MultiVOIPTM SS
Survivable SIP Gateway & Server
User Guide for Voice/IP Gateways
Models: MVP210-SS
MVP410-SS
MVP810-SS
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CONTENTS
CHAPTER 1: OVERVIEW.......................................................................................7
ABOUT THIS MANUAL...............................................................................................8
INTRODUCTION TO ANALOG MULTIVOIPS WITH SIP SURVIVABILITY FEATURES
(MVP-210SS/410SS/810SS) ..................................................................................12
MultiVOIP Front Panel LEDs ............................................................................17
COMPUTER REQUIREMENTS ....................................................................................19
SPECIFICATIONS ......................................................................................................20
INSTALLATION AT A GLANCE ..................................................................................21
RELATED DOCUMENTATION....................................................................................21
CHAPTER 2: QUICK START INSTRUCTIONS.................................................22
INTRODUCTION........................................................................................................23
MULTIVOIP STARTUP TASKS .................................................................................24
Phone/IP Details *Absolutely Needed* Before Starting the Installation............25
Gather IP Information...................................................................................................25
Gather Telephone Information .....................................................................................26
Obtain Email Address for VOIP (for email call log reporting).....................................27
Config Info CheckList..................................................................................................28
Identify Remote VOIP Site to Call...............................................................................29
Identify MVP-SS Unit’s Role in SIP VOIP System.....................................................29
Placement ...........................................................................................................30
Command/Control Computer Setup (Specs & Settings) .....................................30
Quick Hookup for MVP410-SS & MVP810-SS...................................................31
Quick Hookup for MVP210-SS ...........................................................................32
Load MultiVOIP Control Software onto PC.......................................................33
Phone/IP Starter Configuration..........................................................................34
Phonebook Starter Configuration (with remote voip).........................................40
Outbound Phonebook ...................................................................................................40
Inbound Phonebook......................................................................................................44
Phonebook Tips ..................................................................................................47
Phonebook Example ...........................................................................................51
Connectivity Test ................................................................................................56
Troubleshooting..................................................................................................60
CHAPTER 3: MECHANICAL INSTALLATION AND CABLING...................61
INTRODUCTION........................................................................................................62
SAFETY WARNINGS .................................................................................................62
Lithium Battery Caution .....................................................................................62
Safety Warnings Telecom....................................................................................62
UNPACKING YOUR MULTIVOIP..............................................................................63
Unpacking the MVP-410SS/810SS......................................................................64
Unpacking the MVP210-SS.................................................................................65
Safety Recommendations for Rack Installations.................................................67
19-Inch Rack Enclosure Mounting Procedure....................................................68
CABLING PROCEDURE FOR MVP-410SS/810SS......................................................69
Cabling Procedure for MVP210-SS....................................................................73
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MultiVOIP User Guide
CHAPTER 4: SOFTWARE INSTALLATION .....................................................77
INTRODUCTION........................................................................................................78
LOADING MULTIVOIP SOFTWARE ONTO THE PC....................................................78
UN-INSTALLING THE MULTIVOIP CONFIGURATION SOFTWARE.............................85
CHAPTER 5: TECHNICAL CONFIGURATION................................................88
CONFIGURING THE MULTIVOIP..............................................................................89
LOCAL CONFIGURATION..........................................................................................92
Pre-Requisites.....................................................................................................92
IP Parameters................................................................................................................92
Telephony Interface Parameters ...................................................................................93
SMTP Parameters (for email call log reporting)...........................................................94
Config Info CheckList..................................................................................................95
Local Configuration Procedure (Summary) .......................................................96
Local Configuration Procedure (Detailed).........................................................97
Modem Relay ....................................................................................................124
CHAPTER 6: T1 PHONEBOOK CONFIGURATION ......................................205
T1 VERSUS E1 TELEPHONY ENVIRONMENTS.........................................................206
CONFIGURING T1 (NAM) TELEPHONY MULTIVOIP PHONEBOOKS......................206
T1 PHONEBOOK EXAMPLES...................................................................................222
3 Sites, All-T1 Example.....................................................................................222
Configuring Mixed Digital/Analog VOIP Systems ...........................................228
Call Completion Summaries .............................................................................237
Variations in PBX Characteristics....................................................................240
CHAPTER 7: E1 PHONEBOOK CONFIGURATION ......................................241
E1 VERSUS T1 TELEPHONY ENVIRONMENTS.........................................................242
E1-STANDARD INBOUND AND OUTBOUND MULTIVOIP PHONEBOOKS.................242
Free Calls: One VOIP Site to Another.............................................................243
Local Rate Calls: Within Local Calling Area of Remote VOIP.......................244
National Rate Calls: Within Nation of Remote VOIP Site...............................246
Inbound versus Outbound Phonebooks.............................................................247
PHONEBOOK CONFIGURATION PROCEDURE...........................................................251
E1 PHONEBOOK EXAMPLES...................................................................................262
3 Sites, All-E1 Example ....................................................................................262
Configuring Digital & Analog VOIPs in Same System.....................................269
Call Completion Summaries.......................................................................................277
Variations in PBX Characteristics....................................................................280
International Telephony Numbering Plan Resources.......................................281
CHAPTER 8: OPERATION AND MAINTENANCE ........................................283
OPERATION AND MAINTENANCE ...........................................................................284
SIP Server Endpoint Statistics screen...............................................................284
System Information screen................................................................................288
Statistics Screens ..............................................................................................291
About Call Progress..........................................................................................291
About Logs........................................................................................................299
4
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MultiVOIP User Guide
ContentsVOIP
About IP Statistics.............................................................................................306
About Link Management...................................................................................311
About Registered Gateway Details ...................................................................314
About Alternate Server Statistics ......................................................................317
About Packetization Time .................................................................................321
MULTIVOIP PROGRAM MENU ITEMS .....................................................................324
Configuration Option........................................................................................326
Configuration Port Setup..................................................................................326
Date and Time Setup.........................................................................................327
Obtaining Updated Firmware...........................................................................327
Implementing a Software Upgrade ...................................................................331
Identifying Current Firmware Version .......................................................................331
Downloading Firmware..............................................................................................332
Downloading Factory Defaults...................................................................................335
Downloading IFM Firmware............................................................................337
Setting and Downloading User Defaults ..........................................................341
Setting a Password (Windows GUI) .................................................................344
Setting a Password (Web Browser GUI) ..........................................................347
Un-Installing the MultiVOIP Software .............................................................348
Upgrading Software..........................................................................................350
FTP SERVER FILE TRANSFERS (“DOWNLOADS”)...................................................351
WEB BROWSER INTERFACE ...................................................................................361
SYSLOG SERVER FUNCTIONS ................................................................................367
CHAPTER 9 WARRANTY, SERVICE, AND TECH SUPPORT.....................370
LIMITED WARRANTY.............................................................................................371
REPAIR PROCEDURES FOR U.S. AND CANADIAN CUSTOMERS ...............................371
TECHNICAL SUPPORT ............................................................................................373
Contacting Technical Support ..........................................................................373
CHAPTER 10: REGULATORY INFORMATION ............................................374
EMC, Safety, and R&TTE Directive Compliance.............................................375
FCC DECLARATION...............................................................................................375
Industry Canada ...............................................................................................376
FCC Part 68 Telecom.......................................................................................376
Canadian Limitations Notice............................................................................377
WEEE Statement...............................................................................................378
APPENDIX A: CABLE PINOUTS......................................................................379
APPENDIX A: CABLE PINOUTS..............................................................................380
Command Cable ...............................................................................................380
Ethernet Connector...........................................................................................380
T1/E1 Connector...............................................................................................381
Voice/Fax Channel Connectors ........................................................................381
ISDN BRI RJ-45 Pinout Information ................................................................383
ISDN Interfaces: “ST” and “U” .....................................................................384
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Contents
MultiVOIP User Guide
APPENDIX B: TCP/UDP PORT ASSIGNMENTS............................................385
WELL KNOWN PORT NUMBERS.............................................................................386
PORT NUMBER ASSIGNMENT LIST.........................................................................386
INDEX .....................................................................................................................388
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MultiVOIP User Guide
Overview
Chapter 1: Overview
7
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Overview
MultiVOIP User Guide
About This Manual
This manual is about Voice-over-IP products made by Multi-Tech
Systems, Inc. It describes three analog MultiVOIP units with SIP-
survivability features, models MVP810SS, MVP410SS, and MVP210SS
These MultiVOIP units can inter-operate with other contemporary
analog MultiVOIP units (MVP130, MVP130FXS, MVP210, MVP410, and
MVP810), 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 in the MultiVOIP product family.
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).
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MultiVOIP User Guide
Overview
Digital MultiVOIP Products
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
9
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Overview
MultiVOIP User Guide
Analog MultiVOIP Products
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
MVP
810 SS
MVP
410SS
MVP
210SS
Function
analog voip; acts analog voip; acts
analog
as minimal SIP
proxy server
giving SIP
proxy
redundancy to
WAN
as minimal SIP
proxy server
giving SIP proxy
redundancy to
WAN
voip; acts
as minimal
SIP proxy
server
giving SIP
proxy
redundancy
to WAN
Capacity
Chassis/
Mounting rack
mount
8 channels
19” 1U
4 channels
2 channels
19” 1U
rack
mount
table-top
unit
10
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MultiVOIP User Guide
Overview
ISDN/BRI MultiVOIP Products
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.
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Overview
MultiVOIP User Guide
Introduction to Analog MultiVOIPs
with SIP Survivability Features
(MVP-210SS/410SS/810SS)
VOIP: The Free Ride. We proudly present Multi-Tech's MVP-
210SS/410SS/810SS MultiVOIP Voice-over-IP Gateways. These three
models allow voice/fax communication to be transmitted at no
additional expense over your existing IP network, which has ordinarily
been data only. To access this free voice and fax communication, you
simply connect the MultiVOIP to your telephone equipment and your
existing Internet connection. These analog MultiVOIPs inter-operate
readily with T1 or E1 MultiVOIP units.
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Voice/ Fax6
VoiceF/ ax7
Voi ec/ Fax8
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SX
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MT
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RS
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RC
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CV
SX
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CR
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R
R
C
V
X
GS
RS
V
X
S
G
R
Figure 1-1: MVP-410SS/810SS Chassis
Figure 1-2: MVP210SS Chassis
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MultiVOIP User Guide
Overview
Capacity. MultiVOIP model MVP810SS is an eight-channel unit, the
model MVP410SS is a four-channel unit, and the MVP210SS is a two-
channel unit. All three of these MultiVOIP units have a 10/100Mbps
Ethernet interface and a command port for configuration.
SIP Survivability. The MVP210SS, MVP410SS and MVP810SS have a
special capacity that reaches beyond ordinary voip functionality: they
can direct call traffic for phones connected to their channels or phones
connected to channels of other SIP gateways in the network (this is
basic SIP server functionality). The MVP-SS unit would normally be
located at a remote branch office served by a central SIP server (PBX) at
the organization’s main office. The MVP-SS is intended as a backup in
case the network’s main SIP server (often a PBX) fails or loses contact
with the group of gateways at the remote branch office. If the main SIP
server fails, the MVP-SS allows branch office phone users to call each
other and access the PSTN via POTS lines or a key telephone system.
Main Office
Central SIP Server
(Main PBX)
PSTN
Router
Internet
Branch Office
Router
LAN
Ordinary
SIP
Gateway
SIP Phone 1
SIP Phone 2
SIP Survivability
POTS
or KTS
Server &
Gateway
SIP Phone 3
PSTN
Figure 1-3: SIP Survivability MultiVOIP in system
A single MVP210SS, MVP410SS or MVP810SS can provide SIP server
functionality for as many as 500 other voip gateways. However, the
number of phone lines that these units support (4 for the MVP410SS; 8
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Overview
MultiVOIP User Guide
for the MVP810SS) constitutes a practical limitation on their capacity to
support PSTN access for other gateways. Systems must be scaled to
match required capacity by including additional MultiVOIP-SS units.
Mounting. Mechanically, the MVP410SS and MVP810SS MultiVOIPs
are designed for a one-high industry-standard EIA 19-inch rack
enclosure. The product must be installed by qualified service personnel
in a restricted-access area, in accordance with Articles 110-16, 10-17, and
110-18 of the National Electrical Code, ANSI/NFPA 70.
Phone System Transparency. These MultiVOIPs inter-operate with a
telephone switch or PBX, acting as a switching device that directs voice
and fax calls over an IP network. The MultiVOIPs have “phonebooks,”
directories that determine to who calls may be made and the sequences
that must be used to complete calls through the MultiVOIP. The
phonebooks allow the phone user to interact with the VOIP system just
as they would with an ordinary PBX or telco switch. When the
phonebooks are set, special dialing sequences are minimized or
eliminated altogether. Once the call destination is determined, the
phonebook settings determine whether the destination VOIP unit must
strip off or add dialing digits to make the call appear at its destination
to be a local call.
Voip Protocol. The MVP-SS units use the SIP protocol only. (“SIP”
means Session Initiation Protocol.)
Data Compression & Quality of Service. The analog MultiVOIP unit
comes equipped with a variety of data compression capabilities,
including G.723, G.729, and G.711 and features DiffServ quality-of-
service (QoS) capabilities.
PSTN Failover Feature. The MultiVOIP can be programmed to divert
calls to the PSTN temporarily in case the IP network fails.
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).
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
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MultiVOIP User Guide
Overview
MultiVOIP web browser GUI. All of these control software packages
are included on the Product CD.
While the web GUI’s appearance differs slightly, its content and
organization are essentially the same as that of the Windows GUI
(except for logging).
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Overview
MultiVOIP User Guide
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.
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
MultiVOIP Front Panel LEDs
LED Types. The MultiVOIPs have two types of LEDs on their front
panels:
(1) general operation LED indicators (for power, booting, and
ethernet functions), and
(2) channel operation LED indicators that describe the data traffic
and performance in each VOIP data channel.
Active LEDs. On both the MVP410SS and MVP810SS, there are eight
sets of channel-operation LEDs. However, on the MVP410SS, only the
lower four sets of channel-operation LEDs are functional. On the
MVP810SS, all eight sets are functional.
Figure 1-4. MVP-410SS/810SS LEDs
Similarly, the MVP210 has the general-operation indicator LEDs and
two sets of channel-operation LEDs, one for each channel.
Figure 1-5. MVP210SS LEDs
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Overview
MultiVOIP User Guide
LED Descriptions for MultiVOIP-SS Units
Front Panel LED Definitions
LED NAME DESCRIPTION
General Operation LEDs (one set on each MultiVOIP model)
Power
Boot
Indicates presence of power.
After power up, the Boot LED will be on briefly while the
MultiVOIP is booting. It lights whenever the MultiVOIP is
booting or downloading a setup configuration data set.
FDX. LED indicates whether Ethernet connection is
half-duplex or full-duplex (FDX) and, in half-duplex
mode, indicates occurrence of data collisions. LED is
on constantly for full-duplex mode; LED is off
constantly for half-duplex mode. When operating in
half-duplex mode, the LED will flash during data
collisions.
Ethernet
LNK. Link/Activity LED. This LED is lit if Ethernet
connection has been made. It is off when the link is
down (i.e., when no Ethernet connection exists).
While link is up, this LED will flash off to indicate data
activity.
Channel-Operation LEDs (one set for each channel)
Transmit. This indicator blinks when voice packets
are being transmitted to the local area network.
XMT
RCV
XSG
Receive. This indicator blinks when voice packets
are being received from the local area network.
Transmit Signal. This indicator lights when the FXS-
configured channel is off-hook, the FXO-configured
channel is receiving a ring from the Telco, or the M
lead is active on the E&M configured channel. That is,
it lights when the MultiVOIP is receiving a ring from
the PBX.
Receive Signal. This indicator lights when the FXS-
configured channel is ringing, the FXO-configured
channel has taken the line off-hook, or the E lead is
active on the E&M-configured channel.
RSG
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MultiVOIP User Guide
Overview
Computer Requirements
The computer on which the MultiVOIP’s configuration program is
installed must meet these requirements:
•
must be IBM-compatible PC with MS Windows operating
system;
•
must have an available COM port for connection to the
MultiVOIP.
However, this PC does not need to be connected to the MultiVOIP
permanently. It only needs to be connected when local configuration
and monitoring are done. Nearly all configuration and monitoring
functions can be done remotely via the IP network.
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Overview
MultiVOIP User Guide
Specifications
Parameter
/Model
MVP410SS
MVP810SS
MVP210SS
100-240 VAC
1.2 - 0.6 A
100-240 VAC
1.2 - 0.6 A
Operating
Voltage/
Current
External
transformer:
3A @5V
50/60 Hz
50/60 Hz
50/60 Hz
Mains
Frequencies
Power
29 watts
46 watts
19 watts
Consumption
Mechanical
Dimensions
1.75” H x
17.4” W x
8.5” D
1.75” H x
17.4” W x
8.5” D
6.2” W x
9” D x
1.4” H
4.5cm H x
44.2 cm W x
21.6 cm D
7.1 lbs.
4.5cm H x
44.2 cm W x
21.6 cm D
7.7 lbs.
15.8cm W x
22.9cm D x
3.6cm H
1.8lbs (.82kg)
2.6lbs (1.17kg)
with transformer
Weight
(3.2 kg)
(3.5 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 MultiVOIP is shipped with a printed
Cabling Guide.
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. (Note that the
configuration of the MultiTech home page is subject to change. The current User Guide
will be present, in any case.
Entries (organized by model number) in the “knowledge base” and
‘troubleshooting resolutions’ sections of the MultiTech web site (found
under “Support”) constitute another source of help for problems
encountered in the field.
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Quick Start
MultiVOIP User Guide
Chapter 2: Quick Start Instructions
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MultiVOIP User Guide
QS: Intro
Introduction
This chapter contains 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 Instructions 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 Cabling Guide is shipped with the MultiVOIP and an
electronic copy is included on the Product CD.
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MultiVOIP User Guide
QS: Startup Tasks
MultiVOIP Startup Tasks
Task
Summary
ꢀ Collecting Phone/IP
The MultiVOIP must be configured to
interface with your particular phone
system and IP network. To do so,
certain details must be known about
those phone and IP systems.
Details ( vital! )
ꢀ Placement
Decide where you’ll mount the voip.
ꢀ Command/Control
Computer Setup:
Specs & Settings
Some modest minimum specifications
must be met. A COM port must be set
up.
ꢀ Hookup
Connect power, phone, and data cables
per diagram.
ꢀ Software Installation
This is the configuration program.
It’s a standard Windows software
installation.
ꢀ Phone/IP Starter
You will enter phone numbers and IP
addresses. You’ll use default parameter
values where possible to get the system
running quickly.
Configuration
Use “Config Info CheckList” (page 28).
ꢀ Phonebook Starter
The phonebook is where you specify
how calls will be routed. To get the
system running quickly, you’ll make
phonebooks for just two voip sites.
Configuration
ꢀ Connectivity Test
You’ll find out if your voip system can
carry phone calls between two sites.
That means you’re up and running!
ꢀ Troubleshooting
Detect and remedy any problems that
might have prevented connectivity.
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MultiVOIP User Guide
QS: Gathering Phone/IP Details
Phone/IP Details *Absolutely Needed*
Before Starting the Installation
The MultiVOIP will interface with both the IP network and the phone
system. You must gather information about the IP network and about
the phone system so that the MultiVOIP can be configured to operate
with them properly. A summary of this configuration information
appears on page 28 (“Config Info CheckList”).
Gather IP Information
Ask your computer network
administrator.
Info needed to operate:
all MultiVOIP models.
ꢁ
IP Network Parameters:
Record for each VOIP Site
in System
#
• IP Address
• IP Mask
• Gateway
• Domain Name Server (DNS) Info (optional)
• Determine whether or not 802.1p Packet Prioritization
will be used.
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QS: Gathering Phone/IP Details
MultiVOIP User Guide
Phone/IP Details *Absolutely Needed*
Gather Telephone Information
Telephony Parameters
ꢁ
Ask phone company or
telecom manager.
Analog Telephony Interface Parameters:
Record for this VOIP Site
#
• Which interface type is used?
E&M_____ FXS/FXO_____ DID/DPO _____
• If FXS, determine whether the line will be used for a
phone, fax, or KTS (key telephone system)
• If FXO, determine if line will be an analog PBX
extension or an analog line from a telco central office
• If E&M, determine these aspects of the E&M trunk
line from the PBX:
• What is its Type (1, 2, 3, 4, or 5)?
• Is it 2-wire or 4-wire?
• Is it Dial-Tone or Wink?
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MultiVOIP User Guide
QS: Gathering Phone/IP Details
Phone/IP Details Often Needed/Wanted
Obtain Email Address for VOIP (for email call log reporting)
required if log reports of
VOIP call traffic
Optional
are to be sent by email
SMTP Parameters
Preparation Task:
To: I.T. Department
Ask Mail Server
re: email account for VOIP
administrator to set up
email account (with
password) for the
MultiVOIP unit itself.
Be sure to give a unique
identifier to each
individual MultiVOIP
unit.
Get the IP address of the
mail server computer, as
well.
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QS: Gathering Phone/IP Details
MultiVOIP User Guide
Config Info CheckList
Type of Config Info
Gathered
MultiVOIP
Configuration
screen
on which to enter
Config Info
√
IP info for voip unit
● IP address
Ethernet/IP Parameters
● Gateway
● DNS IP (if used)
● 802.1p Prioritization (if used)
Interface Type
(Choices: E&M, FXS/FXO*,
DIP, DPO)
Interface Parameters
*In FXO/FXS systems,
channels used for phone, fax,
or key system are FXS;
channels used for analog
PBX extensions or analog
telco lines are FXO.
E&M info (only if E&M is used)
● Type (1-5) ● 2 or 4 wires?
● Dial Tone or Wink?
Interface Parameters
Country Code
Email address for voip (optional)
Regional Parameters
SMTP Parameters
SIP Operating Mode
SIP Server Configuration
● Survivability ● Stand-Alone
Network Locations of Alternate
SIP Proxy units, if used
SIP Call Signaling
(IP Address or Domain Name)
Alt #1:
Alt #2
Endpoint Info
Device Name Regist Type
SIP Server Predefined
Endpoints
IP Address
Port
--------------------------------------
Device Name Regist Type
IP Address
Port
Reminder: Be sure to Save Setup after entering configuration values.
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MultiVOIP User Guide
QS: Gathering Phone/IP Details
Identify Remote VOIP Site to Call
When you’re done installing the MultiVOIP, you’ll want to confirm that
it is configured and operating properly. To do so, it’s good to have
another voip that you can call for testing purposes. You’ll want to
confirm end-to-end connectivity. You’ll need IP and telephone
information about that remote site.
If this is the very first voip in the system, you’ll want to coordinate the
installation of this MultiVOIP with an installation of another unit at a
remote site.
Identify MVP-SS Unit’s Role in SIP VOIP System
The MVP210-SS/410-SS/810-SS unit always uses the SIP protocol.
However, the MVP-SS units are equipped to play an additional role in
the voip system -- the role of a SIP server. And as a SIP server, the
MVP-SS unit can operate in either “stand-alone” mode or “SIP
survivability” mode.
Stand-Alone Mode. The MVP-SS unit can function as a stand-alone SIP
server that controls the flow of phone traffic to lines connected to
gateways that are registered with the MVP-SS unit. This stand-alone
capability allows the MVP-SS to operate with ‘smart’ SIP phones. Such
smart SIP phones can choose the SIP server under which they operate
and, consequently, can be controlled by either the SIP-based PBX or by
the MVP-SS.
SIP Survivability Mode. The MVP-SS unit can function as a back-up SIP
server that performs SIP server functions when/if the network’s main
SIP server fails or loses contact with the subnetwork in which the
MVP-SS unit is placed.
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QS: Voip Placement & PC Settings
MultiVOIP User Guide
Placement
Mount your MultiVOIP in a safe and convenient location where cables
for your network and phone system are accessible. Rack-mounting
instructions are in Chapter 3: Mechanical Installation & Cabling of the User
Guide.
Command/Control Computer Setup (Specs & Settings)
The computer used for command and control of the MultiVOIP
(a) must be an IBM-compatible PC,
(b) must use a Microsoft operating system,
(c) must be connected to your local network (Ethernet) system, and
(d) must have an available serial COM port.
The configuration tasks and control tasks the PC will have to do with
the MultiVOIP are not especially demanding. Still, we recommend
using a reasonably new computer. The computer that you use to
configure your MultiVOIP need not be dedicated to the MultiVOIP
after installation is complete.
COM port on controller PC. You’ll need an available COM port on the
controller PC. You’ll need to know which COM port is available for use
with the MultiVOIP (COM1, COM2, etc.).
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MultiVOIP User Guide
QS: Quick Hookups
Quick Hookup for MVP410-SS & MVP810-SS
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QS: Quick Hookups
MultiVOIP User Guide
Quick Hookup for MVP210-SS
CH1
CH2
ETHERNET
FXS/FXO
E&M FXS/FXO E&M
RS232
POWER
10/100 COMMAND
10BASET
COMMAND PORT
POWER
Voice/Fax Channel 1 - 2
Connections
E&M FXO/FXS
Power Connection
GND
FXS
E&M
FXO
Command Port Connection
Ethernet Connection
PSTN
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QS: Software Installation
MultiVOIP User Guide
Load MultiVOIP Control Software onto PC
For more details, see Chapter 4: Software Installation in User Guide.
1. MultiVOIP must be properly cabled. Power must be turned on.
2. Insert MultiVOIP CD into drive. Allow 10-20 seconds for Autorun to
start. If Autorun fails, go to
My Computer | CD ROM drive | Open. Click Autorun icon.
3. At first dialog box, click Install Software.
4. At ‘welcome’ screen, click Next.
5. Follow on-screen instructions. Accept default program folder
location and click Next.
6. Accept default icon folder location. Click Next. Files will be copied.
7. Select available COM port on command/control computer.
8. At completion screen, click Finish.
9. At the prompt “Do you want to run MultiVOIP Configuration?,”
click No. Software installation is complete.
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QS: Phone/IP Starter Config.
MultiVOIP User Guide
Phone/IP Starter Configuration
This is a summary. For full details, see “Technical Configuration”
chapter of User Guide.
1. Open MultiVOIP program: Start | MultiVOIP xxx | Configuration.
2. Go to Configuration | Ethernet/IP. Enter the IP parameters for your
voip site. Activate Packet Prioritization (802.1p) if desired. If you use a
Domain Name Server (DNS), specify its IP address. If DNS is used, you
can activate the Service Record (SRV) feature. For details, see the
“Technical Configuration” chapter of the User Guide.
3. Do you want to configure and operate the MultiVOIP unit using the
web browser GUI? (It has the same functionality as the local
Windows GUI, but offers remote access.)
If NO, skip to step 5.
If YES, continue with step 4.
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MultiVOIP User Guide
QS: Phone/IP Starter Config.
4. Web Browser GUI Setup (Optional). To do configuration and
operation procedures using the web browser GUI, you must first set
it up. To do so, follow these steps. (The browser used must be
Internet Explorer 6.0 or above; or Netscape 6.0 or above; or FireFox
1.0 or above.)
A. Be sure an IP address has
been assigned to the
E. Open web browser.
(Note: The PC being used must
be connected to and have an IP
address on the same IP network
that the voip is on.)
MultiVOIP unit (this must be
done in the MultiVOIP
Windows GUI).
B. Save Setup in Windows GUI.
F. Browse to IP address of
MultiVOIP unit.
C. Close the MultiVOIP
G. If username and password
have been established, enter
them when prompted by
voip.
Windows GUI.
D. Install Java program from
MultiVOIP product CD.
(Must be Java Runtime
H. Use web browser GUI to
configure or operate voip.
Environment 1.4.2_01 or above.)
NOTE: Required on first use of
Web Browser GUI only.
Need more See “Web Browser Interface” in Operation &
info? Maintenance chapter of User Guide (on CD).
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.
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QS: Phone/IP Starter Config.
MultiVOIP User Guide
Phone/IP Starter Configuration (continued)
5. Go to Configuration | Voice/Fax. Select Coder | “Automatic.” At
the right-hand side of the dialog box, click OK. If you know any
specific parameter values that will apply to your system, enter them.
Click Copy Channel. Select Copy to All. Click Copy. At main
Voice/Fax Parameters screen, click OK to exit from the dialog box.
6. Enter telephone system information.
Go to Configuration | Interface.
Enter parameters obtained from
phone company or PBX administrator.
7. Go to Configuration | Regional Parameters. Select the
Country/Region that fits your situation. Click OK and confirm.
Click OK to exit from the dialog box.
8. Go to Configuration | Regional Parameters. In the Country
Selection for Built-In Modem field (drop-down list), select the
country that best fits your situation. (This may not be the same as
your selection for the Country/Region field. The selections in the
Country Selection for Built-In Modem field entail more detailed
groupings of telephony parameters than do the Country/Region
values.)
9. Do you want the phone-call logs produced by the MultiVOIP to be
sent out by email (to your Voip Administrator or someone else)?
If NO, skip to step 11.
If YES, continue with step 10.
10. Go to Configuration | SMTP.
SMTP lets you send phone-call log records to the Voip Administrator
by email. Select Enable SMTP.
You should have already obtained an email address for the
MultiVOIP itself (this serves as the origination email account for
email logs that the MultiVOIP can email out automatically).
Enter this email address in the “Login Name” field.
Type the password for this email account.
Enter the IP address of the email server where the MultiVOIP’s email
account is located in the “Mail Server IP Address” field.
Typically the email log reports are sent to the Voip Administrator
but they can be sent to any email address. Decide where you want
the email logs sent and enter that email address in the “Recipient
Address” field.
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MultiVOIP User Guide
QS: Phone/IP Starter Config.
Whenever email log messages are sent out, they must have a
standard Subject line. Something like “Phone Logs for Voip N” is
useful. If you have more than one MultiVoip unit in the building,
you’ll need a unique identifier for each one (select a useful name or
number for “N”). In this “Subject” field, enter a useful subject title for
the log messages.
In the “Reply-To Address” field, enter the email address of your Voip
Administrator.
11. Go to Configuration | Logs.
Select “Enable Console Messages.”
To allow log reports by email (if desired), click SMTP. Click OK.
To do logging with a SysLog client program, click on “SysLog Server
– Enable” in the Logs screen. To implement this function, you must
install a SysLog client program. For more info, see the “SysLog
Server Functions” section of the Operation & Maintenance chapter of
the User Guide.
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QS: Phone/IP Starter Config.
MultiVOIP User Guide
Phone/IP Starter Configuration (continued)
12. Enable premium (H.450) telephony features.
Go to Supplementary Services. Select any features to be used.
For Call Hold, Call Transfer, & Call Waiting, specify the key sequence
that the phone user will press to invoke the feature. For Call Name
Identification, specify the allowed name types to be used and a caller-
id descriptor.
If Call Forwarding is to be used, enable this feature in the
Add/Edit Inbound Phone Book screen.
After making changes, click on OK in the current configuration
screen before moving on to the next configuration screen.
13. RADIUS Support. If you intend to use a RADIUS server for billing or
other accounting purposes, enter the server information in the
RADIUS screen.
14. STUN Support. If you are using the SIP protocol with the UDP
transmission protocol, and if you want the MultiVOIP to operate
behind a NAT (Network Address Translation server) using the STUN
protocol (Simple Traversal of UDP through NAT), enable this feature
in the NAT Traversal screen. You must also specify the IP address
(etc.) of the STUN server you will use. The STUN server could be a
local device or it could be a public STUN server accessible on the
Internet.
15. Network Locations of SIP Servers (Primary & Alternate).
Go to SIP Call Signaling and enter the IP address or domain name for the
primary SIP Server in your system, as well as any alternate SIP servers.
The UserName and Password entered here will be used to
authenticate all inbound phonebook entries that do not already have
their own unique usernames and passwords.
16. Endpoint Info. Go to SIP Server | Predefined Endpoints.
For every other endpoint (gateway) to be registered with the
MultiVOIP-SS unit, enter values for the following parameters.
The parameters required are different for static registrations than for
dynamic registrations, as shown in the table below.
Static Registration
Dynamic Registration
Endpoint Name =
Endpoint Name =
IP Address:
Password:
.
.
.
Port #:
Re-Registrat. Interval (sec):
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MultiVOIP User Guide
QS: Phone/IP Starter Config.
17. Go to Save Setup | Save and Reboot. Click OK. This will save the
parameter values that you have just entered.
The MultiVOIP’s “BOOT” LED will light up while the configuration
file is being saved and loaded into the MultiVOIP. Don’t do anything
to the MultiVOIP until the “BOOT “LED is off (a loss of power at this
point could cause the MultiVOIP unit to lose the configuration
settings you have made).
END OF PROCEDURE.
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QS: Phonebook Starter Config.
MultiVOIP User Guide
Phonebook Starter Configuration (with remote voip)
If the topic of voip phone books is new to you, it may be helpful to read
the PhoneBook Tips section (page 47) before starting this procedure.
To do this part of the quick setup, you need to know of another voip
that you can call to conduct a test. It should be at a remote location,
typically somewhere outside of your building. You must know the
phone number and IP address for that site. We are assuming here that
the MultiVOIP will operate in conjunction with a PBX.
You must configure both the Outbound Phonebook and the Inbound
Phonebook. A starter configuration only means that two voip locations
will be set up to begin the system and establish voip communication.
Outbound Phonebook
1. Open the MultiVOIP program.
( Start | MultiVOIP xxx | Configuration )
2. Go to Phone Book | Outbound Phonebook | Add Entry.
3. On a sheet of paper, write down the calling code of the remote voip
(area code, country code, city code, etc.) that you’ll be calling.
Follow the example that best fits your situation.
North America,
Long-Distance Example
Euro, National Call
Example
Technician in Seattle (area
206) must set up one voip
there, another in Chicago
(area 312, downtown).
Technician in central
London (area 0207) to set
up voip there, another in
Birmingham (area 0121).
Answer: Write down 312.
Answer: write down 0121.
Euro, International Call Example
Technician in Rotterdam (country 31; city 010) to
set up one voip there, another in Bordeaux
(country 33; area 05).
Answer: write down 3305.
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MultiVOIP User Guide
QS: Phonebook Starter Config.
4. Suppose you want to call a phone number outside of your building
using a phone station that is an extension from your PBX system (if
present). What digits must you dial? Often a “9” or “8” must be
dialed to “get an outside line” through the PBX (i.e., to connect to the
PSTN). Generally, “1 “or “11” or “0” must be dialed as a prefix for
calls outside of the calling code area (long-distance calls, national
calls, or international calls).
On a sheet of paper, write down the digits you must dial before you
can dial a remote area code.
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
Seattle voip works with
PBX that uses “8” for all
voip calls. “1” must
immediately precede area
code of dialed number.
London voip works with
PBX that uses “9” for all
out-of-building calls
whether by voip or by
PSTN. “0” must
immediately precede area
code of dialed number.
Answer: write down 81.
Answer: write down 90.
Euro, International Call Example
Rotterdam/Bordeaux system.
Rotterdam voip works with PBX where “9” is
used for all out-of-building calls. “0” must
precede all international calls.
Answer: write down 90.
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QS: Phonebook Starter Config.
MultiVOIP User Guide
5. In the “Destination Pattern” field of the Add/Edit Outbound
Phonebook screen, enter the digits from step 4 followed by the digits
from step 3.
North America,
Long-Distance Example
Seattle-Chicago system.
Euro, National Call
Example
London/Birming. system.
Answer: enter 81312 as
Destination Pat-
tern in Outbound
Phone-book of
Leading zero of
Birmingham area code is
dropped when combined
with national-dialing
access code. (Such
Seattle voip.
practices vary by country.)
Answer: enter 90121 as
Destination Pat-
tern in Outbound
Phonebook of
London voip.
Not 900121.
Euro, International Call Example
Rotterdam/Bordeaux system.
Answer: enter 903305 as Destination Pattern in
Outbound Phonebook of Rotterdam voip.
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MultiVOIP User Guide
QS: Phonebook Starter Config.
6. In the “Remove Prefix” field, enter the initial PBX access digit (“8” or
“9”).
North America,
Long-Distance Example
Seattle-Chicago system.
Euro, National Call
Example
London/Birming. system.
Answer: enter 8 in “Remove
Prefix” field of
Answer: enter 9 in “Remove
Prefix” field of
Seattle Outbound
Phonebook.
London Outbound
Phonebook.
Euro, International Call Example
Rotterdam/Bordeaux system.
Answer: enter 9 in “Remove Prefix” field of Outbound
Phonebook for Rotterdam voip.
Some PBXs will not ‘hand off’ the “8” or “9” to the voip. But for those PBX
units that do, it’s important to enter the “8” or “9” in the “Remove Prefix”
field in the Outbound Phonebook. This precludes the problem of having to
make two inbound phonebook entries at remote voips, one to account for
situations where “8” is used as the PBX access digit, and another for when
“9” is used.
7. In the “SIP” field group, select “Use Proxy” and specify the Transport
Protocol to be used (TCP or UDP). Use the default SIP Port Number
(5060).
8. Click OK to exit from the Add/Edit Outbound Phonebook screen.
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QS: Phonebook Starter Config.
MultiVOIP User Guide
Inbound Phonebook
1. Open the MultiVOIP program.
( Start | MultiVOIP xxx | Configuration )
2. Go to Phone Book | Inbound Phonebook | Add Entry.
3. In the “Remove Prefix” field, enter your local calling code (area code,
country code, city code, etc.) preceded by any other “access digits”
that are required to reach your local site from the remote voip
location (think of it as though the call were being made through the
PSTN – even though it will not be).
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
Seattle is area 206. Chicago
employees must dial 81
before dialing any Seattle
number on the voip system.
Inner London is 0207 area.
Birmingham employees must
dial 9 before dialing any
London number on the voip
system.
Answer: 1206 is prefix to be
removed by local
Answer: 0207 is prefix to be
removed by local
(Seattle) voip.
(London) voip.
Euro, International Call Example
Rotterdam/Bordeaux system.
Rotterdam is country code 31, city code 010. Bordeaux
employees must dial 903110 before dialing any
Rotterdam number on the voip system.
Answer: 03110 is prefix to be removed by local
(Rotterdam) voip.
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MultiVOIP User Guide
QS: Phonebook Starter Config.
4. In the “Add Prefix” field, enter any digits that must be dialed from
your local voip to gain access to the PSTN.
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
On Seattle PBX, “9” is used to
get an outside line.
On London PBX, “9” is used
to get an outside line.
Answer: 9 is prefix to be
added by local
Answer: 9 is prefix to be
added by local
(Seattle) voip.
(London) voip.
Euro, International Call Example
Rotterdam/Bordeaux system.
On Rotterdam PBX, “9” is used to get an outside line.
Answer: 9 is prefix to be added by local (Rotterdam)
voip.
5. In the “Channel Number” field, enter “Hunting.” A “hunting” value
means the voip unit will assign the call to the first available channel.
If desired, specific channels can be assigned to specific incoming calls
(i.e., to any set of calls received with a particular incoming dialing
pattern).
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QS: Phonebook Starter Config.
MultiVOIP User Guide
6. In the “Description” field, it is useful to describe the ultimate
destination of the calls. For example, in a New York City voip
system, “incoming calls to Manhattan office,” might describe a
phonebook entry, as might the descriptor “incoming calls to NYC
local calling area.” The description should make the routing of calls
easy to understand. (40 characters max.)
North America,
Long-Distance Example
Euro, National Call
Example
Seattle-Chicago system.
London/Birming. system.
Possible Description:.
Free Seattle access, all
employees
Possible Description:.
Local-rate London access,
all empl.
Euro, International Call Example
Rotterdam/Bordeaux system.
Possible Description:. Local-rate Rotterdam access, all
empl.
7. In the Add/Edit Inbound Phonebook screen, under “Registration
Options,” enter the special password (if any) that will be used for
this inbound phonebook entry. If you specify a special password that
applies only to this inbound phonebook entry, that password will
override the general password used by endpoints registering with the
SIP server (in the SIP Call Signaling screen).
8. Repeat steps 2-8 for each inbound phonebook entry. When all entries
are complete, go to step 9.
9. Click OK to exit the inbound phonebook screen.
10. Click on Save Setup. Highlight Save and Reboot. Click OK.
Your starter inbound phonebook configuration is complete.
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MultiVOIP User Guide
QS: Phonebook Tips
Phonebook Tips
Preparing the phonebook for your voip system is a complex task that, at
first, seems quite daunting. These tips may make the task easier.
1. Use Dialing Patterns, Not Complete Phone Numbers. You will not
generally enter complete phone numbers in the voip phonebook.
Instead, you’ll enter “destination patterns” that involve area codes and
other digits. If the destination pattern is a whole area code, you’ll be
assigning all calls to that area code to go to a particular voip which has
a unique IP address. If your destination pattern includes an area code
plus a particular local phone exchange number, then the scope of calls
sent through your voip system will be narrowed (only calls within that
local exchange will be handled by the designated voip, not all calls in
that whole area code). In general, when there are fewer digits in your
destination pattern, you are asking the voip to handle calls to more
destinations.
2. The Four Types of Phonebook Digits Used. Important!
“Destination patterns” to be entered in your phonebook will generally
consist of:
(a) calling area codes,
(b) access codes,
(c) local exchange numbers, and
(d) specialized codes.
Although voip phonebook entries may look confusing at first, it’s
useful to remember that all the digits in any phonebook entry must be
of one of these four types.
(a) calling area codes. There are different names for these around the
world: “area codes,” “city codes,” “country codes,” etc. These codes,
are used when making non-local calls. They always precede the phone
number that would be dialed when making a local call.
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QS: Phonebook Tips
MultiVOIP User Guide
(b) access codes. There are digits (PSTN access codes) that must be
dialed to gain access to an operator, to access the publicly switched
‘long-distance’ calling system(North America), to access the publicly
switched ‘national’ calling system (Europe and elsewhere), or to access
the publicly switched ‘international’ calling system (worldwide).
There are digits (PBX access codes) that must be dialed by phones
connected to PBX systems or key systems. Often a “9” must be dialed
on a PBX phone to gain access to the PSTN (‘to get an outside line’).
Sometimes “8” must be dialed on a PBX phone to divert calls onto a
leased line or to a voip system. However, sometimes PBX systems are
‘smart’ enough to route calls to a voip system without a special access
code (so that “9” might still be used for all calls outside of the building).
There are also digits (special access codes) that must be dialed to gain
access to a particular discount long-distance carrier or to some other
closed or proprietary telephone system.
(c) local exchange numbers. Within any calling area there will be many
local exchange numbers. A single exchange may be used for an entire
small town. In cities, an exchange may be used for a particular
neighborhood (although exchanges in cities do not always cover easily
discernible areas). Organizations like businesses, governments,
schools, and universities are also commonly assigned exchange
numbers for their exclusive use. In some cases, these organizational-
assigned exchanges can become non-localized because the exchange is
assigned to one facility and linked, by the organization’s private
network, to other sometimes distant locations.
(d) specialized codes. Some proprietary voip units assign, to sites and
phone stations, numbers that are not compatible with PSTN
numbering. This can also occur in PBX or key systems. These
specialized numbers must be handled on a case-by-case basis.
48
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MultiVOIP User Guide
QS: Phonebook Tips
Example
3. Knowing When to Drop Digits.
When calling area codes and
access codes are used in
combination, a leading “1” or “0”
must sometimes be dropped.
Area code for Inner London is
listed as “0207.” However, in
international calls the leading
“0” is dropped.
U.K.
Country
Code
Phonebook Entry ꢂ
International
Access Code
Leading Zero
Dropped from
Area Code
49
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QS: Phonebook Tips
MultiVOIP User Guide
4. Using a Comma.
Detail
Commas are used in telephone
dialing strings to indicate a pause
to allow a dial tone to appear
(common on PBX and key
systems). Commas may be used
only in the “Add Prefix” field of
the Inbound Phonebook.
,
= 1-second pause
in many PBX systems
(not needed in all)
5. Ease of Use. The phonebook setup determines how easy the voip
system is to use. Generally, you’ll want to make it so dialing a voip call
is very similar to dialing any other number (on the PSTN or through the
PBX).
6. Avoid Unintentional Calls to Official/Emergency Numbers. Dialing a
voip call will typically be somewhat different than ordinary dialing.
Because of this, it’s possible to set up situations, quite unwittingly,
where phone users may be predisposed to call official numbers without
intending to do so. Conversely, a voip/PBX system might also make it
difficult to place an official/emergency call when one intends to do so.
Study your phonebook setup and do some test-dialing on the system to
avoid these pitfalls.
7. Inbound/Outbound Pattern Matching. In general, the Inbound
Phonebook entries of the local voip unit will match the Outbound
Phonebook entries of the remote voip unit. Similarly, the Outbound
Phonebook entries of the local voip unit will match the Inbound
Phonebook entries of the remote voip unit. There will often be non-
matching entries, but it’s nonetheless useful to notice the matching
between the phonebooks.
8. Simulating Network in-lab/on-benchtop. One common method of
configuring a voip network is to set up a local IP network in a lab,
connect voip units to it, and perhaps have phones connected on channel
banks to make test calls.
50
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QS: Phonebook Example
MultiVOIP User Guide
Phonebook Example
One Common Situation
Boise Office
Area: 208
PBX System.
Voip Example. This company has offices in three
different cities. The PBX units all operate alike.
N otably, they all give access to outside lines using
“ 9.” They all are ‘smart’ enough to identify voip calls
w ithout using a special access digit (“ 8” is used in
some systems). Finally, the system operates so that
employees in any office can dial employees in any
other office using only three digits. Hereare the
phonebooks needed for that system.
Main Number:
333-2700
PSTN
90 extensions
204.16.49.73
24-Channel
Digital VoIP
(MVP2410)
Inbound Phonebook
Each Inbound Phonebook contains
two entries. The first entry (4 digits)
specifies how incoming calls from the
other voip sites will be handled if
they go out onto the local PSTN.
Essentially, all those calls come to the
receiving voip with a pattern
beginning with 1+area code. The local
voip removes those four digits
because they aren’t needed w hen
dialing locally. The local voip
attaches a “ 9” at the beginning of the
number to get an outside line. The
PBX then completes the call to the
PSTN .
Santa Fe Office
Area: 505
204.16.49.74
8-Channel
Analog VoIP
(MVP810)
IP
Network
PBX System.
Main Number:
444-3200
40 extensions
The second Inbound Phonebook entry
(1 digit) is for receiving calls from
company employees in the other tw o
cities. The out-of-town employee
simply dials 3 digits. The first of the
three digits is uniquely used at each
site and so acts as a destination
pattern (Boise extensions are 7xx,
Santa Fe extensions 2xx, Flagstaff
extensions 6xx).
PSTN
The local voip sees the pattern in its
inbound phone book and notesthe
first digit (here either 2, 5, or 6).
To make the match, this first digit,
2, 5, or 6 is put in the “ Remove Prefix”
Each Outbound Phonebook contains two
pairs of entries, two entries for each
remote site. Whenever an out-of-tow n
employee dials a 12-digit number
beginning with the listed 5-digit
destination pattern (9+1+area code) of
another company location, the PBX
hands the call to the voip system. The
local voip strips off the “ 9” and directs
the call to the IP address of the remote
voip. The remote voip receives the call
and hands it to its PBX. The PBX then
completes the call to the PSTN.
field. This first digit must then be
added back once again so that the
voip will send all three digits to the
PBX. The PBX can then dial the
specific extension identified by the
three-digit number.
Flagstaff Office
Area: 520
The one-digit Outbound destination
patterns pertain to 3-digit calling
betw een company employees.
204.16.49.75
8-Channel
Analog VoIP
(MVP810)
PBX System.
Main Number:
777-5600
PSTN
30 extensions
51
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QS: Phonebook Example
MultiVOIP User Guide
Voip Sites with Phonebooks
Boise Voip
Inbound Phonebook
Boise Voip
Outbound Phonebook
Boise Office
Area: 208
Prefix Description
to Add Incoming Calls
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Remove
91208
Destin.
Pattern
PBX System.
Main Number:
333-2700
204.
16.49.
74
9,
7
Incoming calls 91505
to PSTN,
12
none
none
none
none
Outgoing calls
to Santa Fe
area
3-digit calls to
Santa Fe
employees
(extensions
200 to 240)
Outgoing calls
PSTN
Boise Area
204.
16.49.
74
7
i ncoming calls
to extensions
of company’s
PBX system
in Boise
2
3
90 extensions
204.16.49.73
91520
6
12
3
none
none
none
none
204.
24-Channel
Digital VoIP
(MVP2410)
16.49. to Flagstaff
75
204.
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
IP
Network
Santa Fe Office
Area: 505
Santa Fe Voip
Santa Fe Voip
Inbound Phonebook
Outbound Phonebook
204.16.49.74
Prefix Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
91505
Destin.
Pattern
8-Channel
Analog VoIP
(MVP810)
to Add Incoming Calls
to Add
Addr
9,
2
Incoming calls 91208
to PSTN,
12
none
none
none
none
204.
Outgoing calls
16.49. to Boise area
73
Santa Fe local
calls
2
Incoming calls
to extensions
of company’s
PBX system
in Santa Fe
7
3
204.
3-digit calls to
PBX System.
Main Number:
444-3200
16.49. Boise
73
employees
(extensions
700-790)
91520
6
12
3
none
none
none
none
204.
Outgoing calls
16.49. to Flagstaff
75
204.
40 extensions
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
PSTN
Flagstaff Voip
Inbound Phonebook
Flagstaff Voip
Outbound Phonebook
Flagstaff Office
Area: 520
Prefix Description
to Add Incoming Calls
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
204.16
.49.74
91520
9
Incoming calls 91505
to PSTN,
Flagstaff local
calls
12
none
none
none
none
Outgoing calls
to Santa Fe
area
204.16.49.75
8-Channel
Analog VoIP
(MVP810)
204.16
.49.74
6
6
Incoming calls
to extensions
of company’s
PBX system
in Flagstaff
2
3
3-digit calls to
Santa Fe
employees
(extensions
200-240)
Outgoing calls
to Boise area
3-digit calls to
Boise
PBX System.
Main Number:
777-5600
204.16
.49.73
91208
7
12
3
none
none
none
none
PSTN
204.16
.49.73
employees
(extensions
700-790)
30 extensions
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MultiVOIP User Guide
QS: Phonebook Example
Sample Phonebooks Enlarged
Boise Voip
Boise Voip
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
91208
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
204.
16.49.
74
9,
7
Incoming calls 91505
to PSTN,
12
none
none
none
none
Outgoing calls
to Santa Fe
area
3-digit calls to
Santa Fe
employees
(extensions
200 to 240)
Outgoing calls
Boise Area
204.
16.49.
74
7
i ncoming calls
to extensions
of company’s
PBX system
in Boise
2
3
91520
6
12
3
none
none
none
none
204.
16.49. to Flagstaff
75
204.
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
Santa Fe Voip
Santa Fe Voip
Inbound Phonebook
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
91505
Destin.
Pattern
to Add Incoming Calls
to Add
Addr
9,
2
Incoming calls 91208
to PSTN,
12
none
none
none
none
204.
Outgoing calls
16.49. to Boise area
73
Santa Fe local
calls
2
Incoming calls
to extensions
of company’s
PBX system
in Santa Fe
7
3
204.
3-digit calls to
16.49. Boise
73
employees
(extensions
700-790)
91520
6
12
3
none
none
none
none
204.
Outgoing calls
16.49. to Flagstaff
75
204.
area
3-digit calls to
16.49. Flagstaff
75
employees
(extensions
600-630)
Flagstaff Voip
Inbound Phonebook
Flagstaff Voip
Outbound Phonebook
Prefix
Description
Total
Digits
Prefix to
Remove
Prefix
to Add
IP
Addr
Description
Outgoing Calls
Prefix to
Destin.
Pattern
to Add Incoming Calls
Remove
204.16
.49.74
91520
9
Incoming calls 91505
to PSTN,
Flagstaff local
12
none
none
none
none
Outgoing calls
to Santa Fe
area
calls
204.16
.49.74
6
6
Incoming calls
to extensions
of company’s
PBX system
in Flagstaff
2
3
3-digit calls to
Santa Fe
employees
(extensions
200-240)
204.16
.49.73
91208
7
12
3
none
none
none
none
Outgoing calls
to Boise area
3-digit calls to
Boise
204.16
.49.73
employees
(extensions
700-790)
53
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QS: Phonebook Example
MultiVOIP User Guide
Phonebook Worksheet
Voip Location/ID:____________________________
Inbound Phonebook Outbound Phonebook
Prefix
Description
Total
Prefix to
Remove
Prefix
IP
Addr
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
to Add Incoming Calls
Digits
to Add
Other Details:
Voip Location/ID:____________________________
Inbound Phonebook Outbound Phonebook
Prefix
Description
Total
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
to Add Incoming Calls
Digits
to Add
Addr
Other Details:
Voip Location/ID:____________________________
Inbound Phonebook Outbound Phonebook
Prefix
Description
Total
Prefix to
Remove
Prefix
IP
Description
Outgoing Calls
Prefix to
Remove
Destin.
Pattern
to Add Incoming Calls
Digits
to Add
Addr
Other Details:
54
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MultiVOIP User Guide
QS: Phonebook Example
Enlarged Phonebook Worksheet
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QS: Connectivity Test
MultiVOIP User Guide
Connectivity Test
The procedures “Phone/IP Starter Configuration” and “Phonebook
Starter Configuration” must be completed before you can do this
procedure.
1. These connections must be made:
ꢀ MultiVOIP to local phone station
–OR--
MultiVOIP to extension of key phone system
ꢀ MultiVOIP to command PC
ꢀ MultiVOIP to Internet
2. Inbound Phonebook and Outbound Phonebook must both be set up
with at least one entry in each. These entries must allow for
connection between two voip units.
3. Console messages must be enabled. (If this has not been done
already, go, in the MultiVOIP GUI, to Configuration | Logs and
select the “Console Messages” checkbox.
56
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MultiVOIP User Guide
QS: Connectivity Test
4. You now need to free up the COM port connection (currently being
used by the MultiVOIP program) so that the HyperTerminal program
can use it. To do this, you can either (a) click on Connection in the
sidebar and select “Disconnect” from the drop-down box, or (b) close
down the MultiVOIP program altogether.
5. Open the HyperTerminal program.
6. Use HyperTerminal to receive and record console messages from the
MultiVOIP unit. To do so, set up HyperTerminal as follows (setup
shown is for Windows NT4; details will differ slightly in other MS
operating systems):
In the upper toolbar of the HyperTerminal screen, click on
the Properties button.
In the “Connect To” tab of the Connection Properties
dialog box, click on the Configure button.
In the next dialog box, on the “General” tab, set
“Maximum Speed” to 115200 bps.
On the “Connection” tab, set connection preferences to:
Data bits: 8
Parity:
none
Stop bits: 1
Click OK twice to exit settings dialog boxes.
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QS: Connectivity Test
MultiVOIP User Guide
7. Make VOIP call. Make call on a local phone line accessing PSTN
directly or through key system.
8. Read console messages recorded on HyperTerminal.
Console Messages from Originating VOIP. The voip unit that
originates the call will send back messages like that shown below.
[00026975] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[1]
TimeStamp : 26975
[00027190] CAS[0] : TX : ABCD = 1, 1, 1, 1
[00027190] PSTN: cas seizure detected on 0
[00027440] CAS[0] : TX : ABCD = 0, 0, 0, 0
[00033290] PSTN:call detected on 0 num=17637175662*
[00033290] H323IF[0]:destAddr =
TA:200.2.10.5:1720,NAME:Mounds
View,TEL:17637175662,17637175662
[00033290] H323IF[0]:srcAddr = NAME:New
York,TA:200.2.9.20
[00033440] H323IF [0]:cmCallStateProceeding
[00033500] H323[0]: Remote Information (Q931): MultiVOIP
- T1
[00033565] CAS[0] : TX : ABCD = 1, 1, 1, 1
[00033675] H323IF [0]: MasterSlaveStatus=Slave
[00033675] H323IF[0]:FastStart Setup Not Used
[00033690] CAS[0] : TX : ABCD = 1, 1, 1, 1
[00033755] H323IF[0]: Coder used 'g7231'
[00033810] PSTN:pstn call connected on 0
58
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MultiVOIP User Guide
QS: Connectivity Test
Console Messages from Terminating VOIP. The voip unit connected
to the phone where the call is answered will send back messages like
that shown below.
[00170860] H323[0]: New incoming call
[00170860] PSTNIF : Placing call on channel 0 Outbound
digit 7175662
[00170885] CAS[0] : TX : ABCD = 1, 1, 1, 1
[00171095] H323IF [0]: MasterSlaveStatus=Master
[00171105] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[7]
TimeStamp : 171105
[00171105] H323IF[0]: Coder used 'g7231'
[00171110] H323IF[0]:FastStart Setup Not Used
[00171110] H323IF[0]: Already opened the outgoing logical
channel
[00171110] H323IF[0]: Coder used 'g7231'
[00171315] CAS[0] : RX : ABCD = 0, 0, 0, 0,Pstn State[9]
TimeStamp : 171315
[00172275] PSTN: dialing digit ended on 0
[00172285] PSTN: pstn proceeding indication on 0
[00172995] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[12]
TimeStamp : 172995
[00173660] CAS[0] : TX : ABCD = 1, 1, 1, 1
[00173760] PSTN:pstn call connected on 0
9. When you see the following message, end-to-end voip connectivity
has been achieved.
“PSTN: pstn call connected on X”
where x is the number of the voip channel carrying the call
10. If the HyperTerminal messages do not confirm connectivity, go to
the Troubleshooting procedure below.
59
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QS: Troubleshooting
MultiVOIP User Guide
Troubleshooting
If you cannot establish connectivity between two voips in the system,
follow the steps below to determine the problem.
1. Ping both MultiVOIP units to confirm connectivity to the network.
2. Verify the telephone connections.
Check cabling. Are connections well seated? To correct receptacle?
Are telephone Interface Parameter settings correct?
3. Verify phonebook configuration.
4. Observe console messages while placing a call. Look for error messages indi-
cating phonebook problems, network problems, voice-coder mismatches, etc.
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MultiVOIP User Guide
Mechanical Installation & Cabling
Chapter 3: Mechanical Installation
and Cabling
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Mechanical Installation & Cabling
MultiVOIP User Guide
Introduction
When MVP410-SS or MVP810-SS units are to be installed into a rack,
two able-bodied persons should participate. The MVP210-SS is a table-
top unit that can generally be handled easily by one person.
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.
9. This product must be disconnected from its power source and
telephone network interface when servicing.
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MultiVOIP User Guide
Mechanical Installation & Cabling
UnpackingYour MultiVOIP
When unpacking your MultiVOIP, check to see that all of the items
shown are included in the box. For the various MultiVOIP models, the
contents of the box will be different. Study the particular illustration
below that is appropriate to the model you have purchased. If any box
contents are missing, contact MultiTech Tech Support at 1-800-972-2439.
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Mechanical Installation & Cabling
MultiVOIP User Guide
Unpacking the MVP-410SS/810SS
TM
MultiVOIP
Cabling
Guide
Voice/Fax
5
Voice/Fax
RC SG
6
2
Voice/Fax
RCV SG
7
3
Voice/Fax
RCV SG
8
4
X
M
T
R
C
V
X
S
G
R
S
G
X
M
T
V
X
RSG
RSG
X
M
T
X
RSG
RSG
X
M
T
X
RSG
Ethernet
CO
Power
Boot
Voice/Fax
1
Voice/Fax
RC SG
Voice/Fax
R CV SG
Voice/Fax
RCV SG
RCV
X
M
T
L
L
N
K
X
M
T
R
C
V
X
S
G
R
S
G
XM
T
V
X
XM
T
X
XM
T
X
RSG
Figure 3-1: Unpacking the MVP-410SS/810SS
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MultiVOIP User Guide
Mechanical Installation & Cabling
Unpacking the MVP210-SS
TM
MultiVOIP
Cabling
Guide
Figure 3-2: Unpacking the MVP210-SS
65
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Mechanical Installation & Cabling
MultiVOIP User Guide
Rack Mounting Instructions for
MVP410-SS & MVP810-SS
The MultiVOIPs can be mounted in an industry-standard EIA 19-inch
rack enclosure, as shown in Figure 3-3.
Figure 3-3: Rack-Mounting (MVP410SS or MVP810SS)
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MultiVOIP User Guide
Mechanical Installation & Cabling
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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Mechanical Installation & Cabling
MultiVOIP User Guide
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 3-4, and then secure unit to rack rails by the
brackets, as shown in Figure 3-5. 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 3-4: Bracket Attachment for Rack Mounting
(MVP410SS & MVP810SS)
Figure 3-5: Attaching MultiVOIP to Rack Rail
(MVP410-SS & MVP810-SS)
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MultiVOIP User Guide
Mechanical Installation & Cabling
Cabling Procedure for MVP-410SS/810SS
Cabling involves connecting the MultiVOIP to your LAN and telephone
equipment.
1. For DID channels only. If all channels of your MultiVOIP will be
using either FXS, FXO, or E&M telephony interfaces, skip to step 2.
For any channel on which you are using the DID interface type, you
must change the jumper on the MultiVOIP circuit card.
a. Disconnect power. Unplug the AC power cord from the wall outlet
or from the receptacle on the MultiVOIP unit.
b. Using a #1 Phillips driver, remove the three screws (at back of unit)
that attach the main circuit card to the chassis of the MultiVOIP.
Screws (3) holding circuit card assembly to chassis.
x
x
MVP410/810
rear panel
x
Figure 3-6. MVP-410SS/810SS Rear Screw Locations
c. Pull the main circuit card out about 5 inches (the power
connection to the board prevents it from being removed entirely
from the chassis).
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d. Identify the channels on which the DID interface will be used.
Jumper Configurations
(enlarged)
Upper Circuit Card
MVP810 only
For DID
Interface type
Ch 7
Ch 6
Ch 8
Ch 5
U10
U7
U9
U8
Jumpers 5-8
For non-DID
Interface type
}
Main Circuit Card
MVP-410/810
Generality:
For channels using the DID
interface, the jumper must
not straddle across the
Ch 1
Ch 2
Ch 3
Ch 4
cross-hatched area between
the jumper posts.
}
For channels using any non-DID
interface, it is acceptable that the
jumper straddles across the
cross-hatched area between
the jumper posts.
Jumpers 1-4
Figure 3-7. MVP-410SS/810SS Channel Jumper Settings
e. Position the jumper for each DID channel so that it does not connect
the two jumper posts. For DID operation of a voip channel, the
MultiVOIP will work properly if you simply remove the jumper
altogether, but that is inadviseable because the jumper might be
needed later if a different telephony interface is used for that voip
channel.
f. Slide the main circuit card back into the MultiVOIP chassis and
replace the three screws.
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2. Connect the power cord supplied with your MultiVOIP to a live AC
outlet and to the power connector on the back of the MultiVOIP as
shown at top right in Figure 3-8.
Command Modem connector
for remote configuration
ETH ERNET
COMMAND
E& M FXS/FX O
E& M FXS/FXO
E &M FXS/FXO
E&M FXSF/ XO
E&M FXS/FXO
E&M FXS /FXO
E&M FXS/FXO
E&M FX S/FXO
COMMAND
MODEM
10 BASET
Voice/Fax Channel Connections
Channels 1-4 Bottom MVP410 /8 10
Channels 5-8 Top MVP810 Only
E&M FXS/FXO
Ethernet Connection
FXS
E&M
FXO
Comm and Port Connec tion
PSTN
Figure 3-8: Cabling for MVP-410SS/810SS
3. Connect the MultiVOIP to a PC by using a DB-25 (male) to DB-9
(female) cable. Plug the DB-25 end of the cable into the Command
port of the MultiVOIP and the other end into the PC serial port. See
Figure 3-8.
4. Connect a network cable to the ETHERNET 10BASET connector on
the back of the MultiVOIP. Connect the other end of the cable to your
network.
5. For an FXS or FXO connection.
(FXS Examples: analog phone, fax machine, Key Telephone System.)
(FXO Examples: PBX extension, POTS line from telco central office.)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO
connector on the back of the MultiVOIP.
Connect the other end to the device or phone jack.
For an E&M connection.
(E&M Example: trunk line from telephone switch.)
Connect one end of an RJ-45 phone cord to the Channel 1 E&M
connector on the back of the MultiVOIP.
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Connect the other end to the trunk line.
Verify that the E&M Type in the E&M Options group of the Interface
dialog box is the same as the E&M trunk type supported by the
telephone switch. See Appendix B for an E&M cabling pinout.
For a DID connection.
(DID Example: DID fax system or DID voice phone lines.)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO
connector on the back of the MultiVOIP.
Connect the other end to the DID jack.
NOTE: DID lines are polarity sensitive. If, during testing, the DID line
rings busy consistently, you will need to reverse the polarity of
one end of the connector (swap the connections of the wires to
the two middle pins of one RJ-11 connector).
6. Repeat step 5 to connect the remaining telephone equipment to each
channel on your MultiVOIP. Although a MultiVOIP’s channels are
often all configured identically, each channel is individually
configurable. So, for example, some channels of a MultiVOIP might
use the FXO interface and others the FXS; some might use the DID
interface and others E&M, etc.
7. If you intend to configure the MultiVOIP remotely using the
MultiVOIP Windows GUI, connect an RJ-11 phone cable between the
Command Modem connector (at the rear of the MultiVOIP) and a
receptacle served by a telco POTS line. See Figure 3-9.
The Command Modem is built into the MultiVOIP unit. To configure
the MultiVOIP remotely using its Windows GUI, you must call into
the MultiVOIP’s Command Modem. Once a connection is made, the
configuration process is identical to local configuration with the
Windows GUI.
Command Modem connector
for remote configuration
ETHERNET
COMMAND
MODEM
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
COMMAND
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
10 BASET
MVP-410SS/810SS
Rear Panel
Grounding Screw
Telco POTS Line
Figure 3-9. MVP-410SS/810SS Voip Connections for GND &
Remote Config Modem
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8. Ensure that the unit is properly connected to earth ground by
verifying that it is reliably grounded when mounted within a rack.
This can be accomplished by connecting a grounding wire between
the chassis grounding screw (see Figure 3-9) and a metallic object that
will provide an electrical ground.
9. Turn on power to the MultiVOIP by placing the ON/OFF switch on
the back panel to the ON position. Wait for the Boot LED on the
MultiVOIP to go off before proceeding. This may take a few minutes.
Proceed to Chapter 4 to load the MultiVOIP software.
Cabling Procedure for MVP210-SS
Cabling involves connecting the MultiVOIP to your LAN and telephone
equipment.
1. For DID channels only. If both channels of your MVP210-SS
MultiVOIP will be using either FXS, FXO, or E&M telephony
interfaces, skip to step 2.
For any channel on which you are using the DID interface type, you
must change the jumper on the MultiVOIP circuit card.
a. Disconnect power. Unplug the AC power cord from the wall outlet
or from the receptacle on the MultiVOIP unit.
b. Using a #1 Phillips driver, remove the screw (at bottom of unit,
near the back-cover end) that attaches the main circuit card to the
chassis of the MVP210-SS.
c. Pull the main circuit card out about half way.
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d. Identify the channels on which the DID interface will be used.
L
E
D1
4
L
E
D1
2
L
E
D7
L
ED1
3
L
E
D11
L
E
D10
LE D 9
LE D 8
L
ED6
LE D 5
LE D4
LE D 3
L
E
D2
L
E
D1
R113
R7
2
R74
R114
R58
R57
R56
R5
5
R2
05
R2
MVP210SS Circuit Board
Ch1
Ch2
as configured
for DID Interface
JP4
Ch 1 Jumper
Block
P7
JP7
as shipped,
for non-DID interfaces
JP8
Ch 2 Jumper
Block
JP1
F
B
3
J3
J
7
J5
J9
J
11
J1
S
1
0
J
15
as configured
for DID Interface
Figure 3-10. MVP210-SS Channel Jumper Settings
e. Position the jumper for each DID channel so that it does not connect
the two jumper posts. For DID operation of a voip channel, the
MultiVOIP will work properly if you simply remove the jumper
altogether, but that is inadviseable because the jumper might be
needed later if a different telephony interface is used for that voip
channel.
f. Slide the main circuit card back into the MultiVOIP chassis and
replace the screw at the bottom of the unit.
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2. Connect the power cord supplied with your MultiVOIP to the power
connector on the back of the MultiVOIP and to a live AC outlet as
shown in Figure 3-11.
Figure 3-11: Cabling for MVP210-SS
3. Connect the MultiVOIP to a PC by using a RJ-45 (male) to DB-9
(female) cable. Plug the RJ-45 end of the cable into the Command port
of the MultiVOIP and the other end into the PC serial port. See Figure
3-11.
4. Connect a network cable to the ETHERNET 10/100 connector on the
back of the MultiVOIP. Connect the other end of the cable to your
network.
5. For an FXS or FXO connection.
(FXS Examples: analog phone, fax machine, Key Telephone System.)
(FXO Examples: PBX extension, POTS line from telco central office.)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO
connector on the back of the MultiVOIP.
Connect the other end to the device or phone jack.
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For an E&M connection.
(E&M Example: trunk line from telephone switch.)
Connect one end of an RJ-45 phone cord to the Channel 1 E&M
connector on the back of the MultiVOIP.
Connect the other end to the trunk line.
Verify that the E&M Type in the E&M Options group of the Interface
dialog box is the same as the E&M trunk type supported by the
telephone switch. See Appendix B for an E&M cabling pinout.
For a DID connection.
(DID Example: DID fax system or DID voice phone lines.)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO
connector on the back of the MultiVOIP.
Connect the other end to the DID jack.
NOTE: DID lines are polarity sensitive. If, during testing, the DID line
rings busy consistently, you will need to reverse the polarity of
one end of the connector (swap the connections of the wires to
the two middle pins of one RJ-11 connector).
6. Repeat the above step to connect the remaining telephone equipment
to the second channel on your MultiVOIP.
7. Ensure that the unit is properly connected to earth ground by
verifying that it is reliably grounded when mounted within a rack.
This can be accomplished by connecting a grounding wire between
the chassis and a metallic object that will provide an electrical
ground.
8. Turn on power to the MultiVOIP by placing the ON/OFF switch on
the back panel to the ON position. Wait for the BOOT LED on the
MultiVOIP to go off before proceeding. This may take a few minutes.
Proceed to the Software Installation chapter to load the MultiVOIP
software.
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Chapter 4: Software Installation
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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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2. Insert the MultiVOIP CD into your CD-ROM drive. The CD should
start automatically. It may take 10 to 20 seconds for the Multi-Tech
CD installation window to display.
If the Multi-Tech Installation CD window does not display
automatically, click My Computer, then right click the CD ROM
drive icon, click Open, and then click the Autorun icon.
3. When the Multi-Tech Installation CD dialog box appears, click the
Install Software icon.
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4. A ‘welcome’ screen appears.
Press Enter or click Next to continue.
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5. Follow the on-screen instructions to install your MultiVOIP software.
The first screen asks you to choose the folder location of the files of
the MultiVOIP software.
Choose a location and click Next.
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6. At the next screen, you must select a program folder location for the
MultiVOIP software program icon.
Click Next. Transient progress screens will appear while files are
being copied.
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7. On the next screen you can select the COM port that the command
PC will use when communicating with the MultiVoip unit. After
software installation, the COM port can be re-set in the MultiVOIP
Software (from the sidebar menu, select Connection | Settings to
access the COM Port Setup screen or use the keyboard shortcut Ctrl
+ G).
NOTE: If the COM port setting made
here conflicts with the actual COM
port resources available in the
command PC, this error message will
appear when the MultiVOIP program
is launched. If this occurs, you must
reset the COM port.
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8. Transient screens will flash by as files are being copied.
Then 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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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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4. A completion screen will appear.
Click Finish.
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Chapter 5:Technical Configuration
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Technical Configuration
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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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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Technical Configuration
Functional Equivalence of Interfaces. The MultiVOIP configuration
program is required to do the initial configuration (that is, setting an IP
address for the MultiVOIP unit) so that the VOIP unit can communicate
with the MultiVoipManager program or with the web browser GUI.
Management of the VOIP after that point can be done from any of these
three programs since they all offer essentially the same functionality.
Functionally, either the MultiVoipManager program or the web
browser GUI can replace the MultiVOIP configuration program after
the initial configuration is complete (with minor exceptions, as noted).
WARNING: Do not attempt to interface the MultiVOIP unit with
two control programs simultaneously (that is, by
accessing the MultiVOIP configuration program via
the Command Port and either the
MultiVoipManager program or the web browser
interface via the Ethernet Port). The results of using
two programs to control a single VOIP
simultaneously would be unpredictable.
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Local Configuration
This manual primarily describes local configuration with the Windows
GUI. After IP addresses have been set locally using the Windows GUI,
most aspects of configuration (logging functions are an exception) can
be handled through the web browser GUI, as well (see the Operation and
Maintenance chapter of this manual). In most aspects of configuration,
the Windows GUI and web-browser GUI differ only graphically, not
functionally. For information on SNMP remote configuration and
management, see the MultiVoipManager documentation.
Pre-Requisites
To complete the configuration of the
MultiVOIP unit, you must know several
things about the overall system.
Before configuring your MultiVOIP Gateway unit, you must know the
values for several IP and telephone parameters that describe the IP
network system and telephony system (PBX or telco central office
equipment) with which the digital MultiVOIP will interact. If you plan
to receive log reports on phone traffic by email (SMTP), you must
arrange to have an email address assigned to the VOIP unit on the
email server on your IP network. A summary of this configuration
information appears on page 58 (“Config Info CheckList”).
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.
ꢁ
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.
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Write down the values for these IP parameters. You will need to enter
these values in the “IP Parameters” screen in the Configuration section
of the MultiVOIP software. You must have this IP information about
every VOIP in the system.
Telephony Interface Parameters
The following parameters must be known about the PBX or telco
central office equipment to which the analog MultiVOIP will connect:
Phone Parameters
ꢁ
Ask phone company or
telecom manager.
Telephony Interface Parameters:
Record for this VOIP Site
#
• Which interface type is to be used?
E&M_____ FXS/FXO_____ DIP/DPO _____
• If FXS, determine whether the line will be used for a
phone, fax, or KTS (key telephone system)
• If FXO, determine if line will be an analog PBX
extension or an analog line from a telco central office
• If E&M, determine these aspects of the E&M trunk
line from the PBX:
• What is its Type (1, 2, 3, 4, or 5)?
• Is it 2-wire or 4-wire?
• Is it Dial Tone or Wink?
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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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Technical Configuration
Config Info CheckList
Type of Configuration
Info Gathered
MultiVOIP
Info
Info
Configuration
screen on which
to enter the Info
Obtained Entered
√
√
IP Info for voip unit
● IP address
Ethernet/IP
Parameters
● Gateway
● DNS IP (if used)
●802.1p Prioritization
(if used)
Interface Type
(Choices: E&M,
Interface
Parameters
FXS/FXO*, DIP, DPO)
*In FXO/FXS
systems, channels
used for phone,
fax, or key
system are FXS;
channels used
for analog PBX
extensions or
analog telco lines
are FXO.
E&M info
(only if E&M is used)
● Type (1-5)
Interface
Parameters
● 2 or 4 wires?
● Dial Tone or Wink?
Country Code
Regional
Parameters
Email address for voip
(optional)
SMTP Parameters
Reminder: Be sure to Save Setup after entering
configuration values.
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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 Telephony Interface Parameters.
10. Set SIP Call Signaling parameters.
12. Set Regional Parameters (Phone Signaling Tones & Cadences and
setup for built-in Remote Configuration/Command Modem).
13. Set Custom Tones and Cadences (optional).
14. Set SMTP Parameters (applicable if Log Reports are via Email).
15. Set Log Reporting Method (GUI, locally in MultiVOIP
Configuration program; or SMTP, via email).
16. Set Supplementary Services Parameters. The Supplementary
Services screen allows voip deployment of features that are normally
found in PBX or PSTN systems (e.g., call transfer and call waiting).
17. Set 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).
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20. Set SIP Server Configuration parameters.
21. Set SIP Server PreDefined Endpoint parameters.
22. View System Info screen and set updating interval (optional).
23. Save the MultiVOIP configuration.
24. 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, four error messages will appear (saying “MultiVOIP-
SS Not Found,” “Phone Database Not Read,” “SIP Endpoint Database
Not Read,” and “Password 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)
Valuee Description
Field Name
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
Dynamic
Description
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
Jitter Buffer
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
Description
Maximum
Jitter Value
60 to 400
ms
The maximum dynamic jitter buffer
of 400 milliseconds is the maximum
delay tolerable over a high jitter
network.
Default = 300 msec
Optimizat-
ion Factor
0 to 12
The Optimization Factor
determines how quickly the length
of the Dynamic Jitter Buffer is
changed based on actual jitter
encountered on the network.
Selecting the minimum value of 0
means low voice delay is desired,
but increases the possibility of jitter-
induced voice quality problems.
Selecting the maximum value of 12
means highest voice quality under
jitter conditions is desired at the
cost of increased voice delay.
Default = 7.
Modem Relay
To place modem traffic onto the voip network (an application called “modem relay”),
use Coder G.711 mu-law at 64kbps.
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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 Telephony Interface Parameters. This dialog box can be
reached by pulldown menu, toolbar icon, keyboard shortcut, or sidebar.
Accessing Telephony Interface Parameters
Pulldown
Icon
--
Shortcut
Sidebar
Ctrl + Alt + N
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In each field, enter the values that fit your particular network.
The kinds of parameters for which values must be chosen depend on
the type of telephony supervisory signaling or interface used (FXO,
E&M, etc.). We present here the various parameters grouped and
organized by interface type.
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Note that Interface parameters are applied on a channel-by-channel
basis. However, once you have established a set of Interface
parameters for a particular channel, you can apply this entire set of
Voice/FAX parameters to another channel by using the Copy Channel
button and its dialog box. To copy a set of Interface parameters to all
channels, select “Copy to All” and click Copy.
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FXS Loop Start Parameters. The parameters applicable to FXS Loop
Start are shown in the figure below and described in the table that
follows.
FXS Loop Start Interface: Parameter Definitions
Field Name
Values
Description
FXS (Loop
Start)
Y/N
Enables FXS Loop Start
interface type.
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FXS Loop Start Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Dialing Options fields
Inter Digit
Timer
1 - 10 seconds
This is the length of time that
the MultiVOIP will wait
between digits. When the time
expires, the MultiVOIP will
look in the outbound
phonebook for the number
entered and place the call
accordingly.
Default = 2.
Message
Waiting
--
Not applicable to FXS Loop
Start interface
Indication
The length of time between the
outputting of DTMF digits.
Default = 100 ms.
Inter Digit
Regeneration
Time
in milliseconds
FXS Options fields
Maximum number of rings that
the MultiVOIP will issue before
giving up the attempted call.
When enabled, the MultiVOIP
will interrupt loop current in
the FXS circuit to initiate a
disconnection. This tells the
device connected to the FXS
port to hang up. The Multi-
VOIP cannot drop the call; the
FXS device must go on hook.
When selected, this option
implements Answer
FXS Ring
Count , FXS
1-99
Current Loss
Y/N
Generate
Current
Reversal
Y/N
Supervision and Disconnect
Supervision to the FXO
interface using current reversal
to indicate events. Applicable
only when FXS and FXO
interfaces are connected back to
back.
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FXS Loop Start Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Flash Hook Options fields
not applicable to FXS interface
Generation
--
Detection
Range
for Min. and Max.,
50 - 1500
For a received flash hook to be
regarded as such by the
milliseconds
MultiVOIP, its duration must
fall between the minimum and
maximum values given here.
When enabled, this parameter
creates an open audio path
through the MultiVOIP.
Pass Through
Enable
Y/N
If the Pass-Through feature is
enabled, the AutoCall feature
must be enabled for this voip
channel in the Voice/Fax
Parameters screen.
Caller ID fields
The MultiVOIP currently
supports only one
Type
Bellcore
implementation of Caller ID.
That implementation is Bellcore
type 1 with Caller ID placed
between the first and second
rings of the call.
Caller ID information is a
description of the remote
calling party received by the
called party. The description
has three parts: name of caller,
phone number of caller, and
time of call. The ‘time-of-call’
portion is always generated by
the receiving MultiVOIP unit
(on FXS channel) based on its
date and time setup.
Enable
Y/N
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FXS Loop Start Interface: Parameter Definitions (cont’d)
Field Name
Values
Description
Caller ID fields
The forms of the ‘Caller Name’
and ‘Caller Phone Number’
differ depending on the IP
transmission protocol used
(H.323, SIP, or SPP) and upon
entries in the phonebook
Enable (cont’d) Y/N
screens of the remote (CID
generating) voip unit. The CID
Name and Number appearing
on the phone at the terminating
FXS end will come either from a
central office switch (showing a
PSTN phone number), or the
phonebook of the remote (CID
sending) voip unit.
The Caller ID feature has dependencies on both the telco central office
and the MultiVOIP phone book. See the diagram series after the FXO
Parameters section below.
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FXO Parameters. The parameters applicable to the FXO telephony
interface type are shown in the figure below and described in the table
that follows.
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FXO Interface: Parameter Definitions*
Field Name
Values
Description
Interface Type
FXO
Enables FXO functionality
Dialing Options
Regeneration
Pulse, DTMF
Determines whether digits
generated and sent out will be
pulse tones or DTMF.
Inter Digit
Timer
1 to 10 seconds
This is the length of time that
the MultiVOIP will wait
between digits. When the time
expires, the MultiVOIP will
look in the phonebook for the
number entered.
Default = 2.
Message
Waiting
--
Not applicable to FXO interface.
Indication
Inter Digit
Regeneration
Time
50 to 20,000
milliseconds
The length of time between the
outputting of DTMF digits.
Default = 100 ms.
FXO Options
FXO Ring
Count
1-99
Number of rings required
before the MultiVOIP answers
the incoming call.
No Response
Timer
1 – 65535
(in seconds)
Length of time before call
connection attempt is
abandoned.
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FXO Interface: Parameter Definitions (cont’d)
Field Name Values Description
Flash Hook Options fields
Generation
50 - 1500
milliseconds
Length of flash hook that will
be generated and sent out when
the remote end initiates a flash
hook and it is regenerated
locally. Default = 600 ms.
Detection
Range
--
Not applicable to FXO.
Caller ID fields
Caller ID Type Bellcore
The MultiVOIP currently
supports only one
implementation of Caller ID.
That implementation is Bellcore
type 1 with caller ID placed
between the first and second
rings of the call.
Caller ID information is a
description of the remote
calling party received by the
called party. The description
has three parts: name of caller,
phone number of caller, and
time of call. The ‘time-of-call’
portion is always generated by
the receiving MultiVOIP unit
(on FXS channel) based on its
date and time setup. The forms
of the ‘Caller Name’ and ‘Caller
Phone Number’ differ
Caller ID
enable
Y/N
depending on the IP
transmission protocol used
(H.323, SIP, or SPP) and upon
entries in the phonebook
screens of the remote (CID
generating) voip unit. The CID
Name and Number appearing
on the phone at the terminating
FXS end will come either from a
central office switch (showing a
PSTN phone number), or the
phonebook of the remote (CID
sending) voip unit.
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The Caller ID feature has dependencies on both the telco central office
and the MultiVOIP phone book. See the diagram series below.
CID Flow
Call is received
here.
Call originates here
at 1:42pm, May 31.
CID
Terminating
VoIP
CID
Generating
VoIP
Central Office
with
standard telephony
Caller ID service
FXO
FXS
IP
Network
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
Clock:
5-31,
1:42pm
phone of:
Display shows:
H.323 or SPP
Melvin Jones
763-555-8794
Protocol
*
CID Number: 763-555-8794
CID Name: Melvin Jones
Time Stamp: Date: 05/31
Time:1:42pm
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
*
Figure 5-1: Voip Caller ID Case #1 – Call, through telco
central office with standard CID, enters voip system
CID Flow
Call is received
Call originates here
at 4:19pm, July 10.
here.
CID
Generating
VoIP
CID
Ch1
Terminating
VoIP
Central Office
without
FXO
FXS
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
standard telephony
Caller ID service
Ch3
Ch4
Clock:
7/10, 4:19pm
phone of:
Display shows:
CID Number: 423
H.323 Protocol
*
Wilda Jameson
763-555-4071
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 7/10
Time: 4:19pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Remove Prefix Add Prefix Forward/Addr
*
Gatekeeper RAS Param
423
748
Figure 5-2: Voip Caller ID Case #2 – Call, through telco
central office without standard CID, enters H.323 voip system
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CID Flow
IP
Call is received
here.
Call originates here
at 5:47pm, Sept 27.
Central Office
Ch1
Generating
Terminating
VoIP
VoIP
FXO
FXS
without
Ch2
x
x
x
y
y
y
z
z
z
z
x
x
x
y
yy
zz
zz
standard telephony
Caller ID service
J. Q. P u bl ci
J. Q. P u bl ci
Network
Ch3
Ch4
Clock:
15:26, 5-31
phone of:
Display shows:
SPP Protocol
Henry Brampton
763-555-4077
CID Number: 423
{Channel 2}
Inbound Phone Book
CID Name: Shipping Dept
Remove Prefix Add Prefix Forward/Addr
Time Stamp: Date: 0927
Time: 1747
... if “Description” field in Add/Edit
Inbound Phone Book is used
423
748
Phone Book Configuration
Anoka-Whse-VP3
Gateway Name:
OR
Add/Edit Inbound Phone Book
Use as default entry
CID Number: 423
Remove Prefix: rs
Add Prefix:
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 0927
Time: 1747
... if “Description” in Add/Edit
Inbound Phone Book is blank
Channel Number: Channel 2
Description: Shipping Dept
Figure 5-3: Voip Caller ID Case #3 – Call, through telco
central office without standard CID, enters SPP voip system
CID Flow
Call is received
Call originates here
at 4:51pm, Oct 3.
here.
CID
Generating
VoIP
CID
Ch1 FXS
401
Terminating
VoIP
xxxyyyzzzz
J.Q. Public
FXS
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
402
403
phone of:
Nigel Thurston
763-555-9401
Ch3
Ch4
Clock:
10/03, 4:51pm
404
Display shows:
H.323 Protocol
*
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 10/03
Time: 4:51pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
Remove Prefix Add Prefix Forward/Addr
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
*
Gatekeeper RAS Param
423
748
Figure 5-4: Voip Caller ID Case #4 – Remote FXS call on
H.323 voip system
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CID Flow
IP
Call is received
Call originates here
at 6:17pm, Nov 15.
here.
CID
CID
Generating
VoIP
Ch1
Terminating
Central Office
VoIP
FXS
DID
without
Ch2
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
standard telephony
Caller ID service
Network
Ch3
Ch4
Clock:
11/15, 6:17pm
phone of:
Display shows:
H.323 Protocol
*
Edwin Smith
763-743-5873
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 11/15
Time: 6:17pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Remove Prefix Add Prefix Forward/Addr
*
Gatekeeper RAS Param
423
748
Figure 5-5: Voip Caller ID Case #5 – Call through telco central
office without standard CID enters DID channel in H.323 voip
system
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FXO Supervision. When the selected Interface type is FXO, the Supervision button
is active. Click on this button to access call answering supervision parameters and
call disconnection parameters that relate to the FXO interface type.
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FXO Supervision Parameter Definitions
Field Name
Values
Description
Answer Supervision fields
Current
Reversal
Y/N
When this option is selected, the
FXO interface sends notice to
make connection upon
detecting current reversal from
the PBX (which occurs when
the called extension goes
offhook).
Answer Delay
Y/N
When this option is selected, the
FXO 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
integer values
(in seconds)
When Answer Delay is enabled,
this value determines when the
FXO interface sends the
Range = 1 - 65535
connection notice.
Tone Detection Y/N
When selected, call
disconnection will be triggered
by a tone sequence.
Available
Tones
dial tone,
ring tone,
List from which tones can be
chosen to signal call answer.
busy tone,
unobtainable
tone (fast busy),
survivability
tone,
re-order tone
Answer Tones
any tone from
Available Tones
list
Currently chosen call-answer
supervision tone.
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FXO Supervision Parameter Definitions
Field Name
Values
Description
Disconnect Supervision fields
There are four possible criteria
for disconnection under FXO:
current reversal, current loss,
tone detection, and silence
detection. Disconnection can be
triggered by more than one of
the three criteria.
Current
Reversal
Y/N
Y/N
Disconnection to be triggered
by reversal of current from the
PBX.
Current Loss
Disconnection to be triggered
by loss of current. That is,
when Current Loss is enabled
(“Y”), the MultiVOIP will hang
up the call at a specified
interval after it detects a loss of
current initiated by the attached
device.
Current Loss
Timer
200 to 2000
(in milliseconds)
Determines the interval after
detection of current loss at which
the call will be disconnected.
Silence
Detection
Enable
Y/N
Enables/disables silence-
detection method of
supervising call disconnection.
Silence
Detection Type Two-Way
One-Way or
Disconnection to be triggered
by silence in one direction only
or in both directions
simultaneously.
Silence Timer
in seconds
integer value
Duration of silence required to
trigger disconnection.
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FXO Supervision Parameter Definitions
Field Name Values Description
Disconnect Supervision fields
DTMF Tone
Enables supervision of call
disconnection using DTMF
tones.
DTMF Tone Pairs
Low Tones
2
5
8
0
3
6
9
#
A
B
C
D
697Hz
770Hz
852Hz
941Hz
1
4
7
*
High Tones 1209Hz 1336Hz 1447Hz 1633Hz
Disconnect
Tone Sequence
1st tone pair
+
2nd tone pair
These are DTMF tone pairs.
Values for first tone pair are:
*, #, 0, 1-9, and A-D.
Values for second tone pair are:
none, 0, 1-9, A-D, *, and #.
The tone pairs 1-9, 0, *, and #
are the standard DTMF pairs
found on phone sets. The tone
pairs A-D are “extended
DTMF” tones, which are used
for various PBX functions.
Tone Detection Y/N
Enables supervision of call
disconnection by detecting
cessation of a pre-specified tone
from the PBX.
Available
Tones
dial tone,
ring tone,
busy tone,
List from which tones can be
chosen to signal call
disconnection.
unobtainable
tone (fast busy),
survivability
tone,
re-order tone
Disconnect
Tones
any tone from
Available Tones
list
Currently chosen disconnection
supervision tone.
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E&M Parameters. The parameters applicable to the E&M telephony
interface type are shown in the figure below and described in the table
that follows.
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E&M Interface Parameter Definitions
Field Name
Interface
Type
Values
E&M
Description
enables E&M functionality
Types 1-5.
Refers to the type of E&M
interface being used.
Mode
Signal
2-wire or 4-wire
Each E&M interface type can be
either 2-wire or 4-wire audio.
Dial Tone or
Wink
When Dial Tone is selected, no
wink is required on the E lead
or M lead in the call initiation or
setup.
When Wink is selected, a wink
is required during call setup.
Wink Timer
(in ms)
integer values,
in milliseconds
This is the length of the wink
for wink signaling.
Applicable only when Signal
parameter is set to “Wink.”
No Response
Timer
integer values (in The value here denotes the time
seconds)
(in seconds) after which the call
attempt would be disconnected
by the FXO Interface because
there was no answer.
Disconnect on
Call Progress
Tone
Y/N
Allows call on FXO port to be
disconnected when a PBX issues a
call-progress tone denoting that
the phone station on the PBX that
has been involved in the call has
been hung up.
Pass Through
Enable
Y/N
When enabled (“Y”), this
feature is used to create an open
audio path for 2- or 4-wire. The
E&M leads are passed through
the voip transparently.
Applicable only for E&M
Signaling with Dial Tone (not
applicable for Wink signaling).
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E&M Interface Parameter Definitions (cont’d)
Field Name Values Description
Dialing Options
Inter Digit integer values,
This is the length of time that
the MultiVOIP will wait
Timer
in seconds
between digits. When the time
expires, the MultiVOIP will
look in the phonebook for the
number entered. Default = 2.
Message
Waiting
Indication
Light or None
Allows MultiVOIP to pass
mode-code sequences between
Avaya Magix PBXs to turn on
and off the message-waiting
light on a PBX extension phone.
Mode codes:
*53 + PBX extension
Î turns message light on.
#53 + PBX extension
Î turns message light off.
Signals to turn message-waiting
lights on/off are not sent to
phones connected directly to
the MultiVOIP on FXS
channels, not to other non-
Avaya Magix PBX phone
stations on the voip network.
Inter Digit
Regeneration
Timer
milliseconds
The length of time between the
outputting of DTMF digits.
Default = 100 ms.
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E&M Interface Parameter Definitions (cont’d)
Field Name Values Description
Dialing Options (cont’d)
Flash Hook Options fields
Generation
integer values, in Length of flash hook that will
milliseconds
be generated and sent out when
the remote end initiates a flash
hook and it is regenerated
locally. Default = 600 ms.
Detection
Range
for Min. and Max.,
50 1500
For a received flash hook to be
regarded as such by the
milliseconds
MultiVOIP, its duration must
fall between the minimum and
maximum values given here.
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DID Parameters. The parameters applicable to the Direct Inward Dial
(DID) telephony interface type are shown in the figure below and
described in the table that follows. The DID interface allows one phone
line to direct incoming calls to any one of several extensions without a
switchboard operator. Of course, one DID line can handle only one call
at a time. The parameters described here pertain to the customer-
premises side of the DID connection (DID-DPO, dial-pulse originating);
the network side of the DID connection (DID-DPT, dial-pulse
terminating) is not supported.
DID Interface Parameter Definitions
Field Name
Values
Description
Interface
DID-DPO
Enables the customer-premises
side of DID functionality
DID Options
MultiVOIP’s use of DID applies
only for incoming DID calls.
The Start Mode used by the
MultiVOIP must match that
used by the originating
telephony equipment, else DID
calls cannot be completed.
Start Modes
Immediate Start,
Wink Start,
Delay Dial
For Immediate Start, the voip
detects the off-hook condition
initiated by the telco central-
office call and becomes ready to
receive dial digits immediately.
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DID Interface Parameter Definitions (cont’d)
Field Name Values Description
DID Options (cont’d)
Start Modes Immediate Start,
For Wink Start, the voip detects
the off-hook condition. Then
the voip reverses battery
polarity for a specified time
(140-290 ms; a “wink”) and then
becomes ready to receive dial
digits.
Wink Start,
Delay Dial
For Delay Dial, the voip detects
detects the off-hook condition.
Then the voip reverses battery
polarity for a specified time
(reverse polarity duration has
wider acceptable range than for
Wink Start) and then becomes
ready to receive dial digits.
Wink Timer
(in ms)
integer values,
in milliseconds
This is the length of the wink
for Wink Start and Delay Dial
signaling modes..
Applicable only when Start
Mode parameter is set to “Wink
Start” or “Delay Dial.”
Dialing Options
Inter Digit
Timer
integer values,
in seconds
This is the length of time that
the MultiVOIP will wait
between digits. When the time
expires, the MultiVOIP will
look in the phonebook for the
number entered.
Default = 2.
Message
Waiting
--
Not applicable to DID-DPO
interface.
Indication
Inter-Digit
Regeneration
Timer
integer values,
in milliseconds
This parameter is applicable
when digits are dialed onto a
DID-DPO channel after the
connection has been made. The
length of time between the
outputting of DTMF digits.
Default = 100 ms.
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10. Set Call Signaling Parameters. This dialog box addresses SIP Call
Signaling parameters. It can be reached by pulldown menu,
keyboard shortcut, or a sidebar menu.
Accessing “Call Signaling Parameters”
Pulldown
Shortcut
Sidebar
Ctrl + Alt + Shft + P
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The tables below describes all fields in the general SIP Call Signaling screen.
SIP Call Signaling Parameter Definitions
Field Name
SIP Proxy Parameters
Signaling Port
Values
Description
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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Technical Configuration
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 MVP410 and MVP810
MultiVOIP unit contains a built-in modem. This modem allows the
MultiVOIP to be configured remotely when a standard POTS line is
connected to the “Command Modem” connector on the back panel of
the MultiVOIP. In the Country Selection for Built-In Modem field
(drop-down list), select the country that best fits your situation. This
may not be the same as your selection for the Country/Region field.
The selections in the Country Selection for Built-In Modem field
entail more detailed groupings of telephony parameters than do the
Country/Region values.
In each field, enter the values that fit your particular system.
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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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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
If you enable console messages, you can customize the types of
messages to be included/excluded in log reports by clicking on the
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“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 two log reporting methods,
GUI, or SMTP, 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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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 also be
used under SIP, but they are implemented differently in SIP than in
H.323. Even though the H.450 standard refers only to H.323,
Supplementary Services are still applicable to the SIP and SPP voip
protocols, in which cases these features are implemented differently.
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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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The Supplementary Services fields are described in the tables below.
Supplementary Services Parameter Definitions
Field Name Values
Description
Select
Channel
1-4 (410SS) The channel to be configured is
1-8 (810SS) selected here.
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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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);
NAT/STUN 3478
)
numeric; The data port (TDM time slot) at
default=
which STUN info will be transmitted
and received.
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-SS
supports the authentication functions. 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
Server
Address
n.n.n.n
0 – 255
IP address of the RADIUS server that
handles accounting 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.
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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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“Custom Fields” Definitions (cont’d)
Field
Description
Field
Description
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 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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20. Set SIP Server Configuration parameters.
Accessing SIP Server Configuration Parameters
Sidebar
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SIP Server Configuration Parameter Definitions
Field Name Values
Description
Operating
Mode
surviv.,
In “Survivability” mode, the MVP-SS
stnd-alone unit can function as a SIP server for
other gateways in its network in case
that network loses contact with the
network’s main SIP server (typically a
PBX). When in “Survivability” mode,
the MVP-SS unit is, essentially, a
backup SIP server.
In “Stand-Alone” mode, the MVP-SS
functions as a primary SIP server for
other gateways. In stand-alone mode,
the MVP-SS operate to technical
advantage with ‘smart’ SIP phones.
Such smart SIP phones can choose the
SIP server under which they operate
and, consequently, can be controlled
by either the SIP-based PBX or by the
MVP-SS.
Survivability
Status Check
Register,
Options
One of two status-check packets is
sent to the main SIP Proxy servers to
which the MVP-SS serves as a backup.
Regardless of the packet type used,
this packet determines whether the
MVP-SS needs to take over SIP server
functions or stay in its normal backup
mode. “Options” and “Register” are
two distinct SIP request “methods.”
The Options method solicits
information but does not set up a
connection. The Register method
conveys information about a user’s
location to the SIP server. The
“Register” method may entail more
data overhead than the “Options”
method. If both of these methods are
supported by your SIP server, it is OK
to use either one. If only one is
supported, use the supported method.
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SIP Server Configuration Parameter Definitions
Field Name Values
Registrar Options
Description
Allow
Undefined
Registrations
Y/N
If undefined registrations are allowed
(value= Y), then gateways other than
those listed in the PreDefined
Endpoints list can register with the
MVP-SS voip unit as it functions in its
SIP server mode. If undefined
registrations are allowed, then
incoming registrations will be allowed
if they originate from endpoints at
accepted domains or accepted IP
addresses (specified below in this
software screen).
Accept
any
Determines whether registrations to
Registrations
for:
domains; the MVP-SS SIP server will be
specific
accepted from any domain or only
from specified domains. Multiple
domains can be listed, separated by
semicolons. The “any domains”
option is intended for private
networks not accessible via Internet or
PSTN.
domains
Domain Names this.com; List (entries separated by semicolon)
that.org
etc.
of domains of endpoints from which
the MVP-SS will accept registrations.
Accept
any IP
Determines whether registrations to
Registrations
for:
addresses; the MVP-SS SIP server will be
specific IP
accepted from any IP address or only
from specified IP addresses. Multiple
IP addresses can be listed, separated
by semicolons. The “any IP
addresses
addresses” option is intended for
private networks not accessible via
Internet or PSTN.
IP Addresses
a.b.c.d;
q.r.s.t;
for
values
0-255
List (entries separated by semicolon)
of IP addresses of endpoints from
which the MVP-SS will accept
registrations.
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SIP Server Configuration Parameter Definitions
Field Name Values
Registrar Options
Description
Re-
integer
The time after which the MultiVOIP
Registration
Time
values; in UserAgent Client is supposed to register
seconds;
default is
3600
with the proxy server.
Expiration of the registration interval
means that the gateway has lost contact
with the main SIP server and that the
MVP-SS unit will enter its ‘survivability’
mode. In survivability mode, the MVP-SS
unit will complete calls acting as a backup
to the main SIP server. Normally,
however, the MVP-SS will initiate re-
registration with some small margin of
time before the interval lapses.
21. Set SIP Server | PreDefined Endpoint parameters. In this screen you
will specify the voip gateways that will depend on the MVP-SS unit
either as their primary SIP server (if the MVP-SS is used in “Stand-
Alone” mode, as set in the SIP Server | Configuration screen) or as
their backup SIP server (if the MVP-SS is used in “Survivability”
mode, as set in the SIP Server |Configuration screen).
Accessing “Predefined Endpoints” Parameters
Pulldown
Icon
--
Shortcut
Sidebar
Ctrl + Alt + 9
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The main screen for Predefined Endpoints is a list. If you click on function
buttons to Add or Edit entries in this list of endpoints, a secondary screen will
appear and allow you to add new endpoints or edit existing endpoint entries.
When your work with the list is complete, click Save.
SIP Server Predefined Endpoints Parameter Definitions
Field Name Values
Description
Endpoint
Name
alpha-
numeric
Identifier for gateway within SIP voip
system. Max. length is 33 characters.
Password
alpha-
numeric
This password is for authentication of
gateway to SIP server.
Registration
Type
Static,
Static registrations are fixed and the
Dynamic contact information for them is
configured by the user and not subject
to removal from the registration list
due to timeouts.
Dynamic registrations are registered
from an external endpoint with the
contact information. Dynamic entries
must re-register before the re-
registration interval expires else they
will be removed from the list.
Endpoints removed from this list can
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neither make nor receive calls.
Re-Registration integer
The time after which the MultiVOIP
Interval
values; in UserAgent Client is supposed to register
seconds;
default is
3600
with the proxy server.
Expiration of the registration interval
means that the gateway has lost contact
with the main SIP server and that the
MVP-SS unit will enter its ‘survivability’
mode. In survivability mode, the MVP-SS
unit will complete calls acting as a backup
to the main SIP server. Normally,
however, the MVP-SS will initiate re-
registration with some small margin of
time before the interval lapses.
Contact Information
Address
a.b.c.d
for
The IP address at which this endpoint
can be reached.
values
0-255
Port
0 – 64000 Digital time slot on which SIP calls
will be made. Default is 5060
Re-
Registration
Time
--
See “Re-Registration Interval” entry
above.
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22. 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
nn.nn.nn. Indicates version of MultiVOIP
Configur-
ation
Version
nn
alpha-
numeric
Configuration software (which
includes screens for IP Parameters,
SMTP Parameters, Regional
Parameters, etc.).
Phone Book
Version
numeric
Indicates the version of the inbound
and outbound phonebook portion of
the MultiVOIP software.
IFM Version numeric
Indicates the version of the firmware
running on the MultiVOIP’s Interface
Module, which is its analog telephony
hardware.
Mac
Address
alpha-
numeric
Denotes the number assigned as the
voip unit’s unique Ethernet address.
Up Time
days:
hours:
mm:ss
Indicates how long the voip has been
running since its last booting.
Hardware
ID
alpha-
numeric
Indicates the version of the
MultiVOIP unit’s circuit board and
components.
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The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
23. 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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24. 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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MultiVOIP User Guide
T1 Phonebook Configuration
Chapter 6:T1 Phonebook
Configuration
(North American Telephony Standards)
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T1 Phonebook Configuration
MultiVOIP User Guide
T1 Versus E1 Telephony Environments
Phonebooks for Series II analog MultiVOIP units (MVP130,
MVP130FXS, MVP210, MVP410, and MVP810) can be operated in
either a North American telephony standards environment (potentially
operating with T1 digital MultiVOIPs) or in a European telephony
standards environment (potentially operating with E1 digital
MultiVOIPs). The configuration of the phonebook is the same in either
case. However, because the telephony environment is different in each
case and the examples used here must reflect those differences, we have
separate chapters for phonebook configuration in North American (T1)
environments (Chapter 6; this chapter) and for that in European (E1)
environments (Chapter 7). Consult the chapter that best fits the needs
of your voip system.
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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T1 PhoneBook Configuration
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
MultiVOIP User Guide
Phonebook configuration screens can be accessed using icons or the
sidebar menu.
Phonebook Icons
Description
Phonebook Configuration
Inbound Phonebook
Entries List
Add Inbound Phonebook
Entry
Edit selected Inbound
Phonebook Entry
Outbound Phonebook
Entries List
Add Outbound
Phonebook Entry
Edit selected Outbound
Phonebook Entry
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T1 PhoneBook Configuration
Phonebook 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 describe various SIP parameters.
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
SIP Proxy routing device is
used.
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 SIP proxy routing
device. 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
n = 0-255
The IP address to which the
call will be directed if it
begins with the destination
pattern given.
Description
alpha-
numeric
Describes the facility or
geographical location at
which the call will be
completed.
Indicates protocol to be used in
outbound transmission. For the
MVP-SS units, only SIP is used.
Protocol Type
SIP or H.323
or SPP
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T1 Phonebook Configuration
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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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MultiVOIP User Guide
T1 PhoneBook Configuration
Clicking on the Advanced button brings up the Alternate Routing secondary screen.
This feature provides an alternate path for calls if the primary IP network cannot carry
the traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN.
However, this feature could also be used to divert traffic to a redundant (backup) unit
in case one voip unit fails. The user must specify the IP address of the alternate route
for each destination pattern entry in the Outbound Phonebook.
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T1 Phonebook Configuration
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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
MultiVOIP User Guide
Add/Edit Inbound Phone Book: Field Definitions
(cont’d)
Field Name
Values
Description
Call Forward Parameters
Unconditional. When selected,
all calls received will be
forwarded.
Busy. When selected, calls
will be forwarded when
station is busy.
Forward
Condition
Uncondit.;
Busy
No Resp.
No Response. When selected,
calls will be forwarded if
called party does not answer
after a specified number of
rings, as specified in Ring
Count field.
Forwarding can be
conditioned on both “Busy”
and “No Response.”
Forward
Destination
Phone number or IP address to which calls
will be directed.
IP address,
For SIP calls, the Forward Destination can be
phone number, one of the following:
port number,
etc.
(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.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Add/Edit Inbound Phone Book: Field Definitions
(cont’d)
Field Name
Values and Description
Ring Count
0, 1, 2, 3, etc. When “No Response” is
condition for forwarding calls, this
determines how many unanswered rings
are needed to trigger the forwarding.
Registration
Option
In a SIP voip system, gateways can register
with the SIP Proxy.
Parameters
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
MultiVOIP User Guide
T1 Phonebook Examples
The following example demonstrates how Outbound and Inbound
PhoneBook entries work in a situation of multiple area codes. Consider
a company with offices in Minneapolis and Baltimore. The system
depicted is H.323. However, the phonebook entries presented are still
applicable for SIP systems.
3 Sites, All-T1 Example
Notice first the area code situation in those two cities: Minneapolis’s
local calling area consists of multiple adjacent area codes; Baltimore’s
local calling area consists of a base area code plus an overlay area code.
Company
VOIP/PBX
5
Baltimore/
SIte
Outstate MD
Overlay
443
NW
Suburbs
St. Paul
& Suburbs
651
763
Mpls
612
Company
VOIP/PBX
SIte
...
5
SW Suburbs
952
Baltimore
410
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MultiVOIP User Guide
T1 PhoneBook Configuration
An outline of the equipment setup in both offices is shown below.
Local-Call
Area Codes:
612, 651,
952
Company HQ.
Minneapolis
North Sub.
area 763
T1
Digital
VoIP
PBX
-5174
200.2.10.3
-5173
-5172
-5171
717-5170
IP
Network
Overlay
Area Code:
443
Baltimore
ꢀ
Sales Ofc.
area 410
ꢀ
Digital
VoIP
R
o
u
t
e
r
T1
PBX
-7003
200.2.9.7
-7002
325-7001
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T1 Phonebook Configuration
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The screen below shows Outbound PhoneBook entries for the VOIP
located in the company’s Baltimore facility.
The entries in the Minneapolis VOIP’s Inbound PhoneBook match the
Outbound PhoneBook entries of the Baltimore VOIP, as shown below.
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T1 PhoneBook Configuration
To call the Minneapolis/St. Paul area, a Baltimore employee must dial
eleven digits. (In this case, we are assuming that the Baltimore PBX
does not require an “8” or “9” to seize an outside phone line.)
If a Baltimore employee dials any phone number in the 612 area code,
the call will automatically be handled by the company’s voip system.
Upon receiving such a call, the Minneapolis voip will remove the digits
“1612”. But before the suburban-Minneapolis voip can complete the
call to the PSTN of the Minneapolis local calling area, it must dial “9”
(to get an outside line from the PBX) and then a comma (which denotes
a pause to get a PSTN dial tone) and then the 10-digit phone number
which includes the area code (612 for the city of Minneapolis; which is
different than the area code of the suburb where the PBX is actually
located -- 763).
A similar sequence of events occurs when the Baltimore employee calls
number in the 651 and 952 area codes because number in both of these
area codes are local calls in the Minneapolis/St. Paul area.
The simplest case is a cal from Baltimore to a phone within the
Minneapolis/St. Paul area code where the company’s voip and PBX are
located, namely 763. In that case, that local voip removes 1763 and
dials 9 to direct the call to its local 7-digit PSTN.
Finally, consider the longest entry in the Minneapolis Inbound
Phonebook, “17637175. Note that the main phone number of the
Minneapolis PBX is 763-717-5170. The destination pattern 17637175
means that all calls to Minneapolis employees will stay within the
suburban Minneapolis PBX and will not reach or be carried on the local
PSTN.
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T1 Phonebook Configuration
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Similarly, the Inbound PhoneBook for the Baltimore VOIP (shown first
below) generally matches the Outbound PhoneBook of the Minneapolis
VOIP (shown second below).
Notice the extended prefix to be removed: 14103257. This entry allows
Minneapolis users to contact Baltimore co-workers as though they were
in the Minneapolis facility, using numbers in the range 7000 to 7999.
Note also that a comma (as in the entry 9,443) denotes a delay in
dialing. A one-second delay is commonly used to allow a second dial
tone to be generated for calls going outside of the facility’s PBX system.
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MultiVOIP User Guide
T1 PhoneBook Configuration
The Outbound PhoneBook for the Minneapolis VOIP is shown below.
The third destination pattern, “7” facilitates reception of co-worker calls
using local-appearing-extensions only. In this case, the “Add Prefix”
field value for this phonebook entry would be “1410325” .
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T1 Phonebook Configuration
MultiVOIP User Guide
Configuring Mixed Digital/Analog VOIP Systems
Analog MultiVOIP units, like the MVP-210/410/810/410SS/810SS 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-210/410xx/810xx
MultiVOIP units (Series II units), legacy analog VOIP units (Series I
units made by MultiTech) may be included in the system, as well.
When legacy VOIP units are included, the VOIP administrator must
handle two styles of phonebooks in the same VOIP network. The
diagram below shows a small-scale system of this kind: one digital
VOIP (the MVP2410) operates with two Series II analog VOIPs (an
MVP210 and an MVP410), and two Series I legacy VOIPs (two MVP200
units).
EXAMPLE:
Site D:
Digital & Analog VOIPs
Pierre, SD
in Same System
Area Code 615
PSTN
PBX
200.2.9.9
Digital
VoIP
MVP2410
T1
Other extensions
x3101 - x3199
Router
615-492-3100
Site E:
Site A:
Cheyenne, WY
Area Code 307
Bismarck, ND
Area Code 701
200.2.9.6
Series #1 Analog MultiVOIP
(Server/Client Phonebook)
MVP200
Series #2 Analog MultiVOIP
MVP210
FXS
Unit
FXS
CH1
#200
CH1
421
201
200.2.9.7
Client
IP
Network
Site F:
Lincoln, NE
Area Code 402
Site B:
Rochester, MN
Area Code 507
200.2.9.5
FXO
Series #1 Analog MultiVOIP
(Server/Client Phonebook)
PSTN
Series #2 Analog MultiVOIP
MVP410
Port #4
102
MVP200
CH2
FXS
FXO
Unit
#100
CH1
FXS Port
FXS Ports
CO Ports
717-5000
200.2.9.8
Host
(Holds phonebook for both
Series #1 analog VOIPs.)
CO Port
Key
System
Other extensions
x7401 - x7429
PSTN
402-263-7400
507-717-5662
Site C:
Suburban Rochester
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MultiVOIP User Guide
T1 PhoneBook Configuration
The Series I analog VOIP phone book resides in the “Host” VOIP unit at
Site B. It applies to both of the Series I analog VOIP units.
Each of the Series II analog MultiVOIPs (the MVP210 and the MVP410)
requires its own inbound and outbound phonebooks. The MVP2410
digital MultiVOIP requires its own inbound and outbound
phonebooks, as well.
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T1 Phonebook Configuration
MultiVOIP User Guide
These seven phone books are shown below.
Phone Book for Series I Analog VOIP Host Unit (Site B)
VOIP Dir #
-OR-
IP Address Channel Comments
Destination
Pattern
102
101
200.2.9.8
200.2.9.8
2
1
Site B, FXS channel.
Site B, FXO
channel.
421
201
200.2.9.6
200.2.9.7
0
1
Site E FXS channel.
Site A, FXS
channel.
1615
xxx
xxxx
200.2.9.9
200.2.9.9
200.2.9.5
200.2.9.5
0
Gives remote voip
(Note 2.) users access to local
PSTN of Site D
(Pierre, SD, area
code 615).
3xxx
0
0
0
Allows remote voip
users to call all PBX
extensions at Site D
(Pierre, SD) using
only four digits.
(Note 1.)
1402
Gives remote voip
users access to local
PSTN of Site F
(Lincoln, NE; area
code 402).
140226374
(Note 1)
(Note 3)
Gives remote voip
users access to key
phone system
extensions at Site F
(Lincoln).
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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
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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
Inbound Phonebook for MVP2410 Digital VOIP (Site D)
Remove Add
Channel
Number
0
Comment
Prefix
Prefix
1615
9,
Allows phone users at remote
voip sites to call non-toll
numbers within the Site D area
code (615; Pierre, SD) over the
VOIP network.
Note 4.
Note 5.
1615
49231
31
0
Allows voip calls directly to
employees at Site D (at
extensions x3101 to x3199).
Note 4. “9” gives PBX station users access to outside line.
Note 5. The comma represents a one-second pause, the
time required for the user to receive a dial tone on
the outside line (PSTN). The comma is only
allowed in the Inbound phonebook.
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T1 Phonebook Configuration
MultiVOIP User Guide
Outbound Phone Book for MVP410 Analog VOIP
(Site F)
IP
Destin.
Pattern
201
Remove
Prefix
Add
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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MultiVOIP User Guide
T1 PhoneBook Configuration
Inbound Phonebook for MVP410 Analog VOIP (Site F)
Remove Add
Channel
Number
4
Comment
Prefix
Prefix
1402
Access to Lincoln local PSTN by
users at remote VOIP locations
via FXO port at Site F.
1402
263740
1402
263741
1402
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
MultiVOIP User Guide
Outbound Phone Book for MVP210 Analog VOIP
(Site E)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A.
1507
1507
101#
Note 3.
200.2.9.8 To originate calls
to any PSTN
phone in
Rochester area
using the FXO
channel (channel
#1) of the Site B
VOIP.
102
200.2.9.8 To originate calls
to phone
connected to FXS
port (channel #2)
of the Site B VOIP.
200.2.9.5 Calls to Lincoln
area PSTN (via
1402
FXO channel,
CH4, of the Site F
VOIP).
7
1402
263
200.2.9.5 Calls to Lincoln
key extensions
with four digits.
200.2.9.9 Calls to Pierre area
PSTN via Site D
PBX.
200.2.9.9 Calls to Pierre PBX
extensions with
four digits.
1615
31
1615
492
Note 3. The pound sign (“#”) is a delimiter separating the
VOIP number from the standard telephony phone number.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Inbound Phonebook for MVP210 Analog VOIP (Site E)
Remove Add
Channel
Number
1
Comment
Prefix
Prefix
421
Call Completion Summaries
Site A calling Site C, Method 1
1. Dial 101.
2. Hear dial tone from Site B.
3. Dial 7175662.
4. Await completion. Talk.
Site A calling Site C, Method 2
1. Dial 101#7175662
2. Await completion. Talk.
Note: Some analog VOIP gateways will allow
completion by Method 2. Others will not.
Site C calling Site A
1. Dial 7175000.
2. Hear dial tone from Site B VOIP.
3. Dial 201.
4. Await completion. Talk.
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T1 Phonebook Configuration
MultiVOIP User Guide
Site D calling Site C
1. Dial 9,15077175662.
2. “9” gets outside line. On some PBXs, an “8” may be used to
direct calls to the VOIP, while “9” directs calls to the PSTN.
However, some PBX units can be programmed to identify the
destination patterns of all calls to be directed to the VOIP.
3. PBX at Site D is programmed to divert all calls made to the 507
area code and exchange 717 into the VOIP network. (It would
also be possible to divert all calls to all phones in area code 507
into the VOIP network, but it may not be desirable to do so.)
4. The MVP2410 removes the prefix “1507” and adds the prefix
“101#” for compatibility with the analog MultiVOIP’s
phonebook scheme. The “#” is a delimiter separating the
analog VOIP’s phone number from the digits that the analog
VOIP must dial onto its local PSTN to complete the call. The
digits “101#7175662” are forwarded to the Site B analog VOIP.
5. The call passes through the IP network (in this case, the
Internet).
6. The call arrives at the Site B VOIP. This analog VOIP receives
this dialing string from the MVP2410: 101#7175662. The analog
VOIP, seeing the “101” prefix, uses its own channel #1 (an FXO
port) to connect the call to the PSTN. Then the analog VOIP
dials its local phone number 7175662 to complete the call.
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MultiVOIP User Guide
T1 PhoneBook Configuration
Site D calling Site F
A voip call from Pierre PBX to extension 7424 on the key telephone system in Lincoln,
Nebraska.
A. The required entry in the Pierre Outbound Phonebook to facilitate
origination of the call, would be 1402263742. The call would be directed to
the Lincoln voip’s IP address, 200.2.9.5.
(Generally on such a call, the caller would have to dial an initial “9.” But
typically the PBX would not pass the initial “9” to the voip. If the PBX did
pass along that “9” however, its removal would have to be specified in the
local Outbound Phonebook.)
B. The corresponding entry in the Lincoln Inbound Phonebook to facilitate
completion of the call would be
1402263742
1402
for calls within the office at Lincoln
for calls to the Lincoln local calling area (PSTN).
Call Event Sequence
1. Caller at Pierre dials 914022637424.
2. Pierre PBX removes “9” and passes 14022637424 to voip.
3. Pierre voip passes remaining string, 14022637424 on to the Lincoln
voip
at IP address 200.2.9.5.
4. The dialed string matches an inbound phonebook entry at the
Lincoln voip, namely 1402263742.
5. The Lincoln voip rings one of the three FXS ports connected to the
Lincoln
key phone system.
6. The call will be routed to extension 7424 either by a human
receptionist/
operator or to an auto-attendant (which allows the caller to specify
the
extension to which they wish to be connected).
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T1 Phonebook Configuration
MultiVOIP User Guide
Site F calling Site D
A voip call from a Lincoln key extension to extension 3117 on the PBX in Pierre, South
Dakota.
A. The required entry in the Lincoln Outbound Phonebook to facilitate
origination of the call, would be “31”. The string “1615492” would have to be
added as a prefix. The call would be directed to the Pierre voip’s IP address,
200.2.9.9.
B. The corresponding entry in the Pierre Inbound Phonebook to facilitate
completion of the call would be 1615492.
1. Caller at Lincoln picks up phone receiver, presses button on key
phone set. This button has been assigned to a particular voip
channel (any one of the three FXS ports).
2. The caller at Lincoln hears dial tone from the Lincoln voip.
3. The caller at Lincoln dials 3117.
4. The Lincoln voip adds the prefix 1615492 and sends the entire
dialing string, 16154923117, to the Pierre voip
at IP address 200.2.9.9.
5. The Pierre voip matches the called digits 16154923117 to its
Inbound Phonebook entry “1615492” .
6. The Pierre PBX dials extension 3117 in the office at Pierre.
Variations in PBX Characteristics
The exact dialing strings needed in the Outbound and Inbound
Phonebooks of the MVP2410 will depend on the capabilities of the PBX.
Some PBXs require trunk access codes (like an “8” or “9” to access an
outside line or to access the VOIP network). Other PBXs can
automatically distinguish between intra-PBX calls, PSTN calls, and
VOIP calls.
Some PBX units can also insert digits automatically when they receive
certain dialing strings from a phone station. For example, a PBX may
be programmable to insert automatically the three-digit VOIP identifier
strings into calls to be directed to analog VOIPs.
The MVP2410 offers complete flexibility for inter-operation with PBX
units so that a coherent dialing scheme can be established to connect a
company’s multiple sites together in a way that is convenient and
intuitive for phone users. When working together with modern PBX
units, the presence of the MVP2410 can be completely transparent to
phone users within the company.
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MultiVOIP User Guide
E1 Phonebook Configuration
Chapter 7: E1 Phonebook
Configuration
(European Telephony Standards)
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E1 Phonebook Configuration
MultiVOIP User Guide
E1 Versus T1 Telephony Environments
Phonebooks for Series II analog MultiVOIP units (MVP130,
MVP130FXS, MVP210, MVP410, MVP810, MVP210SS, MVP410SS, and
MVP810SS) can be operated in either an environment of either North
American telephony standards (potentially operating with T1 digital
MultiVOIPs) or of European telephony standards (potentially operating
with E1 digital MultiVOIPs). The configuration of the phonebook is the
same in either case. However, because the telephony environment is
different in each case and the examples used here must reflect those
differences, we have separate chapters for phonebook configuration in
North American (T1) environments (Chapter 6) and for that in
European (E1) environments (Chapter 7; this shapter). Consult the
chapter that best fits the needs of your voip system.
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
onto a single network. Typically
these are private networks.
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MultiVOIP User Guide
E1 PhoneBook Configuration
Free Calls: One VOIP Site to Another
The most direct use of the VOIP system is making calls between the
offices where the VOIPs are located. Consider, for example, the Wren
Clothing Company. This company has VOIP-equipped offices in
London, Paris, and Amsterdam, each served by its own PBX. VOIP
calls between the three offices completely avoid international long-
distance charges. These calls are free. The phonebooks can be set up to
allow all Wren Clothing employees to contact each other using 3-, 4-, or
5-digit numbers, as though they were all in the same building.
United Kingdom
Wren Clothing Co.
5 Wren Clothing Co.
VOIP/PBX Site
VOIP/PBX Site
5London
Amsterdam
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Paris
Free VOIP Calls
France
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E1 Phonebook Configuration
MultiVOIP User Guide
Local Rate Calls: Within Local Calling Area of Remote
VOIP
In the second use of the VOIP system, the local calling area of each
VOIP location becomes accessible to all of the VOIP system’s users. As
a result, international calls can be made at local calling rates. For
example, suppose that Wren Clothing buys its zippers from The
Bluebird Zipper Company in the western part of metropolitan London.
In that case, Wren Clothing personnel in both Paris and Amsterdam
could call the Bluebird Zipper Company without paying international
long-distance rates. Only London local phone rates would be charged.
This applies to calls completed anywhere in London’s local calling area
(which includes both Inner London and Outer London). Generally,
local calling rates apply only within a single area code, and, for all calls
outside that area code, national rates apply. There are, however, some
European cases where local calling rates extend beyond a single area
code. Local rates between Inner and Outer London are one example of
this. (It is also possible, in some locations, that calls within an area code
may be national calls. But this is rare.)
United Kingdom
Wren Clothing Co.
5Wren Clothing Co.
VOIP/PBX Site
Bluebird Zipper Co.
VOIP/PBX Site
5London
London
Amsterdam
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Calls at London local rates
Paris
Local Calling Area
France
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MultiVOIP User Guide
E1 PhoneBook Configuration
Similarly, the VOIP system allows Wren Clothing employees in London
and Amsterdam to call anywhere in Paris at local rates; it allows Wren
Clothing employees in Paris and London to call anywhere in
Amsterdam at local rates.
United Kingdom
Wren Clothing Co.
VOIP/PBX Site
Amsterdam
Wren Clothing Co.
VOIP/PBX Site
5London
5
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Paris
Calls at Amsterdam local rates
Calls at Paris local rates
Local Calling Areas
France
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E1 Phonebook Configuration
MultiVOIP User Guide
National Rate Calls: Within Nation of Remote VOIP Site
In the third use of the VOIP system, the national calling area of each
VOIP location becomes accessible to all of the VOIP system’s users. As
a result, international calls can be made at national calling rates. Again,
significant savings are possible. For example, suppose that the Wren
Clothing Company buys its buttons from the Chickadee Button
Company in the Dutch city of Rotterdam. In that case, Wren Clothing
personnel in both London and Paris could call the Chickadee Button
Company without paying international long-distance rates; only Dutch
national calling rates would be charged. This applies to calls completed
anywhere in The Netherlands.
United Kingdom
The
Wren Clothing Co.
VOIP/PBX Site
Netherlands
5London
Wren Clothing Co.
5
VOIP/PBX Site
Amsterdam
Chickadee Button Co.
Rotterdam
Wren Clothing Co.5
VOIP/PBX Site
Paris
Calls at Dutch
National Rates
France
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MultiVOIP User Guide
E1 PhoneBook Configuration
Similarly, the VOIP system allows Wren Clothing employees in London
and Amsterdam to call anywhere in France at French national rates; it
allows Wren Clothing employees in Paris and Amsterdam to call
anywhere in the United Kingdom at its national rates.
United Kingdom
Wren Clothing Co.
VOIP/PBX Site
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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E1 Phonebook Configuration
MultiVOIP User Guide
In order for any VOIP phone call to be made, there must be both an
Inbound Phonebook entry and an Outbound Phonebook entry that
describe the end-to-end connection. The phone station originating the
call must be connected to the VOIP system. The Outbound Phonebook
for that VOIP unit must have a destination pattern entry that includes
the ‘called’ phone (that is, the phone completing the call). The Inbound
Phonebook of the VOIP where the call is completed must have a
destination pattern entry that includes the digit sequence dialed by the
originating phone station.
The PhoneBook Configuration procedure below is brief, but it is
followed by an example case. For many people, the example case may
be easier to grasp than the procedure steps. Configuration is not
difficult, but all phone number sequences, destination patterns, and
other information must be entered exactly; otherwise connections will
not be made.
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MultiVOIP User Guide
E1 PhoneBook Configuration
Phonebook configuration screens can be accessed using icons or the
sidebar menu.
Phonebook Icons
Description
Phonebook Configuration
Inbound Phonebook
Entries List
Add Inbound Phonebook
Entry
Edit selected Inbound
Phonebook Entry
Outbound Phonebook
Entries List
Add Outbound
Phonebook Entry
Edit selected Outbound
Phonebook Entry
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E1 Phonebook Configuration
MultiVOIP User Guide
Phonebook Pulldown Menu
Inbound Phonebook Shortcut
Outbound Phonebook
Shortcut
Alt + I
Alt + O
Phonebook Sidebar Menu
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MultiVOIP User Guide
E1 PhoneBook Configuration
Phonebook Configuration Procedure
1. Select Outbound Phone Book/List Entries.
Click Add.
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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
The fields of the Add/Edit Outbound Phone Book screen are described
in the table below.
Add/Edit Outbound Phone Book: Field Definitions
Field Name
Values
Description
Accept Any
Number
Y/N
When checked, “Any
Number” appears as the
value in the Destination
Pattern field.
The Any Number feature
works differently depending
on whether or not an external
routing device is used
(Gatekeeper for H323
protocol, Proxy for SIP
protocol, Registrar for SPP
protocol).
When no external routing
device is used. If Any
Number is selected, calls to
phone numbers not matching
a listed Destination Pattern
will be directed to the IP
Address in the Add/Edit
Outbound Phone Book
screen. “Any Number” can
be used in addition to one or
more Destination Patterns.
When external routing
device is used. If Any
Number is selected, calls to
phone numbers not matching
a listed Destination Pattern
will be directed to the
external routing device used
(Gatekeeper for H323
protocol, Proxy for SIP
protocol, Registrar for SPP
protocol). The IP Address of
the external routing device
must be set in the Phone
Book Configuration screen.
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E1 Phonebook Configuration
MultiVOIP User Guide
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.
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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
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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-
210x/410x/810x) used in Euro-type telephony settings will be the same
in format as entries for the MVP3010.
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MultiVOIP User Guide
E1 PhoneBook Configuration
France Country Code: 33
Lille
Paris: Area 01
Reims
Rouen
Nantes
Strasbourg
Lyon
Bordeaux
Toulouse
Marseille
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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 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 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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
Outbound Phone Book for MVP410 Analog VOIP
(Site F)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A
(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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MultiVOIP User Guide
E1 PhoneBook Configuration
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.
275
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E1 Phonebook Configuration
MultiVOIP User Guide
Outbound Phone Book for MVP210 Analog VOIP
(Site E)
IP
Destin.
Pattern
201
Remove Add
Prefix
Comment
Prefix
Address
200.2.9.7 To originate calls
to Site A
(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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MultiVOIP User Guide
E1 PhoneBook Configuration
Inbound Phonebook for MVP210 Analog VOIP (Site E)
Remove Add
Channel
Number
1
Comment
Prefix
Prefix
421
Call Completion Summaries
Site A calling Site C, Method 1
1.
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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
MultiVOIP User Guide
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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MultiVOIP User Guide
E1 PhoneBook Configuration
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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E1 Phonebook Configuration
MultiVOIP User Guide
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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MultiVOIP User Guide
Operation & Maintenance
Chapter 8: Operation and
Maintenance
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Operation & Maintenance
MultiVOIP User Guide
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. Also, the
SIP Server | Endpoint Statistics screen presents statistical information
unique to the MVP SS MultiVOIP units.
SIP Server Endpoint Statistics screen
This screen shows values previously entered in the Add Predefined
Endpoint screen as well as various measures of the IP phone traffic that
have occurred on each endpoint in the SIP system. This is a screen
whereupon settings may be read and performance data may be read.
However, no parameter values are set on this screen.
Accessing “Endpoint Statistics” screens
Pulldown
Icon
--
Shortcut
Sidebar
Ctrl + Alt + 1
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SIP Server Endpoint Statistics Parameter Definitions
Field Name Values
Description
Endpoint
Name
alpha-
numeric
Identifier for gateway within SIP voip
system. Max. length is 33 characters.
Status
server
Indicates the SIP server that is
identifier controlling traffic for this endpoint.
Max. Expiry numeric
Time in sec.
Indicates the time remaining before
the endpoint’s registration with the
SIP server has expired.
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SIP Server Endpoint Statistics Parameter Definitions
Field Name Values
Description
Initiated
Call Count
numeric
numeric
numeric
Indicates how many calls were
initiated by phones connected to this
endpoint.
Received
Call Count
Indicates how many calls were
received by phones connected to this
endpoint.
No. of
Entries
Indicates how many endpoints are
included in the system.
Registration Static,
Static registrations are fixed and the
Type
Dynamic contact information for them is
configured by the user and not subject
to removal from the registration list
due to timeouts.
Dynamic registrations are registered
from an external endpoint with the
contact information. Dynamic entries
must re-register before the re-
registration interval expires else they
will be removed from the list.
Endpoints removed from this list can
neither make nor receive calls.
Endpoint
Type
pre/un -
defined
Indicates whether the listed endpoint
has been predefined within the SIP
system or is an endpoint using the SIP
server under rules of open access to
endpoints at specified URLs or
domain names.
Contact
Address
a.b.c.d
for
values
0-255
The IP address at which this endpoint
can be reached.
Port
Number
0 – 64000 Indicates the digital time slot on
which SIP calls will be made.
Default is 5060
Remaining
Time
numeric
in sec.
Indicates the time remaining before
the endpoint’s registration with the
SIP server has expired.
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The illustration below shows the SIP Server Endpoint Statistics screen
for an active SIP phone system in web GUI format.
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System Information screen
This screen presents vital system information at a glance. Its primary
use is in troubleshooting. This screen is accessible via the
Configuration pulldown menu, the Configuration sidebar menu, or by
the keyboard shortcut Ctrl + Alt + Y.
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System Information Parameter Definitions
Field Name Values
Description
Boot
Version
nn.nn
alpha-
numeric
Indicates the version of the code that
is used at the startup (booting) of the
voip. The boot code version is
independent of the software version.
Firmware
Version
nn.nn.nn Indicates the version of the
alpha-
MultiVOIP firmware.
numeric
Configur-
ation
Version
nn.nn.
nn.nn
alpha-
numeric
Indicates the version of the
MultiVOIP configuration software.
Phone Book
Version
nn.nn
alpha-
numeric
Indicates the version of the
MultiVOIP phone book being used.
IFM Version nn
alpha-
Indicates version of the IFM module,
the device that performs the
transformation between telephony
signals and IP signals.
numeric
Mac
Address
numeric
Denotes the number assigned as the
voip unit’s unique Ethernet address.
Up Time
days:
hours:
mm:ss
Indicates how long the voip has been
running since its last booting.
Hardware
ID
alpha-
numeric
Indicates version of the MultiVOIP
circuit board assembly being used.
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The frequency with which the System Information screen is updated is
determined by a setting in the Logs screen
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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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The Call Progress Details Screen
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Call Progress Details: Field Definitions
Field Name
Values
Description
Channel
1-n
Number of data channel or time
slot on which the call is carried.
This is the channel for which call-
progress details are being viewed.
Call Details
Duration
Mode
Hours:
Minutes:
Seconds
The length of the call in hours,
minutes, and seconds (hh:mm:ss).
Indicates whether the call being
described was a voice call or a
FAX call.
Voice or FAX
Voice Coder
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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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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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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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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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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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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About Logs
The Logs
Accessing “Statistics: Logs”
Pulldown Icon
Shortcut
Sidebar
Ctrl + O
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The Logs Screen
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Logs Screen Details: Field Definitions
Field Name
Values
Description
Log # column
1 or higher
All calls are assigned an event
number in chronological order,
with the most recent call having
the highest event number.
Start Date,Time
column
dd:mm:yyyy
hh:mm:ss
The starting time of the call (event).
The date is presented as a day
expression of one or two digits, a
month expression of one or two
digits, and a four-digit year. This is
followed by a time-of-day expression
presented as a two-digit hour, a two-
digit minute, and a two-digit seconds
value. (statistics, logs) field
This describes how long the call
(event) lasted in hours, minutes, and
seconds.
Duration column hh:mm:ss
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).
Indicates whether the DTMF dialing
digits are carried "Inband" or "Out of
Band." The corresponding field
values differ for the 3 different voip
protocols.
disconnection.
DTMF Capability inband,
out of band
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".
The digits, sent by MultiVOIP to
PBX/telco, that were
(H.323, SIP, or
SPP).
Outbound Digits 0-9, #, *
Received
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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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.
Packets sent
integer value
integer value
integer value
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.
Packets received
Packets loss
(lost)
The number of voice packets from
this call that were lost after being
received from the IP network.
The number of bytes of data sent
over the IP network in the course of
this call.
Bytes sent
integer value
integer value
Bytes received
The number of bytes of data
received over the IP network in
the course of this call.
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Logs Screen Details: Field Definitions (cont’d)
Field Name Values Description
Call Details (cont’d)
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
Field
Values
Description
Name
UDP versus TCP. (User Datagram
Protocol versus Transmission Control
Protocol). UDP provides
unguaranteed, connectionless
transmission of data across an IP
network. By contrast, TCP provides
reliable, connection-oriented
transmission of data..
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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)
Values Description
Field
Name
Total Packets
(cont’d)
Sum of data packets of all types.
Received integer
Total number of error-laden packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
UDP Packets
User Datagram Protocol packets.
Transmit integer
Number of UDP packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of UDP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden UDP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
TCP Packets
Transmission Control Protocol packets.
Transmit integer
Number of TCP packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of TCP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden TCP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
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IP Statistics: Field Definitions (cont’d)
RTP Packets
Voice signals are transmitted in
Realtime Transport Protocol packets.
RTP packets are a type or subset of
UDP packets.
Transmit integer
Number of RTP packets transmitted by
this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of RTP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden RTP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
RTCP Packets
Realtime Transport Control Protocol
packets convey control information to
assist in the transmission of RTP (voice)
packets. RTCP packets are a type or
subset of UDP packets.
Transmit integer
Number of RTCP packets transmitted
by this VOIP gateway since the last
“clearing” or resetting of the counter
within the MultiVOIP software.
ted
value
Number of RTCP packets received by this
VOIP gateway since the last “clearing” or
resetting of the counter within the
MultiVOIP software.
Received integer
value
Received integer
Number of error-laden RTCP packets
received by this VOIP gateway since the
last “clearing” or resetting of the
with
value
Errors
counter within the MultiVOIP software.
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About Link Management
The Link Management screen is essentially an automated utility for
pinging endpoints on your voip network. This utility generates pings
of variable sizes at variable intervals and records the response to the
pings.
Accessing Link Management
Pulldown
none
Shortcut // Icon
Sidebar
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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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
Icon
Shortcut
Sidebar
Ctrl + Alt + W
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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 Factory Defaults Select this to return the configuration
parameters to the original factory
values.
Download Firmware
Select this to download new versions
of firmware as enhancements become
available.
Download IFM Firmware
Select this to download new versions of
IFM firmware as enhancements become
available. The Interface Module (IFM) is
the telephony interface for analog
MultiVOIP units (MVP130, MVP130FXS,
MVP210, MVP410, MVP810). There is one
IFM for each channel of the MultiVOIP
unit. For each channel, the IFM handles
the analog signals to and from the attached
telephone, PBX or CO line.
Download User Defaults
Set Password
To be used after a full set of parameter
values, values specified by the user,
have been saved (using Save Setup).
This command loads the saved user
defaults into the MultiVOIP.
Select this to create a password for
access to the MultiVOIP software
programs (Program group commands,
Windows GUI, web browser GUI, &
FTP server). Only the FTP Server
function requires a password for access.
The FTP Server function also requires
that a username be established along
with the password.
Uninstall
Select this to uninstall the MultiVOIP
software (most, but not all components
are removed from computer when this
command is invoked).
Upgrade Software
Loads firmware (including H.323
stack) and 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 Factory Defaults
1. The MultiVoip Configuration program must be off when invoking
the Download Factory Defaults command. If it is on, the command
will not work.
2.To invoke the Download Factory Defaults command, go to Start |
Programs | MVP____ x.xx | Download Factory Defaults.
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3. If a password has been established, the Password Verification screen
will appear.
Type in the password and click OK.
4. The MVP____- Firmware screen appears saying “MultiVOIP [model
number] is up. Reboot to Download Firmware?”
Click OK to download the factory defaults.
The “Boot” LED on the MultiVOIP will light up and remain lit during
the file transfer process.
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5. After the PC gets a response from the MultiVOIP, the Dialog – IP
Parameters screen will appear.
The user should verify that the correct IP parameter values are listed
on the screen and revise them if necessary. Then click OK.
6. Progress bars will appear at the bottom of the screen during the data
transfer.
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.
7. The Download Factory Defaults procedure is complete.
Downloading IFM Firmware
The Interface Module (IFM) is the telephony interface for analog
MultiVOIP units (MVP130, MVP130FXS, MVP210, MVP410, MVP810).
There is one IFM for each channel of the MultiVOIP unit. For each
channel, the IFM handles the analog signals to and from the attached
telephone, PBX or CO line. The IFM communicates with the main
processor indicating the status of the telephone line. For example, it
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might indicate that a phone is off hook (FXS) or that an incoming ring is
present (FXO). The IFM receives operating instructions from the voip’s
main processor. For example, the IFM might be instructed to ring the
phone (FXS) or seize the line (FXO). The IFM contains a codec
(coder/decoder) to convert the incoming audio to a PCM stream (pulse
code modulation) which it sends to the DSP (digital signal processor).
The IFM’s codec also converts outgoing PCM to audio.
The firmware of the IFMs will change from time to time and you may
need to upgrade the firmware on your MultiVOIP unit. To do so,
follow these instructions.
1. In the System Information screen of the MultiVOIP Configuration
software, check the version number of the IFM firmware already
installed on the MultiVOIP unit. Write down the version number.
2. Exit the Configuration software program. The MultiVoip
Configuration program must be off when invoking the Download
IFM Firmware command. If it is on, the command will not work.
3.To invoke the Download IFM Firmware command, go to Start |
Programs | MVP____ x.xx | Download IFM Firmware.
4. A warning window will appear: “Downloading IFM Firmware will
reboot the MultiVOIP. Do you want to continue?” Click OK.
4. The “Boot” LED on the front panel of the MultiVOIP will come on.
5. The software will search for an IFM firmware file to use to upgrade
the system. If the file found represents firmware newer than that
already installed on the MultiVOIP (or if you want to overwrite the
same version of firmware) click Open.
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6. The IFM Firmware Download screen will appear. Select “Copy to
All IFMs” and click OK. (Only in very special circumstances would
different IFMs in the same voip be loaded with different IFM
firmware.)
7. The main MultiVOIP Configuration screen will appear. Progress
bars can be seen at the bottom of the screen while files are being
copied.
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8. Then a completion screen entitled IFM Test will appear.
Click OK.
9. The MultiVOIP will reboot itself. When the reboot is complete, the
MultiVOIP Configuration screen will close.
10. The IFM firmware downloading process is complete.
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Setting and Downloading User Defaults
The Download User Defaults command allows you to maintain a known
working configuration that is specific to your VOIP system. You can then
experiment with alterations or improvements to the configurations confident
that a working configuration can be restored if necessary.
1. Before you can invoke the Download User Defaults command, you
must first save a set of configuration parameters by using the Save
Setup command in the sidebar menu of the MultiVOIP software.
2. Before the setup configuration is saved, you will be prompted to save
the setup as the User Default Configuration. Select the checkbox and
click OK.
Save Current Setup as User Default Configuration
MultiVOIP _____ will be brought down.
OK
Cancel
Help
A user default file will be created. 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.
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4. A confirmation screen will appear indicating that this action will
entail rebooting the MultiVOIP.
Click OK.
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.
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.
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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.
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 settings. The
settings can be either Factory Default Settings or Current Configuration
Settings.
NOTE: To upgrade a MultiVOIP from software version 6.04 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 6.08 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 6.04 or earlier versions using FTP. You
must use ‘upgradesoftware’ via the serial port.
Important: These ftp upgrade instructions do not apply to software release
6.05 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 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.
Note, however, that before any CAS file can be used as an update, it
must be renamed to CASFILE.CAS so that it overwrites and replaces
the default CAS file.
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.
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, however, cannot perform logging in the same direct
mode done in the Windows GUI. However, when the web GUI is used,
logging can be done by email (SMTP).
The graphic layout of the web GUI is also somewhat larger-scale than
that of the Windows GUI. For that reason, it’s helpful to use as large of
a video monitor as possible.
The primary advantage of the web GUI is remote access for control and
configuration. The controller PC and the MultiVOIP unit itself must
both be connected to the same IP network and their IP addresses must
be known.
In order to use the web GUI, you must also install a Java application
program on the controller PC. This Java program is included on the
MultiVOIP product CD. ). Java is needed to support drop-down menus
and multiple windows in the web GUI.
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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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Operation & Maintenance
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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Warranty, Service, & Tech Support
MultiVOIP User Guide
Chapter 9 Warranty, Service, and
Tech Support
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Warranty, Service, & Tech Support
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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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
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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Regulatory Information
MultiVOIP User Guide
Chapter 10: Regulatory Information
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Regulatory Information
EMC, Safety, and R&TTE Directive Compliance
The CE mark is affixed to this product to confirm compliance with the
following European Community Directives:
Council Directive 89/336/EEC of 3 May 1989 on the approximation of the
laws of Member States relating to electromagnetic compatibility,
and
Council Directive 73/23/EEC of 19 February 1973 on the harmonization of
the laws of Member States relating to electrical equipment designed for use
within certain voltage limits,
and
Council Directive 1999/5/EC of 9 March 1999 on radio equipment and
telecommunications terminal equipment and the mutual recognition of their
conformity.
FCC Declaration
NOTE: This equipment has been tested and found to comply with the
limits for a Class A digital device, pursuant to Part 15 of the FCC Rules.
These limits are designed to provide reasonable protection against
harmful interference when the equipment is operated in a commercial
environment. This equipment generates, uses and can radiate radio
frequency energy, and if not installed and used in accordance with the
instructions, may cause harmful interference to radio communications.
Operation of this equipment in a residential area is likely to cause
harmful interference in which case the user will be required to correct
the interference at his own expense.
This device complies with Part 15 of the FCC rules.
Operation is subject to the following two conditions:
(1) This device may not cause harmful interference.
(2) This device must accept any interference that may cause
undesired operation.
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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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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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MultiVOIP User Guide
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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MultiVOIP User Guide
Appendix A: Cable Pinouts
Appendix A: Cable Pinouts
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MultiVOIP User Guide
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
1 2 3 4 5 6 7 8
8
3
2
6
1
5
CLEAR TO SEND
TRANSMIT DATA
RECEIVE DATA
To DTE
To Command
Port Connector
Device
(e.g., PC)
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
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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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
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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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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TCP/UDP Port Assignments
Appendix B: TCP/UDP Port
Assignments
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TCP/UDP Port Assignments
MultiVOIP User Guide
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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Index
Index
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Index
MultiVOIP User Guide
INDEX
802.1p Priority Levels .......... 104, 105
abbreviated dialing, inter-office
E1.............................................. 243
T1.............................................. 207
Accept Any Number (inbound)
E1.............................................. 259
T1.............................................. 219
Accept Any Number (outbound) field
E1.............................................. 253
T1.............................................. 212
Accept Registrations for domains
field
accessing Voice/FAX Parameters
screen ........................................112
Accounting Port (RADIUS screen)
field ...........................................189
Add Inbound Phonebook Entry icons
E1..............................................249
T1..............................................208
Add Outbound Phonebook Entry icon
E1..............................................249
T1..............................................208
Add Prefix (inbound) field
E1..............................................259
T1..............................................219
Add Prefix (outbound) field
E1..............................................254
T1..............................................213
Add/Edit Inbound Phonebook field
definitions
SIP Server Configuration
parameters............................. 196
Accept Registrations for IP addresses
field
SIP Server Configuration
parameters............................. 196
accessing Statistics, Logs screen
.................................................. 299
accessing Call Progress (Statistics)
screen........................................ 291
accessing configuration parameter
groups ....................................... 101
accessing Ethernet/IP Parameters
screen........................................ 102
accessing interface parameters...... 126
accessing IP Statistics screen........ 306
accessing Logs (Statistics) screen
.................................................. 299
accessing logs screen .................... 169
accessing Regional Parameters..... 153
accessing Registered Gateway Details
(Statistics) screen...................... 316
accessing Registered Gateway
Details screen.................. 315, 316
accessing RTP Parameters screen. 321
accessing SMTP parameters ......... 162
accessing Supplementary Services
screen........................................ 173
accessing System Information screen
.................................................. 200
E1..............................259, 260, 261
T1..............................219, 220, 221
Add/Edit Inbound Phonebook screen
E1..............................................259
T1..............................................219
Add/Edit Inbound Phonebook screen
fields (E1)
Accept Any Number .................259
Add Prefix.................................259
Channel Number .......................260
Description (callee location) .....260
Enable (Call Forwarding)..........260
Forward Condition....................260
Forward Destination..................261
Registration Option Parameters 261
Remove Prefix ..........................259
Ring Count................................261
Add/Edit Inbound Phonebook screen
fields (T1)
Accept Any Number .................219
Add Prefix.................................219
Channel Number .......................219
Description (callee location) .....219
Enable (Call Forwarding)..........219
Forward Condition....................220
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Index
Forward Destination ................. 220
Registration Option Parameters 221
Remove Prefix .......................... 219
Ring Count................................ 221
Add/Edit Outbound Phonebook field
definitions
E1.............................. 253, 254, 255
T1.............................. 212, 213, 214
Add/Edit Outbound Phonebook fields
(E1)
airflow.............................................67
Alerting Party
Supplementary Services...180, 181,
182
Allow Incoming Calls Through SIP
Proxy Only (SIP Call Signaling)
field ...........................................151
Allow Undefined Registrations field
SIP Server Configuration
parameters.............................196
Allowed Name Type
Accept Any Number................. 253
Add Prefix................................. 254
Advanced button....................... 255
Description................................ 254
destination pattern..................... 254
IP Address................................. 254
Protocol Type............................ 254
Remove Prefix .......................... 254
SIP Port Number....................... 255
SIP URL ................................... 255
Total Digits............................... 254
Transport Protocol (SIP)........... 255
Use Proxy (SIP)........................ 255
Add/Edit Outbound Phonebook fields
(T1)
Accept Any Number................. 212
Add Prefix................................. 213
Advanced button....................... 214
Description................................ 213
Destination Pattern.................... 213
IP Address................................. 213
Protocol Type............................ 213
Remove Prefix .......................... 213
SIP Port Number....................... 214
SIP URL ................................... 214
Total Digits............................... 213
Transport Protocol (SIP)........... 214
Use Proxy (SIP)........................ 214
Add/Edit Outbound Phonebook screen
E1.............................................. 252
T1.............................................. 211
Address (Contact Info)
Alerting Party............180, 181, 182
Calling Party .............................179
Allowed Name Types, Call Name ID
Alerting Party............................180
Busy Party.................................181
Calling Party .............................179
Connected Party........................182
allowing pop-ups with Web GUI ..111
Alternate IP Address field
E1..............................................257
T1..............................................216
Alternate IP Routing
E1..............................................252
T1..............................................211
Alternate Proxy 1 and 2 (SIP Call
Signaling) fields........................151
Alternate Routing
PSTN failover feature, and........216
Alternate Routing field definitions
E1..............................................257
T1..............................................216
Alternate Routing field definitions
(E1)
Alternate IP Address.................257
Round Trip Delay......................257
Alternate Routing field definitions
(T1)
Alternate IP Address.................216
Round Trip Delay......................216
analog voip product family .............10
Answer Delay (FXO answer
SIP Server Predefined Endpoint
Parameters............................. 199
Advanced button, Outbound
Phonebook
T1.............................................. 211
Advanced Features field group..... 119
supervision) field.......................140
Answer Delay Timer (FXO answer
supervision) field.......................140
answer supervision criteria, FXO..140
Answer Tones (FXO answer
supervision) field.......................140
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Append SIP Proxy Domain Name in
User ID (proxy server).............. 151
Auto Disconnect field group......... 125
AutoCall........................................ 120
AutoCall (Voice/Fax Params) and
Pass Through Enable (FXS Loop
Start) ......................................... 120
AutoCall/Offhook Alert field 120, 121
Automatic Disconnection field ..... 125
Available Tones (FXO answer
supervision) field ...................... 140
Available Tones (FXO disconnection
supervision) .............................. 142
bandwidth, coder........................... 118
battery caution ................................ 62
baud rate, default (MultiVOIP
software connection):................ 193
baud rate, fax ................................ 116
baud rate, setting........................... 193
Boot LED........................................ 18
MVP210-SS................................ 76
MVP-410SS/810SS .................... 73
Boot Version
Bytes Sent (RADIUS Attributes) field
..................................................191
Bytes Sent (SMTP logs) field .......166
Bytes sent (statistics, logs) field....303
cabling diagram, quick (210) ..........32
cabling diagram, quick (410/810)...31
cabling problem, fixing.................101
cabling procedure
MVP210-SS................................73
MVP410-SS................................69
MVP810-SS................................69
Cadence 1 (custom) field ..............161
Cadence 2 (custom) field ..............161
Cadence 3 (custom) field ..............161
Cadence 4 (custom) field ..............161
Cadence field ........................157, 158
cadences, custom
T1.E1.........................................161
cadences, signaling........................153
Call Control PHB field..................107
Call Control Priority (Ethernet/IP
parameters) field .......................105
Call Control Status
System Info....................... 201, 289
booting time.................................... 18
box contents
verifying...................................... 63
BRI connector pinout.................... 383
BRI interface types
Call Progress Details (statistics)
field .......................................298
Call Control Status (call progress)
field ...........................................298
Call Direction (SMTP logs) field..166
Call Duration field ........................125
Call Forward Parameters (inbound
phonebook)
ST and U................................... 384
built-in modem
setup in Regional Parameters
screen.............................. 96, 154
busy & no-response (E1)
E1..............................................260
T1..............................................220
Call Forwarded To
forwarding, dual conditions ...... 260
busy & no-response (T1)
logs (statistics) field ..................305
Call Hold.......................................174
Call Hold Enable...........................177
Call Mode (RADIUS Attributes) field
..................................................190
Call Mode (SMTP logs) field........165
Call Name Identification...............174
Call Name Identification
Calling Party .............................179
Call Name Identification
Alerting Party............................180
Call Name Identification
forwarding, dual conditions ...... 220
busy tone, custom ......................... 160
busy-tones..................................... 159
Bytes Received (call progress) field
.................................................. 294
Bytes Received (RADIUS
Attributes) field...................... 191
Bytes Received (SMTP logs) field166
Bytes received (statistics, logs) field
.................................................. 303
Bytes Sent (call progress) field..... 294
Alerting Party............................181
Call Name Identification
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Index
Alerting Party............................ 182
Call On Hold
Call Status (call progress) field.....298
Call Status (RADIUS Attributes) field
..................................................191
Call Status (SMTP logs) field .......166
Call Transfer .................................174
Call Transfer Enable .....................176
Call Transfer music jingle during hold
..................................................176
Call Transferred To
Call Progress Details (statistics)
field....................................... 297
Call on Hold (call progress) field.. 297
Call Progress (Statistics)............... 291
Call Progress Details (statistics) field
definitions 293, 294, 295, 296, 297,
298
Call Progress Details (statistics)
screen field
logs (statistics) field ..................305
Call Type (SMTP logs) field.........166
Call Waiting..................................174
Call Progress Details (statistics)
field ...................................297
Call Progress Details (statistics)
field .......................................297
Call Waiting (call progress) field..297
Call Waiting Enable......................177
Caller ID .......................................174
Call Progress Details (statistics)
field .......................................297
Caller ID (call progress) field .......297
Caller ID (Supplementary Services)
field ...........................................183
Caller ID enable
FXO ..........................................135
FXS Loop Start .................131, 132
Caller ID examples........136, 137, 138
Caller ID fields
FXO ..........................................135
Caller ID Type
FXO ..........................................135
FXS Loop Start .........................131
Caller Name Identification Enable 178
Calling Party
Supplementary Services............179
Canadian Class A requirements ....376
Canadian Limitations Notice
Call On Hold......................... 297
Call Waiting.......................... 297
Caller ID ............................... 297
Call Progress Details (statistics)
screen fields
Channel................................. 293
Duration................................ 293
Mode..................................... 293
Voice Coder.......................... 293
IP Call Type.......................... 293
IP Call Direction................... 293
Packets Sent.......................... 294
Packets Received .................. 294
Bytes Sent............................. 294
Bytes Received ..................... 294
Packets Lost.......................... 294
Outbound Digits Sent............ 296
Outbound Digits Received.... 296
Prefix Matched...................... 296
Server Details........................ 296
DTMF Capability.................. 296
Call On Hold......................... 297
Call Waiting.......................... 297
Caller ID ............................... 297
Call Status............................. 298
Call Control Status................ 298
Silence Compression............. 298
Forward Error Correction ..... 298
Gateway Name (from and to) ... 295
IP Address (from and to) .......... 295
Options (from and to) ............... 295
Gateway Name (from ................... 295
IP Address (from........................... 295
Options (from ............................... 295
Gateway Name (to........................ 295
IP Address (to ............................... 295
Options (to.................................... 295
(regulatory) ...............................377
CD, MultiVOIP...............................21
Channel (call progress) field.........293
channel capacity..............................13
channel capacity (analog voips)......10
channel capacity (digital voips) ........9
channel capacity (ISDN/BRI voips)11
Channel Number (inbound) field
E1..............................................260
T1..............................................219
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MultiVOIP User Guide
Channel Number (RADIUS
compatibility, H.450 with H.323, not
with SIP.......................................14
compression, silence .....................119
Compression, Silence (RADIUS
Attributes) .................................192
Compression, Silence (SMTP logs)
..................................................167
computer requirements....................19
Config Info Checklist
Attributes) field......................... 190
Channel Number (SMTP logs) field
.................................................. 165
channel tracing on/off (logging) ... 172
Checklist of configuration info....... 28
Clear (IP Statistics) button............ 308
Clear command (Link Management)
button........................................ 312
coder
bandwidth, max......................... 118
G.711 ........................................ 118
G.723.1 ..................................... 118
G.726 ........................................ 118
G.727 ........................................ 118
G.729 ........................................ 118
Net Coder.................................. 118
Coder (RADIUS Attributes) field. 191
Coder (SMTP logs) field............... 166
Coder field .................................... 118
coder options
packetization rates and.............. 321
Coder Parameters field group ....... 118
coder types (voice/fax, RTP
packetization)............................ 322
COM port
conflict, resolving ..................... 100
error message............................ 100
on command PC.......................... 83
COM port allocation..................... 193
COM port assignments ................. 193
COM port conflict
error message.............................. 83
COM Port Setup screen .......... 83, 100
command cable pinout.................. 380
command modem
Quick Start Instructions ..............28
configuration of voip
local versus remote................89, 90
Configuration option description
(MultiVOIP program menu) .....324
Configuration Parameter Groups,
accessing ...................................101
Configuration Port Setup option
description (MultiVOIP program
menu) ........................................324
configuration procedure, local
detailed........................................97
summary......................................96
Configuration Version
System Info...............................202
Configur-ation Version
System Information...................289
configuration, local .........................92
configuration, phonebook
E1..............................................248
T1..............................................207
configuration, saving.....................203
user............................................341
configuration, user default ............204
Configuring MultiVOIP phonebooks,
general
E1..............................................242
T1..............................................206
conflicts
and Regional Parameters screen 96,
154
Command Modem
setup for.............................. 96, 154
command PC
COM port....................................83
Connection Problems, Solving......100
connectivity test
COM port assignment (detailed). 83
Command PC
Quick Start Instructions ..............56
Consecutive Packets Lost field .....125
Console Message Settings, Filters for
..................................................172
console messages, enabling...........170
console parameters tracked ...........172
Contact Address
COM port requirement................ 19
non-dedicated use of................... 19
operating system ......................... 19
compatibility, H.450 services with
SIP ............................................ 173
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SIP Server Endpoint Statistics
Parameters............................. 286
contacting technical support ......... 373
coordinated phonebook entries
E1.............................................. 248
T1.............................................. 207
Copy Channel command (Interface
Parameters) ............................... 128
Copy Channel command (Voice/Fax
Parameters) ............................... 114
Copy Channel field....................... 115
Copy Channel, Supplementary
Options......................................192
Options......................................192
Description (callee)...................192
Description (caller) ...................192
Disconnect Reason....................191
From Gateway Number.............192
From IP Address .......................192
Outbound Digits (sent)..............191
Packets Lost ..............................191
Prefix Matched..........................191
Server Details............................191
To Gateway Number.................192
To IP Address ...........................192
Custom Fields, RADIUS Attributes
Call Mode..................................190
Channel Number .......................190
Duration ....................................190
Packets Received.......................190
Packets Sent ..............................190
Select All...................................190
Start Date, Time........................190
Custom Fields, SMTP log email
Services command.................... 175
Copy Channel, Supplementary
Services field ............................ 183
Count of Registered Numbers field
(Registered Gateway Details) ... 316
Country Selection for Built-In Modem
field........................................... 158
Country/Region (tone schemes) field
.......................................... 155, 156
Creating a User Default Configuration
.................................................. 204
Current Loss (FXO disconnect
criteria) field ............................. 141
Current Loss field
FXS Loop Start......................... 130
Current Loss Timer (FXO disconnect
criteria) field ............................. 141
Current Reversal (FXO answer
supervision) field ...................... 140
Current Reversal (FXO disconnect
criteria) field ............................. 141
Custom (tones, Regional)field ...... 157
custom cadences ........................... 161
custom DTMF............................... 160
Custom Fields (RADIUS Attributes)
definitions................................. 190
Custom Fields (RADIUS) definitions
.................................................. 191
Custom Fields (SMTP) definitions
.......................................... 165, 166
Custom Fields, RADIUS Accounting
Attributes
Bytes Received..........................166
Bytes Sent .................................166
Call Direction............................166
Call Mode..................................165
Call Status.................................166
Call Type...................................166
Channel Number .......................165
Coder.........................................166
Options......................................167
Options......................................167
Description (callee)...................167
Description (caller) ...................167
Disconnect Reason....................167
DTMF Capability......................166
Duration ....................................165
From Gateway Number.............167
From IP Address .......................167
Outbound Digits Received........166
Outbound digits sent .................167
Packets Lost ..............................166
Packets Received.......................165
Packets Sent ..............................165
Prefix Matched..........................166
Select All...................................165
Server Details............................ See
Start Date, Time........................165
Bytes Received ......................... 191
Bytes Sent................................. 191
Call Status................................. 191
Coder ........................................ 191
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To Gateway Number................. 167
MultiVOIP User Guide
Description field (Registered Gateway
Details)......................................316
Description, From Details (RADIUS
Attributes) field.........................192
Description, From Details (SMTP
logs) field ..................................167
Description, To Details (RADIUS
Attributes) field.........................192
Description, To Details (SMTP logs)
field ...........................................167
Destination Pattern (outbound) field
E1..............................................254
T1..............................................213
destination patterns, discussion
E1..............................................247
T1..............................................206
Detection Range, Flash Hook Options
field
E&M .........................................146
FXO ..........................................135
FXS Loop Start .........................131
dial tone, custom ...........................160
Dialing Options (E&M) fields ......145
Dialing Options (FXO) fields........134
dial-tones.......................................159
DID interface (MVP210-SS)
To IP Address ........................... 167
Custom Tone-Pair Settings definitions
.......................................... 160, 161
Custom Tone-Pair Settings fields
Cadence 1 ................................. 161
Cadence 2 ................................. 161
Cadence 3 ................................. 161
Cadence 4 ................................. 161
Frequency 1 .............................. 160
Frequency 2 .............................. 160
Gain 1 ....................................... 160
Gain 2 ....................................... 160
Tone Pair................................... 160
customized log email ............ 165, 167
customized RADIUS Accounting. 190
customized RADIUS accounting
parameters................................. 192
data capacity ................................... 13
data capacity (analog voips) ........... 10
data capacity (digital voips).............. 9
data capacity (ISDN/BRI voips)..... 11
data compression ............................ 14
Date & Time Setup (program menu
option), command..................... 327
Date and Time Setup option
description (MultiVOIP program
menu)........................................ 324
debugging messages ..................... 171
Default (Supplementary Services)
field........................................... 183
Default (Voice/FAX) field............ 115
default baud rate (MultiVOIP
software connection)................. 193
default configuration, user............ 204
default values, software ................ 335
delay, packets................................ 123
delay, versus voice quality............ 124
Delete File button
uses of .........................................76
DID interface (MVP-410SS/810SS)
uses of .........................................72
DID Interface Parameter definitions
..................................................148
DID Interface Parameter fields
Message Waiting Indication......148
DID Interface Parameters..............147
DID jumper
MVP210-SS................................73
MVP-410SS/810SS.....................70
DID lines (MVP210-SS)
polarity sensitivity and................76
DID lines (MVP-410SS/810SS)
polarity sensitivity and................72
DID-DPO Interface Parameter
Logs (Statistics) screen............. 301
Description (callee location)
E1.............................................. 260
T1.............................................. 219
Description (callee, outbound
definitions .................................147
DID-DPO Interface Parameter fields
Inter Digit Timer (dialing) ........148
Start Modes...............................147
Wink Timer...............................147
DID-DPO Parameter fields
phonebook)
E1.............................................. 254
T1.............................................. 213
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Index
Inter-Digit Regeneration Timer
(dialing) ................................ 148
DID-DPO vs. DID-DPT ............... 147
DiffServ and IP datagram ............. 108
DiffServ PHB (Per Hop Behavior)
value.......................................... 107
digital voip product family ............... 9
dimensions...................................... 20
Disconnect on Call Progress Tone
(E&M) field .............................. 144
Disconnect Reason (SMTP logs) field
.................................................. 167
Disconnect Reason (statistics, logs)
field........................................... 302
Disconnect Tone Sequence (FXO)
field........................................... 142
Disconnect Tones (FXO
downloading user defaults ............341
downloads vs. uploads (FTP)........351
DTMF
extended....................................142
standard.....................................142
DTMF "Out of Band" and Outbound
Digits Sent.................................167
DTMF Capability (call progress) field
..................................................296
DTMF Capability (SMTP logs) field
..................................................166
DTMF Capability (statistics, logs)
field ...........................................302
DTMF frequency chart..................142
DTMF Gain (High Tones) field ....115
DTMF Gain (Low Tones) field.....115
DTMF Gain field ..........................115
DTMF In/Out of Band field..........116
DTMF inband................................116
DTMF out of band ........................116
DTMF Tone (FXO disconnect
criteria) field..............................142
DTMF, custom tone pairs .............160
Duration (call progress) field ........293
Duration (DTMF) field .................116
Duration (RADIUS Attributes) field
..................................................190
Duration (SMTP logs) field ..........165
Duration (statistics, logs) field......301
Dynamic Jitter Buffer field ...........123
Dynamic Jitter field group ............123
Dynamic Jitter fields.....................124
dynamic registration......................198
E&M interface (MVP210-SS)
disconnection supervision) ....... 142
disconnection criteria, FXO.. 134, 141
DNS Server IP Address (Ethernet/IP
Parameters) field....................... 109
Domain Names acceptable for
registration field
SIP Server Configuration
parameters............................. 196
Download Factory Defaults (program
menu option) , command .......... 335
Download Factory Defaults option
description (MultiVOIP program
menu)........................................ 325
Download Firmware (program menu
option), command............. 331, 332
Download Firmware option
description (MultiVOIP program
menu)........................................ 325
Download IFM Firmware (program
menu option) , command .. 337, 338
Download IFM Firmware option
description (MultiVOIP program
menu)........................................ 325
Download User Defaults (program
menu option) , command .......... 341
Download User Defaults option
description (MultiVOIP program
menu)........................................ 325
downloading firmware, machine
perspective........................ 326, 351
downloading IFM firmware.......... 337
matching telco trunk line.............76
uses of .........................................76
E&M interface (MVP-410SS/810SS)
matching telco trunk line.............72
uses of .........................................72
E&M Interface Parameter fields
Detection Range (flash hook)....146
Disconnect on Call Progress Tone
..............................................144
Flash Hook................................146
Inter Digit Timer (dialing) ........145
Interface ....................................144
Message Waiting Indication......145
Pass Through.............................144
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Regeneration (dialing) .............. 145
MultiVOIP User Guide
Error Correction (RADIUS
Signal........................................ 144
Type.......................................... 144
Wink Timer............................... 144
E&M Parameter definitions. 144, 145,
146
E&M Parameters .......................... 143
Echo Cancellation field................. 119
echo, removing ............................. 119
Edit selected Inbound Phonebook
Entry icon
E1.............................................. 249
T1.............................................. 208
Edit selected Outbound Phonebook
Entry icon
E1.............................................. 249
T1.............................................. 208
email account for voip unit........... 163
email address for voip............. 94, 162
email log reports ........................... 162
email logs, illustration .................. 168
EMC, Safety, R&TTE Directive
Compliance............................... 375
Enable (Call Fwdg)
E1.............................................. 260
T1.............................................. 219
Enable (STUN) field..................... 186
Enable Call Hold........................... 177
Enable Call Transfer..................... 176
Enable Call Waiting...................... 177
Enable Caller Name Identification 178
Enable Console Messages field .... 171
Enable DHCP (Ethernet/IP
Parameters) field....................... 106
Enable DNS (Ethernet/IP Parameters)
field........................................... 109
Enable SMTP field ....................... 163
Enable SRV (Ethernet/IP Parameters)
field........................................... 109
enabling SMTP............................. 162
enabling web browser GUI........... 111
analog.......................................... 35
Endpoint Name
Attributes) .................................192
Error Correction (SMTP logs) ......167
error correction, forward...............119
error message
COM port conflict...............83, 100
MultiVOIP-SS Not Found.........101
Password Phone Database Not
Read ......................................101
Phone Database Not Read.........101
SIP Endpoint Database Not Read
..............................................101
ethernet cable pinout.....................380
Ethernet interface............................13
Ethernet/IP parameter definitions 104,
105, 106, 107, 109
Ethernet/IP Parameter fields
802.1p Priority Levels.......104, 105
Frame Type...............................104
Ethernet/IP Parameter screen fields
Enable DNS ..............................109
Ethernet/IP Parameters screen fields
Call Control (Priority)...............105
Call Control PHB......................107
DiffServ.....................................107
DNS Server IP Address.............109
Enable DHCP............................106
Enable SRV...............................109
FTP Server Enable ....................109
Gateway ....................................106
Gateway Name..........................106
IP Address.................................106
IP Mask.....................................106
Others (Priorities)......................105
Packet Prioritization 802.1p......104
TDM Routing Option................110
Use TDM Routing for Intra-
Gateway Calls .......................110
VLAN ID ..................................105
VoIP Media (Priority) ...............105
Voip Media PHB.......................107
Ethernet/IP Parameters screen,
accessing ...................................102
European Community Directives..375
factory default software settings ...335
factory defaults, downloading.......335
factory repair for customers U.S. &
Canada ......................................371
SIP Server Endpoint Statistics
Parameters............................. 285
Endpoint Type
SIP Server Endpoint Statistics
Parameters............................. 286
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Index
failover (PSTN) feature ................ 216
FAQ for MultiVOIPs........................ 8
fast busy (unobtainable) tones ...... 159
fax baud rate, default .................... 116
Fax Enable field............................ 116
fax machine
Forward upon No Response
E1..............................................260
T1..............................................220
forwarding, dual conditions (E1)
busy & no-response...................260
forwarding, dual conditions, busy &
no-response
connecting to analog voip
(MVP210-SS) ......................... 76
connecting to MVP210-SS voip . 75
connecting to voip (MVP-
T1..............................................220
frame relay, and fax ......................117
Frame Type field...........................104
free calls
E1..............................................243
T1..............................................206
frequencies, touch tone .................142
Frequency 1 (custom tone) field ...160
Frequency 1 (tone pair scheme) ...156,
158
410SS/810SS)................... 71, 72
FAX Parameters............................ 116
fax tones, output level................... 117
Fax Volume field.......................... 117
FCC Declaration........................... 375
FCC Part 68 Telecom rules........... 376
FCC registration number .............. 377
FCC rules, Part 15......................... 375
FDX LED ....................................... 18
Filters (Console Message Settings)172
Filters button (Console Message
Settings).................................... 171
firmware upgrade, implementing.. 331
Firmware Version
System Information .................. 289
Firmware Version (System Info) .. 201
firmware version, identifying ....... 331
firmware, downloading................. 332
firmware, obtaining updated......... 327
Flash Hook Options fields
Frequency 2 (custom tone) field ...160
Frequency 2 (tone pair scheme) ...156,
158
frequency, power.............................20
FRF11 ...........................................117
From (gateway, statistics, logs) field
..................................................301
front panel.......................................18
FTP client program .......................351
FTP client program, obtaining ......353
FTP client programs
graphic vs. textual orientation...360
FTP file transfers
E&M......................................... 146
FXO .......................................... 135
forgotten password................ 344, 347
Forward Condition (Call Fwdg)
E1.............................................. 260
T1.............................................. 220
Forward Destination (Inbound PhBk)
E1.............................................. 261
T1.............................................. 220
Forward Error Correction (call
using FTP client program..........353
using web browser ....................353
FTP Server Enable (Ethernet/IP
Parameters) field .......................109
FTP Server function
as added feature.........................351
enabling.....................................353
FTP Server, contacting..................355
FTP Server, invoking
download/transfer
progress) field........................... 298
Forward Error Correction (RADIUS
Attributes)................................. 192
Forward Error Correction (SMTP
logs) .......................................... 167
Forward Error Correction field..... 119
forward on busy
using FTP client program..........359
using web browser ....................357
FTP Server, logging in..................356
FTP Server, logging out................360
FTP transfers
file types............................351, 354
phonebooks ...............................351
server location...........................351
T1...................................... 220, 260
397
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Index
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FXO Supervision Parameter
definitions .................................140
FXS interface(MVP210-SS)
function tracing on/off (logging) .. 172
FXO Disconnect On fields............ 141
FXO disconnection criteria........... 134
FXO disconnection, triggering of 140,
141
FXO Interface Parameter definitions
.................................................. 134
FXO Interface Parameter fields
Current Loss ............................. 134
Current Loss Detect Timer ....... 134
Detection Range (flash hook) ... 135
Flash Hook................................ 135
Inter Digit Regeneration Timer. 134
Inter Digit Timer (dialing) ........ 134
Message Waiting Indication ..... 134
No Response Timer .................. 134
Regeneration (dialing) .............. 134
Tone Detection.......................... 134
FXO interface(MVP210-SS)
uses of .........................................75
FXS interface(MVP-410SS/810SS)
uses of .........................................71
FXS Loop Start Interface parameter
definitions .................................129
FXS Loop Start Interface Parameter
fields
Caller ID enable ........................132
Caller ID Enable........................131
Caller ID Type ..........................131
Current Loss..............................130
Detection Range (flash hook)....131
Inter Digit Regeneration Timer.130
Inter Digit Timer.......................130
Message Waiting Indication......130
Pass Through Enable.................131
Ring Count................................130
FXS Loop Start Parameter fields
Generate Current Reversal........130
Inter Digit Timer.......................129
Message Waiting Light .............129
FXS Loop Start Parameters...........129
FXS/FXO connector
MVP210-SS................................75
MVP-410SS/810SS.....................71
G711 coders (RTP packetization,
voice/fax) ..................................322
G723 coders (RTP packetization,
voice/fax) ..................................322
G726 coders (RTP packetization,
voice/fax) ..................................322
G727 coders (RTP packetization,
voice/fax) ..................................322
G729 coders (RTP packetization,
voice/fax) ..................................322
Gain 1 (custom tone) field ............160
Gain 1 (tone pair scheme) .....157, 158
Gain 2 (custom tone) field ............160
Gain 2 (tone pair scheme) .....157, 158
Gateway (Ethernet/IP Parameters)
field ...........................................106
Gateway Name (callee, statistics,
logs) field ..................................304
Gateway Name (caller, statistics, logs)
field ...........................................304
uses of......................................... 75
FXO interface(MVP-410SS/810SS)
uses of......................................... 71
FXO Parameter fields
Caller ID enable........................ 135
Caller ID Type.......................... 135
FXO Current Detect Timer....... 140
Tone Detection.......................... 140
FXO Parameters............................ 133
FXO Supervision (answer) fields
Answer Delay ........................... 140
Answer Delay Timer................. 140
Answer Tones........................... 140
Available Tones........................ 140
Current Reversal....................... 140
Tone Detection.......................... 140
FXO Supervision (disconnect) fields
Available Tones........................ 142
Current Loss ............................. 141
Current Loss Timer................... 141
Current Reversal....................... 141
Disconnect Tone Sequence....... 142
Disconnect Tones...................... 142
DTMF Tone.............................. 142
Silence Detection Enable.......... 141
Silence Detection Type............. 141
Silence Timer............................ 141
Tone Detection......................... 142
398
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Index
Gateway Name (Ethernet/IP
icons, phonebook
Parameters) field....................... 106
Gateway Number, From Details
(RADIUS Attributes) field........ 192
Gateway Number, From Details
(SMTP logs) field ..................... 167
Gateway Number, To Details
(RADIUS Attributes) field........ 192
Gateway Number, To Details (SMTP
logs) field.................................. 167
Generate Current Reversal
E1..............................................249
T1..............................................208
identifying current firmware version
..................................................331
IFM (interface module, analog voips
only) description .......................337
IFM firmware, downloading.337, 338
IFM Version
System Info...............................202
System Information...................289
implementing firmware upgrade...331
in band, DTMF..............................116
Inbound Phonebook Entries List icon
E1..............................................249
T1..............................................208
Inbound Phonebook entries, list
E1..............................................257
T1..............................................217
inbound vs. outbound phonebooks
E1..............................................247
T1..............................................206
Industry Canada requirements.......376
info sources
FXS Loop Start......................... 130
Generate Local Dial Tone
(Voice/FAX – AutoCall/Offhook
Alert) field ................................ 121
Generation Flash-Hook Options field
E&M......................................... 146
FXO .......................................... 135
GK Name (H.323 Gatekeepers,
Statistics, Servers) field ............ 318
grounding
in rack installations..................... 67
MVP210...................................... 76
GUI (log reporting type) button.... 171
H.323 coder .................................. 118
H.323 Gatekeepers (Statistics,
Servers)
GK Name.................................. 318
IP Address................................. 318
Port ........................................... 318
Priority...................................... 318
Status ........................................ 318
Type.......................................... 318
H.450 features, compatible with SIP
.................................................. 173
H.450 features, incompatible with SIP
.................................................... 14
H.450 functionality
IP details......................................92
SMTP details...............................94
telephony interface details...........93
voip email account ......................94
Initiated Call Count
SIP Server Endpoint Statistics
Parameters.............................286
Input Gain field.............................115
installation
airflow.........................................67
in a nutshell.................................21
in rack .........................................66
log reports by email.....................94
software (detailed).......................78
voip email account ......................94
installation prerequisites ...........92, 93
installation, mechanical...................14
installing Java vis-a-vis web GUI .364
integrated phone/data networks.....242
Inter Digit Regeneration Time
logs for...................................... 305
Hardware ID
System Info............................... 202
System Information .................. 289
Hold Sequence...................... 174, 177
hold, caller on
musical jingle for ...................... 176
hookup diagram, quick (210).......... 32
hookup diagram, quick (410/810)... 31
IANA ............................................ 386
E&M .........................................145
FXO ..........................................134
FXS Loop Start .........................130
Inter Digit Timer (dialing) field
399
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Index
DID-DPO.................................. 148
MultiVOIP User Guide
IP address, SysLog Server ............171
IP Address, To Details (RADIUS
Attributes) field.........................192
IP Address, To Details (SMTP logs)
field ...........................................167
IP Addresses acceptable for
E&M......................................... 145
FXO .......................................... 134
FXS Loop Start......................... 130
Intercept Tone (Regional Params) and
Offhook Alert (Voice/Fax Params)
.................................................. 155
Intercept Tone and required Interface
& Voice/Fax settings ................ 155
Interface field (DID-DPO)............ 147
Interface field (E&M)................... 144
interface parameters, accessing..... 126
interface parameters, setting ......... 126
interface types, BRI
registration field
SIP Server Configuration
parameters.............................196
IP Call Direction (call progress) field
..................................................293
IP Call Type (call progress) field..293
IP datagram and DiffServ .............108
IP Direction (statistics, logs) field.301
IP Mask field.................................106
IP Statistics field
ST ............................................. 384
U 384
inter-office dialing
IP Address.................................308
IP Statistics field definitions .308, 309
IP Statistics fields
E1.............................................. 243
T1.............................................. 207
inter-operation (analog)
Clear..........................................308
Received (RTCP Packets).........310
Received (RTP Packets)............310
Received (TCP Packets)............309
Received (Total Packets) ..........308
Received (UDP Packets)...........309
Received with errors (RTCP
Packets).................................310
Received with errors (RTP Packets)
..............................................310
Received with errors (TCP Packets)
..............................................309
Received with errors (Total
with T1/E1 voips......................... 12
inter-operation with phone system.. 14
IP Address (callee, statistics, logs)
field........................................... 304
IP Address (caller, statistics, logs)
field........................................... 304
IP Address (Ethernet/IP Parameters)
field........................................... 106
IP Address (H.323 Gatekeepers,
Statistics, Servers) field ............ 318
IP Address (IP Statistics) field...... 308
IP Address (outbound phonebook)
E1.............................................. 254
T1.............................................. 213
IP Address (ping target, Link
Management) field.................... 313
IP Address (SIP Proxies, Statistics,
Servers) field............................. 319
IP Address (SPP Registrars, Statistics,
Servers) field............................. 320
IP Address field (Registered Gateway
Details)...................................... 316
IP Address to Ping (Link
Packets).................................309
Received with errors (UDP
Packets).................................309
Transmitted (RTCP Packets).....310
Transmitted (RTP Packets) .......310
Transmitted (TCP Packets) .......309
Transmitted (Total Packets)......308
Transmitted (UDP Packets).......309
IP Statistics function .....................306
ISDN/BRI voip product family.......11
Java
Management) field.................... 312
IP Address, From Details (RADIUS
Attributes) field......................... 192
IP Address, From Details (SMTP
logs) field.................................. 167
installing....................................364
web GUI and.............................363
jitter buffer ....................................123
Jitter Value (Fax) field..................117
Jitter Value field............................125
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Index
jitter, dynamic............................... 123
jumper, DID
MVP210-SS................................ 73
MVP-410SS/810SS .................... 70
Keep Alive (Timers; NAT/STUN)186
key system
connecting to analog voip (MVP-
410SS/810SS)......................... 71
connecting to MVP210-SS ......... 75
Knowledge Base (online, for
MultiVOIPs) ................................. 8
Last button
Link Management (Statistics) screen
field definitions .................312, 313
Link Status fields
Link Management (Statistics)
screen ....................................313
List of Registered Numbers field
(Registered Gateway Details) ...316
lithium battery caution ....................62
LNK LED........................................18
loading of weight in rack ................67
local configuration ..........................92
local configuration procedure
Logs (Statistics) screen............. 301
Last Error (Link Management) field
.................................................. 313
LED definitions
detailed, analog ...........................97
summary......................................96
local voip configuration ..................89
local Windows GUI vs. web GUI
comparison................................362
local-rate calls to remote voip sites
E1..............................................244
Log # (statistics, logs) field...........301
log report email, customizing 165, 167
log report email, triggering.......164
log reporting method, setting ........169
log reports .......................................94
log reports & SMTP......................162
log reports by email.......................162
logging options..............................170
logging update interval..................170
logging, web GUI and...................363
Login Name (SMTP) field ............163
Logs (Statistics) fields
Boot ............................................ 18
Ethernet....................................... 18
FDX ............................................ 18
LNK............................................ 18
Power.......................................... 18
RCV (channel) .......................... 18
RSG ............................................ 18
XMT (channel).......................... 18
XSG ............................................ 18
LED indicators
channel operation........................ 17
general operation ........................ 17
LED indicators, active .................... 17
LED types....................................... 17
lifting
Bytes recvd................................303
Bytes Sent .................................301
Call Forwarded to......................305
Call Transferred to ....................305
Disconnect Reason....................302
DTMF Capability......................302
Duration ....................................301
From (gateway).........................301
Gateway Name (callee).............304
Gateway Name (caller) .............304
H.450 functionality ...................305
IP Address (callee)....................304
IP Address (caller) ....................304
IP Direction column..................301
Log #.........................................301
Mode.........................................301
Options (callee).........................304
precaution about.......................... 62
limitations notice (regulatory),
Canadian ................................... 377
limited warranty............................ 371
Link Management (Statistics) fields
Clear command button.............. 312
IP Address column.................... 313
IP Address to Ping .................... 312
Last Error.................................. 313
No. of Pings Received .............. 313
No. of Pings Sent...................... 313
Ping Size in Bytes..................... 312
Pings per Test ........................... 312
Response Timeout .................... 312
Round Trip Delay ..................... 313
Start Now command button ...... 312
Timer Interval between Pings... 312
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Options (caller)......................... 304
MultiVOIP User Guide
mail criteria (SMTP), records .......164
Mail Server IP Address (SMTP) field
..................................................164
Mail Type (SMTP logs) field........164
mains frequency..............................20
Max bandwidth (coder).................118
Max Baud Rate field .....................116
Max Expiry Time
Outbound digits ........................ 304
Outbound Digits Recvd ............ 302
Outbound Digits Sent ............... 302
Packets lost ............................... 303
Packets recvd ............................ 303
Packets sent............................... 303
Packets Sent.............................. 301
Server Details............................ 303
Start Date, Time........................ 301
Status ........................................ 301
Supplementary Services info .... 305
To (gateway)............................. 301
Type (call) column.................... 301
Voice coder............................... 302
Logs (Statistics) function........... 299
Logs (Statistics) screen
SIP Server Endpoint Statistics
Parameters.............................285
Maximum Jitter Value field ..........124
Message Waiting Indication (DID-
DPO) .........................................148
Message Waiting Indication (E&M)
and DID.....................................145
Message Waiting Indication field
DID-DPO..................................148
E&M .........................................145
FXO ..........................................134
FXS Loop Start .........................130
Minimum Jitter Value field...........123
Mode (call progress) field.............293
Mode (Fax) field ...........................117
Mode (statistics, logs) field...........301
modem relay..................................124
modem traffic on voip network.....124
modem, command
Delete File button ..................... 301
field definitions 301, 302, 303, 304,
305
First button................................ 301
Last button ................................ 301
Next button ............................... 301
Previous button......................... 301
logs and web browser GUI ........... 170
logs by email, illustration.............. 168
Logs screen definitions................. 170
Logs screen field definitions......... 171
Logs screen parameters
Enable Console Messages......... 171
Filters........................................ 171
GUI........................................... 171
IP Address (SysLog Server) ..... 171
Online Statistics Updation Interval
.............................................. 171
Port (SysLog Server) ................ 171
SMTP........................................ 171
SNMP ....................................... 171
SysLog Server Enable............... 171
Turn Off Logs........................... 171
logs screen, accessing ................... 169
long-distance call savings
and Regional Parameters Country
Selection..........................96, 154
modem, remote
configuration/command
setup for ..............................96, 154
Monitor Link fields
Link Management (Statistics)
screen ....................................312
mounting .........................................14
mounting in rack .............................66
procedure for...............................68
safety.....................................62, 67
mounting options ..............................9
MultiVOIP FAQ (on MTS web site) 8
MultiVOIP Program Menu items..324
MultiVOIP Program Menu options
Configuration ............................324
Configuration Port Setup ..........324
Date & Time Setup ...................324
Download Factory Defaults ......325
Download Firmware .................325
E1.............................................. 242
T1.............................................. 206
lost packets, consecutive............... 125
lost password ........................ 344, 347
Mac Address
System Info....................... 202, 289
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Index
Download IFM Firmware......... 325
Set Password............................. 325
Uninstall.................................... 325
Upgrade Software..................... 325
MultiVOIP program menu, option
descriptions....................... 324, 325
MultiVOIP software
installing ..................................... 78
location of files ........................... 81
program icon location................. 82
uninstalling ......................... 85, 348
MultiVOIP software
moving around in...................... 101
MultiVoipManager ......................... 90
musical jingle during call transfer. 176
MVP210
grounding.................................... 76
unpacking.................................... 65
MVP210-SS
No. of Pings Received (Link
Management) field....................313
No. of Pings Sent (Link Management)
field ...........................................313
no-response & busy forwarding, dual
conditions
E1..............................................260
T1..............................................220
Number of Days (email log criteria)
..................................................164
Number of Records (email log
criteria)......................................164
Number of Retransmissions (RADIUS
screen) field...............................189
numbering plan resources .............281
obtaining updated firmware ..........327
Offhook alert.................................120
Offhook Alert (Voice/Fax Params)
and Intercept Tone (Regional
cabling procedure........................ 73
MVP410-SS
cabling procedure........................ 69
remote configuration modem...... 73
unpacking.................................... 64
MVP810-SS
Params) .....................................120
Offhook Alert Timer (Voice/FAX --
AutoCall/Offhook Alert) field...122
Online Statistics Updation Interval
field (Logs)................................171
Operating Mode field
cabling procedure........................ 69
remote configuration modem...... 73
unpacking.................................... 64
Name/IP (Server) field.................. 186
NAT inter-operation support .......... 14
NAT Traversal screen fields
Enable....................................... 186
Keep Alive (Timers)................. 186
Name/IP (Server)...................... 186
Port ........................................... 186
Port (Server................................... 186
national-rate calls to foreign voip sites
(E1)........................................... 246
Netcoder coders (RTP packetization,
voice/fax).................................. 322
Network Disconnection field........ 125
No Response Timer (E&M) field . 144
No. of Entries
SIP Server Configuration
parameters.............................195
operating system ..........................19
operating temperature .....................67
operating voltage.............................20
Optimization Factor field..............124
Options (callee, statistics, logs) field
..................................................304
Options (caller, statistics, logs) field
..................................................304
Options value
Survivability Status Check........195
Options, From Details (RADIUS
Attributes) field.........................192
Options, From Details (SMTP logs)
field ...........................................167
Options, To Details (RADIUS
Attributes) field.........................192
Options, To Details (SMTP logs) field
..................................................167
Others, Priorities (Ethernet/IP params,
802.1p) field..............................105
out of band, DTMF .......................116
SIP Server Endpoint Statistics
Parameters............................. 286
No. of Entries field (Registered
Gateway Details) ...................... 316
403
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MultiVOIP User Guide
Outbound Digits Received (call
progress) field........................... 296
Outbound Digits Received (statistics,
logs) field.................................. 302
Outbound Digits Received(SMTP
logs) field.................................. 166
Outbound Digits Sent (call progress)
field........................................... 296
Outbound Digits Sent (RADIUS
Attributes) field......................... 191
Outbound Digits Sent (SMTP logs)
field........................................... 167
Outbound Digits Sent (statistics, logs)
field........................................... 302
Outbound Digits Sent and DTMF
"Out of Band" ........................... 167
Outbound Phonebook Entries List
icon
E1.............................................. 249
T1.............................................. 208
Outbound Phonebook entries, list
E1.............................................. 251
T1.............................................. 210
outbound vs. inbound phonebooks
E1.............................................. 247
T1.............................................. 206
Out-of-Band DTMF and Outbound
Digits Sent ................................ 167
Output Gain field.......................... 115
output level, fax tones................... 117
Packet Prioritization 802.1p
Packets received (statistics, logs) field
..................................................303
Packets Sent (call progress) field ..294
Packets Sent (RADIUS Attributes)
field ...........................................190
Packets Sent (SMTP logs) field ....165
Packets sent (statistics, logs) field.303
packets, consecutive lost...............125
parameters tracked by console ......172
Pass Through Enable (FXS Loop Start
interface) and AutoCall (Voice/Fax
Params) .....................................131
Password
SIP Server Predefined Endpoint
Parameters.............................198
Password (proxy server) field .......152
Password (SMTP) field.................164
password, lost/forgotten........344, 347
password, setting...........................344
web browser GUI......................347
patents..............................................2
PBX characteristics, variations in
E1..............................................280
T1..............................................240
PBX interaction...............................14
PC Settings/Specs
Quick Start Instructions ..............30
PC, command
COM port assignment (detailed).83
personnel requirement
for rack installation .....................67
to lift during installation..............68
to lift unit during installation.......62
Phone Book Version
(Ethernet/IP parameters)........... 104
packet priority and DiffServ ......... 108
packetization (RTP), ranges &
increments................................. 322
packetization rates
System Info...............................202
System Information...................289
Phone Number (Voice/FAX –
AutoCall/Offhook Alert) field...122
Phone Signaling Tones & Cadences
..................................................153
phone/IP details
importance of writing down........92
Phone/IP details, gathering
Quick Start Instructions ..............25
phone/IP starter configuration
coder options and...................... 321
Packets Lost (call progress) field.. 294
Packets Lost (RADIUS Attributes)
field........................................... 191
Packets Lost (SMTP logs) field.... 166
Packets lost (statistics, logs) field. 303
Packets Received (call progress) field
.................................................. 294
Packets Received (RADIUS
Attributes) field......................... 190
Packets Received (SMTP logs) field
.................................................. 165
Quick Start Instructions ..............34
phonebook
FTP remote file transfers...........351
404
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Index
phonebook configuration................ 89
phonebook configuration (remote) 351
Phonebook Configuration icon
E1.............................................. 249
T1.............................................. 208
Phonebook Configuration Procedure
E1.............................................. 248
T1.............................................. 207
Phonebook Configuration screen
E1.............................................. 248
T1.............................................. 207
phonebook entries, coordinating
E1.............................................. 248
T1.............................................. 207
phonebook example
command cable .........................380
ethernet cable ............................380
T1/E1 connector........................381
Voice/FAX connector ...............381
placement of voip
Quick Start Instructions ..............30
polarity sensitivity
DID lines and (MVP210-SS) ......76
DID lines and (MVP-
410SS/810SS) .........................72
pop-ups
allowing with Web GUI............111
Port (Contact Info)
SIP Server Predefined Endpoint
Parameters.............................199
Port (H.323 Gatekeepers, Statistics,
Servers) field.............................318
Port (SIP Proxies, Statistics, Servers)
field ...........................................319
Port (SPP Registrars, Statistics,
Servers) field.............................320
Port field (Registered Gateway
Details)......................................316
Port field, SysLog Server..............171
Port Number
SIP Server Endpoint Statistics
Parameters.............................286
Port Number (proxy server) field..151
Port Number (SMTP) field ...........164
power consumption.........................20
power frequency..............................20
Power LED......................................18
Prefix Matched (call progress) field
..................................................296
Prefix Matched (RADIUS Attributes)
field ...........................................191
Prefix Matched (SMTP logs) field 166
prerequisites
Quick Start Instructions .............. 51
phonebook icons
E1.............................................. 249
T1.............................................. 208
phonebook keyboard shortcuts
E1.............................................. 250
T1.............................................. 209
phonebook objectives &
considerations
E1.............................................. 247
phonebook pulldown menu
E1.............................................. 250
T1.............................................. 209
phonebook sidebar menu
E1.............................................. 250
T1.............................................. 209
phonebook starter configuration
Quick Start Instructions .............. 40
phonebook tips
Quick Start Instructions .............. 47
phonebook, objectives &
considerations
E1.............................................. 242
T1.............................................. 206
phonebooks, inbound vs. outbound
E1.............................................. 247
T1.............................................. 206
Ping Size in Bytes (Link
Management) field.................... 312
Pings per Test (Link Management)
field........................................... 312
pinout
for technical configuration..........92
Primary Proxy (SIP Call Signaling)
field ...........................................151
Priority (H.323 Gatekeepers,
Statistics, Servers) field.............318
Priority Levels (802.1p) ........104, 105
product CD......................................21
use in software installation..........78
Program Menu items.....................324
Protocol Type (outbound phonebook)
BRI connector........................... 383
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Index
T1.............................................. 213
Proxy Domain Name / IP Address
field........................................... 151
Proxy Polling Interval (SIP Call
Signaling) field ......................... 152
PSTN failover feature
MultiVOIP User Guide
Received (RTP Packets, IP Stats) field
..................................................310
Received (TCP Packets, IP Stats) field
..................................................309
Received (Total Packets, IP Stats)
field ...........................................308
Received (UDP Packets, IP Stats)
field ...........................................309
Received Call Count
Alternate Routing, and.............. 216
quality-of-service............................ 14
quick hookup diagram (210)
Quick Start Instructions .............. 32
quick hookup diagram (410/810)
Quick Start Instructions.............. 31
Quick Start Instructions
SIP Server Endpoint Statistics
Parameters.............................286
Received with Errors (RTCP Packets,
IP Stats) field.............................310
Received with Errors (RTP Packets,
IP Stats) field.............................310
Received with Errors (TCP Packets,
IP Stats) field.............................309
Received with Errors (Total Packets,
IP Stats) field.............................309
Received with Errors (UDP Packets,
IP Stats) field.............................309
Recipient Address (email logs) field
..................................................164
recovering voice packets...............119
Regeneration (dialing, FXO) field 134
Regional Parameter definitions....155,
156, 157, 158
config info checklist ................... 28
connectivity test.......................... 56
PC settings/specs ........................ 30
phone/IP details, gathering ......... 25
phone/IP starter configuration..... 34
phonebook example.................... 51
phonebook starter configuration . 40
phonebook tips............................ 47
placement of voip ....................... 30
quick hookup diagram (210)....... 32
quick hookup diagram (410/810) 31
software installation.................... 33
startup tasks ................................ 24
troubleshooting ........................... 60
rack mounting
Regional Parameter fields
grounding.................................... 67
safety..................................... 62, 67
rack mounting instructions.............. 66
rack mounting procedure ................ 68
rack, equipment
weight capacity of....................... 67
rack-mountable voip models........... 62
RADIUS accounting parameters,
customizing....................... 190, 192
RADIUS accounting support.......... 14
RADIUS screen field
Cadence.....................................157
Country/Region (tone schemes) 155
Custom (tones)..........................157
Frequency 1...............................156
Frequency 2...............................156
Gain 1........................................156
Gain 2........................................156
Pulse Generation Ratio..............157
type (of tone).............................156
Regional Parameters fields
Country Selection for Built-In
Enable Accounting.................... 189
Retransmission Interval ............ 189
RADIUS screen fields
Accounting Port........................ 189
Server Address.......................... 189
RCV (channel) LED ..................... 18
Received (RTCP Packets, IP Stats)
field........................................... 310
Modem..................................157
regional parameters, setting ..........153
Register Duration field (Registered
Gateway Details).......................316
Register value
Survivability Status Check........195
Registered Gateway Details
(Statistics) screen, accessing.....316
406
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MultiVOIP User Guide
Index
SIP Server Predefined Endpoint
Parameters.............................199
Re-Registration Time
SIP Server Predefined Endpoint
Parameters.............................199
Re-Registration Time (proxy server)
..................................................152
Re-Registration Time field
SIP Server Configuration
parameters.............................197
Resolutions (MultiVOIP
troubleshooting) ............................8
Response Timeout (Link
Management) field....................312
Retransmission Interval (RADIUS
screen) field...............................189
Retrieve Sequence.................174, 177
RFC 2782......................................109
RFC 2833......................................116
RFC 3087......................................214
RFC 3489......................................184
RFC2474.......................................107
RFC2597.......................................107
RFC2833.......................166, 296, 302
RFC3246.......................................107
RFC768.........................................386
RFC793.........................................386
ring cadences, custom ...................161
Ring Count field
FXS Loop Start .........................130
Ring Count forwarding condition
E1..............................................261
T1..............................................221
ring tone, custom...........................160
ring-tones ......................................159
Round Trip Delay (Link
Management) field....................313
Round Trip Delay field
E1..............................................257
T1..............................................216
RSG LED........................................18
RTP packetization, ranges &
increments.................................322
RTP Parameters screen .................322
Safety Recommendations for Rack
Installations.................................67
safety warnings ...............................62
Safety Warnings Telecom..........62
Registered Gateway Details
‘Statistics’ function......... 315, 316
Registered Gateway Details screen316
Registered Gateway Details screen
fields
Description................................ 316
IP Address................................. 316
No. of Entries............................ 316
Port ........................................... 316
Register Duration...................... 316
Status ........................................ 316
Registered Gateway Details screen
fields: ........................................ 316
Registration Option Parameters
(Inbound Phone Book)
E1.............................................. 261
T1.............................................. 221
Registration Type
SIP Server Endpoint Statistics
Parameters............................. 286
SIP Server Predefined Endpoint
Parameters............................. 198
Remaining Time
SIP Server Endpoint Statistics
Parameters............................. 286
remote configuration modem
MVP410-SS................................ 73
MVP810-SS................................ 73
Remote Configuration/Command
Modem
setup for.............................. 96, 154
remote control/configuration
web GUI and............................. 363
remote phonebook configuration.. 351
remote voip configuration............... 89
Remove Prefix (inbound) field
E1.............................................. 259
T1.............................................. 219
Remove Prefix (outbound) field
E1.............................................. 254
T1.............................................. 213
repair procedures for customers U.S.
& Canada.................................. 371
Reply-To Address (email logs)field
.................................................. 164
Requires Authentication (SMTP) field
.................................................. 163
Re-Registration Interval
407
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Index
MultiVOIP User Guide
Save Setup command.................... 203
saving configuration ..................... 203
user ........................................... 341
Saving the MultiVOIP Configuration
.................................................. 203
savings on toll calls
E1.............................................. 242
T1.............................................. 206
Select All (RADIUS Attributes) field
.................................................. 190
Select All (SMTP logs) field ........ 165
Select Attributes (RADIUS) button
.................................................. 189
Select Channel field...................... 115
Select Channel, Supplementary
Services field ............................ 176
Selected Coder field...................... 118
Server Address (RADIUS screen)
field........................................... 189
Server Details (call progress) field 296
Server Details (RADIUS Attributes)
field........................................... 191
Server Details (SMTP logs) field.. 167
Server Details (statistics, logs) field
.................................................. 303
Service Records ............................ 110
Set Baud Rate ............................... 193
Set Log Reporting Method ........... 169
Set Password (program menu option) ,
command .................................. 344
Set Password (web browser GUI) ,
command .................................. 347
Set Password option description
setup, saving user values...............341
Shared Secret (RADIUS screen) field
..................................................189
Signal (type, E&M) field ..............144
signaling cadences.........................153
signaling parameters .....................126
Signaling Port (SIP Call Signaling)
field ...........................................150
signaling tones ..............................153
signaling types
(MVP210-SS) .............................75
(MVP-410SS/810SS)..................72
telephony interfaces (MVP210) ..76
telephony interfaces (MVP-
410SS/810SS) .........................71
Silence Compression (call progress)
field ...........................................298
Silence Compression (RADIUS
Attributes) .................................192
Silence Compression (SMTP logs)167
Silence Compression field ............119
Silence Detection Enable (FXO
disconnect criteria) field............141
Silence Detection Type (FXO) field
..................................................141
Silence Timer (FXO) field ............141
SIP Call Signaling Parameter
definitions .................150, 151, 152
SIP Call Signaling screen fields
Password (proxy server)............152
Proxy Domain Name / IP Address
..............................................151
Proxy Polling Interval...............152
Re-Registration Time (proxy
server) ...................................152
Signaling Number (proxy server)
..............................................151
TTL Value.................................152
Use SIP Proxy...........................150
User Name (proxy server).........151
SIP compatibility with H.450
Supplementary Services............173
SIP Fields (Outbound Phonebook)
E1..............................................255
T1..............................................214
SIP incompatibility with H.450
Supplementary Services..............14
SIP Port Number field
(MultiVOIP program menu) ..... 325
Set Regional Parameters............... 153
Set SMTP Parameters................... 162
Set Supplementary Services
Parameters ................................ 173
Set Telephony Interface Parameters
.................................................. 126
Set Voice/FAX Parameters........... 112
setting Ethernet/IP parameters...... 102
setting password............................ 344
web browser GUI...................... 347
setting RTP Parameters................. 322
setting user defaults ...................... 341
setup, saving ................................. 203
user ........................................... 341
408
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Index
E1.............................................. 255
T1.............................................. 214
SIP port number, standard
SIP survivability..............................13
SIP URL field
E1..............................................255
T1..............................................214
SMTP (log reporting type) button.171
SMTP logs by email, illustration ..168
SMTP Parameters definitions .......164
SMTP Parameters fields
E1.............................................. 255
T1.............................................. 214
SIP Proxies (Statistics, Servers)
IP Address................................. 319
Port ........................................... 319
Status ........................................ 319
Type.......................................... 319
SIP proxy capacity.......................... 13
SIP Proxy Parameters ................... 150
SIP Server Configuration parameters
Accept Registrations for domains
.............................................. 196
Accept Registrations for IP
Enable SMTP............................163
Login Name ..............................163
Mail Server IP Address.............164
Mail Type..................................164
Number of Days........................164
Number of Records...................164
Password ...................................164
Port Number..............................164
Recipient Address .....................164
Reply-To Address .....................164
Requires Authentication............163
Subject ......................................164
SMTP parameters, accessing ........162
SMTP parameters,setting..............162
SMTP port, standard..................164
SMTP prerequisites.........................94
SMTP, enabling ............................162
SNMP (log reporting type) button 171
SNMP agent program......................90
software
Addresses.............................. 196
Allow Undefined Registrations. 196
Domain Names acceptable for
registration ........................... 196
IP Addresses acceptable for
registration ........................... 196
Operating Mode........................ 195
Re-Registration Time................ 197
Survivability Status Check........ 195
SIP Server Endpoint Statistics
Contact Address........................ 286
SIP Server Endpoint Statistics
Parameters
uninstalling (detailed) .................85
updates ........................................90
software (MultiVOIP)
uninstalling................................348
software configuration
Endpoint Name......................... 285
Endpoint Type .......................... 286
Initiated Call Count................... 286
Max Expiry Time...................... 285
No. of Entries............................ 286
Port Number ............................. 286
Received Call Count................. 286
Registration Type...................... 286
Remaining Time ....................... 286
Status ........................................ 285
SIP Server Predefined Endpoint
Parameters
summary......................................78
software installation
detailed........................................78
Quick Start Instructions ..............33
software loading..............................78
software version numbers ...............80
software, MultiVOIP
moving around in ......................101
software, MultiVOIP
screen-surfing in........................101
Solving Common Connection
Problems ...................................100
sound quality, improving ..............119
SPP Registrars (Statistics, Servers)
IP Address.................................320
Address (Contact Info).............. 199
Endpoint Name......................... 198
Password................................... 198
Port (Contact Info).................... 199
Registration Type...................... 198
Re-Registration Interval............ 199
Re-Registration Time................ 199
409
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Index
Port ........................................... 320
Type.......................................... 320
SPP Registrarss (Statistics, Servers)
Status ........................................ 320
SRV record ................................... 110
ST interface (ISDN-BRI)
MultiVOIP User Guide
Call Waiting..............................174
Call Waiting Enable..................177
Caller Name Identification Enable
..............................................178
Calling Party .............................179
Enable Call Hold.......................177
Enable Call Transfer .................176
Enable Call Waiting..................177
Enable Caller Name Identification
..............................................178
Hold Sequence ..........................177
Retrieve Sequence.....................177
Select Channel ..........................176
Transfer Sequence.....................176
Supplementary Services Info
logs for ......................................305
Supplementary Services Parameter
buttons
description ................................ 384
Start Date, Time (RADIUS
Attributes) field......................... 190
Start Date, Time (SMTP logs) field
.................................................. 165
Start Date,Time (statistics, logs) field
.................................................. 301
Start Modes (DID-DPO) field147, 148
Start Now command (Link
Management) button................. 312
starter configuration, phone/IP ....... 34
starter configuration, phonebook .... 40
Startup Tasks
Copy Channel............................183
Default ......................................183
Supplementary Services Parameter
Definitions 176, 177, 178, 179, 180,
181, 182, 183
Quick Start Instructions .............. 24
static registration........................... 198
Status
SIP Server Endpoint Statistics
Parameters............................. 285
Status (H.323 Gatekeepers, Statistics,
Servers) field............................. 318
Status (SIP Proxies, Statistics,
Supplementary Services Parameter
fields
Call Waiting Enable..................177
Hold Sequence ..........................177
Retrieve Sequence.....................177
Supplementary Services Parameter
fields
Call Hold Enable.......................177
Call Transfer Enable .................176
Select Channel ..........................176
Supplementary Services Parameter
fields
Call Name Identification Enable178
Supplementary Services Parameter
fields
Calling Party .............................179
Supplementary Services Parameter
fields
Servers) field............................. 319
Status (SPP Registrars, Statistics,
Servers) field............................. 320
Status (statistics, logs) field .......... 301
Status field (Registered Gateway
Details)...................................... 316
STUN clients and servers ............. 184
STUN support................................. 14
Subject (email logs) field.............. 164
supervisory signaling.................... 127
supervisory signaling parameters.. 126
supervisory signaling types
MVP210-SS.......................... 75, 76
MVP-410SS/810SS .............. 71, 72
Supplementary Services
Allowed Name Types................179
Supplementary Services Parameter
fields
Alerting Party............................180
Supplementary Services Parameter
fields
Alerting Party............ 180, 181, 182
Call Hold................................... 174
Call Hold Enable....................... 177
Call Name Identification........... 174
Call Transfer............................. 174
Call Transfer Enable................. 176
Allowed Name Types................180
410
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Index
Supplementary Services Parameter
fields
Phone Book Version .................289
System Information screen
Busy Party................................. 181
Supplementary Services Parameter
fields
Allowed Name Types ............... 181
Supplementary Services Parameter
fields
Connected Party........................ 182
Supplementary Services Parameter
fields
for op & maint...........................288
System Information screen, accessing
..................................................200
System Information update interval,
setting........................................200
for op & maint...........................290
T1/E1 connector pinout.................381
table-top voip models......................62
TCP/UDP compared
Allowed Name Types ............... 182
Supplementary Services Parameter
fields
Caller ID ................................... 183
Supplementary Services Parameters
fields
E1..............................................255
IP Statistics context...........307, 308
T1..............................................214
TDM Routing Option (Ethernet/IP
Parameters) field .......................110
technical configuration
Transfer Sequence .................... 176
Supplementary Services Parameters
screen, accessing....................... 173
Supplementary Services parameters,
setting........................................ 173
Supplementary Services, compatible
with SIP .................................... 173
Supplementary Services, incompatible
with SIP ...................................... 14
support, technical.......................... 373
Survivability Status Check field
SIP Server Configuration
parameters............................. 195
SysLog client .................................. 16
SysLog client programs
availability ................................ 367
features & presentation types.... 369
SysLog functionality....................... 16
SysLog server ................................. 16
SysLog Server Enable field .......... 171
SysLog Server function
as added feature ........................ 367
capabilities of............................ 369
enabling .................................... 368
location of................................. 367
SysLog Server IP Address field.... 171
SysLog Server, enabling............... 170
System Information Parameters
Boot Version............................. 289
Configuration Version .............. 289
IFM Version ............................. 289
prerequisites to............................92
summary......................................89
technical configuration procedure
detailed........................................97
summary......................................96
technical support ...........................373
telecom safety warnings.............62
telephony interface parameters .......93
telephony interface parameters,
setting........................................126
telephony interfaces
uses of .......................71, 72, 75, 76
telephony signaling cadences........153
telephony signaling tones..............153
telephony toning schemes .............159
temperature
operating .....................................67
timeout interval
voips under SIP proxy server....152
Timer Interval between Pings (Link
Management) field....................312
To (gateway, statistics, logs) field.301
toll-call savings
E1..............................................242
T1..............................................206
Tone Detection (FXO answer
supervision criteria) field ..........140
Tone Detection (FXO disconnection
supervision)...............................142
Tone Pair (custom) field ...............160
tones, signaling .............................153
411
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MultiVOIP User Guide
Total Digits (outbound) field
Undefined Registrations................196
Uninstall (program menu option) ,
command...................................348
Uninstall option description
(MultiVOIP program menu) .....325
uninstalling MultiVOIP software...85,
348
unobtainable tone, custom.............160
unobtainable tones.........................159
unpacking........................................63
MVP210......................................65
MVP410-SS................................64
MVP810-SS................................64
Up Time
System Info.......................202, 289
update interval (logging)...............170
updated firmware, obtaining .........327
Upgrade Software option description
MultiVOIP program menu........325
upgrade, firmware.........................331
uploads vs. downloads (FTP)........351
Use Proxy (SIP) field
E1.............................................. 254
T1.............................................. 213
touch tone frequencies .................. 142
trace on/off (logging).................... 172
Transfer Sequence ................ 174, 176
Transmitted (RTCP Packets, IP Stats)
field........................................... 310
Transmitted (RTP Packets, IP Stats)
field........................................... 310
Transmitted (TCP Packets, IP Stats)
field........................................... 309
Transmitted (Total Packets, IP Stats)
field........................................... 308
Transmitted (UDP Packets, IP Stats)
field........................................... 309
Transport Protocol (SIP) field
E1.............................................. 255
T1.............................................. 214
triggering log report email ....... 164
troubleshooting
Quick Start Instructions .............. 60
Troubleshooting Resolutions for
MultiVOIPs .................................. 8
TTL Value (SIP Call Signaling) field
.................................................. 152
Turn Off Logs field....................... 171
Type (call, statistics, logs) field.... 301
Type (E&M type) field ................. 144
Type (H.323 Gatekeepers, Statistics,
Servers) field............................. 318
Type (of tone, Regional Parameters)
field........................................... 156
Type (SIP Proxies, Statistics, Servers)
field........................................... 319
Type (SPP Registrars, Statistics,
Servers) field............................. 320
Type-of-Service IP header field &
DiffServ .................................... 108
U interface (ISDN-BRI)
E1..............................................255
T1..............................................214
Use SIP Proxy field.......................150
Use TDM Routing for Intra-Gateway
Calls ..........................................110
user default configuration, creating
..................................................204
user defaults, downloading ...........341
user defaults, setting......................341
user name
Windows GUI ...........................344
User Name (proxy server) field.....151
user values (software), saving.......341
variations in PBX characteristics
E1..............................................280
T1..............................................240
version numbers (software).............80
version, firmware ..........................331
VLAN ID (Ethernet/IP Parameters)
field ...........................................105
Voice Coder (call progress) field ..293
Voice coder (statistics, logs) field.302
voice delay ............................123, 124
Voice Gain field............................115
voice packets
description ................................ 384
UDP/TCP compared
E1.............................................. 255
IP Statistics context........... 307, 308
T1.............................................. 214
unconditional forwarding
E1.............................................. 260
T1.............................................. 220
recovering lost/corrupted ..........119
412
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Index
voice packets, consecutive lost ..... 125
voice packets, delayed .......... 123, 124
voice packets, re-assembling ........ 117
voice quality, improving............... 119
voice quality, versus delay............ 124
Voice/FAX connector pinout........ 381
Voice/FAX Parameter definitions 124,
125
Dynamic Jitter Buffer................123
Voice/FAX Parameter fields
Minimum Jitter Value...............123
Voice/FAX Parameter fields
Maximum Jitter Value ..............124
Voice/FAX Parameter fields
Optimization Factor ..................124
Voice/FAX Parameter fields
Voice/FAX Parameter Definitions115,
116, 117, 118, 119, 123
Automatic Disconnection..........125
Voice/FAX Parameter fields
Voice/FAX Parameter fields
Jitter Value................................125
Voice/FAX Parameter fields
Call Duration.............................125
Voice/FAX Parameter fields
AutoCall/Offhook Alert.... 120, 121
AutoCall/Offhook Alert fields. 120,
121
Generate Local Dial Tone......... 121
Offhook Alert Timer................. 122
Out-of-Band Mode (DTMF)..... 115
Phone Number (Auto Call/Offhook
Alert)..................................... 122
Voice/FAX Parameter fields
Copy Channel ........................... 115
Default ...................................... 115
DTMF Gain .............................. 115
DTMF Gain (High Tones)........ 115
DTMF Gain (Low Tones)......... 115
DTMF In/Out of Band.............. 115
Duration (DTMF) ..................... 115
Input Gain................................. 115
Output Gain .............................. 115
Select Channel .......................... 115
Voice Gain................................ 115
Voice/FAX Parameter fields
Fax Enable ................................ 116
Voice/FAX Parameter fields
Max Baud Rate (Fax)................ 116
Voice/FAX Parameter fields
Fax Volume .............................. 117
Voice/FAX Parameter fields
Jitter Value (Fax)...................... 117
Voice/FAX Parameter fields
Consecutive Packets Lost..........125
Voice/FAX Parameter fields
Network Disconnection.............125
Voice/FAX Parameters screen,
accessing ...................................112
Voice/FAX parameters, setting.....112
Voip Caller ID Case #1 –telco
standard CID enters voip system
..................................................136
Voip Caller ID Case #2 – H.323 voip
system, no telco CID.................136
Voip Caller ID Case #3 –SPP .......137
Voip Caller ID Case #4 – Remote
FXS call on H.323 voip system.137
Voip Caller ID Case #5 –DID channel
in H.323 voip system ................138
voip email account ........................163
Voip Media PHB field ..................107
VoIP Media Priority (Ethernet/IP
parameters) field .......................105
voip software
host PC..................................19, 90
voip system example, conceptual (E1)
calls to remote PSTN ................244
foreign calls, national rates .......246
voip site to voip site ..................243
voip system example, digital &
analog, with phonebook details
E1..............................................269
T1..............................................228
voip system example, digital only,
with phonebook details
Mode (Fax) ............................... 117
Voice/FAX Parameter fields
Silence Compression ................ 119
Voice/FAX Parameter fields
Echo Cancellation..................... 119
Voice/FAX Parameter fields
Forward Error Correction ......... 119
Voice/FAX Parameter fields
E1..............................................262
T1..............................................222
413
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MultiVOIP User Guide
voltage, operating ........................... 20
warnings, safety.............................. 62
warranty........................................ 371
web browser GUI and logs ........... 170
web browser GUI, enabling.......... 111
analog.......................................... 35
web browser interface
web GUI, logging and...................363
weight..............................................20
weight loading
in rack .........................................67
weight of unit
lifting precaution.........................62
personnel requirement.................62
Well Known Ports.........................386
well-known port number, SMTP
..................................................164
well-known port, SIP
E1..............................................255
T1..............................................214
wink signaling (DID-DPO)...........148
wink signaling (E&M) ..................144
Wink Timer (DID-DPO) field.......148
Wink Timer (E&M) field..............144
XMT (channel) LED .....................18
XSG LED........................................18
browser version requirement ... 361,
365
general ...................................... 361
Java requirement....................... 361
prerequisite local assigning of IP
address .................................. 362
video useability......................... 361
web GUI
Java and .................................... 363
remote control/configuration and
.............................................. 363
web GUI vs. local Windows GUI
comparison................................ 362
414
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S000393B
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