).
disconnection.
DTMF Capability inband,
out of band
Indicates whether the DTMF
dialing digits are carried
"Inband" or "Out of Band." The
corresponding field values differ
for the 3 different voip protocols.
Expressions
differ slightly
for different
Call Signaling
protocols
For H.323, this field can display
"Out of Band" or "Inband". For
SIP it can display either "Out of
Band RFC2833" or "Out of Band
SIP INFO" to indicate the out-of-
band condition or "Inband" to
indicate the in-band condition.
For SPP it can display "Out of
Band RFC2833" or "Inband".
(H.323, SIP, or
SPP).
Outbound Digits 0-9, #, *
Received
The digits, sent by MultiVOIP to
PBX/telco, that were
acknowledged as having been
received by the remote voip
gateway.
Outbound Digits 0-9, #, *
Sent
The digits transmitted by the
MultiVOIP to the PBX/telco for
this call.
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Operation and Maintenance
MultiVOIP User Guide
Logs Screen Details: Field Definitions (cont’d)
Field Name
Values
Description
Call Details
Server Details
n.n.n.n
When the MultiVOIP is
for n= 0-255
operating in the non-direct mode
(with Gatekeeper in H.323 mode;
with proxy in SIP mode; or in the
client/server configuration of
SPP mode), this field shows the
IP address of the server that is
directing IP phone traffic.
The number of data packets sent
over the IP network in the course
of this call.
The number of data packets
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.
Packets sent
integer value
integer value
integer value
Packets received
Packets loss
(lost)
Bytes sent
integer value
integer value
The number of bytes of data sent
over the IP network in the course
of this call.
The number of bytes of data
received over the IP network in
the course of this call.
Bytes received
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Logs Screen Details: Field Definitions (cont’d)
Field Name Values Description
Call Details (cont’d)
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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About IP Statistics
Accessing IP Statistics
Pulldown
Icon
Shortcut
Sidebar
Ctrl + P
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IP Statistics Screen
IP Statistics: Field Definitions
Values Description
Field
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.
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IP Statistics: Field Definitions
Field
Values
Description
Name
UDP versus TCP (continued).
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).
IP address of the MultiVOIP. For an IP
address to be displayed here, the
MultiVOIP must have DHCP enabled.
Its IP address, in such a case, is
IP
Address
n.n.n.n
0 - 255
assigned by the DHCP server.
“Clear”
button
--
Clears packet tallies from memory.
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)
Field
Values
Description
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
Shortcut // Icon
Sidebar
Ctrl + 2 // 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.
Pings per Test 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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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.
Status
Signaling has been frozen at the most
recent values due to loss of frame
alignment, loss of multiframe
Freeze
Signaling
Active
alignment or due to a receive slip.
Line
Line loopback deactivation signal has
been detected in the receive bit stream.
Loopback
Deactivation
Signal
A short exists between the transmit pair for
at least 32 consecutive pulses.
Transmit
Line Short
Transmit
Data
Overflow
For use by MTS Technical Support
personnel.
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
Receive
Timeslot 16
Alarm
Indication
Signal
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
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.
Transmit
Line Open
At least 32 consecutive zeroes were
transmitted.
Transmit
Data
Underrun
For use by MTS Technical Support
Personnel.
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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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 Call Signaling screen and in the Add/Edit
Outbound PhoneBook screen.
Accessing Registered Gateway Details
Pulldown
Shortcut
Ctrl + Alt + W
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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About Alternate Server Statistics
Accessing Alternate Server Statistics
Pulldown
Shortcut
Sidebar
Ctrl + Alt + 4
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H.323 Gatekeepers (Statistics, Servers): Field Definitions
Field
Values
Description
Name
Column Headings
IP Address
Port
n.n.n.n,
The IP address of the gatekeeper.
for n = 0-255
TDMA time slot used for communication
between MultiVOIP unit and the
gatekeeper that serves it.
GK Name
Type
alpha-numeric
string
Identifier for gatekeeper.
This field describes the type of gateway as
which the MultiVOIP is defined with
respect to the gatekeeper.
Primary,
Predefined
Priority refers to … .
Priority
Status
The current status of the gateway, either
registered or unregistered.
registered, not
registered
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SIP Proxies (Statistics, Servers): Field Definitions
Field
Values
Description
Name
Column Headings
IP Address
Port
n.n.n.n,
The IP address of the SIP proxy by which
the MultiVOIP is governed.
for n = 0-255
TDMA time slot used for communication
between MultiVOIP unit and the SIP Proxy
that governs it.
This field describes the type of gateway as
which the MultiVOIP is defined with
respect to the gatekeeper.
Type
Primary,
Alternate
The current status of the MultiVOIP
gateway with respect to the SIP proxy,
either registered or unregistered.
Status
registered,
not registered
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SPP Registrars (Statistics, Servers): Field Definitions
Field
Values
Description
Name
Column Headings
IP Address
Port
n.n.n.n,
The IP address of the gatekeeper.
for n = 0-255
TDMA time slot used for communication
between MultiVOIP unit and the
gatekeeper that serves it.
This field describes the type of gateway as
which the MultiVOIP is defined with
respect to the gatekeeper.
Type
Primary,
Predefined
The current status of the gateway, either
registered or unregistered.
Status
registered, not
registered
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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}
Increments (in Kbps)
G711, G726, G727
G723
G729
5-120
{5}
5
30-120
10-120
20-120
{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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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 CAS Protocol
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.
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 User Defaults
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.
Set Password
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 settings from the controller
PC to the MultiVOIP unit. User can
choose whether to load Factory
Default Settings or Current
Configuration settings.
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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 CAS Protocol
1. The MultiVoip Configuration program may be on or off when
invoking the Download CAS Protocol command.
2.To invoke the Download CAS Protocol command, go to Start |
Programs | MVP____ x.xx | Download CAS Protocol.
3. A message screen will appear warning that the download will entail
a rebooting of the MultiVOIP.
Click OK.
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4. The directory containing the CAS protocol files (extension is .cas) will
appear.
Select the CAS protocol needed for your system. Click Open.
5. The chosen CAS protocol file will be loaded from the PC to the
MultiVOIP unit. Progress bars will appear at the bottom of the screen
while the download occurs. When the download is complete, the
MultiVOIP will complete its rebooting process.
6. The MultiVOIP software will be closed when the download is
complete. You will have to launch the MultiVOIP software again to
continue using it.
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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.
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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. The MultiVOIP unit will reboot
itself.
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3. To download the user defaults, go to
Start | Programs | MultiVOIP xxx | Download User Defaults.
4. A confirmation screen will appear indicating that this action will
entail rebooting the MultiVOIP.
Click OK.
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5. 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.
NOTE: To upgrade a MultiVOIP from software version 4.04 or earlier, an ftp primer
file must first be sent to the VOIP. This file is located in the
Software/ftp_Primer folder on the CD and the file name is
"FTP_Primer.bin". Before uploading this file, it must be renamed
"mvpt1ftp.bin". The VoIP will only accept files of this name. This is a
safety precaution to prevent the wrong files from being uploaded to the
VoIP. Once the primer file has been uploaded, upload the FTP firmware file.
If you accepted the defaults during the software loading process, this file is
located on your local drive at C:\Program Files\Multi-Tech
Systems\MultiVOIP 4.08 where the X is the software number and the .08 is
the version number of the MultiVOIP software on your local drive. Of
course the firmware file is named ‘mvpt1ftp.bin’.
Important: You cannot go back to 4.04 or earlier versions using FTP. You
must use ‘upgradesoftware’ via the serial port.
Important: These ftp upgrade instructions do not apply to software release
4.05 and above.
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FTP Server File Transfers (“Downloads”)
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,
Netscape or FireFox, 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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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 Ethernet/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. Any CAS file
used must first be
renamed to
“CASFILE.CAS.”
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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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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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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Web Browser Interface
You can control the MultiVOIP unit with a graphic user interface (GUI)
based on the common web browser platform. Qualifying browsers are
InternetExplorer6, Netscape6, and Mozilla FireFox 1.0.
Pop-Ups. Note that the MultiVOIP Web GUI uses pop-up windows
extensively. You must configure the browser to allow pop-ups when
using the MultiVOIP Web GUI.
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; or
Mozilla Firefox 1.0 or higher
Java Requirement
Java Runtime Environment
version 1.4.0_01 or higher
(this application program is
included with MultiVOIP)
Video Usability
large video monitor recommended
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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.
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The web GUI 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).
The web GUI gives easy access to Console Messages. Whereas with
the Windows GUI console messages must be viewed using a
communications program like HyperTerminal, with the Web GUI, it’s
easy: just click on STATISTICS | CONSOLE MESSAGES and a pop-up
window appears.
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
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a video monitor as possible in order to see all of a screen’s contents with
minimal scrolling.
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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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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SysLog Server Functions
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.
Read the End-User License Agreement carefully and observe license
requirements. See www.kiwisyslog.com. SysLog client programs
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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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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Chapter 9 Warranty, Service, and
Tech Support
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Limited Warranty
Multi-Tech Systems, Inc. (“MTS”) warrants that its products will be free
from defects in material or workmanship for a period of two years from
the date of purchase, or if proof of purchase is not provided, two years
from date of shipment. MTS MAKES NO OTHER WARRANTY,
EXPRESSED OR IMPLIED, AND ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
ARE HEREBY DISCLAIMED. This warranty does not apply to any
products which have been damaged by lightning storms, water, or
power surges or which have been neglected, altered, abused, used for a
purpose other than the one for which they were manufactured, repaired
by the customer or any party without MTS’s written authorization, or
used in any manner inconsistent with MTS’s instructions.
MTS’s entire obligation under this warranty shall be limited (at MTS’s
option) to repair or replacement of any products which prove to be
defective within the warranty period, or, at MTS’s option, issuance of a
refund of the purchase price. Defective products must be returned by
Customer to MTS’s factory—transportation prepaid.
MTS WILL NOT BE LIABLE FOR CONSEQUENTIAL DAMAGES
AND UNDER NO CIRCUMSTANCES WILL ITS LIABILITY EXCEED
THE PURCHASE PRICE FOR DEFECTIVE PRODUCTS.
Repair Procedures for U.S. and Canadian
Customers
In the event that service is required, products may be shipped, freight
prepaid, to our Mounds View, Minnesota factory:
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, MN 55112
Attn: Repairs, Serial # ________________
A Returned Materials Authorization (RMA) is not required. Return
shipping charges (surface) will be paid by MTS.
Please include, inside the shipping box, a description of the problem, a
return shipping address (it must be a street address, not a P.O. Box
number), your telephone number, and if the product is out of warranty,
a check or purchase order for repair charges.
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Warranty, Service, & Tech Support
For out-of-warranty repair charges, go to www.
multitech.com/documents/warranties
Extended two-year overnight replacement service agreements are
available for selected products. Please call MTS at (888) 288-5470,
extension 5308, or visit our web site at
www.multitech.com/programs/orc
for details on rates and coverages.
Please direct your questions regarding technical matters, product
configuration, verification that the product is defective, etc., to our
Technical Support department at (800) 972-2439 or email
expediting, receiving, shipping, billing, etc., to our Repair Accounting
department at (800) 328-9717 or (763) 717-5631, or email
Repairs for damages caused by lightning storms, water, power surges,
incorrect installation, physical abuse, or used-caused damages are
billed on a time-plus-materials basis.
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Warranty, Service, & Tech Support
MultiVOIP User Guide
Technical Support
Multi-Tech Systems has an excellent staff of technical support personnel
available to help you get the most out of your Multi-Tech product. If
you have any questions about the operation of this unit, or experience
difficulty during installation you can contact Tech Support via the
following:
Contacting Technical Support
Country By E-mail
By telephone
France
India
U.K.
(33) 1-64 61 09
81
support@
multitechindia.com
(91) 124-340778
(44) 118 959 7774
(800) 972-2439
(763) 785-3500
support@
multitech.co.uk
U.S. &
Canada
tsupport@
multitech.com
Rest of
World
support@
multitech.com
Internet: http://www.multitech.com/ _forms/email_tech_support.htm
Please have your product information available, including model and
serial number.
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Chapter 10: Regulatory Information
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Regulatory Information
MultiVOIP User Guide
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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Regulatory Information
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
MultiVOIP User Guide
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:
Multi-Tech Systems, Inc.
MultiVOIP
Trade name:
Model number:
MVP-810/410/210
US: AU7DDNAN46050
RJ-48C
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, MN 55112
Tel: (763) 785-3500
FAX: (763) 785-9874
FCC registration number:
Modular jack (USOC):
Service center in USA:
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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Regulatory Information
WEEE Statement
(Waste Electrical and Electronic Equipment)
July, 2005
The WEEE directive places an obligation on EU-based manufacturers,
distributors, retailers and importers to take-back electronics products at the
end of their useful life. A sister Directive, ROHS (Restriction of Hazardous
Substances) compliments the WEEE Directive by banning the presence of
specific hazardous substances in the products at the design phase. The
WEEE Directive covers all Multi-Tech products imported into the EU as of
August 13, 2005. EU-based manufacturers, distributors, retailers and
importers are obliged to finance the costs of recovery from municipal
collection points, reuse, and recycling of specified percentages per the WEEE
requirements.
Instructions for Disposal of WEEE by Users in the European Union
The symbol shown below is on the product or on its packaging, which
indicates that this product must not be disposed of with other waste. Instead,
it is the user’s responsibility to dispose of their waste equipment by handing it
over to a designated collection point for the recycling of waste electrical and
electronic equipment. The separate collection and recycling of your waste
equipment at the time of disposal will help to conserve natural resources and
ensure that it is recycled in a manner that protects human health and the
environment. For more information about where you can drop off your waste
equipment for recycling, please contact your local city office, your household
waste disposal service or where you purchased the product.
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Appendix A: Cable Pinouts
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Cable Pinouts
Appendix A: Cable Pinouts
Command Cable
RJ-45 Connector
End-to-End Pin Info
RJ-45
DB9F
PIN NO.
PIN NO.
1
2
3
4
5
6
7
8
4
7
8
3
2
6
1
5
1 2 3 4 5 6 7 8
CLEAR TO SEND
To DTE
To Command
TRANSMIT DATA
Port Connector
Device
(e.g., PC)
RECEIVE DATA
SIGNAL GROUND
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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MultiVOIP User Guide
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
Telco
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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Appendix B: TCP/UDP Port
Assignments
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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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Appendix C: Installation
Instructions for MVP428
Upgrade Card
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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 C-2: Removing screws from blank cover plate
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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 C-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 C-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.
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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 C-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 C-6: Standoffs at LED edge of board (top view)
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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 C-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.
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MultiVOIP User Guide
Index
Index
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Index
MultiVOIP User Guide
INDEX
Alternate Phone Number field, SPP
E1.............................................. 224
Alternate Phone Number, SPP
T1.............................................. 181
802.1p Priority Levels .............. 67, 68
abbreviated dialing, inter-office
E1.............................................. 210
T1.............................................. 172
Accept Any Number (inbound)
E1.............................................. 228
T1.............................................. 186
Accept Any Number (outbound) field
E1.............................................. 220
T1.............................................. 177
access to network.......................... 123
access to remote PSTN
accessing T1/E1/ISDN Parameters
screen ..........................................89
accessing Voice/FAX Parameters
screen ..........................................75
Accounting Port (RADIUS screen)
field ...........................................160
Add Inbound Phonebook Entry icons
E1..............................................216
T1..............................................173
Add Outbound Phonebook Entry icon
E1..............................................216
T1..............................................173
Add Prefix (inbound) field
E1..............................................228
T1..............................................186
Add Prefix (outbound) field
E1................................................ 17
T1.................................................. 9
accessing Statistics, Logs screen
.................................................. 264
accessing Call Progress (Statistics)
screen........................................ 256
accessing configuration parameter
groups ......................................... 64
accessing Ethernet/IP Parameters
screen.......................................... 65
accessing IP Statistics screen........ 271
accessing Logs (Statistics) screen
.................................................. 264
accessing logs screen .................... 140
accessing Regional Parameters..... 124
accessing Registered Gateway Details
(Statistics) screen...................... 289
accessing Registered Gateway
Details screen.................. 287, 289
accessing RTP Parameters screen. 294
accessing SMTP parameters ......... 133
accessing SNMP parameters......... 121
accessing Supplementary Services
screen........................................ 144
accessing System Information screen
.................................................. 165
E1..............................................221
T1..............................................178
Add/Edit Inbound Phonebook field
definitions
E1..............................228, 229, 230
T1..............................186, 187, 188
Add/Edit Inbound Phonebook screen
E1..............................................228
T1..............................................186
Add/Edit Inbound Phonebook screen
fields (E1)
Accept Any Number .................228
Add Prefix.................................228
Channel Number .......................229
Description (callee location) .....229
Enable (Call Forwarding)..........229
Forward Condition....................229
Forward Destination..................230
Registration Option Parameters 230
Remove Prefix ..........................228
Ring Count................................230
Add/Edit Inbound Phonebook screen
fields (T1)
Accept Any Number .................186
Add Prefix.................................186
Channel Number .......................186
Description (callee location) .....186
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Index
Enable (Call Forwarding) ......... 186
Forward Condition.................... 187
Forward Destination ................. 187
Registration Option Parameters 188
Remove Prefix .......................... 186
Ring Count................................ 188
Add/Edit Outbound Phonebook field
definitions
E1.............. 220, 221, 222, 223, 224
T1.............. 177, 178, 179, 180, 181
Add/Edit Outbound Phonebook fields
(E1)
Use Gatekeeper .................179, 181
Use Proxy (SIP) ........................180
Add/Edit Outbound Phonebook screen
E1..............................................219
T1..............................................176
Add/Edit Outbound Phonebook SPP
Fields
E1..............................................224
T1..............................................181
add-on module (4-to-8 channel),
installation.................................361
Address (SNMP) field...................123
Advanced button, Outbound
Accept Any Number................. 220
Add Prefix................................. 221
Advanced button....................... 223
Description................................ 221
destination pattern..................... 221
Gateway H.323 ID.................... 222
Gateway Prefix ......................... 222
H.323 Port Number................... 222
IP Address................................. 221
Protocol Type............................ 221
Remote Device is [legacy]
Phonebook
E1..............................................224
T1..............................................181
Advanced Features field group .......82
airflow.............................................34
Alerting Party
Supplementary Services...151, 152,
153
Allow Incoming Calls Through
Gatekeeper Only (H.323 Call
MultiVOIP............................ 224
Remove Prefix .......................... 221
SIP Port Number....................... 223
SIP URL ................................... 223
Total Digits............................... 221
Transport Protocol (SIP)........... 223
Use Gatekeeper................. 222, 224
Use Proxy (SIP)........................ 223
Add/Edit Outbound Phonebook fields
(T1)
Signaling) field..........................109
Allow Incoming Calls Through SIP
Proxy Only (SIP Call Signaling)
field ...........................................115
Allowed Name Type
Alerting Party............151, 152, 153
Calling Party .............................150
Allowed Name Types, Call Name ID
Alerting Party............................151
Busy Party.................................152
Calling Party .............................150
Connected Party........................153
allowing pop-ups with Web GUI ....74
Alternate GK (Gatekeepers) 1 and 2
(H.323 Call Signaling) fields ....110
Alternate IP Address field
E1..............................................226
T1..............................................183
Alternate IP Routing
E1..............................................219
T1..............................................176
Alternate Phone Number field, SPP
E1..............................................224
Alternate Phone Number, SPP
(Add/Edit Outbound Phonebook)
Accept Any Number................. 177
Add Prefix................................. 178
Advanced button....................... 180
Description................................ 178
Destination Pattern.................... 178
Gateway H.323 ID.................... 179
Gateway Prefix ......................... 179
IP Address................................. 178
Protocol Type............................ 178
Q.931 Port Number................... 179
Remove Prefix .......................... 178
SIP Port Number....................... 180
SIP URL ................................... 180
Total Digits............................... 178
Transport Protocol (SIP)........... 180
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Index
E1.............................................. 224
MultiVOIP User Guide
baud rate, default (MultiVOIP
T1.............................................. 181
Alternate Proxy 1 and 2 (SIP Call
Signaling) fields........................ 115
Alternate Registrar 1 and 2 (SPP Call
Signaling) fields........................ 119
Alternate Routing
PSTN failover feature, and ....... 183
Alternate Routing field definitions
E1.............................................. 226
T1.............................................. 183
Alternate Routing field definitions
(E1)
software connection):................164
baud rate, fax...................................79
baud rate, setting ...........................164
Behind Proxy/NAT device............120
Bipolar Violation (E1 stats) field ..286
Bipolar Violation (T1 stats) field ..283
Blue Alarm (E1 stats) field ...........284
Blue Alarm (T1 stats) field ...........281
Boot LED
on MVP-2410/3010.....................37
Boot Version
System Info.......................166, 254
booting time
Alternate IP Address................. 226
Round Trip Delay ..................... 226
Alternate Routing field definitions
(T1)
E1................................................24
T1................................................16
box contents
Alternate IP Address................. 183
Round Trip Delay ..................... 183
Annex E field................................ 113
Answer Delay (Enable)
E1.............................................. 102
T1................................................ 94
Answer Delay Timer
verifying......................................32
BRI connector pinout....................356
BRI interface types
ST and U ...................................357
built-in modem
setup in Regional Parameters
screen ..............................59, 125
busy & no-response (E1)
E1.............................................. 102
T1................................................ 94
answer supervision criteria, FXS (E1)
.................................................. 103
answer supervision criteria, FXS (T1)
.................................................... 95
Answer Tones (FXS answer
supervision) field ................ 95, 103
Append SIP Proxy Domain Name in
User ID (proxy server).............. 115
Auto Disconnect field group........... 88
AutoCall.......................................... 83
AutoCall (Voice/Fax Params) and
Pass Through Enable (FXS Loop
Start) ........................................... 83
AutoCall/Offhook Alert field.... 83, 84
Automatic Disconnection field ....... 88
Available Tones (FXS answer
forwarding, dual conditions ......229
busy & no-response (T1)
forwarding, dual conditions ......187
busy tone, custom..........................131
busy-tones .....................................130
Bytes Received (call progress) field
..................................................259
Bytes Received (RADIUS
Attributes) field ......................162
Bytes Received (SMTP logs) field137
Bytes received (statistics, logs) field
..................................................268
Bytes Sent (call progress) field .....259
Bytes Sent (RADIUS Attributes) field
..................................................162
Bytes Sent (SMTP logs) field .......137
Bytes sent (statistics, logs) field....268
cable length, maximum span
supervision, E1) field................ 103
Available Tones (FXS answer
supervision, T1) field.................. 95
bandwidth, coder............................. 81
battery caution ................................ 31
E1..............................................100
T1................................................92
cabling problem, fixing...................64
cabling procedure
MVP2410....................................36
368
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Index
MVP3010.................................... 36
Cadence 1 (custom) field.............. 132
Cadence 2 (custom) field.............. 132
Cadence 3 (custom) field.............. 132
Cadence 4 (custom) field.............. 132
Cadence field ........................ 128, 129
cadences, custom
T1.E1 ........................................ 132
cadences, signaling ....................... 124
Call Control PHB field ................... 70
Call Control Priority (Ethernet/IP
parameters) field ......................... 68
Call Control Status
Call Progress Details (statistics)
field....................................... 263
Call Control Status (call progress)
field........................................... 263
Call Direction (SMTP logs) field.. 137
Call Duration field .......................... 88
Call Forward Parameters (inbound
phonebook)
Call Progress Details (statistics) field
definitions 258, 259, 260, 261, 262,
263
Call Progress Details (statistics)
screen field
Call On Hold.........................262
Call Waiting..........................262
Caller ID................................262
Call Progress Details (statistics)
screen fields
Channel .................................258
Duration ................................258
Mode .....................................258
Voice Coder ..........................258
IP Call Type ..........................258
IP Call Direction ...................258
Packets Sent ..........................259
Packets Received...................259
Bytes Sent .............................259
Bytes Received......................259
Packets Lost ..........................259
Outbound Digits Sent............261
Outbound Digits Received....261
Prefix Matched......................261
Server Details........................261
DTMF Capability..................261
Call On Hold.........................262
Call Waiting..........................262
Caller ID................................262
Call Status .............................263
Call Control Status................263
Silence Compression.............263
Forward Error Correction......263
Gateway Name (from and to)....260
IP Address (from and to)...........260
Options (from and to)................260
Gateway Name (from....................260
IP Address (from...........................260
Options (from................................260
Gateway Name (to ........................260
IP Address (to ...............................260
Options (to ....................................260
Call Status (call progress) field.....263
Call Status (RADIUS Attributes) field
..................................................162
Call Status (SMTP logs) field .......137
Call Transfer .................................145
E1................................................23
E1.............................................. 229
T1.............................................. 187
Call Forwarded To
logs (statistics) field.................. 270
Call Hold....................................... 145
E1................................................ 23
T1................................................ 15
Call Hold Enable........................... 148
Call Mode (RADIUS Attributes) field
.................................................. 161
Call Mode (SMTP logs) field ....... 136
Call Name Identification............... 145
E1................................................ 23
T1................................................ 15
Call Name Identification
Calling Party............................. 150
Call Name Identification
Alerting Party............................ 151
Call Name Identification
Alerting Party............................ 152
Call Name Identification
Alerting Party............................ 153
Call On Hold
Call Progress Details (statistics)
field....................................... 262
Call on Hold (call progress) field.. 262
Call Progress (Statistics)............... 256
369
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Index
T1................................................ 15
MultiVOIP User Guide
CAS Protocol field
Call Transfer Enable..................... 147
Call Transfer music jingle during hold
.................................................. 147
Call Transferred To
E1..............................................101
T1................................................93
CAS Protocol, downloading..........308
CAS vs. CCS
logs (statistics) field.................. 270
Call Type (SMTP logs) field ........ 137
Call Waiting.................................. 145
Call Progress Details (statistics)
field................................... 262
Call Progress Details (statistics)
field....................................... 262
E1................................................ 23
T1................................................ 15
Call Waiting (call progress) field.. 262
Call Waiting Enable...................... 148
Called Party Number Plan
T1........................................93, 101
CCS vs. CAS
T1........................................93, 101
CD, MultiVOIP...............................27
Channel (call progress) field.........258
channel capacity................................8
E1................................................17
T1..................................................9
Channel Number (inbound) field
E1..............................................229
T1..............................................186
Channel Number (RADIUS
E1.............................................. 104
T1................................................ 96
Called Party Number Type
Attributes) field.........................161
Channel Number (SMTP logs) field
..................................................136
channel tracing on/off (logging)....143
Clear (IP Statistics) button ............273
Clear command (Link Management)
button ........................................278
Client Options fields..................119
Clocking field
E1.............................................. 104
T1................................................ 96
Caller ID ....................................... 145
Call Progress Details (statistics)
field....................................... 262
Caller ID (call progress) field ....... 262
Caller ID (Supplementary Services)
field........................................... 154
Caller ID Enable
E1..............................................105
T1................................................97
coder
E1.............................................. 105
T1................................................ 97
Caller Name Identification Enable 149
Calling Number Prefix (Caller ID, E1)
.................................................. 105
Calling Number Prefix (Caller ID, T1)
.................................................... 97
Calling Number Suffix (Caller ID,
E1) ............................................ 105
Calling Number Suffix (Caller ID,
T1) .............................................. 97
Calling Party
bandwidth, max...........................81
G.711...........................................81
G.723.1........................................81
G.726...........................................81
G.727...........................................81
G.729...........................................81
Net Coder....................................81
Coder (RADIUS Attributes) field .162
Coder (SMTP logs) field...............137
Coder field ......................................81
coder options
packetization rates and..............294
Coder Parameters field group..........81
coder types (voice/fax, RTP
Supplementary Services............ 150
Calling Party Number Type
E1.............................................. 104
T1................................................ 96
Canadian Class A requirements.... 349
Canadian Limitations Notice
packetization)............................295
COM port
conflict, resolving........................63
error message ..............................63
on command PC..........................44
(regulatory) ............................... 350
370
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COM port allocation..................... 164
COM port assignments ................. 164
COM port conflict
configuration, user default ............169
Configuring MultiVOIP phonebooks,
general
error message.............................. 44
COM Port Setup screen ............ 44, 63
command cable pinout.................. 353
command modem
and Regional Parameters screen 59,
125
Command Modem
E1..............................................209
T1..............................................171
conflicts
COM port....................................44
Connection Problems, Solving........63
Consecutive Packets Lost field .......88
Console Message Settings, Filters for
..................................................143
console messages, enabling...........141
console parameters tracked ...........143
contacting technical support..........346
coordinated phonebook entries
E1..............................................215
T1..............................................172
Copy Channel command (Voice/Fax
Parameters) .................................77
Copy Channel field .........................78
Copy Channel, Supplementary
setup for.............................. 59, 125
command PC
COM port assignment (detailed). 44
community (voip) defined ............ 123
Community Name 1 (SNMP) field123
compatibility, Fast Start................ 109
compatibility, H.450 with H.323, not
with SIP .................................... 144
E1................................................ 18
T1................................................ 10
compression standard
E1.............................................. 106
T1................................................ 98
compression, silence....................... 82
Compression, Silence (RADIUS
Attributes)................................. 163
Compression, Silence (SMTP logs)
.................................................. 138
configuration of voip
local versus remote ............... 50, 51
Configuration option description
(MultiVOIP program menu) ..... 297
Configuration Parameter Groups,
accessing..................................... 64
Configuration Port Setup option
description (MultiVOIP program
menu)........................................ 297
configuration procedure, local
detailed........................................ 60
summary ..................................... 59
Configuration Version
Services command ....................146
Copy Channel, Supplementary
Services field.............................154
Count of Registered Numbers field
(Registered Gateway Details) ...289
country
ISDN type and...........................107
switch type and ISDN ...............107
Country field (ISDN)
E1/ISDN....................................104
T1/ISDN......................................96
Country Selection for Built-In Modem
field ...........................................129
Country/Region (tone schemes) field
..........................................126, 127
CRC and ESF frame format (T1)....92
CRC Check field
E1..............................................100
T1................................................92
Creating a User Default Configuration
..................................................169
Custom (tones, Regional)field ......128
custom cadences............................132
custom DTMF...............................131
Custom Fields (RADIUS Attributes)
definitions .................................161
System Info............................... 167
configuration, local......................... 53
configuration, phonebook
E1.............................................. 215
T1.............................................. 172
configuration, saving .................... 168
user ........................................... 313
371
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MultiVOIP User Guide
Custom Fields (RADIUS) definitions
.................................................. 162
Custom Fields (SMTP) definitions
.......................................... 136, 137
Custom Fields, RADIUS Accounting
Attributes
From IP Address .......................138
Outbound Digits Received........137
Outbound digits sent .................138
Packets Lost ..............................137
Packets Received.......................136
Packets Sent ..............................136
Prefix Matched..........................137
Select All...................................136
Server Details............................ See
Start Date, Time........................136
To Gateway Number.................138
To IP Address ...........................138
Custom Tone-Pair Settings definitions
..........................................131, 132
Custom Tone-Pair Settings fields
Cadence 1..................................132
Cadence 2..................................132
Cadence 3..................................132
Cadence 4..................................132
Frequency 1...............................131
Frequency 2...............................131
Gain 1........................................131
Gain 2........................................131
Tone Pair...................................131
customized log email ............136, 138
customized RADIUS Accounting.161
customized RADIUS accounting
parameters.................................163
data capacity......................................8
E1................................................17
T1..................................................9
data compression
Bytes Received ......................... 162
Bytes Sent................................. 162
Call Status................................. 162
Coder ........................................ 162
Options...................................... 163
Options...................................... 163
Description (callee)................... 163
Description (caller) ................... 163
Disconnect Reason.................... 162
From Gateway Number ............ 163
From IP Address....................... 163
Outbound Digits (sent).............. 162
Packets Lost.............................. 162
Prefix Matched.......................... 162
Server Details............................ 162
To Gateway Number................. 163
To IP Address ........................... 163
Custom Fields, RADIUS Attributes
Call Mode ................................. 161
Channel Number....................... 161
Duration.................................... 161
Packets Received ...................... 161
Packets Sent.............................. 161
Select All .................................. 161
Start Date, Time........................ 161
Custom Fields, SMTP log email
Bytes Received ......................... 137
Bytes Sent................................. 137
Call Direction............................ 137
Call Mode ................................. 136
Call Status................................. 137
Call Type .................................. 137
Channel Number....................... 136
Coder ........................................ 137
Options...................................... 138
Options...................................... 138
Description (callee)................... 138
Description (caller) ................... 138
Disconnect Reason.................... 138
DTMF Capability...................... 137
Duration.................................... 136
From Gateway Number ............ 138
E1................................................18
T1................................................10
Date & Time Setup (program menu
option), command .....................300
Date and Time Setup option
description (MultiVOIP program
menu) ........................................297
debugging messages......................142
Default (Supplementary Services)
field ...........................................154
Default (Voice/FAX) field..............78
default baud rate (MultiVOIP
software connection).................164
default configuration, user ............169
default values, software.................310
delay, packets..................................86
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Download CAS Protocol option
delay, versus voice quality.............. 87
Delete File button
Logs (Statistics) screen............. 266
Description (callee location)
E1.............................................. 229
T1.............................................. 186
Description (callee, outbound
description (MultiVOIP program
menu) ........................................298
Download Factory Defaults (program
menu option) , command...........310
Download Factory Defaults option
description (MultiVOIP program
menu) ........................................298
Download Firmware (program menu
option), command .............304, 305
Download Firmware option
description (MultiVOIP program
menu) ........................................298
Download User Defaults (program
menu option) , command...........313
Download User Defaults option
phonebook)
E1.............................................. 221
T1.............................................. 178
Description field (Registered Gateway
Details)...................................... 289
Description, From Details (RADIUS
Attributes) field......................... 163
Description, From Details (SMTP
logs) field.................................. 138
Description, To Details (RADIUS
Attributes) field......................... 163
Description, To Details (SMTP logs)
field........................................... 138
Destination Pattern (outbound) field
E1.............................................. 221
T1.............................................. 178
destination patterns, discussion
E1.............................................. 214
T1.............................................. 171
Detection Flash Hook field
description (MultiVOIP program
menu) ........................................298
downloading firmware, machine
perspective ........................299, 324
downloading user defaults ............313
downloads vs. uploads (FTP)........324
DTMF "Out of Band" and Outbound
Digits Sent.................................138
DTMF Capability (call progress) field
..................................................261
DTMF Capability (SMTP logs) field
..................................................137
DTMF Capability (statistics, logs)
field ...........................................267
DTMF Gain (High Tones) field ......78
DTMF Gain (Low Tones) field.......78
DTMF Gain field ............................78
DTMF In/Out of Band field............79
DTMF inband..................................79
DTMF out of band ..........................79
DTMF, custom tone pairs .............131
Duration (call progress) field ........258
Duration (DTMF) field ...................79
Duration (RADIUS Attributes) field
..................................................161
Duration (SMTP logs) field ..........136
Duration (statistics, logs) field......266
Dynamic Jitter Buffer field .............86
Dynamic Jitter field group ..............86
Dynamic Jitter fields.......................87
E1 Parameter definitions......100, 101,
102, 103, 104, 105, 106
E1.............................................. 105
T1................................................ 97
Detection Time field
E1.............................................. 105
T1................................................ 97
dial tone, custom........................... 131
dial-tones ...................................... 130
DiffServ and IP datagram ............... 71
DiffServ PHB (Per Hop Behavior)
value............................................ 70
dimensions
E1 models ................................... 26
Disconnect Reason (SMTP logs) field
.................................................. 138
Disconnect Reason (statistics, logs)
field........................................... 267
DNS Server IP Address (Ethernet/IP
Parameters) field......................... 72
Download CAS Protocol (program
menu option) , command .......... 308
373
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Answer Delay (Enable)............. 102
MultiVOIP User Guide
Receive Timeslot 16 Alarm
Answer Delay Timer................. 102
FXS Current Detect Timer........ 103
E1 Parameter definitions (FXS
Supervision)
Tone Detection.......................... 103
E1 Parameter fields
Indication Signal ...................285
Receive Timeslot 16 Loss of
MultiFrame Alignment..........286
Receive Timeslot 16 Loss of Signal
..............................................285
Red Alarm.................................284
Status Freeze Signalling Active 285
Transmit Data Overflow ...........285
Transmit Data Underrun ...........286
Transmit Line Open ..................286
Transmit Line Short ..................285
Transmit Slip.............................286
Transmit Slip Negative .............286
Transmit Slip Positive...............285
Yellow Alarm............................285
E1 telephony parameters.................56
Echo Cancellation field...................82
echo, removing................................82
Edit selected Inbound Phonebook
Entry icon
E1..............................................216
T1..............................................173
Edit selected Outbound Phonebook
Entry icon
E1..............................................216
T1..............................................173
email account for voip unit ...........134
email address for voip.............57, 133
email log reports ...........................133
email logs, illustration...................139
EMC, Safety, R&TTE Directive
Compliance ...............................348
Enable (Call Fwdg)
E1..............................................229
T1..............................................186
Enable (STUN) field.....................157
Enable Accounting (RADIUS) field
..................................................160
Enable Call Hold...........................148
Enable Call Transfer .....................147
Enable Call Waiting......................148
Enable Caller ID (E1) ...................105
Enable Caller ID (T1) .....................97
Enable Caller Name Identification 149
Enable Console Messages field.....142
Enable DHCP (Ethernet/IP
CAS Protocol........................ 101
FXS Options – No Response
Timer ................................ 101
No Response Timer (FXS
Options) ............................ 101
Answer Tones........................... 103
Available Tones (List) .............. 103
Caller ID Enable ....................... 105
Calling Number Prefix.............. 104
Calling Number Prefix (Caller ID)
.............................................. 105
Calling Number Suffix.............. 104
Calling Number Suffix (Caller ID)
.............................................. 105
Clocking.................................... 105
Country..................................... 104
CRC Check............................... 101
Detect Flash Hook .................... 105
Detection Time ......................... 105
Enable Caller ID ....................... 104
Frame Format............................ 101
Generation Time ....................... 105
Line Build-Out.......................... 105
Line Coding .............................. 106
Long-Haul Mode ...................... 101
Operator.................................... 104
PCM Law.................................. 106
Pulse Shape Level..................... 105
Tone Detection (Enable)........... 103
Yellow Alarm Format............... 106
E1 Parameters screen...................... 99
E1 Statistics field definitions284, 285,
286
E1 Statistics fields
Bipolar Variation ...................... 286
Blue Alarm ............................... 284
Excessive Zeroes ...................... 286
Loss of Frame Alignment ......... 284
Loss of MultiFrame Alignment 285
Receive Slip.............................. 286
Parameters) field .........................69
374
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Index
Enable DNS (Ethernet/IP Parameters)
field............................................. 72
Enable ISDN-PRI field
Voip Media PHB.........................70
Ethernet/IP Parameters screen,
accessing .....................................65
European Community Directives..348
Excessive Zeroes (E1 stats) field ..286
Excessive Zeroes (T1 stats) field ..281
expansion card (4-to-8 channel)
installation.................................361
factory default software settings ...310
factory defaults, downloading.......310
factory repair for customers U.S. &
Canada ......................................344
failover (PSTN)
E1/ISDN ................................... 103
T1/ISDN ..................................... 95
Enable SMTP field ....................... 134
Enable SNMP Agent..................... 121
Enable SNMP Agent field ............ 123
Enable SRV (Ethernet/IP Parameters)
field............................................. 72
enabling SMTP............................. 133
enabling web browser GUI............. 74
Error Correction (RADIUS
Attributes)................................. 163
Error Correction (SMTP logs) ...... 138
error correction, forward................. 82
error message
COM port conflict................. 44, 63
MultiVOIP Not Found................ 64
Phone Database Not Read........... 64
ESF and CRC frame format (T1).... 92
ethernet cable pinout..................... 353
Ethernet/IP parameter definitions .. 67,
68, 69, 70, 72
E1 models....................................18
T1 models....................................10
failover (PSTN) feature.................183
FAQ for MultiVOIPs ........................7
fast busy (unobtainable) tones.......130
Fast Connect.......... 113, See Fast Start
Fast Start compatibility .................109
Fast Start plus H.245 Tunneling field
..................................................113
fax baud rate, default.......................79
Fax Enable field ..............................79
FAX Parameters..............................79
fax tones, output level.....................80
Fax Volume field ............................80
FCC Declaration ...........................348
FCC Part 68 Telecom rules...........349
FCC registration number...............350
FCC rules, Part 15.........................348
FDX LED
E1................................................24
T1................................................16
Filters (Console Message Settings)143
Filters button (Console Message
Settings) ....................................142
firmware upgrade, implementing..304
Firmware Version (System Info) ..166
firmware version, identifying........304
firmware, downloading .................305
firmware, obtaining updated .........300
forgotten password................316, 320
Forward Address/Number
Ethernet/IP Parameter fields
802.1p Priority Levels........... 67, 68
Frame Type................................. 67
Ethernet/IP Parameter screen fields
Enable DNS ................................ 72
Ethernet/IP Parameters screen fields
Call Control (Priority)................. 68
Call Control PHB........................ 70
DiffServ ...................................... 70
DNS Server IP Address .............. 72
Enable DHCP.............................. 69
Enable SRV ................................ 72
FTP Server Enable...................... 72
Gateway...................................... 69
Gateway Name............................ 69
IP Address................................... 69
IP Mask....................................... 69
Others (Priorities) ....................... 68
Packet Prioritization 802.1p........ 67
TDM Routing Option ................. 73
Use TDM Routing for Intra-
T1..............................................187
Forward Condition (Call Fwdg)
Gateway Calls......................... 73
VLAN ID.................................... 68
VoIP Media (Priority)................. 68
E1..............................................229
T1..............................................187
375
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MultiVOIP User Guide
Forward Destination (Inbound PhBk)
E1.............................................. 230
Forward Error Correction (call
progress) field........................... 263
Forward Error Correction (RADIUS
Attributes)................................. 163
Forward Error Correction (SMTP
logs) .......................................... 138
Forward Error Correction field....... 82
forward on busy
graphic vs. textual orientation...333
FTP file transfers
using FTP client program..........326
using web browser ....................326
FTP Server Enable (Ethernet/IP
Parameters) field .........................72
FTP Server function
as added feature.........................324
enabling.....................................326
FTP Server, contacting..................328
FTP Server, invoking
T1...................................... 187, 229
Forward upon No Response
download/transfer
E1.............................................. 229
T1.............................................. 187
forwarding, dual conditions (E1)
busy & no-response .................. 229
forwarding, dual conditions (T1)
busy & no-response .................. 187
Frame Format field
E1.............................................. 100
T1................................................ 92
frame relay, and fax ........................ 80
Frame Search Restart Flag (T1 stats)
field........................................... 282
Frame Type field............................. 67
free calls
E1.............................................. 210
T1.............................................. 171
Frequency 1 (custom tone) field ... 131
Frequency 1 (tone pair scheme)... 127,
129
Frequency 2 (custom tone) field ... 131
Frequency 2 (tone pair scheme)... 127,
129
using FTP client program..........332
using web browser ....................330
FTP Server, logging in..................329
FTP Server, logging out................333
FTP transfers
file types............................324, 327
phonebooks ...............................324
server location...........................324
function tracing on/off (logging)...143
FXS (E1) disconnection, triggering of
..................................................103
FXS (T1) disconnection, triggering of
....................................................95
FXS Ground Start Supervision
Parameters
E1......................................102, 103
T1..........................................94, 95
FXS Options (E1) fields
No Response Timer...................101
FXS Options (T1) fields
No Response Timer.....................93
G711 coders (RTP packetization,
voice/fax) ..................................295
G723 coders (RTP packetization,
voice/fax) ..................................295
G726 coders (RTP packetization,
voice/fax) ..................................295
G727 coders (RTP packetization,
voice/fax) ..................................295
G729 coders (RTP packetization,
voice/fax) ..................................295
Gain 1 (custom tone) field ............131
Gain 1 (tone pair scheme) .....128, 129
Gain 2 (custom tone) field ............131
Gain 2 (tone pair scheme) .....128, 129
frequency, power
E1 models ................................... 26
FRF11............................................. 80
From (gateway, statistics, logs) field
.................................................. 266
front panel
E1................................................ 24
MVP2400.................................... 15
MVP2410.................................... 15
MVP3010.................................... 24
T1................................................ 15
FTP client program....................... 324
FTP client program, obtaining...... 326
FTP client programs
376
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Index
Gatekeeper Discovery Polling Interval
(H.323 Call Signaling) field...... 111
gatekeeper interaction
compatibility (E1 models)...........18
compatibility (T1 models)...........10
H.323 Annex E field .....................113
H.323 Call Signaling Parameter
definitions ......... 109, 111, 112, 113
H.323 Call Signaling screen fields
Allow Incoming Calls Through
Gatekeeper Only ...................109
Alternate GK 1 and 2 ................110
Annex E (H.323, UDP
E1 models ............................. 18, 19
T1 models ............................. 10, 11
Gatekeeper IP Address (H.323 Call
Signaling) field ......................... 110
Gatekeeper Name (H.323 Call
Signaling) fields........................ 110
GateKeeper RAS Parameters... 110,
111
Gateway (Ethernet/IP Parameters)
field............................................. 69
Gateway H.323 ID (Outbound
Phonebook) field
T1...................................... 179, 222
Gateway Name (callee, statistics,
logs) field.................................. 269
Gateway Name (caller, statistics, logs)
field........................................... 269
Gateway Name (Ethernet/IP
Parameters) field......................... 69
Gateway Number, From Details
(RADIUS Attributes) field........ 163
Gateway Number, From Details
(SMTP logs) field ..................... 138
Gateway Number, To Details
(RADIUS Attributes) field........ 163
Gateway Number, To Details (SMTP
logs) field.................................. 138
Gateway Prefix (outbound
multiplexing).........................113
Gatekeeper Discovery Polling
Interval..................................110
H.245 Tunneling .......................112
H.323 Multiplexing...................112
Parallel H.245 (Tunneling with Fast
Start)......................................113
Primary GK...............................110
RAS TTL Value........................111
H.323 Call Signaling screen fields
Register with GateKeeper .........109
Signaling Port............................109
Use Fast Start............................109
H.323 Call Signaling screen fields
Gatekeeper IP Address..............110
H.323 Call Signaling screen fields
RAS Port (Gatekeeper) .............110
H.323 Call Signaling screen fields
Gatekeeper Name......................110
H.323 Call Signaling screen fields
Gateway Name..........................110
Primary GK (Gatekeeper ..............110
H.323 coder.....................................81
H.323 fields (Outbound Phonebook)
E1..............................................222
T1..............................................179
H.323 Gatekeepers (Statistics,
phonebook) field
E1.............................................. 222
T1.............................................. 179
General Options fields .................. 118
Generate Local Dial Tone
(Voice/FAX – AutoCall/Offhook
Alert) field .................................. 84
Generation Time field
E1.............................................. 105
T1................................................ 97
GK Name (H.323 Gatekeepers,
Statistics, Servers) field ............ 291
grounding
in rack installations..................... 34
GUI (log reporting type) button.... 142
H.245 Tunneling field................... 112
H.323
Servers)
GK Name ..................................291
IP Address.................................291
Port............................................291
Priority ......................................291
Status.........................................291
Type ..........................................291
H.323 Multiplexing field...............112
H.323 Port Number (outbound
phonebook) field
E1..............................................222
377
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MultiVOIP User Guide
H.323 version 4 features
airflow.........................................34
E1 prerequisites...........................56
expansion card (4-to-8 channel) 361
in a nutshell.................................27
in rack .........................................33
log reports by email.....................57
software (detailed).......................39
T1 prerequisites...........................55
upgrade card (4-to-8 channel) ...361
voip email account ......................57
installation prerequisites .................53
installation, mechanical
E1 models....................................17
T1 models......................................9
installing Java vis-a-vis web GUI .337
integrated phone/data networks.....210
Intercept Tone (Regional Params) and
Offhook Alert (Voice/Fax Params)
..................................................126
Intercept Tone and required Interface
& Voice/Fax settings.................126
interface types, BRI
E1................................................ 18
T1................................................ 10
H.323 Version 4 Parameters.... 113
H.450 features, incompatible with SIP
.................................................. 144
E1................................................ 18
T1................................................ 10
H.450 functionality
logs for...................................... 270
H.450 standard
E1................................................ 23
T1................................................ 15
Hardware ID
System Info............................... 167
Hold Sequence...................... 145, 148
hold, caller on
musical jingle for ...................... 147
IANA ............................................ 359
icon
variable version........................... 41
icons, phonebook
E1.............................................. 216
T1.............................................. 173
identifying current firmware version
.................................................. 304
IFM Version
ST..............................................357
U 357
inter-office dialing
E1..............................................210
T1..............................................172
inter-operation with phone system
E1 models....................................17
T1 models......................................9
IP Address (callee, statistics, logs)
field ...........................................269
IP Address (caller, statistics, logs)
field ...........................................269
IP Address (Ethernet/IP Parameters)
field .............................................69
IP Address (H.323 Gatekeepers,
Statistics, Servers) field.............291
IP Address (IP Statistics) field......273
IP Address (outbound phonebook)
E1..............................................221
T1..............................................178
IP Address (ping target, Link
System Info............................... 167
implementing firmware upgrade... 304
in band, DTMF ............................... 79
Inbound Phonebook Entries List icon
E1.............................................. 216
T1.............................................. 173
Inbound Phonebook entries, list
E1.............................................. 226
T1.............................................. 184
inbound vs. outbound phonebooks
E1.............................................. 214
T1.............................................. 171
Industry Canada requirements ...... 349
info sources
E1 telephony details.................... 56
IP details ..................................... 54
SMTP details .............................. 57
T1 telephony details.................... 55
voip email account...................... 57
Input Gain field............................... 78
installation
Management) field....................279
IP Address (SIP Proxies, Statistics,
Servers) field.............................292
IP Address (SPP Registrars, Statistics,
Servers) field.............................293
378
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Index
IP Address field (Registered Gateway
Details)...................................... 289
IP Address to Ping (Link
T1..........................................95, 96
ISDN parameters, setting ..............107
ISDN-PRI
Management) field.................... 278
IP Address, From Details (RADIUS
Attributes) field......................... 163
IP Address, From Details (SMTP
logs) field.................................. 138
IP address, SysLog Server ............ 142
IP Address, To Details (RADIUS
Attributes) field......................... 163
IP Address, To Details (SMTP logs)
field........................................... 138
IP Call Direction (call progress) field
.................................................. 258
IP Call Type (call progress) field.. 258
IP datagram and DiffServ ............... 71
IP Direction (statistics, logs) field 266
IP Mask field .................................. 69
IP Statistics field
types supported .........................107
ISDN-PRI implementations ..........107
Java
installing....................................337
web GUI and.............................337
jitter buffer ......................................86
Jitter Value (Fax) field....................80
Jitter Value field..............................88
jitter, dynamic .................................86
Keep Alive (Timers; NAT/STUN)157
Keep Alive field............................119
Knowledge Base (online, for
MultiVOIPs) .................................7
Last button
Logs (Statistics) screen .............266
Last Error (Link Management) field
..................................................279
LED definitions
E1................................................24
MVP2400....................................15
MVP2410....................................16
MVP3010....................................24
T1................................................16
LED definitions (analog, T1)
LNK ............................................16
LED definitions (digital, E1)
LNK ............................................24
LED definitions (E1)
IP Address................................. 273
IP Statistics field definitions 272, 273,
274
IP Statistics fields
Clear.......................................... 273
Received (RTCP Packets)......... 275
Received (RTP Packets) ........... 275
Received (TCP Packets) ........... 274
Received (Total Packets) .......... 273
Received (UDP Packets)........... 274
Received with errors (RTCP
Packets)................................. 275
Received with errors (RTP Packets)
.............................................. 275
Received with errors (TCP Packets)
.............................................. 274
Received with errors (Total
Boot.............................................24
E1................................................24
FDX ............................................24
IC ................................................24
LC ...............................................24
LS................................................24
ONL ............................................24
Power ..........................................24
PRI ..............................................24
LED definitions (T1)
Packets)................................. 274
Received with errors (UDP
Packets)................................. 274
Transmitted (RTCP Packets) .... 275
Transmitted (RTP Packets)....... 275
Transmitted (TCP Packets)....... 274
Transmitted (Total Packets)...... 273
Transmitted (UDP Packets) ...... 274
IP Statistics function..................... 271
ISDN Parameters
Boot.............................................16
FDX ............................................16
IC ................................................16
LC ...............................................16
LS................................................16
ONL ............................................16
Power ..........................................16
E1...................................... 103, 104
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Index
PRI.............................................. 16
MultiVOIP User Guide
local configuration procedure
LED indicators
detailed, analog ...........................60
summary......................................59
local voip configuration ..................50
local Windows GUI vs. web GUI
comparison................................335
local-rate access (E1)
E1................................................ 23
T1................................................ 15
LED indicators, active
E1................................................ 23
T1................................................ 15
lifting
precaution about.......................... 31
limitations notice (regulatory),
Canadian ................................... 350
limited warranty............................ 344
Line Build Out field
E1.............................................. 105
T1................................................ 97
Line Coding field
E1.............................................. 106
T1................................................ 98
Line Loopback Activation Signal (T1
stats) field ................................. 282
Line Loopback Deactivation Signal
(T1 stats) field........................... 281
Link Management (Statistics) fields
Clear command button.............. 278
IP Address column.................... 279
IP Address to Ping .................... 278
Last Error.................................. 279
No. of Pings Received .............. 279
No. of Pings Sent...................... 279
Ping Size in Bytes..................... 278
Pings per Test ........................... 278
Response Timeout .................... 278
Round Trip Delay ..................... 279
Start Now command button ...... 278
Timer Interval between Pings... 278
Link Management (Statistics) screen
field definitions................. 278, 279
Link Status fields
to remote PSTN...........................17
local-rate calls to remote voip sites
E1..............................................211
Log # (statistics, logs) field...........266
log report email, customizing 136, 138
log report email, triggering.......135
log reporting method, setting ........140
log reports .......................................57
log reports & SMTP......................133
log reports by email.......................133
logging options..............................141
logging update interval..................141
logging, web GUI and...................336
Login Name (SMTP) field ............134
Logs (Statistics) fields
Bytes recvd................................267
Bytes Sent .................................266
Call Forwarded to......................270
Call Transferred to ....................270
Disconnect Reason....................267
DTMF Capability......................267
Duration ....................................266
From (gateway).........................266
Gateway Name (callee).............269
Gateway Name (caller) .............269
H.450 functionality ...................270
IP Address (callee)....................269
IP Address (caller) ....................269
IP Direction column..................266
Log #.........................................266
Mode.........................................266
Options (callee).........................269
Options (caller) .........................269
Outbound digits.........................269
Outbound Digits Recvd.............267
Outbound Digits Sent................267
Packets lost................................267
Packets recvd ............................267
Packets sent...............................267
Packets Sent ..............................266
Server Details............................267
Link Management (Statistics)
screen.................................... 279
List of Registered Numbers field
(Registered Gateway Details) ... 289
lithium battery caution.................... 31
LNK LED
E1................................................ 24
T1................................................ 16
loading of weight in rack ................ 34
local configuration.......................... 53
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Index
Start Date, Time........................ 266
Status ........................................ 266
Supplementary Services info .... 270
To (gateway)............................. 266
Type (call) column.................... 266
Voice coder............................... 267
Logs (Statistics) function........... 264
Logs (Statistics) screen
Delete File button ..................... 266
field definitions. 266, 267, 269, 270
First button................................ 266
Last button ................................ 266
Next button ............................... 266
Previous button......................... 266
logs and web browser GUI ........... 141
logs by email, illustration.............. 139
Logs screen definitions................. 141
Logs screen field definitions......... 142
Logs screen parameters
Enable Console Messages......... 142
Filters........................................ 142
GUI........................................... 142
IP Address (SysLog Server) ..... 142
Online Statistics Updation Interval
.............................................. 142
Port (SysLog Server) ................ 142
SMTP........................................ 142
SNMP ....................................... 142
SysLog Server Enable............... 142
Turn Off Logs........................... 142
logs screen, accessing ................... 140
long distance call savings
Mac Address
System Info.......................167, 254
mail criteria (SMTP), records .......135
Mail Server IP Address (SMTP) field
..................................................135
Mail Type (SMTP logs) field........135
mains frequency
E1 models....................................26
management (E1 models)
local.............................................19
remote (SNMP)...........................19
remote (web browser GUI) .........19
management of voips, remote .......121
Max bandwidth (coder)...................81
Max Baud Rate field .......................79
Max Retransmission (SPP, General
Options) field ............................118
maximum cable span
E1..............................................100
T1................................................92
Maximum Jitter Value field ............87
Minimum Jitter Value field.............86
Mode (call progress) field.............258
Mode (Fax) field .............................80
Mode (SPP) field...........................118
Mode (statistics, logs) field...........266
model descriptions
E1................................................17
modem relay....................................87
modem traffic on voip network.......87
modem, command
and Regional Parameters Country
Selection..........................59, 125
modem, remote
T1.............................................. 171
long-distance call savings
configuration/command
setup for ..............................59, 125
Monitor Link fields
Link Management (Statistics)
screen ....................................278
mounting
E1 models....................................17
T1 models......................................9
mounting in rack .............................33
procedure for...............................35
safety.....................................31, 34
mounting options ..............................8
Multiplexed UDP field..................113
MultiVOIP configuration software
E1.............................................. 209
Long-Haul Mode field
E1.............................................. 100
T1................................................ 92
Loss of Frame Alignment (E1 stats)
field........................................... 284
Loss of Frame Alignment (T1 stats)
field........................................... 281
Loss of MultiFrame Alignment (E1
stats) field ................................. 285
Loss of MultiFrame Alignment (T1
stats) field ................................. 282
lost packets, consecutive................. 88
lost password ........................ 316, 320
381
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Index
E1 models ................................... 19
MultiVOIP User Guide
Netcoder coders (RTP packetization,
voice/fax) ..................................295
network access ..............................123
Network Disconnection field ..........88
network/terminal settings, voip and
PBX
E1/ISDN....................................103
T1/ISDN......................................95
No. of Entries field (Registered
Gateway Details).......................289
No. of Pings Received (Link
Management) field....................279
No. of Pings Sent (Link Management)
field ...........................................279
no-response & busy(E1)
forwarding, dual conditions ......229
no-response & busy(T1)
forwarding, dual conditions ......187
Number of Days (email log criteria)
..................................................135
Number of Records (email log
criteria)......................................135
Number of Retransmissions (RADIUS
screen) field...............................160
Number Plan, Called Party
T1 models ................................... 11
MultiVOIP FAQ (on MTS web site) 7
MultiVOIP Program Menu items.. 297
MultiVOIP Program Menu options
Configuration............................ 297
Configuration Port Setup .......... 297
Date & Time Setup ................... 297
Download Factory Defaults...... 298
Download Firmware ................. 298
Set Password............................. 298
Uninstall.................................... 298
Upgrade Software..................... 298
MultiVOIP program menu, option
descriptions....................... 297, 298
MultiVOIP software
installing ..................................... 39
location of files ........................... 42
program icon location................. 43
uninstalling ......................... 46, 321
MultiVOIP software
moving around in........................ 64
MultiVoipManager ......................... 51
MultiVoipManager software
E1 models ................................... 19
T1 models ................................... 11
musical jingle during call transfer. 147
MVP2410
E1..............................................104
T1................................................96
Number Type, Called Party
cabling procedure........................ 36
unpacking.................................... 32
MVP-2410
E1..............................................104
T1................................................96
Number Type, Calling Party
remote configuration modem...... 37
MVP3010
E1..............................................104
T1................................................96
numbering plan resources .............250
obtaining updated firmware ..........300
Offhook alert...................................83
Offhook Alert (Voice/Fax Params)
and Intercept Tone (Regional
cabling procedure........................ 36
remote configuration modem...... 37
unpacking.................................... 32
Name/IP (Server) field.................. 157
NAT inter-operation support
E1 models ................................... 19
T1 models ................................... 11
NAT Traversal screen fields
Enable....................................... 157
Keep Alive (Timers)................. 157
Name/IP (Server)...................... 157
Port) .......................................... 157
Port (Server................................... 157
national-rate calls to foreign voip sites
E1.............................................. 213
Params) .......................................83
Offhook Alert Timer (Voice/FAX --
AutoCall/Offhook Alert) field.....85
Online Statistics Updation Interval
field (Logs)................................142
operating temperature .....................34
operating voltage
T1 models....................................26
Operator (ISDN) field
E1/ISDN....................................104
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Index
T1/ISDN ..................................... 96
Optimization Factor field................ 87
Options (callee, statistics, logs) field
.................................................. 269
Options (caller, statistics, logs) field
.................................................. 269
Options, From Details (RADIUS
Attributes) field......................... 163
Options, From Details (SMTP logs)
field........................................... 138
Options, To Details (RADIUS
Attributes) field......................... 163
Options, To Details (SMTP logs) field
.................................................. 138
Others, Priorities (Ethernet/IP params,
802.1p) field................................ 68
out of band, DTMF......................... 79
Outbound Digits Received (call
progress) field........................... 261
Outbound Digits Received (statistics,
logs) field.................................. 267
Outbound Digits Received(SMTP
logs) field.................................. 137
Outbound Digits Sent (call progress)
field........................................... 261
Outbound Digits Sent (RADIUS
Attributes) field......................... 162
Outbound Digits Sent (SMTP logs)
field........................................... 138
Outbound Digits Sent (statistics, logs)
field........................................... 267
Outbound Digits Sent and DTMF
"Out of Band" ........................... 138
Outbound Phonebook Entries List
icon
E1.............................................. 216
T1.............................................. 173
Outbound Phonebook entries, list
E1.............................................. 218
T1.............................................. 175
outbound vs. inbound phonebooks
E1.............................................. 214
T1.............................................. 171
Out-of-Band DTMF and Outbound
Digits Sent ................................ 138
Output Gain field............................ 78
output level, fax tones..................... 80
Packet Prioritization 802.1p
(Ethernet/IP parameters) .............67
packet priority and DiffServ............71
packetization (RTP), ranges &
increments.................................295
packetization rates
coder options and ......................294
Packets Lost (call progress) field ..259
Packets Lost (RADIUS Attributes)
field ...........................................162
Packets Lost (SMTP logs) field ....137
Packets lost (statistics, logs) field .268
Packets Received (call progress) field
..................................................259
Packets Received (RADIUS
Attributes) field.........................161
Packets Received (SMTP logs) field
..................................................136
Packets received (statistics, logs) field
..................................................268
Packets Sent (call progress) field ..259
Packets Sent (RADIUS Attributes)
field ...........................................161
Packets Sent (SMTP logs) field ....136
Packets sent (statistics, logs) field.268
packets, consecutive lost.................88
Parallel H.245 field .......................113
parameters tracked by console ......143
Password (proxy server) field .......116
Password (SMTP) field.................135
password, lost/forgotten........316, 320
password, setting...........................316
web browser GUI......................320
patents..............................................2
PBX characteristics, variations in
E1..............................................249
T1..............................................207
PBX interaction
E1 models....................................17
T1 models......................................9
PC, command
COM port assignment (detailed).44
PCM Law field
E1..............................................106
T1................................................98
Permissions (SNMP) field ............123
personnel requirement
for rack installation .....................34
383
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to lift during installation ............. 35
MultiVOIP User Guide
Phonebooks, objectives &
to lift unit during installation ...... 31
Phone Book Version
considerations
E1..............................................209
Ping Size in Bytes (Link
System Info............................... 167
Phone Number (Voice/FAX –
AutoCall/Offhook Alert) field .... 85
Phone Signaling Tones & Cadences
.................................................. 124
phone switch types
ISDN implementations in ......... 107
phone/IP details
importance of writing down........ 53
phonebook
FTP remote file transfers .......... 324
phonebook configuration................ 50
phonebook configuration (remote) 324
Phonebook Configuration icon
E1.............................................. 216
T1.............................................. 173
Phonebook Configuration Procedure
E1.............................................. 215
T1.............................................. 172
Phonebook Configuration screen
T1.............................................. 172
phonebook entries, coordinating
E1.............................................. 215
T1.............................................. 172
phonebook icons
E1.............................................. 216
T1.............................................. 173
phonebook keyboard shortcuts
E1.............................................. 217
T1.............................................. 174
phonebook objectives &
considerations
E1.............................................. 214
phonebook pulldown menu
E1.............................................. 217
T1.............................................. 174
phonebook sidebar menu
E1.............................................. 217
T1.............................................. 174
phonebooks, inbound vs. outbound
E1.............................................. 214
T1.............................................. 171
phonebooks, objectives &
Management) field....................278
Pings per Test (Link Management)
field ...........................................278
pinout
BRI connector ...........................356
command cable .........................353
ethernet cable ............................353
T1/E1 connector........................354
Voice/FAX connector ...............354
Polling Interval (SPP Call Signaling)
field ...........................................119
pop-ups
allowing with Web GUI..............74
pop-ups and Web GUI ..................334
Port (H.323 Gatekeepers, Statistics,
Servers) field.............................291
Port (SIP Proxies, Statistics, Servers)
field ...........................................292
Port (SPP Registrars, Statistics,
Servers) field.............................293
Port (SPP, General Options) field .118
Port field (Registered Gateway
Details)......................................289
Port field, SysLog Server..............142
Port Number (proxy server) field..115
Port Number (SMTP) field ...........135
port number (SNMP) field ............123
Port Number field, SPP (Outbound
Phonebook)
E1..............................................224
T1..............................................181
power consumption
E1 models....................................26
power frequency
E1 models....................................26
Prefix Matched (call progress) field
..................................................261
Prefix Matched (RADIUS Attributes)
field ...........................................162
Prefix Matched (SMTP logs) field 137
prerequisites
for technical configuration..........53
PRI
considerations
T1.............................................. 171
ISDN implementations..............107
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Index
RADIUS Accounting parameters,
customizing...............................161
RADIUS accounting support
Primary Proxy (SIP Call Signaling)
field........................................... 115
Primary Registrar (SPP Call
Signaling) field ......................... 119
Priority (H.323 Gatekeepers,
EI models ....................................19
TI models ....................................11
RADIUS screen field
Enable Accounting....................160
Retransmission Interval.............160
RADIUS screen fields
Accounting Port ........................160
Server Address..........................160
RAS Port (H.323 Call Signaling) field
..................................................110
RAS TTL Value (Gatekeeper RAS)
field ...........................................111
Receive Slip (E1 Stats) field .........286
Receive Slip (T1 Stats) field .........283
Receive Timeslot 16 Alarm Indication
Signal (E1 stats) field................285
Receive Timeslot 16 Loss of
MultiFrame Alignment (E1 stats)
field ...........................................286
Receive Timeslot 16 Loss of Signal
(E1 stats) field...........................285
Received (RTCP Packets, IP Stats)
field ...........................................275
Received (RTP Packets, IP Stats) field
..................................................275
Received (TCP Packets, IP Stats) field
..................................................274
Received (Total Packets, IP Stats)
field ...........................................273
Received (UDP Packets, IP Stats)
field ...........................................274
Received with Errors (RTCP Packets,
IP Stats) field.............................275
Received with Errors (RTP Packets,
IP Stats) field.............................275
Received with Errors (TCP Packets,
IP Stats) field.............................274
Received with Errors (Total Packets,
IP Stats) field.............................274
Received with Errors (UDP Packets,
IP Stats) field.............................274
Recipient Address (email logs)field
..................................................135
recovering voice packets.................82
Red Alarm (E1 stats) field ............284
Statistics, Servers) field ............ 291
Priority Levels (802.1p)............ 67, 68
product CD ..................................... 27
use in software installation ......... 39
Product CD
E1 models ................................... 19
T1 models ................................... 11
product family................................... 8
Program Menu items..................... 297
Protocol Type (outbound phonebook)
E1.............................................. 221
T1.............................................. 178
Proxy Domain Name / IP Address
field........................................... 115
Proxy Polling Interval (SIP Call
Signaling) field ......................... 116
Proxy/NAT Device Parameters –
Public IP Address ..................... 120
PSTN failover feature
Alternate Routing, and.............. 183
E1 models ................................... 18
T1 models ................................... 10
Pulse Density Violation (T1 stats)
field........................................... 282
Pulse Shape Level field
E1.............................................. 105
T1................................................ 97
Q.931 Port Number (outbound
phonebook) field
T1.............................................. 179
quality-of-service
E1................................................ 18
T1................................................ 10
rack mounting
grounding.................................... 34
safety..................................... 31, 34
rack mounting instructions.............. 33
rack mounting procedure ................ 35
rack, equipment
weight capacity of....................... 34
rack-mountable voip models........... 31
RADIUS accounting parameters,
customizing............................... 163
385
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Index
MultiVOIP User Guide
Red Alarm (T1 stats) field ............ 281
Regional Parameter definitions.... 126,
127, 128, 129
Remote Configuration/Command
Modem
setup for ..............................59, 125
remote control/configuration
Regional Parameter fields
Cadence .................................... 128
Country/Region (tone schemes) 126
Custom (tones).......................... 128
Frequency 1 .............................. 127
Frequency 2 .............................. 127
Gain 1 ....................................... 127
Gain 2 ....................................... 127
Pulse Generation Ratio ............. 128
type (of tone)............................. 127
Regional Parameters fields
web GUI and.............................337
Remote Device is [legacy voip]
(Outbound Phonebook)
T1..............................................181
Remote Device is [legacy] MultiVOIP
checkbox ...................................224
remote phonebook configuration ..324
remote voip configuration...............50
Remote Voip Management ...........121
Remove Prefix (inbound) field
Country Selection for Built-In
E1..............................................228
T1..............................................186
Remove Prefix (outbound) field
E1..............................................221
T1..............................................178
repair procedures for customers U.S.
& Canada ..................................344
Reply-To Address (email logs)field
..................................................135
Requires Authentication (SMTP) field
..................................................134
Re-Registration Time (proxy server)
..................................................116
Resolutions (MultiVOIP
Modem.................................. 128
regional parameters, setting .......... 124
Register Duration field (Registered
Gateway Details) ...................... 289
Register with Gatekeeper (H.323 Call
Signaling) field ......................... 109
Registered Gateway Details
(Statistics) screen, accessing..... 289
Registered Gateway Details
‘Statistics’ function......... 287, 289
Registered Gateway Details screen289
Registered Gateway Details screen
fields
Description................................ 289
IP Address................................. 289
No. of Entries............................ 289
Port ........................................... 289
Register Duration...................... 289
Status ........................................ 289
Registered Gateway Details screen
fields: ........................................ 289
Registrar IP Address field............. 119
Registrar Options (SPP Call
Signalining fields ................... 119
Registrar Port field........................ 119
Registration Option Parameters
(Inbound Phone Book)
E1.............................................. 230
T1.............................................. 188
remote configuration modem
troubleshooting) ............................7
Response Timeout (Link
Management) field....................278
Retransmission (SPP, General
Options) field ............................118
Retransmission Interval (RADIUS
screen) field...............................160
Retrieve Sequence.................145, 148
RFC 2782........................................72
RFC 2833........................................79
RFC 3087......................................180
RFC 3489......................................155
RFC2474.........................................70
RFC2597.........................................70
RFC2833.......................137, 261, 267
RFC3246.........................................70
RFC768.........................................359
RFC793.........................................359
ring cadences, custom ...................132
Ring Count forwarding condition
MVP-2410 .................................. 37
MVP3010.................................... 37
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Index
E1.............................................. 230
T1.............................................. 188
ring tone, custom .......................... 131
ring-tones...................................... 130
Round Trip Delay (Link
Management) field.................... 279
Round Trip Delay field
E1.............................................. 226
T1.............................................. 183
RTP packetization, ranges &
Set Password (program menu option) ,
command...................................316
Set Password (web browser GUI) ,
command...................................320
Set Password option description
(MultiVOIP program menu) .....298
Set Regional Parameters ...............124
Set SMTP Parameters ...................133
Set SNMP Parameters...................121
Set Supplementary Services
increments................................. 295
RTP Parameters screen................. 295
Safety Recommendations for Rack
Installations................................. 34
safety warnings............................... 31
Safety Warnings Telecom.......... 31
Save Setup command.................... 168
saving configuration ..................... 168
user ........................................... 313
Saving the MultiVOIP Configuration
.................................................. 168
savings on toll calls
Parameters.................................144
Set T1/E1/ISDN Parameters ...........89
Set Voice/FAX Parameters .............75
setting Ethernet/IP parameters ........65
setting password............................316
web browser GUI......................320
setting RTP Parameters.................295
setting user defaults.......................313
setup, saving..................................168
user............................................313
setup, saving user values...............313
Shared Secret (RADIUS screen) field
..................................................160
signaling cadences.........................124
Signaling Port (H.323 Call Signaling)
field ...........................................109
Signaling Port (SIP Call Signaling)
field ...........................................114
signaling tones ..............................124
Silence Compression (call progress)
field ...........................................263
Silence Compression (RADIUS
Attributes) .................................163
Silence Compression (SMTP logs)138
Silence Compression field ..............82
Single-Port Protocol, general
E1.............................................. 209
T1.............................................. 171
scale-ability
E1................................................ 17
T1.................................................. 9
Select All (RADIUS Attributes) field
.................................................. 161
Select All (SMTP logs) field ........ 136
Select Attributes (RADIUS) button
.................................................. 160
Select Channel field........................ 78
Select Channel, Supplementary
Services field ............................ 147
Selected Coder field........................ 81
Server Address (RADIUS screen)
field........................................... 160
Server Details (call progress) field 261
Server Details (RADIUS Attributes)
field........................................... 162
Server Details (SMTP logs) field.. 138
Server Details (statistics, logs) field
.................................................. 268
Service Records .............................. 73
Set Baud Rate ............................... 164
Set ISDN Parameters .................... 107
Set Log Reporting Method ........... 140
description
E1................................................18
T1................................................10
SIP
compatibility
E1 models................................18
T1 models................................10
SIP Call Signaling Parameter
definitions .................114, 115, 116
SIP Call Signaling screen fields
Password (proxy server)............116
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Proxy Domain Name / IP Address
MultiVOIP User Guide
SMTP parameters, accessing ........133
SMTP parameters,setting..............133
SMTP port, standard..................135
SMTP prerequisites.........................57
SMTP, enabling ............................133
SNMP (log reporting type) button 142
SNMP agent program......................51
SNMP agent, enabling ..................121
SNMP Parameter fields
Address .....................................123
Community Name (2) ...............123
Community Name 1..................123
Enable SNMP Agent.................123
Permissions (1)..........................123
Permissions (2)..........................123
Port Number..............................123
SNMP Parameters, setting ............121
software
.............................................. 115
Proxy Polling Interval............... 116
Re-Registration Time (proxy
server) ................................... 116
Signaling Number (proxy server)
.............................................. 115
TTL Value ................................ 116
Use SIP Proxy........................... 114
User Name (proxy server)......... 115
SIP Fields (Outbound Phonebook)
E1.............................................. 223
T1.............................................. 180
SIP incompatibility with H.450
Supplementary Services............ 144
E1................................................ 18
T1................................................ 10
SIP Port Number field
E1.............................................. 223
T1.............................................. 180
SIP port number, standard
uninstalling (detailed) .................46
updates ........................................51
software (MultiVOIP)
E1.............................................. 223
T1.............................................. 180
SIP Proxies (Statistics, Servers)
IP Address................................. 292
Port ........................................... 292
Status ........................................ 292
Type.......................................... 292
SIP Proxy Parameters ................... 114
SIP URL field
E1.............................................. 223
T1.............................................. 180
SMTP (log reporting type) button. 142
SMTP logs by email, illustration .. 139
SMTP Parameters definitions....... 135
SMTP Parameters fields
uninstalling................................321
software configuration
summary......................................39
software installation
detailed........................................39
software loading..............................39
software version numbers ...............41
software, MultiVOIP
moving around in ........................64
software, MultiVOIP
screen-surfing in..........................64
Solving Common Connection
Problems .....................................63
sound quality, improving ................82
specifications
Enable SMTP............................ 134
Login Name .............................. 134
Mail Server IP Address............. 135
Mail Type ................................. 135
Number of Days........................ 135
Number of Records................... 135
Password................................... 135
Port Number ............................. 135
Recipient Address..................... 135
Reply-To Address..................... 135
Requires Authentication ........... 134
Subject ...................................... 135
E1 models....................................26
T1 models....................................25
SPP Call Signaling screen
Mode (SPP Protocol) ................118
SPP Call Signaling screen fields
Alternate Registrars 1 and 2......119
Client Options...........................119
General Options ........................118
Keep Alive ................................119
Max Retransmission (SPP, General
Options).................................118
Polling Interval..........................119
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Primary Registrar...................... 119
Registrar IP Address................. 119
Registrar Options...................... 119
Registrar Port............................ 119
Retransmission (SPP, General
Status Freeze Signalling Active (T1
stats) field..................................281
STUN clients and servers..............155
STUN support
E1 models....................................19
T1 models....................................11
Subject (email logs) field ..............135
Supplementary (Telephony) Services
E1................................................23
T1................................................15
Supplementary Services
Alerting Party............151, 152, 153
Call Hold...................................145
Call Hold Enable.......................148
Call Name Identification...........145
Call Transfer .............................145
Call Transfer Enable .................147
Call Waiting..............................145
Call Waiting Enable..................148
Caller Name Identification Enable
..............................................149
Calling Party .............................150
Enable Call Hold.......................148
Enable Call Transfer .................147
Enable Call Waiting..................148
Enable Caller Name Identification
..............................................149
Hold Sequence ..........................148
Retrieve Sequence.....................148
Select Channel ..........................147
Transfer Sequence.....................147
Supplementary Services Info
Options) ................................ 118
Signaling Port (SPP, General
Options) ................................ 118
SPP Fields (Outbound Phonebook)
E1.............................................. 224
T1.............................................. 181
SPP Fields (Phonebook Configuration
screen)....................................... 118
SPP Registrars (Statistics, Servers)
IP Address................................. 293
Port ........................................... 293
Type.......................................... 293
SPP Registrarss (Statistics, Servers)
Status ........................................ 293
SPP, general description
E1................................................ 18
T1................................................ 10
SPP, strengths & compatibilities of
E1................................................ 18
T1................................................ 10
SRV record ..................................... 73
ST interface (ISDN-BRI)
description ................................ 357
Start Date, Time (RADIUS
Attributes) field......................... 161
Start Date, Time (SMTP logs) field
.................................................. 136
Start Date,Time (statistics, logs) field
.................................................. 266
Start Now command (Link
logs for ......................................270
Supplementary Services Parameter
buttons
Management) button................. 278
Status (H.323 Gatekeepers, Statistics,
Servers) field............................. 291
Status (SIP Proxies, Statistics,
Servers) field............................. 292
Status (SPP Registrars, Statistics,
Servers) field............................. 293
Status (statistics, logs) field .......... 266
Status field (Registered Gateway
Details)...................................... 289
Status Freeze Signalling Active (E1
stats) field ................................. 285
Copy Channel............................154
Default ......................................154
Supplementary Services Parameter
Definitions 147, 148, 149, 150, 151,
152, 153, 154
Supplementary Services Parameter
fields
Call Waiting Enable..................148
Hold Sequence ..........................148
Retrieve Sequence.....................148
Supplementary Services Parameter
fields
Call Hold Enable.......................148
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Call Transfer Enable................. 147
Select Channel .......................... 147
Supplementary Services Parameter
fields
Call Name Identification Enable149
Supplementary Services Parameter
fields
Calling Party............................. 150
Supplementary Services Parameter
fields
Allowed Name Types ............... 150
Supplementary Services Parameter
fields
Alerting Party............................ 151
Supplementary Services Parameter
fields
Allowed Name Types ............... 151
Supplementary Services Parameter
fields
MultiVOIP User Guide
T1................................................14
SysLog client programs
availability.................................340
features & presentation types....342
SysLog functionality
E1................................................22
T1................................................14
SysLog server
E1................................................22
T1................................................14
SysLog Server Enable field...........142
SysLog Server function
as added feature.........................340
capabilities of............................342
enabling.....................................341
location of .................................340
SysLog Server IP Address field ....142
SysLog Server, enabling ...............141
System Information screen
Busy Party................................. 152
Supplementary Services Parameter
fields
Allowed Name Types ............... 152
Supplementary Services Parameter
fields
Connected Party........................ 153
Supplementary Services Parameter
fields
for op & maint...........................253
System Information screen, accessing
..................................................165
System Information update interval,
setting........................................165
for op & maint...........................255
T1 model descriptions.......................9
T1 Parameter definitions....92, 93, 94,
95, 96, 97, 98
Allowed Name Types ............... 153
Supplementary Services Parameter
fields
Caller ID ................................... 154
Supplementary Services Parameters
fields
Transfer Sequence .................... 147
Supplementary Services Parameters
screen, accessing....................... 144
Supplementary Services parameters,
setting........................................ 144
Supplementary Services, incompatible
with SIP .................................... 144
E1................................................ 18
T1................................................ 10
support, technical.......................... 346
switch types (phone) and ISDN-PRI
.................................................. 107
SysLog client
Answer Delay (Enable)...............94
Answer Delay Timer...................94
FXS Current Detect Timer..........95
T1 Parameter definitions (FXS
Supervision)
Tone Detection............................95
T1 Parameter fields
CAS Protocol ..........................93
FXS Options – No Response
Timer...................................93
No Response Timer (FXS
Options)...............................93
Answer Tones .............................95
Available Tones (List) ................95
Caller ID Enable..........................97
Calling Number Prefix................96
Calling Number Prefix (Caller ID)
................................................97
Calling Number Suffix................96
E1................................................ 22
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Index
Calling Number Suffix (Caller ID)
................................................ 97
Clocking...................................... 97
Country....................................... 96
CRC Check................................. 92
Detect Flash Hook ...................... 97
Detection Time ........................... 97
Enable Caller ID ......................... 96
Enable ISDN-PRI ............... 95, 103
Frame Format.............................. 92
Generation Time ......................... 97
Line Build-Out............................ 97
Line Coding ................................ 98
Long-Haul Mode ........................ 92
Operator...................................... 96
PCM Law.................................... 98
Pulse Shape Level....................... 97
T1/E1/ISDN................................ 92
Terminal Network............... 95, 103
Tone Detection (Enable)............. 95
Yellow Alarm Format................. 98
T1 Parameters screen...................... 91
T1 Statistics field definitions 282, 283
T1 Statistics fields
Bipolar Violation ...................... 283
Frame Search Restart Flag........ 282
Line Loopback Activation Signal
.............................................. 282
Loss of MultiFrame Alignment 282
Pulse Density Violation ............ 282
Receive Slip.............................. 283
Transmit Data Underrun........... 282
Transmit Line Open.................. 282
Transmit Slip ............................ 282
Transmit Slip Negative............. 282
Yellow Alarm ........................... 282
T1 telephony parameters................. 55
T1/E1 connector pinout ................ 354
T1/E1/ISDN field
T1..............................................180
TDM Routing Option (Ethernet/IP
Parameters) field .........................73
technical configuration
prerequisites to............................53
summary......................................50
technical configuration procedure
detailed........................................60
summary......................................59
technical support ...........................346
telco authorities and ISDN............107
telecom safety warnings.............31
telephony signaling cadences........124
telephony signaling tones..............124
telephony toning schemes .............130
temperature
operating .....................................34
Terminal Network field
E1/ISDN....................................103
T1/ISDN......................................95
terminal/network settings, voip and
PBX
E1/ISDN....................................103
T1/ISDN......................................95
timeout interval
voips under H.323 gatekeeper...111
voips under SIP proxy server....116
Timer Interval between Pings (Link
Management) field....................278
To (gateway, statistics, logs) field.266
toll-call savings
E1..............................................209
T1..............................................171
toll-free access (T1)
to remote PSTN.............................9
within voip network ......................9
toll-free access (within voip network)
E1................................................17
T1..................................................9
Tone Detection (FXO answer
E1.............................................. 100
T1................................................ 92
T1/E1/ISDN Parameters screen,
accessing..................................... 89
T1/E1/ISDN parameters, setting..... 89
table-top voip models ..................... 31
TCP/UDP compared
supervision criteria, E1) field....103
Tone Detection (FXO answer
supervision criteria, T1) field......95
Tone Pair (custom) field ...............131
tones, signaling .............................124
Total Digits (outbound) field
E1.............................................. 223
IP Statistics context................... 273
E1..............................................221
T1..............................................178
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MultiVOIP User Guide
trace on/off (logging).................... 143
Transfer Sequence ................ 145, 147
Transmit Data Overflow (E1 stats)
field........................................... 285
Transmit Data Overflow (T1 stats)
field........................................... 281
Transmit Data Underrun (E1 stats)
field........................................... 286
Transmit Data Underrun (T1 stats)
field........................................... 282
Transmit Line Open (E1 stats) field
.................................................. 286
Transmit Line Open (T1 stats) field
.................................................. 282
Transmit Line Short (E1 stats) field
.................................................. 285
Transmit Line Short (T1 stats) field
.................................................. 281
Transmit Slip (E1 stats) field........ 286
Transmit Slip (T1 stats) field........ 282
Transmit Slip Negative (E1 stats) field
.................................................. 286
Transmit Slip Negative (T1 stats) field
.................................................. 282
Transmit Slip Positive (E1 stats) field
.................................................. 285
Transmit Slip Positive (T1 stats) field
.................................................. 281
Transmitted (RTCP Packets, IP Stats)
field........................................... 275
Transmitted (RTP Packets, IP Stats)
field........................................... 275
Transmitted (TCP Packets, IP Stats)
field........................................... 274
Transmitted (Total Packets, IP Stats)
field........................................... 273
Transmitted (UDP Packets, IP Stats)
field........................................... 274
Transport Protocol (SIP) field
Type (H.323 Gatekeepers, Statistics,
Servers) field.............................291
Type (of tone, Regional Parameters)
field ...........................................127
Type (SIP Proxies, Statistics, Servers)
field ...........................................292
Type (SPP Registrars, Statistics,
Servers) field.............................293
Type-of-Service IP header field &
DiffServ.......................................71
U interface (ISDN-BRI)
description.................................357
UDP multiplexed (H.323 Annex E)
field ...........................................113
UDP/TCP compared
E1..............................................223
IP Statistics context...................273
T1..............................................180
unconditional forwarding
E1..............................................229
T1..............................................187
Uninstall (program menu option) ,
command...................................321
Uninstall option description
(MultiVOIP program menu) .....298
uninstalling MultiVOIP software...46,
321
unobtainable tone, custom.............131
unobtainable tones.........................130
unpacking
MVP2410....................................32
MVP3010....................................32
Up Time
System Info.......................167, 254
update interval (logging)...............141
updated firmware, obtaining .........300
upgrade
E1................................................17
T1..................................................9
upgrade card (4-to-8 channel)
installation.................................361
Upgrade Software option description
MultiVOIP program menu........298
upgrade, firmware.........................304
uploads vs. downloads (FTP)........324
Use Fast Start (H.323 Call Signaling)
field ...........................................109
E1.............................................. 223
T1.............................................. 180
triggering log report email ....... 135
Troubleshooting Resolutions for
MultiVOIPs .................................. 7
TTL Value (SIP Call Signaling) field
.................................................. 116
Turn Off Logs field....................... 142
Type (call, statistics, logs) field.... 266
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Index
Use Gatekeeper (Outbound
AutoCall/Offhook Alert ........83, 84
AutoCall/Offhook Alert fields ...83,
84
Phonebook) field
E1.............................................. 222
T1.............................................. 179
Use Online Alternate Gatekeeper List
(H.323 Call Signaling) field...... 111
Use Proxy (SIP) field
E1.............................................. 223
T1.............................................. 180
Use Registrar field (Outbound
Phonebook)
E1.............................................. 224
T1.............................................. 181
Use SIP Proxy field ...................... 114
Use TDM Routing for Intra-Gateway
Calls............................................ 73
user default configuration, creating
.................................................. 169
user defaults, downloading ........... 313
user defaults, setting ..................... 313
user name
Generate Local Dial Tone...........84
Offhook Alert Timer ...................85
Out-of-Band Mode (DTMF) .......78
Phone Number (Auto Call/Offhook
Alert).......................................85
Voice/FAX Parameter fields
Copy Channel..............................78
Default ........................................78
DTMF Gain.................................78
DTMF Gain (High Tones) ..........78
DTMF Gain (Low Tones)...........78
DTMF In/Out of Band ................78
Duration (DTMF)........................78
Input Gain ...................................78
Output Gain.................................78
Select Channel ............................78
Voice Gain ..................................78
Voice/FAX Parameter fields
Windows GUI........................... 316
User Name (proxy server) field .... 115
user values (software), saving....... 313
variations in PBX characteristics
E1.............................................. 249
T1.............................................. 207
version numbers (software) ............ 41
version, firmware.......................... 304
VLAN ID (Ethernet/IP Parameters)
field............................................. 68
Voice Coder (call progress) field.. 258
Voice coder (statistics, logs) field. 267
voice delay................................ 86, 87
Voice Gain field.............................. 78
voice packets
recovering lost/corrupted............ 82
voice packets, consecutive lost ....... 88
voice packets, delayed .............. 86, 87
voice packets, re-assembling .......... 80
voice quality, improving................. 82
voice quality, versus delay.............. 87
Voice/FAX connector pinout........ 354
Voice/FAX Parameter definitions.. 87,
88
Fax Enable ..................................79
Voice/FAX Parameter fields
Max Baud Rate (Fax)..................79
Voice/FAX Parameter fields
Fax Volume.................................80
Voice/FAX Parameter fields
Jitter Value (Fax) ........................80
Voice/FAX Parameter fields
Mode (Fax)..................................80
Voice/FAX Parameter fields
Silence Compression...................82
Voice/FAX Parameter fields
Echo Cancellation .......................82
Voice/FAX Parameter fields
Forward Error Correction............82
Voice/FAX Parameter fields
Dynamic Jitter Buffer..................86
Voice/FAX Parameter fields
Minimum Jitter Value.................86
Voice/FAX Parameter fields
Maximum Jitter Value ................87
Voice/FAX Parameter fields
Optimization Factor ....................87
Voice/FAX Parameter fields
Voice/FAX Parameter Definitions. 78,
79, 80, 81, 82, 86
Automatic Disconnection............88
Voice/FAX Parameter fields
Voice/FAX Parameter fields
Jitter Value..................................88
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MultiVOIP User Guide
Voice/FAX Parameter fields
Call Duration .............................. 88
Voice/FAX Parameter fields
prerequisite local assigning of IP
address...................................335
video useability .........................334
Consecutive Packets Lost ........... 88
Voice/FAX Parameter fields
web GUI
Java and.....................................337
remote control/configuration and
..............................................337
Web GUI and pop-ups ..................334
web GUI vs. local Windows GUI
comparison................................335
web GUI, logging and...................336
Web GUI, Windows GUI compared
E1................................................20
T1................................................12
weight
Network Disconnection .............. 88
Voice/FAX Parameters screen,
accessing..................................... 75
Voice/FAX parameters, setting....... 75
voip email account........................ 134
voip management, remote............. 121
Voip Media PHB field.................... 70
VoIP Media Priority (Ethernet/IP
parameters) field ......................... 68
voip software
host PC........................................ 51
voip system example, conceptual (E1)
calls to remote PSTN................ 211
foreign calls, national rates....... 213
voip site to voip site.................. 210
voip system example, digital &
analog, with phonebook details
E1.............................................. 238
T1.............................................. 195
voip system example, digital only,
with phonebook details
E1 models....................................26
T1 models....................................25
weight loading
in rack .........................................34
weight of unit
lifting precaution.........................31
personnel requirement.................31
Well Known Ports.........................359
well-known port number, SMTP
..................................................135
well-known port, gatekeeper
E1.............................................. 231
T1.............................................. 189
voip(E1)
basic functions of........................ 18
voip(T1)
registration ................................110
well-known port, H.323 params
E1..............................................222
T1..............................................179
well-known port, Q.931 params,
H.323.........................................109
well-known port, SIP
basic functions of........................ 10
voltage, operating
E1 models ................................... 26
warnings, safety.............................. 31
warranty........................................ 344
web browser GUI and logs ........... 141
web browser GUI, enabling............ 74
web browser interface
E1..............................................223
T1..............................................180
well-known port, SNMP ...............123
Windows GUI, Web GUI compared
E1................................................20
T1................................................12
Yellow Alarm (E1 stats) field .......285
Yellow Alarm (T1 stats) field .......282
Yellow Alarm Format field (E1)...106
Yellow Alarm Format field (T1).....98
browser version requirement ... 334,
338
general ...................................... 334
Java requirement....................... 334
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S000384A
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