Multi Tech Systems Scanner MVP2400 User Manual
Voice / Fax over IP Networks  
User Guide for Voice/IP Gateways  
Digital Models  
(T1, E1, ISDN-PRI):  
MVP2400  
MVP2410  
MVP3010  
Analog Models:  
MVP210  
MVP410  
MVP810  
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CONTENTS  
CONTENTS ................................................................................................................3  
CHAPTER 1: OVERVIEW.......................................................................................7  
ABOUT THIS MANUAL...............................................................................................8  
INTRODUCTION TO TI MULTIVOIPS (MVP2400, MVP2410, & MVP24-48).........10  
T1 Front Panel LEDs..........................................................................................12  
T1 LED Descriptions ..........................................................................................13  
INTRODUCTION TO EI MULTIVOIPS (MVP3010 & MVP30-60)............................14  
E1 Front Panel LEDs .........................................................................................16  
E1 LED Descriptions..........................................................................................16  
INTRODUCTION TO ANALOG MULTIVOIPS (MVP-210/410/810 & MVP428)........18  
Analog MultiVOIP Front Panel LEDs................................................................20  
Analog MultiVOIP LED Descriptions ................................................................21  
COMPUTER REQUIREMENTS.....................................................................................22  
SPECIFICATIONS.......................................................................................................23  
Specs for Digital T1 MultiVOIP Units................................................................23  
Specs for Digital E1 MultiVOIP Units................................................................24  
Specs for Analog MultiVOIP Units.....................................................................25  
INSTALLATION AT A GLANCE ..................................................................................26  
RELATED DOCUMENTATION....................................................................................26  
CHAPTER 2: QUICK START INSTRUCTIONS.................................................27  
INTRODUCTION........................................................................................................28  
MULTIVOIP STARTUP TASKS .................................................................................28  
Phone/IP Details *Absolutely Needed* Before Starting the Installation............29  
Gather IP Information...................................................................................................29  
Gather Telephone Information .....................................................................................29  
Gather Telephone Information .....................................................................................30  
Gather Telephone Information .....................................................................................30  
Obtain Email Address for VOIP (for email call log reporting).....................................31  
Identify Remote VOIP Site to Call...............................................................................31  
Identify VOIP Protocol to be Used...............................................................................31  
Placement ...........................................................................................................32  
The Command/Control Computer (Specs & Settings) ........................................32  
Quick Hookups....................................................................................................33  
Load MultiVOIP Control Software onto PC.......................................................35  
Phone/IP Starter Configuration..........................................................................36  
Phonebook Starter Configuration (with remote voip).........................................39  
Outbound Phonebook...................................................................................................39  
Inbound Phonebook......................................................................................................43  
Phonebook Tips ..................................................................................................46  
Phonebook Example ...........................................................................................49  
Connectivity Test.................................................................................................54  
Troubleshooting..................................................................................................58  
3
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Contents  
MultiVOIP User Guide  
CHAPTER 3: MECHANICAL INSTALLATION AND CABLING...................59  
INTRODUCTION........................................................................................................60  
SAFETY WARNINGS .................................................................................................60  
Lithium Battery Caution .....................................................................................60  
Safety Warnings Telecom....................................................................................60  
UNPACKING YOUR MULTIVOIP..............................................................................61  
Unpacking the MVP2410/3010...........................................................................61  
Unpacking the MVP2400....................................................................................62  
Unpacking the MVP410/810...............................................................................63  
Unpacking the MVP210......................................................................................64  
RACK MOUNTING INSTRUCTIONS FOR MVP2410/3010 & MVP410/810................65  
Safety Recommendations for Rack Installations.................................................66  
19-Inch Rack Enclosure Mounting Procedure....................................................67  
CABLING..................................................................................................................68  
Cabling Procedure for MVP2410/3010..............................................................68  
Cabling Procedure for MVP2400.......................................................................69  
Cabling Procedure for MVP410/810..................................................................70  
Cabling Procedure for MVP210.........................................................................72  
CHAPTER 4: SOFTWARE INSTALLATION .....................................................74  
INTRODUCTION........................................................................................................75  
LOADING MULTIVOIP SOFTWARE ONTO THE PC....................................................75  
UN-INSTALLING THE MULTIVOIP CONFIGURATION SOFTWARE.............................82  
CHAPTER 5: TECHNICAL CONFIGURATION FOR DIGITAL T1/E1  
MULTIVOIPS (MVP2400, MVP2410, MVP3010)................................................86  
CONFIGURING THE DIGITAL T1/E1 MULTIVOIP.....................................................87  
LOCAL CONFIGURATION..........................................................................................89  
Pre-Requisites.....................................................................................................89  
IP Parameters................................................................................................................89  
T1 Telephony Parameters (for MVP2400 & MVP2410)..............................................90  
E1 Telephony Parameters (for MVP3010) ...................................................................91  
SMTP Parameters (for email call log reporting)...........................................................92  
Local Configuration Procedure (Summary) .......................................................93  
Local Configuration Procedure (Detailed).........................................................94  
CHAPTER 6: TECHNICAL CONFIGURATION FOR ANALOG  
MULTIVOIPS (MVP210/410/810)........................................................................161  
CONFIGURING THE ANALOG MULTIVOIP .............................................................162  
LOCAL CONFIGURATION........................................................................................165  
Pre-Requisites...................................................................................................165  
IP Parameters..............................................................................................................165  
Analog Telephony Interface Parameters (for MVP210/410/810)..............................166  
SMTP Parameters (for email call log reporting).........................................................167  
Local Configuration Procedure (Summary) .....................................................168  
Local Configuration Procedure (Detailed).......................................................169  
4
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MVP3000 MultiVOIP User GuideMultiVOIP  
Overview  
CHAPTER 7: T1 PHONEBOOK CONFIGURATION ......................................235  
CONFIGURING THE MVP2400/2410 MULTIVOIP PHONEBOOKS..........................236  
T1 PHONEBOOK EXAMPLES...................................................................................254  
3 Sites, All-T1 Example.....................................................................................254  
Configuring Mixed Digital/Analog VOIP Systems ...........................................260  
Call Completion Summaries .............................................................................270  
Variations in PBX Characteristics....................................................................273  
CHAPTER 8: E1 PHONEBOOK CONFIGURATION ......................................274  
MVP3010 INBOUND AND OUTBOUND MULTIVOIP PHONEBOOKS .......................275  
Free Calls: One VOIP Site to Another.............................................................276  
Local Rate Calls: Within Local Calling Area of Remote VOIP.......................277  
National Rate Calls: Within Nation of Remote VOIP Site...............................279  
Inbound versus Outbound Phonebooks.............................................................280  
PHONEBOOK CONFIGURATION PROCEDURE...........................................................284  
E1 PHONEBOOK EXAMPLES...................................................................................298  
3 Sites, All-E1 Example ....................................................................................298  
Configuring Digital & Analog VOIPs in Same System.....................................305  
Call Completion Summaries.......................................................................................314  
Variations in PBX Characteristics....................................................................317  
International Telephony Numbering Plan Resources.......................................318  
CHAPTER 9: ANALOG PHONEBOOK CONFIGURATION .........................320  
CHAPTER 10: OPERATION AND MAINTENANCE ......................................322  
OPERATION AND MAINTENANCE ...........................................................................323  
System Information screen................................................................................323  
Statistics Screens...............................................................................................325  
About Call Progress..........................................................................................325  
About Logs........................................................................................................331  
About Reports ...................................................................................................334  
About IP Statistics.............................................................................................335  
About T1/E1 Statistics.......................................................................................339  
MULTIVOIP PROGRAM MENU ITEMS .....................................................................347  
Date and Time Setup.........................................................................................349  
Obtaining Updated Firmware...........................................................................349  
Implementing a Software Upgrade...................................................................353  
Identifying Current Firmware Version .......................................................................353  
Downloading Firmware..............................................................................................354  
Downloading CAS Protocol.......................................................................................357  
Downloading Factory Defaults...................................................................................360  
Setting and Downloading User Defaults ..........................................................362  
Setting a Password (Windows GUI) .................................................................364  
Setting a Password (Web Browser GUI) ..........................................................367  
Un-Installing the MultiVOIP Software .............................................................368  
Upgrading Software..........................................................................................370  
FTP SERVER FILE TRANSFERS (“DOWNLOADS”)...................................................371  
5
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Contents  
MultiVOIP User Guide  
WEB BROWSER INTERFACE ...................................................................................381  
SYSLOG SERVER FUNCTIONS ................................................................................386  
CHAPTER 11: WARRANTY, SERVICE, AND TECH SUPPORT..................389  
LIMITED WARRANTY.............................................................................................390  
REPAIR PROCEDURES FOR U.S. AND CANADIAN CUSTOMERS ...............................390  
TECHNICAL SUPPORT.............................................................................................392  
Contacting Technical Support ..........................................................................392  
CHAPTER 12: REGULATORY INFORMATION ............................................393  
EMC, Safety, and R&TTE Directive Compliance.............................................394  
FCC DECLARATION...............................................................................................394  
Industry Canada ...............................................................................................395  
FCC Part 68 Telecom.......................................................................................395  
Canadian Limitations Notice............................................................................396  
APPENDIX A: EXPANSION CARD INSTALLATION  
(MVP24-48 & MVP30-60)......................................................................................397  
INSTALLATION.......................................................................................................398  
OPERATION............................................................................................................400  
APPENDIX B: CABLE PINOUTS ......................................................................401  
APPENDIX B: CABLE PINOUTS..............................................................................402  
Command Cable ...............................................................................................402  
Ethernet Connector...........................................................................................402  
T1/E1 Connector...............................................................................................403  
Voice/Fax Channel Connectors........................................................................403  
APPENDIX C: TCP/UDP PORT ASSIGNMENTS ...........................................405  
WELL KNOWN PORT NUMBERS.............................................................................406  
PORT NUMBER ASSIGNMENT LIST.........................................................................406  
APPENDIX D: INSTALLATION INSTRUCTIONS  
FOR MVP428 UPGRADE CARD.........................................................................407  
INDEX .....................................................................................................................413  
6
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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 product groups.  
1. T1 Digital MultiVOIP units, models MVP2400, MVP2410, and  
the capacity-doubling add-on expansion card, model MVP24-  
48.  
2. E1 Digital MultiVOIP units, models, MVP3010 and the  
capacity-doubling add-on expansion card, model MVP30-60.  
3. Analog MultiVOIP units, models MVP810, MVP410, and  
MVP210.  
The table below describes the vital characteristics of these various  
models.  
MultiVOIP Product Family  
MVP  
2400  
MVP  
2410  
MVP  
24-48  
MVP  
3010  
MVP  
30-60  
Description  
Model  
Function  
T1  
T1  
T1  
E1  
E1  
digital  
VOIP  
unit  
digital digital  
digital  
VOIP  
unit  
digital  
VOIP  
add-on  
card  
VOIP  
unit  
VOIP  
add-on  
card  
Capacity  
Chassis/  
24  
24  
24  
30  
30  
channels channels added  
channels  
channels added  
channels  
table  
19” 1U circuit  
rack  
mount  
19” 1U  
rack  
mount  
circuit  
card  
only  
Mounting top  
card  
only  
Description  
MVP  
810  
MVP  
428  
MVP  
410  
MVP  
210  
Model  
Function  
analog  
voip  
add-on analog  
analog  
voip  
card  
voip  
Capacity  
8
4 added  
4
2
channels channels  
channels  
channels  
Chassis/  
Mounting rack  
mount  
19” 1U  
circuit  
card  
only  
19” 1U  
rack  
mount  
table  
top  
8
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MultiVOIP User Guide  
Overview  
Variable Model/Version Icon and Typography. The MultiVOIP  
product family is a coordinated set of products that can operate with  
each other in a seamless fashion. For example, both the digital and  
analog MultiVOIP units use the same graphic user interface (GUI) in  
the MultiVOIP configuration software and both operate under a single  
GUI in the MultiVoipManager remote management software. Because  
this is the case, the various model numbers and version numbers of  
MultiVOIP family products will each appear in various dialog boxes  
and commands. But instead of showing these dialog boxes once for  
each model in this manual, we substitute the following icon.  
Figure 1-1: Variable Model/Version Icon  
It indicates that, whatever MultiVOIP model you are using, all details  
except the very model and version numbers themselves will be the  
same regardless of the MultiVOIP model used. Also, in some cases, we  
will use other typographic devices, like blank underlining  
(“MultiVOIP ____”) to denote information that applies to any  
and all of the products in this product family.  
9
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Overview  
MultiVOIP User Guide  
Introduction to TI MultiVOIPs (MVP2400,  
MVP2410, & MVP24-48)  
We proudly present MultiTech’s T1 Digital Multi-VOIP products.  
The MVP2400 is a table-top model; the MVP2410 is a rack-mount  
model; and the MVP24-48 is an add-on expansion card that doubles the  
capacity of the MVP2410 without adding another chassis. All of these  
voice-over-IP products have fax capabilities. All adhere to the North  
American standard of T1 trunk telephony using digital 24-channel  
time-division multiplexing, which allows 24 phone conversations to  
occur on the T1 line simultaneously. All can also accommodate T1 lines  
of the ISDN Primary Rate Interface type (ISDN-PRI).  
Scale-ability. The MVP2400 and MVP2410 are tailored to companies  
needing more than a few voice-over-IP lines, but not needing carrier-  
class equipment. When expansion is needed, the MVP2410 can be field-  
upgraded into a dual T1 unit by installing the MVP24-48 kit, which is  
essentially a second MultiVOIP motherboard that fits in an open  
expansion-card slot in the MVP2410. The upgraded dual unit then  
accommodates two T1 lines.  
T1 VOIP Traffic. The MVP2400/ 2410 accepts its outbound traffic from  
a T1 trunk that’s connected to either a PBX or to a telco/ carrier. The  
MVP2400/ 2410 transforms the telephony signals into IP packets for  
transmission on LANs, WANs, or the Internet. Inbound IP data traffic  
is converted to telephony data and signaling.  
When connected to PBX. When connected to a PBX, the  
MVP2400/ 2410 creates a network node served by 10/ 100-Base T  
connections. Local PBX phone extensions gain toll-free access to all  
phone stations directly connected to the VOIP network. Phone  
extensions at any VOIP location also gain toll-free access to the entire  
local public-switched telephone network (PSTN) at every other VOIP  
location in the system.  
When connected to PSTN. When the T1 line(s) connected to the  
MVP2400/ 2410 are connected directly to the PSTN, the unit becomes a  
Point-of-Presence server dedicated to local calls off-net.  
H.323 & SIP. Being H.323 compatible, the MVP2400/ 2410 can place  
calls to telephone equipment at remote IP network locations that also  
contain H.323 compatible voice-over-IP gateways. It will interface with  
H.323 software and H.323 gatekeeper units. H.323 specifications also  
bring to voip telephony many special features common to conventional  
telephony. H.323 features of this kind that have been implemented into  
the MuliVOIP include Call Hold, Call Waiting, Call Name  
10  
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MultiVOIP User Guide  
Overview  
Identification, Call Forwarding (from the H.450 standard), and Call  
Transfer (H.450.2 from H.323 Version 2). The fourth version of the  
H.323 standard improves system resource usage (esp. logical port or  
socket usage) by handling call signaling more compactly and allowing  
use of the low-overhead UDP protocol instead of the error-correcting  
TCP protocol where possible.  
The MultiVOIP is also SIP-compatible. However, H.450 Supplementary  
Services features can be used under H.323 only and not under SIP.  
The MultiVOIP2400/ 2410 comes equipped with a variety of data  
compression capabilities, including G.723, G.729, and G.711 and  
features DiffServ quality-of-service (QoS) capabilities.  
VOIP Functions. The MultiVOIP MVP2400/ 2410 gateway performs  
four basic functions: (a) it converts a dialed number into an IP address,  
(b) it sends voice over the data network, (c) it establishes a connection  
with another VOIP gateway at a remote site, and (d) it receives voice  
over the data network. Voice is handled as IP packets with a variety of  
compression options. Each T1 connection to the MultiVOIP provides 24  
time-slot channels to connect to the telco or to serve phone or fax  
stations connected to a PBX.  
Ports. The MVP2400/ 2410 also has a 10/ 100 Mbps Ethernet LAN  
interface, and a Command port for configuration. An MVP2410  
upgraded with the MVP24-48 kit will have two Ethernet LAN interfaces  
and two Command ports.  
Management. Configuration and system management can be done  
locally with the MultiVOIP configuration software. After an IP address  
has been assigned locally, other configuration can be done remotely  
using the MultiVOIP web browser GUI. Remote system management  
can be done with the MultiVoipManager SNMP software or via the  
MultiVOIP web browser GUI. All of these control software packages  
are included on the Product CD.  
11  
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Overview  
MultiVOIP User Guide  
T1 Front Panel LEDs  
The MVP2400, MVP2410, and MVP24-48 all use a common main circuit  
board or motherboard. Consequently the LED indicators are the same  
for all.  
Figure 1-2. MultiVOIP MVP2400 Front Panel  
Active LEDs. The MVP2410 front panel has two sets of identical LEDs.  
In the MVP2410 as shipped (that is, without an expansion card), the  
left-hand set of LEDs is functional whereas the right-hand set is not.  
When the MVP2410 has been upgraded with an MVP24-48 kit, the  
right-hand set of LEDs will also become active.  
Figure 1-3. MultiVOIP MVP2410 Chassis  
12  
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MultiVOIP User Guide  
Overview  
T1 LED Descriptions  
The descriptions below apply to all digital T1 MultiVOIP units.  
MVP2400/2410 Front Panel LED Definitions  
MVP2400/2410 Front Panel LED Definitions  
LED NAME  
Power  
DESCRIPTION  
Indicates presence of power.  
After power up, the Boot LED will be on for about 10  
seconds while the MVP2400/2410 is booting.  
Receive. Lights when receiving data on Ethernet  
port.  
Boot  
RCV  
XMT  
LNK  
Transmit. Lights when transmitting data on  
Ethernet port.  
Link. When lit, VOIP “sees” the hub or network  
via the Ethernet connection.  
COL  
T1  
Collision. Lit when data collisions occur.  
When lit, indicates presence of T1 connection.  
E1. Not supported.  
E1  
PRI  
ONL  
PRI. On if T1 line is of ISDN-Primary-Rate type.  
Online. This LED is on when frame  
synchronization has been established on the  
T1/ E1 link.  
IC  
IC LED is on when Internal Clocking is selected in  
T1/ E1 configuration.  
LC  
Indicates Loss of Carrier.  
Indicates Loss of Signal.  
For testing purposes only.  
LS  
Test  
13  
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Overview  
MultiVOIP User Guide  
Introduction to EI MultiVOIPs  
(MVP3010 & MVP30-60)  
We proudly present MultiTech’s E1 Digital Multi-VOIP products. The  
MVP3010 is a rack-mount model and the MVP30-60 is an add-on  
expansion card that doubles the capacity of the MVP3010 without  
adding another chassis. All of these voice-over-IP products have fax  
capabilities. All adhere to the European standard of E1 trunk telephony  
using digital 30-channel time-division multiplexing, which allows 30  
phone conversations to occur on the E1 line simultaneously. All can  
also accommodate E1 lines of the ISDN Primary Rate Interface type  
(ISDN-PRI).  
Scale-ability. The MVP3010 is tailored to companies needing more  
than a few voice-over-IP lines, but not needing carrier-class equipment.  
When expansion is needed, the MVP3010 can be field-upgraded into a  
dual E1 unit by installing the MVP30-60 kit, which is essentially a  
second MultiVOIP motherboard that fits into an open expansion-card  
slot in the MVP3010. The upgraded dual unit then accommodates two  
E1 lines.  
E1 VOIP Traffic. The MVP3010 accepts its outbound traffic from a E1  
trunk that’s connected to either a PBX or to a telco/ carrier. The  
MVP3010 transforms the telephony signals into IP packets for  
transmission on LANs, WANs, or the Internet. Inbound IP data traffic  
is converted to telephony data and signaling.  
When connected to PBX. When connected to a PBX, the MVP3010  
creates a network node served by 10/ 100-Base T connections. Local  
PBX phone extensions gain toll-free access to all phone stations directly  
connected to the VOIP network. Phone extensions at any VOIP location  
also gain local-rate access to the entire local public-switched telephone  
network (PSTN) at every other VOIP location in the system.  
When connected to PSTN. When the E1 line(s) connected to the  
MVP3010 are connected directly to the PSTN, the unit becomes a Point-  
of-Presence server dedicated to local calls off-net.  
H. 323 & SIP. Being H.323 compatible, the MVP3010 can place calls to  
telephone equipment at remote IP network locations that also contain  
H.323 compatible voice-over-IP gateways. It will interface with H.323  
software and H.323 gatekeeper units. H.323 specifications also bring to  
voip telephony many special features common to conventional  
telephony. H.323 features of this kind that have been implemented into  
the MuliVOIP include Call Hold, Call Waiting, Call Identification, Call  
Forwarding (from the H.450 standard), and Call Transfer (H.450.2 from  
14  
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MultiVOIP User Guide  
Overview  
H.323 Version 2). The fourth version of the H.323 standard improves  
system resource usage (esp. logical port or socket usage) by handling  
call signaling more compactly and allowing use of the low-overhead  
UDP protocol instead of the error-correcting TCP protocol where  
possible.  
The MultiVOIP is also SIP-compatible. However, H.450  
Supplementary Services features can be used under H.323 only and not  
under SIP.  
The MultiVOIP3010 comes equipped with a variety of data  
compression capabilities, including G.723, G.729, and G.711 and  
features DiffServ quality-of-service (QoS) capabilities.  
VOIP Functions. The MultiVOIP MVP3010 gateway performs four  
basic functions: (a) it converts a dialed number into an IP address, (b) it  
sends voice over the data network, (c) it establishes a connection with  
another VOIP gateway at a remote site, and (d) it receives voice over  
the data network. Voice is handled as IP packets with a variety of  
compression options. Each E1 connection to the MultiVOIP provides 30  
time-slot channels to connect to the telco or to serve phone or fax  
stations connected to a PBX.  
Ports. The MVP3010 also has a 10/ 100 Mbps Ethernet LAN interface,  
and a Command port for configuration. An MVP3010 upgraded with  
the MVP30-60 kit will have two Ethernet LAN interfaces and two  
Command ports.  
Management. Configuration and system management can be done  
locally with the MultiVOIP configuration software. After an IP address  
has been assigned locally, other configuration can be done remotely  
using the MultiVOIP web browser GUI. Remote system management  
can be done with the MultiVoipManager SNMP software or via the  
MultiVOIP web browser GUI. All of these control software packages  
are included on the Product CD.  
15  
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Overview  
MultiVOIP User Guide  
E1 Front Panel LEDs  
Because the MVP3010 and MVP30-60 both use a common main circuit  
card or motherboard, the LED indicators are the same for both.  
Figure 1-4. MultiVOIP MVP3010 Chassis  
Active LEDs. The MVP3010 front panel has two sets of identical LEDs.  
In the MVP3010 as shipped (that is, without an expansion card), the  
left-hand set of LEDs is functional whereas the right-hand set is not.  
When the MVP3010 has been upgraded with an MVP30-60 kit, the  
right-hand set of LEDs will also become active.  
E1 LED Descriptions  
MVP3010 Front Panel LED Definitions  
MVP3010 Front Panel LED Definitions  
LED NAME  
Power  
DESCRIPTION  
Indicates presence of power.  
After power up, the Boot LED will be on for about 10  
seconds while the MVP3010 is booting.  
Receive. Lights when receiving data on Ethernet  
port.  
Boot  
RCV  
XMT  
LNK  
COL  
Transmit. Lights when transmitting data on  
Ethernet port.  
Link. When lit, VOIP “sees” the hub or network  
via the Ethernet connection.  
Collision. Lit when data collisions occur.  
16  
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MultiVOIP User Guide  
Overview  
MVP3010 Front Panel LED Definitions (cont’d)  
T1  
E1  
T1. Not supported.  
E1. When lit, indicates presence of E1  
connection.  
PRI  
PRI. On if E1 line is of ISDN-Primary-Rate type.  
ONL  
Online. This LED is on when frame  
synchronization has been established on the  
T1/ E1 link.  
IC  
IC LED is on when Internal Clocking is selected  
in T1/ E1 configuration.  
LC  
Indicates Loss of Carrier.  
Indicates Loss of Signal.  
LS  
Test  
For testing purposes only. For testing purposes  
only.  
17  
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Overview  
MultiVOIP User Guide  
Introduction to Analog MultiVOIPs  
(MVP-210/410/810 & MVP428)  
VOIP: The Free Ride. We proudly present Multi-Tech's MVP-  
210/ 410/ 810 generation of MultiVOIP Voice-over-IP Gateways. They  
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 MultiVOIPs units.  
Capacity. The MultiVOIP model MVP810 is a eight-channel unit, the  
MVP410 a four-channel unit, and the MVP210 a two-channel unit. All  
of these MultiVOIP units have a 10/ 100Mbps Ethernet interface and a  
command port for configuration. The MVP428 is an expansion circuit  
card for the four-channel MVP410 that turns it into an eight-channel  
voip.  
Mounting. Mechanically, the MVP410 and MVP810 MultiVOIPs are  
designed for a one-high industry-standard EIA 19-inch rack enclosure.  
By contrast, the MVP210 is a table-top unit. 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 which 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.  
H.323 & SIP. The MultiVOIP supports the H.323 standards-based  
protocol enabling your MultiVOIP to participate in real-time  
conferencing with other third-party VOIP Gateways or endpoints that  
support the H.323 protocol (for example, Microsoft NetMeeting® ). The  
H.323 standard defines how endpoints make and receive calls, how  
endpoints negotiate a common set of audio and data capabilities, how  
information is formatted and sent over the network, and how endpoints  
communicate with their respective Gatekeepers.  
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MultiVOIP User Guide  
Overview  
H.323 specifications also bring to voip telephony many special features  
common to conventional telephony. H.323 features of this kind that  
have been implemented into the MuliVOIP include Call Hold, Call  
Waiting, Call Identification, Call Forwarding (from the H.450 standard),  
and Call Transfer (H.450.2 from H.323 Version 2). The fourth version of  
the H.323 standard improves system resource usage (esp. logical port or  
socket usage) by handling call signaling more compactly and allowing  
use of the low-overhead UDP protocol instead of the error-correcting  
TCP protocol where possible.  
The MultiVOIP is also SIP-compatible. However, H.450 Supplementary  
Services features can be used under H.323 only and not under SIP.  
Gatekeepers. Gatekeeper software is optional and when used in a  
network, it typically resides on a designated PC. It acts as the central  
point for all calls within its zone and provides call control services to all  
registered endpoints. In addition, Gatekeepers can perform bandwidth  
management through support for Bandwidth Request, Confirm, and  
Reject messages.  
Management. Configuration and system management can be done  
locally with the MultiVOIP configuration software. After an IP address  
has been assigned locally, other configuration can be done remotely  
using the MultiVOIP web browser GUI. Remote system management  
can be done with the MultiVoipManager SNMP software or via the  
MultiVOIP web browser GUI. All of these control software packages  
are included on the Product CD.  
Voi ceF/ ax5  
Voice/ Fax6  
V
oiceF/ ax7  
Voi ce/ Fax8  
RC  
X
X
M
M
T
T
R
VC  
SX  
G
RS  
RS  
G
G
X
X
MT  
MT  
CR  
V
X
S
G
R
SG  
SG  
XM  
XM  
T
T
R
C
V
X
SG  
RS  
G
G
X
X
TM  
TM  
V
X
S
G
R
GS  
GS  
Power  
Boot  
Ether net  
Voic e/Fax1  
Voice/ Fax2  
V
oiceF/ ax  
3
Voi ce/ Fax4  
RC  
R
C
V
X
M
T
C
LO  
LN  
K
R
VC  
SX  
G
CR  
V
X
S
G
R
R
C
V
X
SG  
RS  
V
X
S
G
R
Figure 1-5: MVP-410/810 Chassis  
Figure 1-6: MVP-210 Chassis  
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Overview  
MultiVOIP User Guide  
Analog MultiVOIP Front Panel LEDs  
LED Types. The MultiVOIPs have two types of LEDs on their front  
panels:  
(1) general operation LED indicators (for power, booting, and  
ethernet functions), and  
(2) channel operation LED indicators which describe the data  
traffic and performance in each VOIP data channel.  
Active LEDs. On both the MVP410 and MVP810, there are eight sets of  
channel-operation LEDs. However, on the MVP410, only the lower  
four sets of channel-operation LEDs are functional. On the MVP810, all  
eight sets are functional.  
Voice/Fax  
5
Voice/Fax  
RC SG  
6
Voice/Fax  
7
Voice/Fax  
8
XM  
T
RC SG RSG  
V
X
X
M
T
V
X
RSG  
XM  
T
RCV XSG RSG  
XM  
T
R
CV SG RSG  
X
Power  
Boot  
Ethernet  
OL  
Voice/Fax  
RC SG RSG  
1
Voice/Fax  
CV SG RSG  
4
Voice/Fax  
RC SG  
2
Voice/Fax  
3
RCV  
X
M
T
C
LNK  
X
M
T
V
X
X
M
T
V
X
RSG  
XM  
T
RCV XSG RSG  
X
M
T
R
X
Figure 1-7. MVP410/810 Front Panel  
Similarly, the MVP210 has the general-operation indicator LEDs and  
two sets of channel-operation LEDs, one for each channel.  
Figure 1-8. MVP210 Front Panel  
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MultiVOIP User Guide  
Overview  
Analog MultiVOIP LED Descriptions  
MVP-210/410/810 Front Panel LED Definitions  
LED NAME DESCRIPTION  
General Operation LEDs (one set on each MultiVOIP model)  
Power  
Boot  
Indicates presence of power.  
After power up, the Boot LED will be on briefly while the  
MultiVOIP is booting. It lights whenever the MultiVOIP is  
booting or downloading a setup configuration data set.  
Ethernet  
RCV. Receive. Lights (blinks) when receiving data on  
Ethernet port.  
XMT. Transmit. Lights (blinks) when transmitting  
data on Ethernet port. ..  
LNK. Link. When lit, VOIP “sees” the hub or network  
via the Ethernet connection. ..  
COL. Collision. Lit when data collisions occur. ..  
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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Overview  
MultiVOIP User Guide  
Computer Requirements  
The computer on which the MultiVOIP’s configuration program is  
installed must meet these requirements:  
•
must be IBM-compatible PC with MS Windows operating  
system;  
•
must have an available COM port for connection to the  
MultiVOIP.  
However, this PC does not need to be connected to the MultiVOIP  
permanently. It only needs to be connected when local configuration  
and monitoring are done. Nearly all configuration and monitoring  
functions can be done remotely via the IP network.  
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MultiVOIP User Guide  
Overview  
Specifications  
Specs for Digital T1 MultiVOIP Units  
Digital T1 MultiVOIP Specifications  
MVP-2410  
Parameter  
……/Model  
w/ MVP24-48  
Expansion  
Card  
MVP-2400  
MVP-2410  
Operating  
Voltage(s)  
External  
transformer:  
1.6A@5v  
100-240 VAC  
1.2 - 0.6 A  
100-240 VAC  
1.2 - 0.6 A  
Mains  
Frequencies  
Power  
50/60 Hz  
50/60 Hz  
17 watts  
50/60 Hz  
27 watts  
13 watts  
Consumption  
Mechanical  
Dimensions  
6.2” W x  
9” D x  
1.4” H  
1.75”H x  
17.4”W x  
8.75”D  
1.75”H x  
17.4”W x  
8.75”D  
15.8cm W x  
22.9cm D x  
3.6cm H  
4.5cm H x  
44.2 cm W x  
22.2 cm D  
7.1 lbs.  
4.5cm H x  
44.2 cm W x  
22.2 cm D  
7.5 lbs.  
Weight  
1.8lbs  
(.82kg)  
(3.2 kg)  
(3.4 kg)  
2.2lbs (.98kg)  
with transformer  
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Overview  
MultiVOIP User Guide  
Specs for Digital E1 MultiVOIP Units  
Digital E1 MultiVOIP Specifications  
Parameter  
……/Model  
MVP-3010  
MVP-3010  
w/ MVP30-60  
Expansion  
Card  
Operating  
Voltage(s)  
100-240 VAC  
1.2 - 0.6 A  
50/60 Hz  
100-240 VAC  
1.2 - 0.6 A  
50/60 Hz  
Mains  
Frequencies  
Power  
17 watts  
27 watts  
Consumption  
Mechanical  
Dimensions  
1.75”H x  
17.4”W x  
8.75”D  
1.75”H x  
17.4”W x  
8.75”D  
4.5cm H x  
44.2 cm W x  
22.2 cm D  
7.1 lbs.  
4.5cm H x  
44.2 cm W x  
22.2 cm D  
7.5 lbs.  
Weight  
(3.2 kg)  
(3.4 kg)  
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MultiVOIP User Guide  
Overview  
Specs for Analog MultiVOIP Units  
Analog MultiVOIP Specifications  
Parameter  
……/Model  
MVP210  
MVP410  
MVP810  
or  
MVP410 + 428  
100-240 VAC  
1.2 - 0.6 A  
Operating  
Voltage(s)  
External  
transformer:  
3A @5V  
100-240 VAC  
1.2 - 0.6 A  
Mains  
Frequencies  
50/60 Hz  
50/60 Hz  
50/60 Hz  
Power  
19 watts  
29 watts  
46 watts  
Consumption  
Mechanical  
Dimensions  
6.2” W x  
9” D x  
1.4” H  
1.75”H x  
17.4”W x  
8.5”D  
1.75”H x  
17.4”W x  
8.5”D  
15.8cm W x  
22.9cm D x  
3.6cm H  
1.8lbs (.82kg)  
2.6lbs (1.17kg)  
with transformer  
4.5cm H x  
44.2 cm W x  
21.6 cm D  
7.1 lbs.  
4.5cm H x  
44.2 cm W x  
21.6 cm D  
7.7 lbs.  
Weight  
(3.2 kg)  
(3.5 kg)  
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Overview  
MultiVOIP User Guide  
Installation at a Glance  
The basic steps of installing your MultiVOIP network involve  
unpacking the units, connecting the cables, and configuring the units  
using management software (MultiVOIP Configuration software) and  
confirming connectivity with another voip site. This process results in a  
fully functional Voice-Over-IP network.  
Related Documentation  
The MultiVOIP User Guide (the document you are now reading) comes  
in electronic form and is included on your system CD. It presents in-  
depth information on the features and functionality of Multi-Tech’s  
MultiVOIP Product Family.  
The CD media is produced using Adobe AcrobatTM for viewing and  
printing the user guide. To view or print your copy of a user guide,  
load Acrobat ReaderTM on your system. The Acrobat Reader is included  
on the MultiVOIP CD and is also a free download from Adobe’s Web  
Site:  
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. To select theMultiVOIP  
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.  
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Chapter 2: Quick Start Instructions  
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Quick Start Instructions  
MultiVOIP User Guide  
Introduction  
This chapter gets the MultiVOIP up and running quickly. The details  
we’ve skipped to make this brief can be found elsewhere in the manual  
(see Table of Contents and Index).  
MultiVOIP Startup Tasks  
Task  
Summary  
G 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!)  
G Placement  
Decide where you’ll mount the voip.  
G The Command/Control  
Computer:  
Some modest minimum specifications  
must be met. A COM port must be set  
up.  
Specs & Settings  
G Hookup  
Connect power, phone, and data cables  
per diagram.  
G Software Installation  
This is the configuration program.  
It’s a standard Windows software  
installation.  
G 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  
G 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  
G Connectivity Test  
You’ll find out if your voip system can  
carry phone calls between two sites.  
That means you’re up and running!  
G Troubleshooting  
Detect and remedy any problems that  
might have prevented connectivity.  
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MultiVOIP User Guide  
Quick Start Instructions  
Phone/IP Details *Absolutely Needed*  
Before Starting the Installation  
Gather IP Information  
Ask your computer network  
administrator.  
Info needed to operate:  
all MultiVOIP models.  
G
IP Network Parameters:  
Record for each VOIP Site  
in System  
ꢀ
• IP Address  
• IP Mask  
• Gateway  
• Domain Name Server (DNS) Info  
(not implemented; for future use)  
Gather Telephone Information  
T1 Phone Parameters  
Info needed to operate:  
MVP2400  
G
Ask phone company or  
PBX maintainer.  
MVP2410  
T1 Telephony Parameters:  
Record for this VOIP Site  
ꢀ
• Which frame format is used? ESF___ or D4___  
• Which CAS or PRI protocol is used? ______________  
• Clocking: Does the PBX or telco switch use  
internal or external clocking? _________________  
Note that the setting used in the voip unit will be the  
opposite of the setting used by the telco/PBX.  
• Which line coding is used? AMI___ or B8ZS___  
• Pulse shape level?: (most commonly 0 to 40 meters)  
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Quick Start Instructions  
MultiVOIP User Guide  
Phone/IP Details *Absolutely Needed* (cont’d)  
Gather Telephone Information  
E1 Phone Parameters  
Info needed to operate:  
G
MVP3010  
Ask phone company or  
PBX maintainer.  
E1 Telephony Parameters:  
Record for this VOIP Site  
ꢀ
• Which frame format is used? Double Frame_____  
MultiFrame w/ CRC4_____  
MultiFrame w/ CRC4 modified_____  
• Which CAS or PRI protocol is used? ______________  
• Clocking: Does the PBX or telco switch use  
internal or external clocking? _________________  
Note that the setting used in the voip unit will be the  
opposite of the setting used by the telco/PBX.  
• Which line coding is used? AMI___ or HDB3___  
• Pulse shape level?: (most commonly 0 to 40 meters)  
Gather Telephone Information  
Analog Phone Parameters  
Needed for:  
MVP810  
MVP410  
MVP210  
G
Ask phone company or  
telecom manager.  
Analog Telephony Interface Parameters:  
Record for this VOIP Site  
ꢀ
• Which interface type (or “signaling”) is used?  
E&M_____ FXS/FXO_____  
• If FXS, determine whether the line will be used for a  
phone, fax, or KTS (key telephone system)  
• If FXO, determine if line will be an analog PBX  
extension or an analog line from a telco central office  
• If E&M, determine these aspects of the E&M trunk  
line from the PBX:  
• What is its Type (1, 2, 3, 4, or 5)?  
• Is it 2-wire or 4-wire?  
• Is it Dial-Tone or Wink?  
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MultiVOIP User Guide  
Quick Start Instructions  
Phone/IP Details Often Needed/Wanted  
Obtain Email Address for VOIP (for email call log reporting)  
required if log reports of  
Optional  
VOIP call traffic  
are to be sent by email  
SMTP Parameters  
Preparation Task:  
T
o
e
:
I . T  
.
D
e
p
a
r
t
m
e
n
t
t
Ask Mail Server  
r
:
e
m
a
i l  
a
c
c
o
u
n
f
o
r
V
O
I P  
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.  
voip-unit2@biggytech.com  
Get the IP address of the  
mail server computer, as  
well.  
Identify Remote VOIP Site to Call  
When you’re done installing the MultiVOIP, you’ll want to confirm that  
it is configured and operating properly. To do so, it’s good to have  
another voip that you can call for testing purposes. You’ll want to  
confirm end-to-end connectivity. You’ll need IP and telephone  
information about that remote site.  
If this is the very first voip in the system, you’ll want to coordinate the  
installation of this MultiVOIP with an installation of another unit at a  
remote site.  
Identify VOIP Protocol to be Used  
Will you use H.323 or SIP? Each has advantages and disadvantages.  
Although it is possible to mix protocols in a single VOIP system, it is  
highly desirable to use the same VOIP protocol for all VOIP units in  
the system.  
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Quick Start Instructions  
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.  
The Command/Control Computer (Specs & Settings)  
The computer used for command and control of the MultiVOIP  
(a) must be an IBM-compatible PC,  
(b) must use a Microsoft operating system,  
(c) must be connected to your local network (Ethernet) system, and  
(d) must have an available serial COM port.  
The configuration tasks and control tasks the PC will have to do with  
the MultiVOIP are not especially demanding. Still, we recommend  
using a reasonably new computer. The computer that you use to  
configure your MultiVOIP need not be dedicated to the MultiVOIP  
after installation is complete.  
COM port on controller PC. You’ll need an available COM port on the  
controller PC. You’ll need to know which COM port is available for use  
with the MultiVOIP (COM1, COM2, etc.).  
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MultiVOIP User Guide  
Quick Start Instructions  
Quick Hookups  
Hookup for MVP2410 & MVP3010  
T1/E1 MultiVOIP Hookup  
(MVP-2410/3010)  
Grounding  
Screw  
Ca bling to your IP netw ork.  
RJ -45 connector.  
Ca bling to computer running  
MultiVOIP softw a re.  
RJ -45 to s eria l connector (DB9).  
T1/ E1/ PRI ca bling to your PBX,  
a nd/ or to the PSTN.  
RJ -45 connector.  
Ethernet Command  
Digital Voice  
Trunk  
l
10 /100  
RS-232  
O
Power Ca ble  
Recepta cle  
On/ Off Switch  
Hookup for MVP410 & MVP810  
Analog MultiVOIP Hookup  
MVP-410/810  
Pow er Ca ble  
Recepta cle  
Ca bling to computer running  
MultiVOIP s oftw a re.  
Connector a t MultiVOIP: DB-25.  
Connector a t computer: DB-9.  
MVP810 ha s 8 connector pa irs.  
MVP410 ha s 4 connector pa irs.  
Only 1 connector of a ny pa ir is  
us ed a t a time.  
Grounding  
Screw :  
Connect to  
Ea rth Ground  
E&M FXS/FXO E&M FXS/FXO  
FXS/FXO  
E&M  
E&M FXS/FXO  
Ethernet  
Command  
FXS/FXO  
E&M FXS/FXO  
E&M  
E&M FXS/FXO  
E&M FXS/FXO  
On/ Off  
Sw itch  
Ca bling to phone equipment.  
Ca bling to your IP network.  
RJ -45 connector.  
E& M (RJ -45 connector):  
connects to E&M trunk line  
from PBX or telco office.  
FXS (RJ -11 connector):  
connects to phone, fa x,  
or key phone s ys tem.  
FXO (RJ -11 connector):  
connects to a na log phone line  
or a na log PBX extens ion.  
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Quick Start Instructions  
MultiVOIP User Guide  
Hookup for MVP2400  
DIGITAL VOICE  
COMMAND  
RS232  
ETHERNET  
1
0
TRUNK  
Power Connection  
10/100  
POWER  
T1  
PBX  
Command Port Connection  
PSTN  
ꢀ
Telephony Connection  
Network Connection  
Hub  
Hookup for MVP210  
CH1  
CH2  
ETHERNET  
FXS/FXO  
E&MFXS/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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MultiVOIP User Guide  
Quick Start Instructions  
Load MultiVOIP Control Software onto PC  
For more details, see Chapter 4: Software Installation.  
1. MultiVOIP must be properly cabled. Power must be turned on.  
2. Insert MultiVOIP CD into drive. Allow 10-20 seconds for Autorun to  
start. If Autorun fails, go to  
My Computer | CD ROM drive | Open. Click Autorun icon.  
3. At first dialog box, click Install Software.  
4. At ‘welcome’ screen, click Next.  
5. Follow on-screen instructions. Accept default program folder  
location and click Next.  
6. Accept default icon folder location. Click Next. Files will be copied.  
7. Select available COM port on command/ control computer.  
8. At completion screen, click Finish.  
9. At the prompt “Do you want to run MultiVOIP Configuration?,”  
click No. Software installation is complete.  
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Quick Start Instructions  
MultiVOIP User Guide  
Phone/IP Starter Configuration  
Full details here:  
MVP2400  
MVP2410  
MVP3010  
Chapter 5: Technical Configuration for  
Digital T1/E1 MultiVOIPs  
MVP210  
MVP410  
MVP810  
Chapter 6: Technical Configuration for  
Analog MultiVOIPs  
1. Open MultiVOIP program: Start | MultiVOIP xxx | Configuration.  
2. Go to Configuration | IP. Enter the IP parameters for your voip site.  
3. Do you want to configure and operate the MultiVOIP unit using the  
web browser GUI? (It has the same functionality as the local  
Windows GUI, but offers remote access.)  
If NO, skip to step 5.  
If YES, continue with step 4.  
4. Enable Web Browser GUI (Optional). To do configuration and  
operation procedures using the web browser GUI, you must first  
enable it. To do so, follow these steps.  
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.  
NOTE: Required on first use of  
Web Browser GUI only.  
H. Use web browser GUI to  
configure or operate voip.  
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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5. Go to Configuration | Voice/Fax. Select Coder | “Automatic.” At  
the right-hand side of the dialog box, click Default. If you know any  
specific parameter values that will apply to your system, enter them.  
Click Copy Channel. Select Copy to All. Click Copy. At main  
Voice/ Fax Parameters screen, click OK to exit from the dialog box.  
6. Enter telephone system information.  
Analog MultiVOIPs  
Digital MultiVOIPs  
MVP-210/ 410/ 810  
MVP-2400/ 2410/ 3010  
Go to  
Go to  
Configuration | Interface.  
Enter parameters obtained from  
phone company or PBX  
administrator.  
Configuration | T1/E1/ISDN.  
Enter parameters obtained from  
phone company or PBX  
administrator.  
7. Go to Configuration | Regional Parameters. Select the  
Country/Region that fits your situation. Click Default and confirm.  
Click OK to exit from the dialog box.  
8. 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 10.  
If YES, continue with step 9.  
9. 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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Phone/IP Starter Configuration (continued)  
9. (continued) 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.  
10. Go to Configuration | Logs.  
Select “Enable Console Messages.” (Not applicable if using Web GUI.)  
To allow log reports by email (if desired), click SMTP. Click OK.  
To do logging with a SysLog client program, click on “SysLog Server  
– Enable” in the Logs screen. To implement this function, you must  
install a SysLog client program. For more info, see the “SysLog  
Server Functions” section of the Operation & Maintenance chapter of  
the User Guide.  
11. 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.  
12. 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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Quick Start Instructions  
Phonebook Starter Configuration (with remote voip)  
To do this part of the quick setup, you need to know of another voip  
that you can call to conduct a test. It should be at a remote location,  
typically somewhere outside of your building. You must know the  
phone number and IP address for that site. We are assuming here that  
the MultiVOIP will operate in conjunction with a PBX.  
You must configure both the Outbound Phonebook and the Inbound  
Phonebook. A starter configuration only means that two voip locations  
will be set up to begin the system and establish voip communication.  
Outbound Phonebook  
1. Open the MultiVOIP program  
(Start | MultiVOIP xxx | Configuration  
2. Go to Phone Book | PhoneBook Modify | Outbound Phonebook  
| Add Entry.  
3. On a sheet of paper, write down the calling code of the remote voip  
(area code, country code, city code, etc.) that you’ll be calling.  
Follow the example that best fits your situation.  
North America,  
Long-Distance Example  
Euro, National Call  
Example  
Technician in Seattle (area  
206) must set up one voip  
there, another in Chicago  
(area 312, downtown).  
Technician in central  
London (area 0207) to set  
up voip there, another in  
Birmingham (area 0121).  
Answer: Write down 312.  
Answer: write down 0121.  
Euro, International Call Example  
Technician in Rotterdam (country 31; city 010) to  
set up one voip there, another in Bordeaux  
(country 33; area 05).  
Answer: write down 3305.  
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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  
London voip works with  
PBX that uses “9” for all  
out-of-building calls  
immediately precede area  
code of dialed number.  
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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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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6. Tally up the number of digits that must be dialed to reach the remote  
voip site (including prefix digits of all types). Enter this number in  
the “Total Digits” field.  
North America,  
Long-Distance Example  
Seattle-Chicago system.  
Euro, National Call  
Example  
London/ Birming. system.  
To complete Seattle-to-  
To complete London-to-  
Birmingham call, 90121 must  
be followed by the 7-digit  
local phone number in  
Birmingham.  
Chicago call, 81312 must be  
followed by the 7-digit local  
phone number in Chicago.  
Answer: enter 12 as number  
of Total Digits in  
Outbound Phone-  
book of Seattle  
Answer: enter 12 as number  
of Total Digits in  
Outbound Phone-  
book of London  
voip.  
voip.  
Euro, International Call Example  
Rotterdam/ Bordeaux system.  
To complete Rotterdam-to-Bordeaux call, 903305 must  
be followed by 8-digit local phone number in Bordeaux.  
Answer: enter 14 as number of Total Digits in  
Outbound Phonebook of Rotterdam voip.  
7. In the “Remove Prefix” field, enter the initial PBX access digit  
(“8” or “9”).  
North America,  
Long-Distance Example  
Seattle-Chicago system.  
Euro, National Call  
Example  
London/ Birming. system.  
Answer: enter 8 in “Remove  
Prefix” field of  
Answer: enter 9 in “Remove  
Prefix” field of  
Seattle Outbound  
Phonebook.  
London Outbound  
Phonebook.  
Euro, International Call Example  
Rotterdam/ Bordeaux system.  
Answer: enter 9 in “Remove Prefix” field of Outbound  
Phonebook for Rotterdam voip.  
Some PBXs will not ‘hand off’ the “8” or “9” to the voip. But for those PBX  
units that do, it’s important to enter the “8” or “9” in the “Remove Prefix”  
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field in the Outbound Phonebook. This precludes the problem of having to  
make two inbound phonebook entries at remote voips, one to account for  
situations where “8” is used as the PBX access digit, and another for when  
“9” is used.  
8. Select the voip protocol that you will use (H.323 or SIP).  
9. Click OK to exit from the Add/Edit Outbound Phonebook screen.  
Inbound Phonebook  
1. Open the MultiVOIP program.  
(Start | MultiVOIP xxx | Configuration  
2. Go to Phone Book | PhoneBook Modify | Inbound Phonebook  
| Add Entry.  
3. In the “Remove Prefix” field, enter your local calling code (area code,  
country code, city code, etc.) preceded by any other “access digits”  
that are required to reach your local site from the remote voip  
location (think of it as though the call were being made through the  
PSTN – even though it will not be).  
North America,  
Long-Distance Example  
Euro, National Call  
Example  
Seattle-Chicago system.  
London/ Birming. system.  
Seattle is area 206. Chicago  
employees must dial 81  
before dialing any Seattle  
number on the voip system.  
Inner London is 0207 area.  
Birmingham employees must  
dial 9 before dialing any  
London number on the voip  
system.  
Answer: 1206 is prefix to be  
removed by local  
Answer: 0207 is prefix to be  
removed by local  
(Seattle) voip.  
(London) voip.  
Euro, International Call Example  
Rotterdam/ Bordeaux system.  
Rotterdam is country code 31, city code 010. Bordeaux  
employees must dial 903110 before dialing any  
Rotterdam number on the voip system.  
Answer: 03110 is prefix to be removed by local  
(Rotterdam) voip.  
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4. In the “Add Prefix” field, enter any digits that must be dialed from  
your local voip to gain access to the PSTN.  
North America,  
Long-Distance Example  
Euro, National Call  
Example  
Seattle-Chicago system.  
London/ Birming. system.  
On Seattle PBX, “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 “0.” A zero value means the  
voip unit will assign the call to an available channel. If desired,  
specific channels can be assigned to specific incoming calls (i.e., to  
any set of calls received with a particular incoming dialing pattern).  
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6. In the “Description” field, it is useful to describe the ultimate  
destination of the calls. For example, in a New York City voip  
system, “incoming calls to Manhattan office,” might describe a  
phonebook entry, as might the descriptor “incoming calls to NYC  
local calling area.” The description should make the routing of calls  
easy to understand. (40 characters max.)  
North America,  
Long-Distance Example  
Euro, National Call  
Example  
Seattle-Chicago system.  
London/ Birming. system.  
Possible Description:.  
Free Seattle access, all  
employees  
Possible Description:.  
Local-rate London access,  
all empl.  
Euro, International Call Example  
Rotterdam/ Bordeaux system.  
Possible Description:. Local-rate Rotterdam access, all  
empl.  
7. Repeat steps 2-6 for each inbound phonebook entry. When all entries  
are complete, go to step 8.  
8. Click OK to exit the inbound phonebook screen.  
9. Click on Save Setup. Highlight Save and Reboot. Click OK.  
Your starter inbound phonebook configuration is complete.  
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Phonebook Tips  
Preparing the phonebook for your voip system is a complex task that, at  
first, seems quite daunting. These tips may make the task easier.  
1. Use Dialing Patterns, Not Complete Phone Numbers. You will not  
generally enter complete phone numbers in the voip phonebook.  
Instead, you’ll enter “destination patterns” that involve area codes and  
other digits. If the destination pattern is a whole area code, you’ll be  
assigning all calls to that area code to go to a particular voip 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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(b) access codes. There are digits (PSTN access codes) that must be  
dialed to gain access to an operator, to access the publicly switched  
‘long-distance’ calling system(North America), to access the publicly  
switched ‘national’ calling system (Europe and elsewhere), or to access  
the publicly switched ‘international’ calling system (worldwide).  
There are digits (PBX access codes) that must be dialed by phones  
connected to PBX systems or key systems. Often a “9” must be dialed  
on a PBX phone to gain access to the PSTN (‘to get an outside line’).  
Sometimes “8” must be dialed on a PBX phone to divert calls onto a  
leased line or to a voip system. However, sometimes PBX systems are  
‘smart’ enough to route calls to a voip system without a special access  
code (so that “9” might still be used for all calls outside of the building).  
There are also digits (special access codes) that must be dialed to gain  
access to a particular discount long-distance carrier or to some other  
closed or proprietary telephone system.  
(c) local exchange numbers. Within any calling area there will be many  
local exchange numbers. A single exchange may be used for an entire  
small town. In cities, an exchange may be used for a particular  
neighborhood (although exchanges in cities do not always cover easily  
discernible areas). Organizations like businesses, governments,  
schools, and universities are also commonly assigned exchange  
numbers for their exclusive use. In some cases, these organizational-  
assigned exchanges can become non-localized because the exchange is  
assigned to one facility and linked, by the organization’s private  
network, to other sometimes distant locations.  
(d) specialized codes. Some proprietary voip units assign, to sites and  
phone stations, numbers that are not compatible with PSTN  
numbering. This can also occur in PBX or key systems. These  
specialized numbers must be handled on a case-by-case basis.  
3. Knowing When to Drop Digits.  
Example  
When calling area codes and  
access codes are used in  
combination, a leading “1” or “0”  
must sometimes be dropped.  
Area code for Inner London is  
listed as “0207.” However, in  
international calls the leading  
“0” is dropped.  
U.K.  
Country  
Code  
Phonebook Entry G  
International  
Access Code  
Leading Zero  
Dropped from  
Area Code  
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4. Using a Comma.  
Detail  
Commas are used in telephone  
dialing strings to indicate a pause  
to allow a dial tone to appear  
(common on PBX and key  
systems). Commas may be used  
only in the “Add Prefix” field of  
the Inbound Phonebook.  
,
= 1-second pause  
in many PBX systems  
(not needed in all)  
5. Ease of Use. The phonebook setup determines how easy the voip  
system is to use. Generally, you’ll want to make it so dialing a voip call  
is very similar to dialing any other number (on the PSTN or through the  
PBX).  
6. Avoid Unintentional Calls to Official/Emergency Numbers. Dialing a  
voip call will typically be somewhat different than ordinary dialing.  
Because of this, it’s possible to set up situations, quite unwittingly,  
where phone users may be predisposed to call official numbers without  
intending to do so. Conversely, a voip/ PBX system might also make it  
difficult to place an official/ emergency call when one intends to do so.  
Study your phonebook setup and do some dialing on the system to  
avoid these pitfalls.  
7. Inbound/Outbound Pattern Matching. In general, the Inbound  
Phonebook entries of the local voip unit will match the Outbound  
Phonebook entries of the remote voip unit. Similarly, the Outbound  
Phonebook entries of the local voip unit will match the Inbound  
Phonebook entries of the remote voip unit. There will often be non-  
matching entries, but it’s nonetheless useful to notice the matching  
between the phonebooks.  
8. Simulating Network in-lab/on-benchtop. One common method of  
configuring a voip network is to to set up a local IP network in a lab,  
connect voip units to it, and perhaps have phones connected on channel  
banks to make test calls.  
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Phonebook Example  
One Common Situation  
Boise Office  
Area: 208  
PBX System.  
Voip Example. This company has offices in three  
different cities. The PBX units all operate alike.  
Notably, they all give access to outside lines using  
“9.” They all are ‘smart’ enough to identify voip calls  
without using a special access digit (“8” is used in  
some systems). Finally, the system operates so that  
employees in any office can dial employees in any  
other office using only three digits. Here are the  
phonebooks needed for that system.  
Main Number:  
333-2700  
PSTN  
90 extensions  
204.16.49.73  
24-Channel  
Digital VoIP  
(MVP2410)  
Inbound Phonebook  
Each Inbound Phonebook contains  
two entries. The first entry (4 digits)  
specifies how incoming calls from the  
other voip sites will be handled if  
they go out onto the local PSTN.  
Essentially, all those calls come to the  
receiving voip with a pattern  
beginning with 1+area code. The local  
voip removes those four digits  
because they aren’t needed when  
dialing locally. The local voip  
attaches a “9” at the beginning of the  
number to get an outside line. The  
PBX then completes the call to the  
PSTN.  
Santa Fe Office  
Area: 505  
204.16.49.74  
8-Channel  
Analog VoIP  
(MVP810)  
IP  
Network  
PBX System.  
Main Number:  
444-3200  
40 extensions  
The second Inbound Phonebook entry  
(8 digits) is for receiving calls from  
company employees in the other two  
cities. The out-of-town employee  
simply dials 3 digits. The first of the  
three digits is uniquely used at each  
site and so acts as a destination  
pattern (Boise extensions are 7xx,  
Santa Fe extensions 2xx, Flagstaff  
extensions 6xx).  
PSTN  
As the remote voip sends out the call,  
it automatically attaches all of the  
foregoing digits that would normally  
have to be dialed using the PSTN.  
Each Outbound Phonebook contains two  
pairs of entries, two entries for each  
remote site. Whenever an out-of-town  
employee dials a 12-digit number  
beginning with the listed 5-digit  
destination pattern (9+1+area code) of  
another company location, the PBX  
hands the call to the voip system. The  
local voip strips off the “9” and directs  
the call to the IP address of the remote  
voip. The remote voip receives the call  
and hands it to its PBX. The PBX then  
completes the call to the PSTN.  
The local (receiving) voip sees the  
extended pattern in its Inbound  
Phonebook and so strips off the long  
telltale pattern of digits needed for 3-  
digit calling. It must finally add back  
the last digit before handing the call  
to the PBX, which completes the call  
to a specific extension.  
Flagstaff Office  
Area: 520  
The one-digit Outbound destination  
patterns pertain to 3-digit calling  
between company employees.  
204.16.49.75  
8-Channel  
Analog VoIP  
(MVP810)  
PBX System.  
Main Number:  
777-5600  
PSTN  
30 extensions  
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Boise Office  
Boise Voip  
Boise Voip  
Outbound Phonebook  
Area: 208  
PBX System.  
Inbound Phonebook  
Main Number:  
Prefix  
Description  
Total  
Digits  
Prefix to  
Remove  
Prefix  
to Add  
IP  
Addr  
Description  
Outgoing Calls  
Prefix to  
Remove  
1208  
Destin.  
Pattern  
to Add Incoming Calls  
333-2700  
PSTN  
204.16  
.49.74  
9
2
Incoming calls 91505  
to PSTN,  
Boise Area  
Incoming calls  
to extensions  
of company’s  
PBX system  
in Boise  
12  
9
none  
Outgoing calls  
to Santa Fe  
area  
3-digit calls to  
Santa Fe  
90 extensions  
204.16.49.73  
204.16  
.49.74  
12083332  
2
3
none  
1505  
444  
3
employees  
24-Channel  
Digital VoIP  
(MVP2410)  
91520  
6
12  
3
9
none  
204.1 Outgoing calls  
6.49.7 to Flagstaff  
5
204.1 3-digit calls to  
6.49.7 Flagstaff  
5
area  
none  
1520  
777  
5
employees  
IP  
Network  
Santa Fe Office  
Area: 505  
Santa Fe Voip  
Santa Fe Voip  
Outbound Phonebook  
204.16.49.74  
Inbound Phonebook  
Prefix  
Description  
Total  
Digits  
Prefix to  
Remove  
Prefix  
to Add  
IP  
Addr  
Description  
Outgoing Calls  
Prefix to  
Remove  
1505  
Destin.  
Pattern  
8-Channel  
Analog VoIP  
(MVP810)  
to Add Incoming Calls  
9,  
2
Incoming calls 91208  
to PSTN,  
Santa Fe local  
calls  
Incoming calls  
to extensions  
of company’s  
PBX system  
in Santa Fe  
12  
9
none  
204.  
Outgoing calls  
16.49. to Boise area  
73  
150544432  
7
3
none  
1208  
333  
2
204.1 Outgoing calls  
6.49.  
73  
PBX System.  
Main Number:  
444-3200  
to extensions  
of company’s  
Boise PBX (3-  
digit dialing)  
Outgoing calls  
91520  
6
12  
3
9
none  
204.  
40 extensions  
16.49. to Flagstaff  
75  
204.  
16.49. Flagstaff  
75 employees  
area  
3-digit calls to  
none  
1520  
777  
5
PSTN  
Flagstaff Office  
Area: 520  
Flagstaff Voip  
Flagstaff Voip  
Outbound Phonebook  
204.16.49.75  
Inbound Phonebook  
8-Channel  
Analog VoIP  
(MVP810)  
Prefix  
Description  
Total  
Digits  
Prefix to  
Remove  
Prefix  
to Add  
IP  
Addr  
Description  
Outgoing Calls  
Prefix to  
Remove  
1520  
Destin.  
Pattern  
to Add Incoming Calls  
204.16  
.49.74  
9
5
Incoming calls 91505  
to PSTN,  
Flagstaff local  
12  
9
none  
Outgoing calls  
to Santa Fe  
area  
calls  
204.16  
.49.74  
15207775  
Incoming calls  
to extensions  
of company’s  
PBX system  
in Flagstaff  
2
3
none  
1505  
444  
3
3-digit calls to  
Santa Fe  
employees  
PBX System.  
Main Number:  
777-5600  
PSTN  
204.16  
.49.73  
91208  
7
12  
3
9
none  
Outgoing calls  
to Boise area  
3-digit calls to  
Boise  
30 extensions  
204.16  
.49.73  
none  
1208  
333  
2
employees  
50  
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Quick Start Instructions  
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  
1208  
Destin.  
Pattern  
to Add Incoming Calls  
to Add  
Addr  
204.  
16.49.  
74  
9,  
7
Incoming calls 91505  
to PSTN,  
12  
9
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  
120833327  
Incoming calls  
to extensions  
of company’s  
PBX system  
in Boise  
2
3
none  
1505  
444  
3
91520  
6
12  
3
9
none  
204.  
16.49. to Flagstaff  
75  
area  
3-digit calls to  
none  
1520  
777  
5
204.  
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  
1505  
Destin.  
Pattern  
to Add Incoming Calls  
to Add  
Addr  
9,  
2
Incoming calls 91208  
to PSTN,  
12  
9
none  
204.  
Outgoing calls  
16.49. to Boise area  
73  
Santa Fe local  
calls  
150544432  
Incoming calls  
to extensions  
of company’s  
PBX system  
in Santa Fe  
7
3
none  
1208  
333  
2
204.  
16.49. Boise  
73  
3-digit calls to  
employees  
(extensions  
700-790)  
91520  
6
12  
3
9
none  
204.  
Outgoing calls  
16.49. to Flagstaff  
75  
204.  
16.49. Flagstaff  
75  
area  
3-digit calls to  
none  
1520  
777  
5
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  
Remove  
1520  
Destin.  
Pattern  
to Add Incoming Calls  
204.16  
.49.74  
9,  
6
Incoming calls 91505  
to PSTN,  
12  
9
none  
Outgoing calls  
to Santa Fe  
area  
Flagstaff local  
calls  
204.16  
.49.74  
152077756  
Incoming calls  
to extensions  
of company’s  
PBX system  
in Flagstaff  
2
3
none  
1505  
444  
3
3-digit calls to  
Santa Fe  
employees  
(extensions  
200-240)  
204.16  
.49.73  
91208  
7
12  
3
9
none  
Outgoing calls  
to Boise area  
3-digit calls to  
Boise  
employees  
(extensions  
700-790)  
204.16  
.49.73  
none  
1208  
333  
2
51  
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Quick Start Instructions  
MultiVOIP User Guide  
Phonebook Worksheet  
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  
Prefix  
IP  
Description  
Outgoing Calls  
Prefix to  
Remove  
Destin.  
Pattern  
to Add Incoming Calls  
Digits  
Remove  
to Add  
Addr  
Other Details:  
Voip Location/ID:____________________________  
Inbound Phonebook Outbound Phonebook  
Prefix Description  
to Add Incoming Calls  
Total  
Digits  
Prefix to  
Remove  
Prefix  
to Add  
IP  
Addr  
Description  
Outgoing Calls  
Prefix to  
Remove  
Destin.  
Pattern  
Other Details:  
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Quick Start Instructions  
Enlarged Phonebook Worksheet  
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Quick Start Instructions  
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:  
Connections  
for digital MultiVOIPs  
(MVP-2400/2410/3010  
for analog MultiVOIPs  
(MVP-210/410/810)  
MultiVOIP to local PBX  
MultiVOIP to local phone  
station  
–OR--  
MultiVOIP to extension of  
key phone system  
MultiVOIP to command PC  
MultiVOIP to Internet  
MultiVOIP to command PC  
MultiVOIP to Internet  
2. Inbound Phonebook and Outbound Phonebook must both be set up  
with at least one entry in each. These entries must allow for  
connection between two voip units.  
3. Console messages must be enabled. (If this has not been done  
already, go, in the MultiVOIP GUI, to Configuration | Logs and  
select the “Console Messages” checkbox.  
4. You now need to free up the COM port connection (currently being  
used by the MultiVOIP program) so that the HyperTerminal program  
can use it. To do this, you can either (a) click on Connection in the  
sidebar and select “Disconnect” from the drop-down box, or (b) close  
down the MultiVOIP program altogether.  
5. Open the HyperTerminal program.  
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Quick Start Instructions  
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.  
7. Make VOIP call.  
for digital MultiVOIPs  
(MVP-2400/2410/3010  
for analog MultiVOIPs  
(MVP-210/410/810)  
Make call from an extension  
of the local PBX.  
Make call on a local phone  
line accessing PSTN directly  
or through key system  
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8. Read console messages recorded on HyperTerminal.  
Console Messages from Originating VOIP. The voip unit that  
originates the call will send back messages like that shown below.  
[00026975] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[1]  
TimeStamp : 26975  
[00027190] CAS[0] : TX : ABCD = 1, 1, 1, 1  
[00027190] PSTN: cas seizure detected on 0  
[00027440] CAS[0] : TX : ABCD = 0, 0, 0, 0  
[00033290] PSTN:call detected on 0 num=17637175662*  
[00033290] H323IF[0]:destAddr =  
TA:200.2.10.5:1720,NAME:Mounds  
View,TEL:17637175662,17637175662  
[00033290] H323IF[0]:srcAddr = NAME:New  
York,TA:200.2.9.20  
[00033440] H323IF [0]:cmCallStateProceeding  
[00033500] H323[0]: Remote Information (Q931): MultiVOIP  
- T1  
[00033565] CAS[0] : TX : ABCD = 1, 1, 1, 1  
[00033675] H323IF [0]: MasterSlaveStatus=Slave  
[00033675] H323IF[0]:FastStart Setup Not Used  
[00033690] CAS[0] : TX : ABCD = 1, 1, 1, 1  
[00033755] H323IF[0]: Coder used 'g7231'  
[00033810] PSTN:pstn call connected on 0  
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Console Messages from Terminating VOIP. The voip unit connected  
to the phone where the call is answered will send back messages like  
that shown below.  
[00170860] H323[0]: New incoming call  
[00170860] PSTNIF : Placing call on channel 0 Outbound  
digit 7175662  
[00170885] CAS[0] : TX : ABCD = 1, 1, 1, 1  
[00171095] H323IF [0]: MasterSlaveStatus=Master  
[00171105] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[7]  
TimeStamp : 171105  
[00171105] H323IF[0]: Coder used 'g7231'  
[00171110] H323IF[0]:FastStart Setup Not Used  
[00171110] H323IF[0]: Already opened the outgoing logical  
channel  
[00171110] H323IF[0]: Coder used 'g7231'  
[00171315] CAS[0] : RX : ABCD = 0, 0, 0, 0,Pstn State[9]  
TimeStamp : 171315  
[00172275] PSTN: dialing digit ended on 0  
[00172285] PSTN: pstn proceeding indication on 0  
[00172995] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[12]  
TimeStamp : 172995  
[00173660] CAS[0] : TX : ABCD = 1, 1, 1, 1  
[00173760] PSTN:pstn call connected on 0  
9. When you see the following message, end-to-end voip connectivity  
has been achieved.  
“PSTN: pstn call connected on X”  
where x is the number of the voip channel carrying the call  
10. If the HyperTerminal messages do not confirm connectivity, go to  
the Troubleshooting procedure below.  
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Troubleshooting  
If you cannot establish connectivity between two voips in the system,  
follow the steps below to determine the problem.  
1. Ping both MultiVOIP units to confirm connectivity to the network.  
2. Verify the telephone connections.  
A. For MVP2400, MVP2410, or MVP3010.  
Check cabling. Are connections well seated? To correct receptacle?  
Is the ONL LED on?  
(If on, ONL indicates that the MultiVOIP is online on the  
network.)  
Are T1/E1/PRI Parameter settings correct?  
B. For MVP210, MVP410, or MVP810.  
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 indicating phonebook problems, network problems, voice-  
coder mismatches, etc.  
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Chapter 3: Mechanical Installation  
and Cabling  
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Mechanical Installation & Cabling  
MultiVOIP User Guide  
Introduction  
The MultiVOIP models MVP210 and MVP2400 are table-top units and  
can be handled easily by one person. However, the MVP410, MVP810,  
MVP2410, and MVP3010 are somewhat heavier units. When these  
units are to be installed into a rack, two able-bodied persons should  
participate.  
Please read the safety notices before beginning installation.  
Safety Warnings  
Lithium Battery Caution  
A lithium battery on the voice/ fax channel board provides backup  
power for the timekeeping capability. The battery has an estimated life  
expectancy of ten years.  
When the battery starts to weaken, the date and time may be incorrect.  
If the battery fails, the board must be sent back to Multi-Tech Systems  
for battery replacement.  
Warning: There is danger of explosion if the battery is incorrectly  
replaced.  
Safety Warnings Telecom  
1. Never install telephone wiring during a lightning storm.  
2. Never install a telephone jack in wet locations unless the jack is  
specifically designed for wet locations.  
3. This product is to be used with UL and cUL listed computers.  
4. Never touch uninsulated telephone wires or terminals unless the  
telephone line has been disconnected at the network interface.  
5. Use caution when installing or modifying telephone lines.  
6. Avoid using a telephone (other than a cordless type) during an  
electrical storm. There may be a remote 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 26 AWG or larger  
telecommunication line cord.  
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Unpacking Your MultiVOIP  
When unpacking your MultiVOIP, check to see that all of the items  
shown are included in the box. For the various MultiVOIP models, the  
contents of the box will be different. Study the particular illustration  
below that is appropriate to the model you have purchased. If any box  
contents are missing, contact MultiTech Tech Support at 1-800-972-2439.  
Unpacking the MVP2410/3010  
Figure 3-1: Unpacking the MVP2410/3010  
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MultiVOIP User Guide  
Unpacking the MVP2400  
200  
Voice/Fax over IP Networks  
Quick Start  
Guide  
Figure 3-2: Unpacking the MVP2400  
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Unpacking the MVP410/810  
Voice/Fax over IP Networks  
Quick Start  
Guide  
Voice/Fax  
RCV SG  
5
1
Voice/Fax  
RCV SG  
6
2
Voice/Fax  
7
Voice/Fax  
RCV SG  
8
4
X
M
T
X
RSG  
RSG  
X
M
T
X
RSG  
RSG  
X
M
T
RCV XSG RSG  
X
M
T
X
RSG  
RSG  
Ethernet  
COL  
Power  
Boot  
Voice/Fax  
RCV SG  
Voice/Fax  
RCV SG  
Voice/Fax  
3
Voice/Fax  
RCV SG  
RCV  
X
M
T
L
N
K
XM  
T
X
XM  
T
X
XM  
T
RCV XSG RSG  
XM  
T
X
Figure 3-3: Unpacking the MVP410/810  
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MultiVOIP User Guide  
Unpacking the MVP210  
200  
Voice/Fax over IP Networks  
Quick Start  
Guide  
Figure 3-4: Unpacking the MVP210  
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Mechanical Installation & Cabling  
Rack Mounting Instructions for  
MVP2410/3010 & MVP410/810  
The MultiVOIPs can be mounted in an industry-standard EIA 19-inch  
rack enclosure, as shown in Figure 2-5.  
Figure 3-5: Rack-Mounting (MVP2410/3010 or MVP410/810)  
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Safety Recommendations for Rack Installations  
Ensure proper installation of the unit in a closed or multi-unit enclosure  
by following the recommended installation as defined by the enclosure  
manufacturer. Do not place the unit directly on top of other equipment  
or place other equipment directly on top of the unit. If installing the  
unit in a closed or multi-unit enclosure, ensure adequate airflow within  
the rack so that the maximum recommended ambient temperature is  
not exceeded. Ensure that the unit is properly connected to earth  
ground by verifying that it is reliably grounded when mounted within  
a rack. If a power strip is used, ensure that the power strip provides  
adequate grounding of the attached apparatus.  
When mounting the equipment in the rack, make sure mechanical  
loading is even to avoid a hazardous condition, such as loading heavy  
equipment in rack unevenly. The rack used should safely support the  
combined weight of all the equipment it supports.  
Ensure that the mains supply circuit is capable of handling the load of  
the equipment. See the power label on the equipment for load  
requirements (full specifications for MultiVOIP models are presented in  
chapter 1 of this manual).  
Maximum ambient temperature for the unit is 40 degrees Celsius (104  
degrees Fahrenheit). This equipment should only be installed by  
properly qualified service personnel. Only connect like circuits. In other  
words, connect SELV (Secondary Extra Low Voltage) circuits to SELV  
circuits and TN (Telecommunications Network) circuits to TN circuits.  
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19-Inch Rack Enclosure Mounting Procedure  
Attaching the MultiVOIP to a rack-rail of an EIA 19-inch rack enclosure  
will certainly require two persons. Essentially, the technicians must  
attach the brackets to the MultiVOIP chassis with the screws provided,  
as shown in Figure 3-6, and then secure unit to rack rails by the  
brackets, as shown in Figure 3-7. Because equipment racks vary, screw  
for rack-rail mounting are not provided. Follow the instructions of the  
rack manufacturer and use screws that fit.  
1. Position the right rack mounting bracket on the MultiVOIP  
using the two vertical mounting screw holes.  
2. Secure the bracket to the MultiVOIP using the two screws  
provided.  
3. Position the left rack mounting bracket on the MultiVOIP  
using the two vertical mounting screw holes.  
4. Secure the bracket to the MultiVOIP using the two screws  
provided.  
5. Remove feet (4) from the MultiVOIP unit.  
6. Mount the MultiVOIP in the rack enclosure per the rack  
manufacture’s mounting procedure.  
x
x
Figure 3-6: Bracket Attachment for Rack Mounting  
(MVP2410/3010 or MVP410/810)  
Figure 3-7: Attaching MultiVOIP to Rack Rail  
(MVP2410/3010 or MVP410/810)  
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MultiVOIP User Guide  
Cabling  
Cabling Procedure for MVP2410/3010  
Cabling your MultiVOIP entails making the proper connections for  
power, command port, phone system (T1/ E1 line connected to PBX or  
telco office), and Ethernet network. Figure 3-8 shows the back panel  
connectors and the associated cable connections. The following  
procedure details the steps necessary for cabling your MultiVOIP.  
1. Connect the power cord to a live AC outlet, then connect it to the  
MultiVOIP’s power receptacle shown at top right in Figure 3-8.  
DIGITAL VOICE  
ETHERNET COMMAND  
TRUNK  
10BASET  
RS232  
DIGITAL VOICE  
ETHERNET COMMAND  
T1  
ꢀ
Hub  
Command Port Connection  
PBX  
PSTN  
Network Connection  
Telephony Connection  
Figure 3-8. Cabling for MVP2410/3010  
2. Connect the MultiVOIP to the PC (the computer that will hold the  
MultiVOIP software) using the RJ-45 to DB9 (female) cable provided  
with your unit. Plug the RJ-45 end of the cable into the Command  
port of the MultiVOIP and connect the other end (the DB9 connector)  
to the PC serial port you are using (typically COM1 or COM2). See  
Figure 3-8.  
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3. Connect a network cable to the Ethernet connector on the back of the  
MultiVOIP. Connect the other end of the cable to your network.  
4. Turn on power to the MultiVOIP by setting the power switch on the  
right side panel to the ON position. Wait for the Boot LED on the  
MultiVOIP to go off before proceeding. This may take a couple of  
minutes.  
Proceed to Chapter 4 “Software Installation.”  
Cabling Procedure for MVP2400  
Cabling your MultiVOIP entails making the proper connections for  
power, command port, phone system (T1 line connected to PBX or telco  
office), and Ethernet network. Figure 3-9 shows the back panel  
connectors and the associated cable connections. The following  
procedure details the steps necessary for cabling your MultiVOIP.  
1. Connect the power supply to a live AC outlet, then connect it to the  
MultiVOIP as shown in Figure 3-9.  
DIGITAL VOICE  
TRUNK  
COMMAND  
RS232  
ETHERNET  
10/100  
1
0
Power Connection  
POWER  
T1  
PBX  
Command Port Connection  
PSTN  
ꢀ
Telephony Connection  
Network Connection  
Hub  
Figure 3-9: Cabling for MVP2400  
2. Connect the MultiVOIP to the PC (the computer that will hold the  
MultiVOIP software) using the RJ-45 to DB9 (female) cable provided  
with your unit. Plug the RJ-45 end of the cable into the Command  
port of the MultiVOIP and connect the other end (the DB9 connector)  
to the PC serial port you are using (typically COM1 or COM2). See  
Figure 3-9.  
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3. Connect a network cable to the Ethernet connector on the back of the  
MultiVOIP. Connect the other end of the cable to your network.  
4. Turn on power to the MultiVOIP by setting the power switch on the  
right side panel to the ON position. Wait for the Boot LED on the  
MultiVOIP to go off before proceeding. This may take a couple of  
minutes.  
Proceed to Chapter 4 “Software Installation.”  
Cabling Procedure for MVP410/810  
Cabling involves connecting the MultiVOIP to your LAN and telephone  
equipment.  
1. Connect the power cord supplied with your MultiVOIP to a live AC  
outlet and to the power connector on the back of the MultiVOIP as  
shown at top right in Figure 3-10.  
ETHERNET  
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  
Voice/Fax Channel Connections  
Channels 1-4 Bottom MVP410/810  
Channels 5-8 Top MVP810 Only  
E&M FXS/FXO  
Ethernet Connection  
FXS  
E&M  
FXO  
Command Port Connection  
PSTN  
Figure 3-10: Cabling for MVP410/810  
2. Connect the MultiVOIP to a PC by using a DB-25 (male) to DB-9  
(female) cable. Plug the DB-25 end of the cable into the Command  
port of the MultiVOIP and the other end into the PC serial port. See  
Figure 3-10.  
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3. Connect a network cable to the ETHERNET 10BASET connector on  
the back of the MultiVOIP. Connect the other end of the cable to your  
network.  
4. If you are connecting a station device such as an analog telephone, a  
fax machine, or a Key Telephone System (KTS) (FXS interface), or a  
PBX extension (FXO interface) to your MultiVOIP, connect one end of  
an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the back  
of the MultiVOIP and the other end to the device or phone jack. You  
will define the interface in the Interface dialog box in the software  
when you configure the unit.  
If you are connecting an E&M trunk from a telephone switch to your  
MultiVOIP, connect one end of an RJ-45 phone cord to the Channel 1  
E&M connector on the back of the MultiVOIP and the other end to  
the trunk. 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 support  
by the telephone switch. See Appendix B for an E&M cabling pinout.  
5. Repeat the above step to connect the remaining telephone equipment  
to each Channel on your MultiVOIP.  
6. Ensure that the unit is properly connected to earth ground by  
verifying that it is reliably grounded when mounted within a rack.  
This can be accomplished by connecting a grounding wire between  
the chassis and a metallic object that will provide an electrical  
ground.  
7. Turn on power to the MultiVOIP by placing the ON/ OFF switch on  
the back panel to the ON position. Wait for the Boot LED on the  
MultiVOIP to go off before proceeding. This may take a few minutes.  
Proceed to Chapter 4 to load the MultiVOIP software.  
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Mechanical Installation & Cabling  
MultiVOIP User Guide  
Cabling Procedure for MVP210  
Cabling involves connecting the MultiVOIP to your LAN and telephone  
equipment.  
1. Connect the power cord supplied with your MultiVOIP to the power  
connector on the back of the MultiVOIP and a live AC outlet as  
shown in Figure 3-11.  
CH1  
CH2  
ETHERNET  
FXS/FXO  
E&MFXS/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  
Figure 3-11: Cabling for MVP210  
2. Connect the MultiVOIP to a PC by using an 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.  
3. Connect a network cable to the ETHERNET 10/100 connector on the  
back of the MultiVOIP. Connect the other end of the cable to your  
network.  
4. If you are connecting a station device such as an analog telephone, a  
fax machine, or a Key Telephone System (KTS) (FXS interface), or a  
PBX extension (FXO interface) to your MultiVOIP, connect one end of  
an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the  
back MultiVOIP and the other end to the device or phone jack. You  
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Mechanical Installation & Cabling  
will define the interface in the Interface dialog box in the software  
when you configure the unit.  
If you are connecting an E&M trunk from a telephone switch to your  
MultiVOIP, connect one end of an RJ-45 phone cord to the Channel 1  
E&M connector on the back of the MultiVOIP and the other end to  
the trunk. 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 support  
by the telephone switch. See Appendix B for an E&M cabling pinout.  
5. Repeat the above step to connect the remaining telephone equipment  
to the second channel on your MultiVOIP.  
6. Ensure that the unit is properly connected to earth ground by  
verifying that it is reliably grounded when mounted within a rack.  
This can be accomplished by connecting a grounding wire between  
the chassis and a metallic object that will provide an electrical  
ground.  
7. Turn on power to the MultiVOIP by placing the ON/ OFF switch on  
the back panel to the ON position. Wait for the BOOT LED on the  
MultiVOIP to go off before proceeding. This may take a few minutes.  
Proceed to Chapter 4 to load the MultiVOIP software.  
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Chapter 4: Software Installation  
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MultiVOIP User Guide  
Software Installation  
Introduction  
Configuring software for your MultiVOIP entails three tasks:  
(1) loading the software onto the PC (this is “Software Installation and  
is discussed in this chapter),  
(2) setting values for telephony and IP parameters that will fit your  
system (this is “Technical Configuration” and it is discussed in Chapter  
5 for T1/ E1 MultiVOIP units and in Chapter 6 for analog MultiVOIP  
units), and  
(3) establishing “phonebooks” that contain the various dialing patterns  
for VOIP calls made to different locations (this is “Phonebook  
Configuration” and it is discussed in Chapters 7, 8, and 9 for T1, E1,  
and analog MultiVOIP units respectively).  
Loading MultiVOIP Software onto the PC  
The software loading procedure does not present every screen or option  
in the loading process. It is assumed that someone with a thorough  
knowledge of Windows and the software loading process is performing  
the installation.  
The MultiVOIP software and User Guide are contained on the  
MultiVOIP product CD. Because the CD is auto-detectable, it will start  
up automatically when you insert it into your CD-ROM drive. When  
you have finished loading your MultiVOIP software, you can view and  
print the User Guide by clicking on the View Manuals icon.  
1. Be sure that your MultiVOIP has been properly cabled and that the  
power is turned on.  
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MultiVOIP User Guide  
2. Insert the MultiVOIP CD into your CD-ROM drive. The CD should  
start automatically. It may take 10 to 20 seconds for the Multi-Tech  
CD installation window to display.  
If the Multi-Tech Installation CD window does not display  
automatically, click My Computer, then right click the CD ROM  
drive icon, click Open, and then click the Autorun icon.  
3. When the Multi-Tech Installation CD dialog box appears, click the  
Install Software icon.  
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4. A ‘welcome’ screen appears.  
Press Enter or click Next to continue.  
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MultiVOIP User Guide  
5. Follow the on-screen instructions to install your MultiVOIP software.  
The first screen asks you to choose the folder location of the files of  
the MultiVOIP software.  
Choose a location and click Next.  
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6. At the next screen, you must select a program folder location for the  
MultiVOIP software program icon.  
Click Next. Transient progress screens will appear while files are  
being copied.  
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7. On the next screen you can select the COM port that the command  
PC will use when communicating with the MultiVoip unit. After  
software installation, the COM port can be re-set in the MultiVoip  
Software (from the sidebar menu, select Connection | Settings to  
access the COM Port Setup screen or use the keyboard shortcut Ctrl  
+ G).  
NOTE: If the COM port setting made  
here conflicts with the actual COM  
port resources available in the  
command PC, this error message will  
appear when the MultiVOIP program  
is launched. If this occurs, you must  
reset the COM port.  
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8. A completion screen will appear.  
Click Finish.  
9. When setup of the MultiVOIP software is complete, you will be  
prompted to run the MultiVOIP software to configure the VOIP.  
Software installation is complete at this point. You may proceed with  
Technical Configuration now or not, at your convenience.  
Technical Configuration instructions are in the next two chapters of  
this manual: Chapter 5 for T1/ E1 MultiVOIP units and Chapter 6 for  
Analog MultiVOIP units.  
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Un-Installing the MultiVOIP Configuration  
Software  
1. To un-install the MultiVOIP configuration software, go to Start |  
Programs and locate the entry for the MultiVOIP program. Select  
Uninstall.  
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2. Two confirmation screens will appear. Click Yes and OK when you  
are certain you want to continue with the uninstallation process.  
3. A special warning message similar to that shown below may appear  
concerning the MultiVOIP software’s “.bin” file. Click Yes.  
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4. A completion screen will appear.  
Click Finish.  
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Chapter 5: Technical Configuration  
for Digital T1/E1 MultiVOIPs  
(MVP2400, MVP2410, MVP3010)  
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Technical Configuration (Digital Voips)  
Configuring the Digital T1/E1 MultiVOIP  
There are two ways in which the MultiVOIP must be configured before  
operation: technical configuration and phonebook configuration.  
Technical Configuration. First, the MultiVOIP must be configured to  
operate with technical parameter settings that will match the  
equipment with which it interfaces. There are seven types of technical  
parameters that must be set.  
These technical parameters pertain to  
(1) its operation in an IP network,  
(2) its operation with T1/ E1 telephony equipment,  
(3) its transmission of voice and fax messages,  
(4) its interaction with SNMP (Simple Network Management Protocol)  
network management software (MultiVoipManager),  
(5) certain telephony attributes that are common to particular nations or  
regions,  
(6) its operation with a mail server on the same IP network (per SMTP  
parameters) such that log reports about VoIP telephone call traffic can  
be sent to the administrator by email,  
(7) implementing some common premium telephony features (Call  
Transfer, Call Hold, Call Waiting, Call ID – “Supplementary Services”),  
and  
(8) selecting the method by which log reports will be made accessible.  
The process of specifying values for the various parameters in these  
seven categories is what we call “technical configuration” and it is  
described in this chapter.  
Phonebook Configuration. The second type of configuration 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 it is described in Chapter 7: T1  
Phonebook Configuration and Chapter 8: E1 Phonebook Configuration of this  
manual. Chapter 2, the Quick Start Instructions, presents additional  
examples relevant to the T1/ E1 voips.  
Local/Remote Configuration. The MultiVOIP must be configured  
locally at first (to establish an IP address for the MultiVOIP unit). But  
changes to this initial configuration can be done either locally or  
remotely.  
Local configuration is done through a connection between the  
“Command” port of the MultiVOIP and the COM port of the computer;  
the MultiVOIP configuration program is used.  
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Technical Configuration (Digital Voips)  
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).  
Functional Equivalence. 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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MultiVOIP User Guide  
Technical Configuration (Digital)  
Local Configuration  
This manual describes local configuration only. For information on  
remote configuration and management, see the MultiVoipManager  
documentation.  
Pre-Requisites  
To complete the configuration of the  
MultiVOIP unit, you must know several  
things about the overall system.  
Before configuring your MultiVOIP Gateway unit, you must know the  
values for several IP and T1/ E1 parameters that describe the IP  
network system and telephony system (PBX or telco central office  
equipment) with which the digital MultiVOIP will interact. If you plan  
to receive log reports on phone traffic by email (SMTP), you must  
arrange to have an email address assigned to the VOIP unit on the  
email server on your IP network.  
IP Parameters  
The following parameters must be known about the network (LAN,  
WAN, Internet, etc.) to which the MultiVOIP will connect:  
Ask your computer network  
administrator.  
Info needed to operate:  
all MultiVOIP models.  
G
IP Network Parameters:  
Record for each VOIP Site  
in System  
ꢀ
• IP Address  
• IP Mask  
• Gateway  
• Domain Name Server (DNS) Info  
(not implemented; for future use)  
Write down the values for these IP parameters. You will need to enter  
these values in the “IP Parameters” screen in the Configuration section  
of the MultiVOIP software. You must have this IP information about  
every VOIP in the system.  
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Technical Configuration (Digital Voips)  
T1 Telephony Parameters (for MVP2400 & MVP2410)  
The following parameters must be known about the PBX or telco  
central office equipment to which the T1 MultiVOIP will connect:  
T1 Phone Parameters  
Info needed to operate:  
MVP2400  
G
Ask phone company or  
PBX maintainer.  
MVP2410  
T1 Telephony Parameters:  
Record for this VOIP Site  
ꢀ
• Which frame format is used? ESF___ or D4___  
• Which CAS or PRI protocol is used? ______________  
• Clocking: Does the PBX or telco switch use  
internal or external clocking? _________________  
Note that the setting used in the voip unit will be the  
opposite of the setting used by the telco/PBX.  
• Which line coding is used? AMI___ or B8ZS___  
Write down the values for these T1 parameters. You will need to enter  
these values in the “T1/ E1 Parameters” screen in the Configuration  
section of the MultiVOIP software.  
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Technical Configuration (Digital)  
E1 Telephony Parameters (for MVP3010)  
The following parameters must be known about the PBX or telco  
central office equipment to which the E1 MultiVOIP will connect:  
E1 Phone Parameters  
Info needed to operate:  
G
MVP3010  
Ask phone company or  
PBX maintainer.  
E1 Telephony Parameters:  
Record for this VOIP Site  
ꢀ
• Which frame format is used? Double Frame_____  
MultiFrame w/ CRC4_____  
MultiFrame w/ CRC4 modified_____  
• Which CAS or PRI protocol is used? ______________  
• Clocking: Does the PBX or telco switch use  
internal or external clocking? _________________  
Note that the setting used in the voip unit will be the  
opposite of the setting used by the telco/PBX.  
• Which line coding is used? AMI___ or HDB3___  
• Pulse shape level?: (most commonly 0 to 40 meters)  
Write down the values for these E1 parameters. You will need to enter  
these values in the “T1/ E1 Parameters” screen in the Configuration  
section of the MultiVOIP software.  
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SMTP Parameters (for email call log reporting)  
required if log reports of  
Optional  
VOIP call traffic  
are to be sent by email  
SMTP Parameters  
Preparation Task:  
T
o
e
:
I . T  
.
D
e
p
a
r
t
m
e
n
t
t
Ask Mail Server  
r
:
e
m
a
i l  
a
c
c
o
u
n
f
o
r
V
O
I P  
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. .  
voip-unit2@biggytech.com  
Get the IP address of the  
mail server computer, as  
well.  
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Technical Configuration (Digital)  
Local Configuration Procedure (Summary)  
After the MultiVOIP configuration software has been installed in the  
‘Command’ PC (which is connected to the MultiVOIP unit), several  
steps must be taken to configure the MultiVOIP to function in its  
specific setting. Although the summary below includes all of these  
steps, some are optional.  
1. Check Power and Cabling.  
2. Start MultiVOIP Configuration Program.  
3. Confirm Connection.  
4. Solve Common Connection Problems.  
A. Fixing a COM Port Problem.  
B. Fixing a Cabling Problem.  
5. Familiarize yourself with configuration parameter screens and how  
to access them.  
6. Set IP Parameters.  
7. Enable web browser GUI (optional).  
8. Set Voice/ Fax Parameters.  
9. Set T1/ E1 Parameters.  
10. Set ISDN Parameters (if applicable).  
11. Set SNMP Parameters (applicable if MultiVoipManager remote  
management software is used).  
12. Set Regional Parameters (Phone Signaling Tones and Cadences).  
13. Set Custom Tones and Cadences (optional).  
14. Set SMTP Parameters (applicable if Log Reports are via Email).  
15. Set Log Reporting Method (GUI, locally in MultiVOIP  
Configuration program; SNMP, remotely in MultiVoipManager  
program; or SMTP, via email).  
16. Set Supplementary Services Parameters. The Supplementary  
Services screen allows voip deployment of features that are normally  
found in PBX or PSTN systems (e.g., call transfer and call waiting).  
17. Set Baud Rate (of COM port connection to ‘Command’ PC).  
18. View System Information and set updating interval (optional).  
19. Save the MultiVOIP configuration.  
20. Create a User Default Configuration (optional).  
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Technical Configuration (Digital Voips)  
Local Configuration Procedure (Detailed)  
You can begin the configuration process as a continuation of the  
MultiVOIP software installation. You can establish your configuration  
or modify it at any time by launching the MultiVOIP program from the  
Windows Start menu.  
1. Check Power and Cabling. Be sure the MultiVOIP is turned on and  
connected to the computer via the MultiVOIP’s Command Port (DB9  
connector at computer’s COM port; RJ45 connector at MultiVOIP).  
You must allow the MultiVOIP to finish booting before you launch  
the MultiVOIP Configuration Program. The RED boot LED turns  
itself off when the booting process is completed.  
2. Start MultiVOIP Configuration Program. Launch the MultiVOIP  
program from the Windows Start menu (from the folder location  
determined during installation).  
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3. Confirm Connection. If the MultiVOIP is set for an available COM  
port and is correctly cabled to the PC, the MultiVOIP main screen will  
appear. (If the main screen appears grayed out and seems inaccessible,  
go to step 4.)  
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In the lower left corner of the screen, the connection status of the  
MultiVOIP will be displayed. The messages in the lower left corner  
will change as detection occurs. The message “MultiVOIP Found”  
confirms that the MultiVOIP is in contact with the MultiVOIP  
configuration program. Skip to step 5.  
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4. Solving Common Connection Problems.  
A. Fixing a COM Port Problem. If the MultiVOIP main screen appears  
but is grayed out and seems inaccessible, the COM port that was  
specified for its communication with the PC is unavailable and must  
be changed. An error message will appear.  
To change the COM port setting, use the COM Port Setup dialog box,  
which is accessible via the keyboard shortcut Ctrl + G or by going to  
the Connection pull-down menu and choosing “Settings.” In the  
“Select Port” field, select a COM port that is available on the PC. (If  
no COM ports are currently available, re-allocate COM port resources  
in the computer’s MS Windows operating system to make one  
available.)  
Ctrl + G  
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4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by  
the computer, two error messages will appear (saying “Multi-VOIP  
Not Found” and “Phone Database Not Read”).  
In this case, the MultiVOIP is simply disconnected from the network.  
For instructions on MultiVOIP cable connections, see the “Cabling”  
section of Chapter 3.  
5. Configuration Parameter Groups: Getting Familiar, Learning  
About Access. The first part of configuration concerns IP parameters,  
Voice/ FAX parameters, T1/ E1 parameters, SNMP parameters,  
Regional parameters, SMTP parameters, Supplementary Services  
parameters, Logs, and System Information. In the MultiVOIP software,  
these seven types of parameters are grouped together under  
“Configuration” and each has its own dialog box for entering values.  
Generally, you can reach the dialog box for these parameter groups in  
one of four ways: pulldown menu, toolbar icon, keyboard shortcut, or  
sidebar..  
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6. Set IP Parameters. This dialog box can be reached by pulldown  
menu, toolbar icon, keyboard shortcut, or sidebar.  
Accessing “IP Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt + I  
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Technical Configuration (Digital Voips)  
In each field, enter the values that fit your particular network.  
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Technical Configuration (Digital)  
The IP Parameters fields are described in the table below.  
IP Parameter Definitions  
Field Name  
Values  
Description  
Enable  
Diffserv  
Y/ N  
Diffserv is used for QoS  
(quality of service).  
When enabled, we  
configure the TOS (Type  
of Service) bits in the IP  
header so routers  
supporting Diffserv can  
give priority to the  
VOIP’s IP packets.  
Disabled by default.  
Frame Type  
Type II, SNAP  
Must be set to match  
network’s frame type.  
Default is Type II.  
IP Address  
IP Mask  
4-places, 0-255  
4-places, 0-255  
The unique LAN IP address  
assigned to the MultiVOIP.  
Subnetwork address that  
allows for sharing of IP  
addresses within a LAN.  
Gateway  
4-places, 0-255  
The IP address of the device  
that connects your  
MultiVOIP to the Internet.  
Enable DNS  
Y/ N  
Enables Domain Name  
Space/ System function  
where computer names  
are resolved using a  
worldwide distributed  
database.  
(feature not yet  
implemented; for  
future use)  
DNS Server IP  
Address  
4-places, 0-255.  
(feature not yet  
implemented; for  
future use)  
IP address of specific  
DNS server to be used to  
resolve Internet  
computer names.  
FTP Server  
Enable  
Y/ N  
See “FTP Server  
MultiVOIP unit has an  
FTP Server function so  
File Transfers” in that firmware and other  
Operation &  
Maintenance  
chapter.  
important operating  
software files can be  
transferred to the voip  
via the network.  
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7. Enable Web Browser GUI (Optional). After an IP address for the  
MultiVOIP unit has been established, you can choose to do any further  
configuration of the unit (a) by using the MultiVOIP web browser GUI,  
or (b) by continuing to use the MultiVOIP Windows GUI. If you want  
to do configuration work using the web browser GUI, you must first  
enable it. To do so, follow the steps below.  
A. Set IP address of MultiVOIP unit using the MultiVOIP  
Configuration program (the Windows GUI).  
B. Save Setup in Windows GUI.  
C. Close Windows GUI.  
D. Install Java program from MultiVOIP product CD (required on  
first use only).  
E. Open web browser.  
F. Browse to IP address of MultiVOIP unit.  
G. If username and password have been established, enter them  
when when prompted.  
H. Use web browser GUI to configure or operate MultiVOIP unit.  
The configuration screens in the web browser GUI will have the same  
content as their counterparts in the Windows GUI; only the graphic  
presentation will be different.  
For more details on enabling the MultiVOIP web GUI, see the “Web  
Browser Interface” section of the Operation & Maintenance chapter of  
this manual.  
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8. Set Voice/FAX Parameters. This dialog box can be reached by  
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.  
Accessing “Voice/FAX Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + H  
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MultiVOIP User Guide  
Technical Configuration (Digital Voips)  
In each field, enter the values that fit your particular network.  
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MultiVOIP User Guide  
Technical Configuration (Digital)  
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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MultiVOIP User Guide  
Technical Configuration (Digital Voips)  
The Voice/FAX Parameters fields are described in the tables below.  
Voice/Fax Parameter Definitions  
Field Name Values  
Description  
Default  
--  
When this button is clicked, all  
Voice/ FAX parameters are set to their  
default values.  
Select  
Channel  
1-24 (T1)  
1-30 (E1)  
Channel to be configured is selected  
here.  
Copy  
Channel  
--  
Copies the Voice/ FAX attributes of  
one channel to another channel.  
Attributes can be copied to multiple  
channels or all channels at once.  
Voice Gain  
Input Gain  
--  
Signal amplification (or attenuation)  
in dB.  
+31dB  
to  
–31dB  
Modifies audio level entering voice  
channel before it is sent over the  
network to the remote VOIP. The  
default & recommended value is 0 dB.  
Output Gain +31dB  
Modifies audio level being output to  
the device attached to the voice  
channel. The default and  
to  
–31dB  
recommended value is 0 dB.  
DTMF Parameters  
DTMF Gain  
--  
The DTMF Gain (Dual Tone Multi-  
Frequency) controls the volume level  
of the digital tones sent out for Touch-  
Tone dialing.  
DTMF Gain, +3dB to  
High Tones  
Default value: -4 dB. Not to be  
-31dB & changed except under supervision of  
“mute”  
DTMF Gain, +3dB to  
Low Tones  
MultiTech’s Technical Support.  
Default value: -7 dB. Not to be  
-31dB & changed except under supervision of  
“mute” MultiTech’s Technical Support.  
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Field Name  
Technical Configuration (Digital)  
Voice/Fax Parameter Definitions (cont’d)  
Values  
DTMF Parameters  
Description  
Duration  
(DTMF)  
60 – 3000  
ms  
When DTMF: Out of Band is  
selected, this setting determines  
how long each DTMF digit ‘sounds’  
or is held. Default = 100 ms.  
DTMF  
In/ Out of  
Band  
Out of  
Band, or  
Inband  
When DTMF Out of Band is selected  
(checked), the MultiVOIP detects DTMF  
tones at its input and regenerates them  
at its output. When DTMF Inband is  
selected, the DTMF digits are passed  
through the MultiVOIP unit as they are  
received.  
FAX Parameters  
Fax Enable  
Y/ N  
Enables or disables fax capability  
for a particular channel.  
Max Baud  
Rate  
(Fax, bps)  
2400, 4800,  
7200, 9600,  
12000, 14400  
Set to match baud rate of fax machine  
connected to channel (see Fax machine’s  
user manual). Default = 14400 bps.  
Fax Volume -18.5 dB  
Controls output level of fax tones. To  
be changed only under the direction of  
Multi-Tech’s Technical Support.  
Default =  
-9.5 dB  
to –3.5 dB  
Jitter Value  
(Fax)  
Default =  
400 ms  
Defines the inter-arrival packet  
deviation (in milliseconds) for the  
fax transmission. A higher value  
will increase the delay, allowing a  
higher percentage of packets to be  
reassembled. A lower value will  
decrease the delay allowing fewer  
packets to be reassembled.  
Mode (Fax)  
FRF 11;  
T.38  
(T.38 not  
currently  
sup-  
FRF11 is frame-relay FAX standard  
using these coders: G.711, G.728,  
G.729, 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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MultiVOIP User Guide  
Technical Configuration (Digital Voips)  
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  
G.723.1 @ digital voice is compressed to 6.3K,  
5.3 kbps,  
6.3 kbps;  
G.729,  
8kbps;  
Net Coder  
@
allowing several simultaneous  
conversations over the same  
bandwidth that would otherwise  
carry only one.  
To make selections from the  
6.4, 7.2, 8,  
8.8, 9.6  
kbps  
Selected Coder drop-down list, the  
Manual option must be enabled.  
Max  
bandwidth  
(coder)  
11 – 128  
kbps  
This drop-down list enables you to  
select the maximum bandwidth  
allowed for this channel. The Max  
Bandwidth drop-down list is  
enabled only if the Coder is set to  
Automatic.  
If coder selected automatically, then  
enter a value for maximum  
bandwidth, as directed by VOIP  
administrator.  
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Technical Configuration (Digital)  
Voice/Fax Parameter Definitions (cont’d)  
Field Name Values  
Advanced Features  
Description  
Silence  
Compression  
Y/ N  
Determines whether silence  
compression is enabled (checked) for  
this voice channel.  
With Silence Compression enabled, the  
MultiVOIP will not transmit voice  
packets when silence is detected,  
thereby reducing the amount of  
network bandwidth that is being used  
by the voice channel.  
Default = off.  
Echo  
Cancellation  
Y/ N  
Y/ N  
Determines whether echo cancellation is  
enabled (checked) for this voice  
channel.  
Echo Cancellation removes echo and  
improves sound quality. Default = on.  
Forward  
Error  
Correction  
Determines whether forward error  
correction is enabled (checked) for this  
voice channel.  
Forward Error Correction enables  
some of the voice packets that were  
corrupted or lost to be recovered. FEC  
adds an additional 50% overhead to the  
total network bandwidth consumed by  
the voice channel.  
Default = Off  
Auto Call  
Enable  
Y/ N  
The Auto Call 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.  
Phone No.  
(Auto Call)  
--  
Phone number used for Auto Call  
function. A corresponding phone  
number must be listed in the Outbound  
Phonebook.  
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MultiVOIP User Guide  
Technical Configuration (Digital Voips)  
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  
Jitter Buffer  
between configurable minimum  
and maximum values. An  
Optimization Factor adjustment  
controls how quickly the length of  
the Jitter Buffer is increased when  
jitter increases on the network. The  
length of the jitter buffer directly  
effects the voice delay between  
MultiVOIP gateways.  
The default minimum dynamic  
jitter buffer of 60 milliseconds is the  
minimum delay that would be  
acceptable over a low jitter network.  
The default maximum dynamic  
jitter buffer of 300 milliseconds is  
the maximum delay tolerable over a  
high jitter network.  
Minimum  
Jitter Value  
60 to 400  
ms  
The default minimum dynamic  
jitter buffer of 60 milliseconds is the  
minimum delay that would be  
acceptable over a low jitter network.  
Default = 60 msec  
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MultiVOIP User Guide  
Technical Configuration (Digital)  
Voice/Fax Parameter Definitions (cont’d)  
Field Name Values  
Dynamic Jitter  
Maximum 60 to 400  
Description  
The default maximum dynamic  
jitter buffer of 300 milliseconds is  
the maximum delay tolerable over a  
high jitter network.  
Jitter Value  
ms  
Default = 300 msec  
Optimizat-  
ion Factor  
0 to 12  
The Optimization Factor  
determines how quickly the length  
of the Dynamic Jitter Buffer is  
changed based on actual jitter  
encountered on the network.  
Selecting the minimum value of 0  
means low voice delay is desired,  
but increases the possibility of jitter-  
induced voice quality problems.  
Selecting the maximum value of 12  
means highest voice quality under  
jitter conditions is desired at the  
cost of increased voice delay.  
Default = 7.  
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Technical Configuration (Digital Voips)  
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 150 milliseconds. A higher  
value means voice transmission will  
be more accepting of jitter. A lower  
value is less tolerant of jitter.  
Inactive by default. When active,  
default = 150 ms. However, value  
must equal or exceed Dynamic  
Minimum Jitter Value.  
Call  
Duration  
1-65535  
seconds  
Call Duration defines the  
maximum length of time (in  
seconds) that a call remains  
connected before the call is  
automatically disconnected.  
Inactive by default.  
When active, default = 180 sec.  
This may be too short for most  
configurations, requiring upward  
adjustment.  
Consecutive 1-65535  
Packets Lost  
Consecutive Packets Lost defines  
the number of consecutive packets  
that are lost after which the call is  
automatically disconnected.  
Inactive by default.  
When active, default = 30  
Network  
Discon-  
nection  
1 to 65535  
seconds;  
Default =  
300 sec.  
Specifies how long to wait before  
disconnecting the call when IP  
network connectivity with the  
remote site has been lost.  
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MultiVOIP User Guide  
Technical Configuration (Digital)  
9. Set T1/E1/ISDN Parameters. This dialog box can be reached by  
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.  
Accessing “T1/E1/ISDN Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + T  
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MultiVOIP User Guide  
Technical Configuration (Digital Voips)  
In each field, enter the values that fit your particular network.  
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Technical Configuration (Digital)  
T1 Parameters. The parameters applicable to T1 and their values are  
shown in the figure below. These T1 Parameter fields are described in  
the tables that follow.  
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Technical Configuration (Digital Voips)  
T1 Parameter Definitions  
Field Name  
Values  
Description  
T1/ E1/ ISDN  
T1  
North American standard.  
Long-Haul  
Mode  
Y/ N  
In Long-Haul Mode, the  
MultiVOIP auto-matically  
recovers received signals as low  
as –36 dB. The maximum  
reachable length with 22 AWG  
cable is 2000 meters. When  
Long-Haul Mode is disabled,  
signals as low as –10 dB can be  
received.  
Default: disabled.  
CRC Check  
Y/ N  
When enabled, allows  
generation and checking of  
CRC bits. If not enabled, all  
check bits in the transmit  
direction are set. Only applies  
to ESF frame format.  
(Cyclic  
Redundancy  
Check)  
Default: enabled.  
Frame Format of MultiVOIP  
should match that used by PBX  
or telco. ESF and D4 are  
commonly used.  
Frame Format  
F4, D4, ESF,  
SLC96  
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Technical Configuration (Digital)  
T1 Parameter Definitions (cont’d)  
Values Description  
E&M Immed Strt Channel Associated Signaling  
Field Name  
CAS Protocol  
E&M Wink Start  
(CAS) is a method of  
incorporating telephony  
signaling info into a T1  
E&M Wink with  
dial tone  
voice/ data stream. In CAS, the  
signaling bits (the A, B, C, and  
D bits) are multiplexed into the  
signal stream of each T1  
FXO Ground Strt  
FXO Loop Start  
FXS Ground Strt  
FXS Loop Start  
channel. (By contrast, in  
Common Channel Signaling  
(CCS), one channel handles  
signaling for all other channels.)  
Each CAS protocol defines the  
states of the signaling bits  
during the various stages of a  
call (IDLE, SEIZED, ANSWER,  
RING-ON, RING-OFF).  
The CAS protocol code allows  
the VOIP to interact properly  
with the PBX or central-office  
switch that it serves. The need  
to download CAS protocols  
arises for only a small minority  
of VOIP users, and only when  
PBX/ switch is found to be  
incompatible with standard  
protocols.  
Match this parameter to the  
setting of PBX or central-office  
switch.  
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Technical Configuration (Digital Voips)  
T1 Parameter Definitions (cont’d)  
ISDN Parameters  
Field Name  
Values  
Description  
Enable  
ISDN-PRI  
Y/ N  
If digital connection is ISDN-  
PRI type, this box should be  
checked. When ISDN is  
enabled, the “CAS Protocols”  
field is grayed out (ISDN has its  
own signaling method).  
Terminal/  
Network  
either  
“Terminal” or  
“Network”  
When “Terminal” is selected, it  
indicates that the MultiVOIP  
should emulate the subscriber  
(terminal) side of the digital  
connection. When “Network”  
is selected, it indicates that the  
MultiVOIP should emulate the  
central office (network) side of  
the digital connection.  
Setting used for MultiVOIP  
must be opposite to the setting  
used in the PBX. For example,  
if the PBX is set to “Terminal,”  
then the MultiVOIP must be set  
to “Network.”  
Country  
Operator  
see table, later  
this chapter  
Country in which MultiVOIP is  
operating with ISDN.  
see table, later  
this chapter  
Indicates phone switch  
manufacturer/ model or refers  
to telco so as to specify the  
switching system in question.  
ISDN is implemented  
somewhat differently in  
different switches.  
Note on  
__  
[ISDN implementation options  
are shown, arranged by  
country, in a table below – soon  
after E1 Parameter Definitions.]  
Country &  
Operator  
options.  
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Field Name  
Technical Configuration (Digital)  
T1 Parameter Definitions (cont’d)  
Values  
Description  
Line Build Out 0 dB, -7.5 dB,  
-15 dB, -22.5 dB  
To reduce the crosstalk on  
received signals, a transmit  
attenuator can be placed in the  
data path. Transmit attenuation  
is selectable. Default: O dB  
Refers to length of cable  
between MultiVOIP and  
PBX/telco in meters. Most  
common will be 0 to 40m.  
Pulse Shape  
Level  
0 to 40 Meters  
40 to 81 m  
81 to 122 m  
122 to 162 m  
162 to 200 m  
Clocking  
External/ Internal Set opposite to telco/ PBX  
setting. Example: if telco  
clocking internal, set VOIP  
clocking as external.  
Line Coding  
PCM Law  
AMI / B8ZS  
Match to PBX or telco.  
A-Law/ Mu-Law  
Match to PBX or telco. “  
Mu-law” is analog-to-digital  
compression/ expansion  
standard used in North  
America. “A-law” is European  
standard.  
Yellow Alarm  
Format  
Bit 2 / 1111…  
Depending on the Frame  
Format used, there are choices  
of Yellow Alarm format, as  
follows:  
D4: -Bit2 = 0 in every speech  
channel  
-FS bit of frame 12 is forced  
to one.  
ESF: -Bit2 = 0 in every speech  
channel  
–1111111100000000 pattern  
in data link channel.  
Check with your PBX/ telco  
administrator for the correct  
setting or use the default value  
(1111 … ).  
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MultiVOIP User Guide  
Technical Configuration (Digital Voips)  
E1 Parameters. The parameters applicable to E1 and their values are  
shown in the figure below. These E1 Parameter fields are described in  
the tables that follow.  
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Technical Configuration (Digital)  
E1 Parameter Definitions  
Field Name  
Values  
E1  
Description  
T1/ E1/ ISDN  
European standard.  
Long-Haul  
Mode  
Y/ N  
In Long-Haul Mode, the  
MultiVOIP auto-matically  
recovers received signals as low  
as –36 dB. The maximum  
reachable length with 22 AWG  
cable is 2000 meters. When  
Long-Haul Mode is disabled,  
signals as low as –10 dB can be  
received.  
Default: disabled.  
CRC Check  
--  
Not applicable to E1.  
(Cyclic  
Redundancy  
Check)  
Frame Format of MultiVOIP  
should match that used by PBX  
or telco.  
Frame Format  
Double Frame;  
MultiFrame  
(with CRC4);  
MultiFrame  
(w/ CRC4,  
modified)  
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Technical Configuration (Digital Voips)  
E1 Parameter Definitions (cont’d)  
Values Description  
E&M Immed Strt Channel Associated Signaling  
Field Name  
CAS Protocol  
E&M Wink Start  
(CAS) is a method of  
incorporating telephony  
signaling info into an E1  
E&M Wink with  
dial tone  
voice/ data stream. In CAS, the  
signaling bits (the A, B, C, and  
D bits) are multiplexed into the  
signal stream of each E1  
FXO Ground Strt  
FXO Loop Start  
FXS Ground Strt  
FXS Loop Start  
channel. (By contrast, in  
MFR2ITU  
Common Channel Signaling  
(CCS), one channel handles  
signaling for all other channels.)  
Each CAS protocol defines the  
states of the signaling bits  
during the various stages of a  
call (IDLE, SEIZED, ANSWER,  
RING-ON, RING-OFF).  
MFR2 China  
MFR2 ANI  
The CAS protocol code allows  
the VOIP to interact properly  
with the PBX or central-office  
switch that it serves. The need  
to download CAS protocols  
arises for only a small minority  
of VOIP users, and only when  
PBX/ switch is found to be  
incompatible with standard  
protocols.  
Match this parameter to the  
setting of PBX or central-office  
switch.  
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Technical Configuration (Digital)  
E1 Parameter Definitions (cont’d)  
ISDN Parameters  
Field Name  
Values  
Description  
Enable  
ISDN-PRI  
Y/ N  
If digital connection is ISDN-  
PRI type, this box should be  
checked. When ISDN is  
enabled, the “CAS Protocols”  
field is grayed out (ISDN has its  
own signaling method).  
Terminal/  
Network  
either  
“Terminal” or  
“Network”  
When “Terminal” is selected, it  
indicates that the MultiVOIP  
should emulate the subscriber  
(terminal) side of the digital  
connection. When “Network”  
is selected, it indicates that the  
MultiVOIP should emulate the  
central office (network) side of  
the digital connection.  
Setting used for MultiVOIP  
must be opposite to the setting  
used in the PBX. For example,  
if the PBX is set to “Terminal,”  
then the MultiVOIP must be set  
to “Network.”  
Country  
Operator  
see table, later  
this chapter  
Country in which MultiVOIP is  
operating with ISDN.  
see table, later  
this chapter  
Indicates phone switch  
manufacturer/ model or refers  
to telco so as to specify the  
switching system in question.  
ISDN is implemented  
somewhat differently in  
different switches.  
Note on  
__  
[ISDN implementation options  
are shown, arranged by  
country, in a table below – soon  
after E1 Parameter Definitions.]  
Country &  
Operator  
options.  
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Field Name  
Technical Configuration (Digital Voips)  
E1 Parameter Definitions (cont’d)  
Values  
Description  
Line Build Out 0 dB, -7.5 dB,  
-15 dB, -22.5 dB  
To reduce the crosstalk on  
received signals, a transmit  
attenuator can be placed in the  
data path. Transmit attenuation  
is selectable. Default: O dB  
Refers to length of cable  
between MultiVOIP and  
PBX/telco in meters. Most  
common will be 0 to 40m.  
Pulse Shape  
Level  
0 to 40 Meters  
40 to 81 m  
81 to 122 m  
122 to 162 m  
162 to 200 m  
Clocking  
External/ Internal Set opposite to telco/ PBX  
setting. Example: if telco  
clocking internal, set VOIP  
clocking as external.  
Line Coding  
PCM Law  
AMI / HDB3  
Match to PBX or telco.  
A-Law/ Mu-Law  
Match to PBX or telco.  
“A-law” is analog-to-digital  
compression/ expansion  
standard used in Europe.  
“Mu-law” is North American  
standard.  
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Technical Configuration (Digital)  
10. Set ISDN Parameters (if applicable). These parameters are acces-  
sible in the T1/E1/ISDN Parameters screen. If your T1 or E1 phone line  
is a Primary Rate Interface ISDN line, enable ISDN-PRI and set it for the  
particular implementation of ISDN that your telco uses. The ISDN  
types supported by the digital MultiVOIP units (at press time) are listed  
below, organized by country.  
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Technical Configuration (Digital Voips)  
11. Set SNMP Parameters (Remote Voip Management). This dialog  
box can be reached by pulldown menu, keyboard shortcut, or  
sidebar. To make the MultiVOIP controllable by a remote PC  
running the MultiVoipManager software, check the “Enable SNMP  
Agent” box on the SNMP Parameters screen.  
Accessing “SNMP Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + M  
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Technical Configuration (Digital)  
In each field, enter the values that fit your particular system.  
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Technical Configuration (Digital Voips)  
The SNMP Parameter fields are described in the table below.  
SNMP Parameter Definitions  
Field Name  
Values  
Y/ N  
Description  
Enable SNMP  
Agent  
Enables the SNMP code in the  
firmware of the MultiVOIP. This  
must be enabled for the MultiVOIP  
to communicate with and be  
controllable by the  
MultiVoipManager software.  
Default: disabled  
Trap Manager Parameters  
Address  
4 places; n.n.n.n  
n = 0-255  
IP address of MultiVoipManager  
PC.  
Community  
Name  
--  
A “community” is a group of VOIP  
endpoints that can communicate  
with each other. Often “public” is  
used to designate a grouping where  
all end users have access to entire  
VOIP network. However, calling  
permissions can be configured to  
restrict access as needed.  
The default port number of the  
SNMP manager receiving the traps  
is the standard port 162.  
Port Number  
162  
Community  
Name 1  
Length = 19  
characters (max.)  
Case sensitive.  
First community grouping.  
Permissions  
Read-Only,  
Read/ Write  
If this community needs to change  
MultiVOIP settings, select  
Read/ Write. Otherwise, select  
Read-Only to view settings.  
Community  
Name 2  
Length = 19  
characters (max.)  
Case sensitive.  
Second community grouping  
If this community needs to change  
MultiVOIP settings, select  
Read/ Write. Otherwise, select  
Read-Only to view settings.  
Permissions  
Read-Only,  
Read/ Write  
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Technical Configuration (Digital)  
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), and ring tone.  
In each field, enter the values that fit your particular system.  
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Technical Configuration (Digital)  
The Regional Parameters fields are described in the table below.  
“Regional Parameter” Definitions  
Field Name  
Values  
Description  
Country/  
Region  
USA, Japan, UK,  
Custom  
Name of a country or region that  
uses a certain set of tone pairs for  
dial tone, ring tone, busy tone, and  
‘unobtainable’ tone (fast busy  
tone). In some cases, the tone-pair  
scheme denoted by a country name  
may also be used outside of that  
country. The “Custom” option  
(button) assures that any tone-  
pairing scheme worldwide can be  
accommodated.  
Type column  
dial tone,  
ring tone,  
busy tone,  
Type of telephony tone-pair for  
which frequency, gain, and  
cadence are being presented.  
unobtainable  
tone (fast busy)  
Frequency 1  
Frequency 2  
frequency in  
Hertz  
Lower frequency of pair.  
Higher frequency of pair.  
frequency in  
Hertz  
Gain 1  
gain in dB  
+3dB to –31dB  
and “mute”  
setting  
Amplification factor of lower  
frequency of pair.  
This figure describes amplification  
that the MultiVOIP  
applies to outbound tones entering  
the MultiVOIP at  
the T1 port. Default: -16dB  
Gain 2  
gain in dB  
+3dB to –31dB  
and “mute”  
setting  
Amplification factor of higher  
frequency of pair.  
This figure describes amplification  
that the MultiVOIP  
applies to outbound tones entering  
the MultiVOIP at  
the T1 port. Default: -16dB  
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“Regional Parameter” Definitions (cont’d)  
Field Name  
Values  
Description  
Cadence  
n/ n/ n/ n  
On/ off pattern of tone durations  
used to denote phone ringing,  
phone busy, connection  
(msec) On/ Off four integer time  
values in  
milli-seconds;  
zero value for  
dial-tone  
unobtainable (fast busy), and dial  
tone (continuous and described as  
“0“). Default values differ for  
different countries/ regions.  
indicates  
continuous tone  
Although most cadences have only  
two parts (an “on” duration and  
an “off” duration), some telephony  
cadences have four parts. Most  
cadences, then, are expressed as  
two iterations of a two-part  
sequence. Although this is  
redundant, it is necessary to allow  
for expression of 4-part cadences.  
Click on the “Custom” button to  
bring up the Custom Tone Pair  
Settings screen. This screen allows  
the user to specify tone pair  
Custom  
(button)  
--  
attributes that are not found in any  
of the standard national/ regional  
telephony toning schemes.  
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Technical Configuration (Digital)  
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) for your system.  
This screen allows the user to specify tone-pair attributes that are not  
found in any of the standard national/ regional telephony toning  
schemes. To access this customization feature, click on the Custom  
button on the Regional Parameters screen.  
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Technical Configuration (Digital Voips)  
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.  
TONE PAIR VALUES  
About Defaults: US telephony  
values are used as defaults on  
this screen. However, since this  
dialog box is provided to allow  
custom tone-pair settings,  
default values are essentially  
irrelevant.  
Frequency 1  
Frequency 2  
Gain 1  
frequency in  
Hertz  
Frequency of lower tone of pair.  
This outbound tone pair enters  
the MultiVOIP at the T1/ E1 port.  
frequency in  
Hertz  
Frequency of higher tone of pair.  
This outbound tone pair enters  
the MultiVOIP at the T1/ E1 port.  
Amplification factor of lower  
frequency of pair. This figure  
describes amplification that the  
MultiVOIP applies to outbound  
tones entering the MultiVOIP at  
the T1 port. Default = -16dB  
gain in dB  
+3dB to –31dB  
and “mute”  
setting  
Gain 2  
gain in dB  
+3dB to –31dB  
and “mute”  
setting  
Amplification factor of higher  
frequency of pair. This figure  
describes amplification that the  
MultiVOIP applies to outbound  
tones entering the MultiVOIP at  
the T1 port. Default = -16dB  
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Technical Configuration (Digital)  
Custom Tone-Pair Settings Definitions  
Field Name  
Values  
Description  
Cadence 1  
integer time  
value in  
milli-seconds;  
zero value for  
dial-tone  
On/ off pattern of tone durations  
used to denote phone ringing,  
phone busy, connection  
unobtainable tone (fast busy),  
and dial tone (which is  
indicates  
continuous tone  
continuous and described as  
“0“). 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).  
Cadence 2  
duration in  
milliseconds  
Cadence 2 is duration of first  
“off” period in signaling  
cadence.  
Cadence 3  
Cadence 4  
duration in  
milliseconds  
Cadence 3 is duration of second  
“on” period in signaling cadence.  
Cadence 4 is duration of second  
“off” period in the signaling  
cadence, after which the 4-part  
cadence pattern of the telephony  
signal repeats.  
duration in  
milliseconds  
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Technical Configuration (Digital Voips)  
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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Technical Configuration (Digital)  
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.  
Login Name  
Password  
alpha-  
numeric, per  
email domain  
This is the User Name for the  
MultiVOIP unit’s email account.  
alpha-  
numeric  
Login password for MultiVOIP  
unit’s email account.  
This mail server must be accessible  
on the IP network to which the  
MultiVOIP is connected.  
Mail Server IP  
Address  
n.n.n.n  
for n= 0 to  
255  
25 is a standard port number for  
SMTP.  
Port Number  
25  
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......  
“SMTP Parameters” Definitions (cont’d)  
Field Name  
Values  
Description  
Mail Type  
text or html  
Mail type in which log reports will  
be sent.  
Subject  
text  
User specified. Subject line that will  
appear for all emailed log reports for  
this MultiVOIP unit.  
User specified. This email address  
functions as a source email identifier  
for the MultiVOIP, which, of course,  
cannot usefully receive email  
messages. The Reply-To address  
provides a destination for returned  
messages indicating the status of  
messages sent by the MultiVOIP  
(esp. to indicate when log report  
email was undeliverable or when an  
error has occurred).  
Reply-To  
Address  
email address  
User specified. Email address at  
which VOIP administrator will  
receive log reports.  
Recipient  
Address  
email address  
Criteria for sending log summary by  
email.  
Mail Criteria  
The log summary email will be sent  
out either when the user-specified  
number of log messages has  
accumulated, or once every day or  
multiple days, which ever comes first.  
This is the number of log records  
that must accumulate to trigger the  
sending of a log-summary email.  
This is the number of days that must  
pass before triggering the sending of  
a log-summary email.  
Number of  
Records  
integer  
integer  
Number of  
Days  
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Technical Configuration (Digital)  
The SMTP Parameters dialog box has a secondary dialog box, Custom  
Fields, that allows you to customize email log messages for the  
MultiVOIP. The MultiVOIP software logs data about many aspects of  
the call traffic going through the MultiVOIP. The Custom Fields screen  
lets you pick which aspects will be included in the email log reports.  
“Custom Fields” Definitions  
Field  
Description  
Field  
Description  
Select All Log report to  
include all fields  
shown.  
Channel  
Number  
Data channel  
carrying call.  
Start  
Date,  
Time  
Call  
Date and time the  
phone call began.  
Duration  
Length of call.  
Voice or fax.  
Mode  
Total packets  
received in call.  
Packets  
Sent  
Total packets sent  
in call.  
Packets  
Received  
Bytes  
Bytes  
Sent  
Total bytes sent in  
call.  
Total bytes received  
in call.  
Received  
Packets  
Lost  
Packets lost in  
call.  
Coder  
Voice Coder  
/ Compression Rate  
used for call will be  
listed in log.  
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“Custom Fields” Definitions (cont’d)  
Field  
Description  
Field  
Description  
Outbound  
Digits  
Digits put out by  
MultiVOIP onto  
the T1 or E1 line.  
Prefix  
Matched  
When selected, the  
phonebook prefix  
matched in  
processing call will be  
listed in log.  
Call  
Status  
Successful or  
unsuccessful.  
To Details  
From Details  
Completing or  
terminating gateway  
Gateway  
Number  
IP Addr  
Originating  
gateway  
Gatew N.  
IP Addr  
IP address where call  
was completed or  
terminated.  
IP address where  
call originated.  
Identifier of site  
where call was  
completed or  
Descript  
Options  
Identifier of site  
where call  
originated.  
Descript  
Options  
terminated.  
When selected, log  
will not use/ non-use  
of Silence  
Compression and  
Forward Error  
Correction by call  
terminator.  
When selected, log  
will not use/ non-  
use of Silence  
Compression and  
Forward Error  
Correction by call  
originator.  
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Technical Configuration (Digital)  
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Technical Configuration (Digital Voips)  
15. Set Log Reporting Method. The Logs screen lets you choose how  
the VoIP administrator will receive log reports about the MultiVOIP’s  
performance and the phone call traffic that is passing through it. Log  
reports can be received in one of three ways:  
A. in the MultiVOIP program (GUI),  
B. via email (SMTP), or  
C. at the MultiVoipManager remote voip system  
management program (SNMP).  
Accessing “Logs” Screen  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt + O  
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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).  
“Logs” Screen Definitions  
Field Name  
Values  
Description  
Enable  
Console  
Messages  
Y/ N  
Allows MultiVOIP debugging messages  
to be read via a basic  
telecommunications program like  
HyperTerminal ™ or similar application.  
Normally, this should be disabled  
because it consumers MultiVOIP  
processing resources. Console messages  
are intended for use by tech support  
personnel.  
Turn Off Logs  
Logs Buttons  
Y/ N  
Y/ N  
Disables log reporting function.  
Only one of these three log reporting  
methods, GUI, SMTP, or SNMP, may be  
chosen.  
GUI  
User must view logs at the MultiVOIP  
configuration program.  
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SNMP  
SMTP  
Y/ N  
Y/ N  
Y/ N  
Log messages will be delivered to the  
MultiVoipManager application program.  
Log messages will be sent to user-  
specified email address.  
SysLog Server  
Enable  
This box must be checked if logging is to  
be done in conjunction with a SysLog  
Server program. For more on SysLog  
Server, see Operation & Maintenance  
chapter.  
IP Address  
Port  
n.n.n.n  
for n=  
0-255  
IP address of computer, connected to  
voip network, on which SysLog Server  
program is running.  
514  
Logical port for SysLog Server. 514 is  
commonly used.  
Online  
integer  
Set the interval (in seconds) at which  
logging information will be updated.  
Statistics  
Updation  
Interval  
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Technical Configuration (Digital)  
16. Set Supplementary Services Parameters. This dialog box can be  
reached by pulldown menu, keyboard shortcut, or sidebar.  
Accessing “Supplementary Services Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt +H  
Supplementary Services features derive from the H.450 standard, which  
brings to voip telephony functionality once only available with PSTN or  
PBX telephony. Supplementary Services features can be used under  
H.323 only and not under SIP.  
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In each field, enter the values that fit your particular network.  
Of the features implemented under Supplementary Services, three are  
very closely related: Call Transfer, Call Hold, and Call Waiting. Call  
Name Identification is similar but not identical to the premium PSTN  
feature commonly known as Caller ID.  
Call Transfer. Call Transfer allows one party to re-connect the party  
with whom they have been speaking to a third party. The first party  
is disconnected when the third party becomes connected. Feature is  
invoked by a program-mable phone keypad sequence (for example,  
#7).  
Call Hold. Call Hold allows one party to maintain an idle (non-  
talking) connection with another party while receiving another call  
(Call Waiting), while initiating another call (Call Transfer), or while  
performing some other call management function. Invoked by  
keypad sequence.  
Call Waiting. Call Waiting notifies an engaged caller of an  
incoming call and allows them to receive a call from a third party  
while the party with whom they have been speaking is put on hold.  
Invoked by keypad sequence.  
Call Name Identification. When enabled for a given voip unit (the  
‘home’ voip), this feature gives notice to remote voips involved in  
calls. Notification goes to the remote voip administrator, not to  
individual phone stations. When the home voip is the caller, a plain  
English descriptor will be sent to the remote (callee) voip identifying  
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the channel over which the call is being originated (for example,  
“Calling Party - Omaha Sales Office Line 2”). If that voip channel is  
dedicated to a certain individual, the descriptor could say that, as  
well (for example “Calling Party - Harold Smith in Omaha”). When  
the home voip receives a call from any remote voip, the home voip  
sends a status message back to that caller. This message confirms  
that the home voip’s phone channel is either busy or ringing or that  
a connection has been made (for example, “Busy Party - Omaha  
Sales Office Line ”). These messages appear in the Statistics – Call  
Progress screen of the remote voip.  
Note that Supplementary Services parameters are applied on a channel-  
by-channel basis. However, once you have established a set of  
Supplementary parameters for a particular channel, you can apply this  
entire set of parameters to another channel by using the Copy Channel  
button and its dialog box. To copy a set of Supplementary Services  
parameters to all channels, select “Copy to All” and click Copy.  
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Technical Configuration (Digital Voips)  
The Supplementary Services fields are described in the tables below.  
Supplementary Services Parameter Definitions  
Field Name Values  
Description  
Select  
Channel  
1-2 (210); The channel to be configured is  
1-4 (410); selected here.  
1-8 (810)  
Call  
Transfer  
Enable  
Y/ N  
Select to enable the Call Transfer  
function in the voip unit.  
This is a “blind” transfer and the  
sequence of events is as follows:  
Callers A and B are having a  
conversation.  
Caller A wants to put B into contact  
with C.  
Caller A dials call transfer sequence.  
Caller A hears dial tone and dials  
number for caller C.  
Caller A gets disconnected while  
Caller B gets connected to caller C.  
The numbers and/or symbols that the  
caller must press on the phone keypad to  
initiate a call transfer.  
The call-transfer sequence can be 1 to 4  
characters in length using any  
Transfer  
Sequence  
any  
phone  
keypad  
character  
combination of digits or characters  
(* or #).  
The sequences for call transfer, call  
hold, and call waiting can be from 1  
to 4 digits in length consisting of any  
combination of digits 1234567890*#.  
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Technical Configuration (Digital)  
Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Call Hold  
Enable  
Y/ N  
Select to enable Call Hold function in  
voip unit.  
Call Hold allows one party to  
maintain an idle (non-talking)  
connection with another party while  
receiving another call (Call Waiting),  
while initiating another call (Call  
Transfer), or while performing some  
other call management function.  
Hold  
Sequence  
phone  
keypad  
characters  
The numbers and/ or symbols that the  
caller must press on the phone  
keypad to initiate a call hold.  
The call-hold sequence can be 1 to 4  
characters in length using any  
combination of digits or characters  
(* or #).  
Call Waiting Y/ N  
Enable  
Select to enable Call Waiting function  
in voip unit.  
Retrieve  
Sequence  
phone  
keypad  
The numbers and/ or symbols that the  
caller must press on the phone  
characters, keypad to initiate retrieval of a  
two  
waiting call.  
characters  
in length  
The call-waiting retrieval sequence  
can be 1 to 4 characters in length  
using any combination of digits or  
characters  
(* or #).  
This is the phone keypad sequence  
that a user must press to retrieve a  
waiting call. Customize-able.  
Sequence should be distinct from  
sequence that might be used to  
retrieve a waiting call via the PBX or  
PSTN.  
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Technical Configuration (Digital Voips)  
Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Call Name  
Identification  
Enable  
Enables CNI function. Call Name  
Identification is not the same as Caller  
ID. When enabled on a given voip  
unit currently being controlled by the  
MultiVOIP GUI (the ‘home voip’),  
Call Name Identification sends an  
identifier and status information to  
the administrator of the remote voip  
involved in the call. The feature  
operates on a channel-by-channel  
basis (each channel can have a  
separate identifier).  
If the home voip is originating the  
call, only the Calling Party field is  
applicable. If the home voip is  
receiving the call, then the Alerting  
Party, Busy Party, and Connected  
Party fields are the only applicable  
fields (and any or all of these could be  
enabled for a given voip channel). The  
status information confirms back to  
the originator that the callee (the  
home voip) is either busy, or ringing,  
or that the intended call has been  
completed and is currently connected.  
The identifier and status information  
are made available to the remote voip  
unit and appear in the Caller ID field  
of its Statistics – Call Progress screen.  
(This is how MultiVOIP units handle  
CNI messages; in other voip brands,  
H.450 may be implemented  
differently and then the message  
presentation may vary.)  
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Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Calling  
Party,  
Allowed  
Name Type  
(CNI)  
If the ‘home’ voip unit is originating  
the call and Calling Party is selected,  
then the identifier (from the Caller Id  
field) will be sent to the remote voip  
unit being called. The Caller Id field  
gives the remote voip administrator a  
plain-language identifier of the party  
that is originating the call occurring  
on a specific channel.  
This field is applicable only when the  
‘home’ voip unit is originating the call.  
Example. Suppose a voip system has  
offices in both Denver and Omaha. In  
the Omaha voip unit (the ‘home’ voip  
in this example), Call Name  
Identification has been enabled,  
Calling Party has been enabled as an  
Allowed Name Type, and “Omaha  
Sales Office Voipchannel 2” has been  
entered in the Caller Id field.  
When channel 2 of the Omaha voip is  
used to make a call to any other voip  
phone station (for example, the  
Denver office), the message  
“Calling Party - Omaha Sales Office  
Voipchannel 2” will appear in the  
“Caller Id” field of the  
Statistics - Call Progress screen  
of the Denver voip.  
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Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Alerting  
Party,  
Allowed  
Name Type  
(CNI)  
If the ‘home’ voip unit is receiving the  
call and Alerting Party is selected,  
then the identifier (from the Caller Id  
field) will tell the originating remote  
voip unit that the call is ringing.  
This field is applicable only when the  
‘home’ voip unit is receiving the call.  
Example. Suppose a voip system has  
offices in both Denver and Omaha. In  
the Omaha voip unit (the ‘home’ voip  
unit in this example), Call Name  
Identification has been enabled,  
Alerting Party has been enabled as an  
Allowed Name Type, and “Omaha  
Sales Office Voipchannel 2” has been  
entered in the Caller Id field of the  
Supplementary Services screen.  
When channel 2 of the Omaha voip  
receives a call from any other voip  
phone station (for example, the  
Denver office), the message “Alerting  
Party - Omaha Sales Office  
Voipchannel 2” will be sent back and  
will appear in the Caller Id field of  
the Statistics – Call Progress screen of  
the Denver voip. This confirms to the  
Denver voip that the phone is ringing  
in Omaha.  
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Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Busy Party,  
Allowed  
Name Type  
(CNI)  
If the ‘home’ voip unit is receiving a  
call directed toward an already  
engaged channel or phone station and  
Busy Party is selected, then the  
identifier (from the Caller Id field)  
will tell the originating remote voip  
unit that the channel or called party is  
busy.  
This field is applicable only when the  
‘home’ voip unit is receiving the call.  
Example. Suppose a voip system has  
offices in both Denver and Omaha. In  
the Omaha voip unit (the ‘home’ voip  
unit in this example), Call Name  
Identification has been enabled, Busy  
Party has been enabled as an Allowed  
Name Type, and “Omaha Sales Office  
Voipchannel 2” has been entered in  
the Caller Id field of the  
Supplementary Services screen.  
When channel 2 of the Omaha voip is  
busy but still receives a call attempt  
from any other voip phone station  
(for example, the Denver office), the  
message “Busy Party - Omaha Sales  
Office Voipchannel 2” will be sent  
back and will appear in the Caller Id  
field of the Statistics – Call Progress  
screen of the Denver voip. This  
confirms to the Denver voip that the  
channel or phone station is busy in  
Omaha.  
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Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Connected  
Party,  
Allowed  
Name Type  
(CNI)  
If the ‘home’ voip unit is receiving a  
call and Connected Party is selected,  
then the identifier (from the Caller Id  
field) will tell the originating remote  
voip unit that the attempted call has  
been completed and the connection is  
made.  
This field is applicable only when the  
‘home’ voip unit is receiving the call.  
Example. Suppose a voip system has  
offices in both Denver and Omaha. In  
the Omaha voip unit (the ‘home’ voip  
unit in this example), Call Name  
Identification has been enabled,  
Connected Party has been enabled as  
an Allowed Name Type, and  
“Omaha Sales Office Voipchannel 2”  
has been entered in the Caller Id field  
of the Supplementary Services  
screen.  
When channel 2 of the Omaha voip  
completes an attempted call from any  
other voip phone station (for example,  
the Denver office), the message  
“Connected Party - Omaha Sales  
Office Voipchannel 2” will be sent  
back and will appear in the Caller Id  
field of the Statistics – Call Progress  
screen of the Denver voip. This  
confirms to the Denver voip that the  
call has been completed to Omaha.  
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Supplementary Services Definitions (cont’d)  
Field Name Values Description  
Caller ID  
This is the identifier of a specific  
channel of the ‘home’ voip unit. The  
Caller Id field typically describes a  
person, office, or location, for  
example, “Harry Smith,” or “Bursar’s  
Office,” or “Barnesville Factory.”  
Default  
--  
--  
When this button is clicked, all  
Supplementary Service parameters  
are set to their default values.  
Copy  
Channel  
Copies the Supplementary Service  
attributes of one channel to another  
channel. Attributes can be copied to  
multiple channels or all channels at  
once.  
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17. Set Baud Rate. The Connection option in the sidebar menu has a  
“Settings” item that includes the baud-rate setting for the COM port  
of the computer running the MultiVOIP software.  
First, it is important to note that the default COM port established by  
the MultiVOIP program is COM1. Do not accept the default value  
until you have checked the COM port allocation on your PC. To do  
this, check for COM port assignments in the system resource dialog  
box(es) of your Windows operating system. If COM1 is not available,  
you must change the COM port setting to COM2 or some other COM  
port that you have confirmed as being available on your PC.  
The default baud rate is 115,200 bps.  
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18. View System Information screen and set updating interval (optional).  
This dialog box can be reached by pulldown menu, keyboard shortcut,  
or sidebar.  
Accessing the “System Information” Screen  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt +Y  
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This screen presents vital system information at a glance. It’s primary  
use is in troubleshooting.  
System Information Parameter Definitions  
Field Name Values  
Description  
Boot Code  
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.  
Mac  
Address  
alpha-  
numeric  
Denotes the number assigned as the  
voip unit’s unique Ethernet address.  
Up Time  
hours:  
mm:ss  
Indicates how long the voip has been  
running since its last booting.  
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The frequency with which the System Information screen is updated is  
determined by a setting in the Logs screen  
19. Saving the MultiVOIP Configuration. When values have been set  
for all of the MultiVOIP’s various operating parameters, click on Save  
Setup in the sidebar.  
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20. Creating a User Default Configuration. When a “Setup” (complete  
grouping of parameters) is being saved, you will be prompted about  
designating that setup as a “User Default” setup. A User Default  
setup may be useful as a baseline of site-specific values to which you  
can easily revert. Establishing a User Default Setup is optional.  
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Chapter 6: Technical Configuration  
for Analog MultiVOIPs  
(MVP210/410/810)  
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Configuring the Analog 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 required  
for the MultiVOIP pertains to the phone number dialing sequences that  
it will receive and transmit when handling calls. Dialing patterns will  
be affected by both the PBX/ telephony equipment and the other VOIP  
devices that the MultiVOIP unit interacts with. We call this  
“Phonebook Configuration,” and, for analog MultiVOIP units, it is  
described nominally in Chapter 9: Analog Phonebook Configuration of this  
manual. But, in fact, nearly all of the descriptions and examples for  
analog phonebook configuration are to be found in Chapter 7 if the  
analog voip is operating under the North American telephony scheme,  
or in Chapter 8 if the analog voip is operating under a European  
telephony scheme. Chapter 2, the Quick Start Instructions, presents  
additional examples relevant to the analog voips.  
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Local/Remote Configuration. The MultiVOIP must be configured  
locally at first (to establish an IP address for the MultiVOIP unit). But  
changes to this initial configuration can be done either locally or  
remotely.  
Local configuration is done through a connection between the  
“Command” port of the MultiVOIP and the COM port of the computer;  
the MultiVOIP configuration program is used.  
Remote configuration is done through a connection between the  
MultiVOIP’s Ethernet (network) port and a computer connected to the  
same network. The computer could be miles or continents away from  
the MultiVOIP itself. There are two ways of doing remote  
configuration and operation of the MultiVOIP unit: (1) using the  
MultiVoipManager SNMP program, or (2) using the MultiVOIP web  
browser interface program.  
MultiVoipManager. MultiVoipManager is an SNMP agent program  
(Simple Network Management Protocol) that extends the capabilities of  
the MultiVOIP configuration program: MultiVoipManager allows the  
user to manage any number of VOIPs on a network, whereas the  
MultiVOIP configuration program can manage only the VOIP to which  
it is directly/ locally connected. The MultiVoipManager can configure  
multiple VOIPs simultaneously, whereas the MultiVOIP configuration  
program can configure only one at a time.  
MultiVoipManager may (but does not need to) reside on the same PC  
as the MultiVOIP configuration program. The MultiVoipManager  
program is on the MultiVOIP Product CD. Updates, when applicable,  
may be posted at on the MultiTech FTP site. To download, go to  
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).  
Functional Equivalence 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).  
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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 describes local configuration only. For information on  
remote configuration and management, see the MultiVoipManager  
documentation.  
Pre-Requisites  
To complete the configuration of the  
MultiVOIP unit, you must know several  
things about the overall system.  
Before configuring your MultiVOIP Gateway unit, you must know the  
values for several IP and telephone parameters that describe the IP  
network system and telephony system (PBX or telco central office  
equipment) with which the digital MultiVOIP will interact. If you plan  
to receive log reports on phone traffic by email (SMTP), you must  
arrange to have an email address assigned to the VOIP unit on the  
email server on your IP network.  
IP Parameters  
The following parameters must be known about the network (LAN,  
WAN, Internet, etc.) to which the MultiVOIP will connect:  
Ask your computer network  
administrator.  
Info needed to operate:  
all MultiVOIP models.  
G
IP Network Parameters:  
Record for each VOIP Site  
in System  
ꢀ
• IP Address  
• IP Mask  
• Gateway  
• Domain Name Server (DNS) Info  
(not implemented; for future use)  
Write down the values for these IP parameters. You will need to enter  
these values in the “IP Parameters” screen in the Configuration section  
of the MultiVOIP software. You must have this IP information about  
every VOIP in the system.  
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Analog Telephony Interface Parameters  
(for MVP210/410/810)  
The following parameters must be known about the PBX or telco  
central office equipment to which the analog MultiVOIP will connect: :  
Analog Phone Parameters  
Needed for:  
MVP810  
MVP410  
MVP210  
G
Ask phone company or  
telecom manager.  
Analog Telephony Interface Parameters:  
Record for this VOIP Site  
ꢀ
• Which interface type (or “signaling”) is used?  
E&M_____ FXS/FXO_____  
• If FXS, determine whether the line will be used for a  
phone, fax, or KTS (key telephone system)  
• If FXO, determine if line will be an analog PBX  
extension or an analog line from a telco central office  
• If E&M, determine these aspects of the E&M trunk  
line from the PBX:  
• What is its Type (1, 2, 3, 4, or 5)?  
• Is it 2-wire or 4-wire?  
• Is it Dial-Tone or Wink?  
Write down the values for these telephony parameters. You will need  
to enter these values in the “Interface” screen in the Configuration  
section of the MultiVOIP software.  
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SMTP Parameters (for email call log reporting)  
required if log reports of  
Optional  
VOIP call traffic  
are to be sent by email  
SMTP Parameters  
Preparation Task:  
T
o
e
:
I . T  
.
D
e
p
a
r
t
m
e
n
t
t
Ask Mail Server  
r
:
e
m
a
i l  
a
c
c
o
u
n
f
o
r
V
O
I P  
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. .  
voip-unit2@biggytech.com  
Get the IP address of the  
mail server computer, as  
well.  
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Local Configuration Procedure (Summary)  
After the MultiVOIP configuration software has been installed in the  
‘Command’ PC (which is connected to the MultiVOIP unit), several  
steps must be taken to configure the MultiVOIP to function in its  
specific setting. Although the summary below includes all of these  
steps, some are optional.  
1. Check Power and Cabling.  
2. Start MultiVOIP Configuration Program.  
3. Confirm Connection.  
4. Solve Common Connection Problems.  
A. Fixing a COM Port Problem.  
B. Fixing a Cabling Problem.  
5. Familiarize yourself with configuration parameter screens and how  
to access them.  
6. Set IP Parameters.  
7. Enable web browser GUI (optional).  
8. Set Voice/ Fax Parameters.  
9. Set Telephony Interface Parameters.  
10. Set SNMP Parameters (applicable if MultiVoipManager remote  
management software is used).  
11. Set Regional Parameters (Phone Signaling Tones and Cadences).  
12. Set Custom Tones and Cadences (optional).  
13. Set SMTP Parameters (applicable if Log Reports are via Email).  
14. Set Log Reporting Method (GUI, locally in MultiVOIP  
Configuration program; SNMP, remotely in MultiVoipManager  
program; or SMTP, via email).  
15. Set Supplementary Services Parameters. The Supplementary  
Services screen allows voip deployment of features that are normally  
found in PBX or PSTN systems (e.g., call transfer and call waiting).  
16. Set Baud Rate (of COM port connection to ‘Command’ PC).  
17. View System Info screen and set updating interval (optional).  
18. Save the MultiVOIP configuration.  
19. Create a User Default Configuration (optional).  
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Local Configuration Procedure (Detailed)  
You can begin the configuration process as a continuation of the  
MultiVOIP software installation. You can establish your configuration  
or modify it at any time by launching the MultiVOIP program from the  
Windows Start menu.  
1. Check Power and Cabling. Be sure the MultiVOIP is turned on and  
connected to the computer via the MultiVOIP’s Command Port (DB9  
connector at computer’s COM port; RJ45 connector at MultiVOIP).  
2. Start MultiVOIP Configuration Program. Launch the MultiVOIP  
program from the Windows Start menu (from the folder location  
determined during installation).  
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3. Confirm Connection. If the MultiVOIP is set for an available COM  
port and is correctly cabled to the PC, the MultiVOIP main screen will  
appear. (If the main screen appears grayed out and seems inaccessible,  
go to step 4.)  
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In the lower left corner of the screen, the connection status of the  
MultiVOIP will be displayed. The messages in the lower left corner  
will change as detection occurs. The message “MultiVOIP Found”  
confirms that the MultiVOIP is in contact with the MultiVOIP  
configuration program. Skip to step 5.  
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4. Solving Common Connection Problems. .  
A. Fixing a COM Port Problem. If the MultiVOIP main screen appears  
but is grayed out and seems inaccessible, the COM port that was  
specified for its communication with the PC is unavailable and must  
be changed. An error message will appear.  
To change the COM port setting, use the COM Port Setup dialog box,  
which is accessible via the keyboard shortcut Ctrl + G or by going to  
the Connection pull-down menu and choosing “Settings.” In the  
“Select Port” field, select a COM port that is available on the PC. (If  
no COM ports are currently available, re-allocate COM port resources  
in the computer’s MS Windows operating system to make one  
available.)  
Ctrl + G  
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4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by  
the computer, two error messages will appear (saying “Multi-VOIP  
Not Found” and “Phone Database Not Read”).  
In this case, the MultiVOIP is simply disconnected from the network.  
For instructions on MultiVOIP cable connections, see the Cabling  
section of Chapter 3.  
5. Configuration Parameter Groups: Getting Familiar, Learning  
About Access. The first part of configuration concerns IP parameters,  
Voice/ FAX parameters, Telephony Interface parameters, SNMP  
parameters, Regional parameters, SMTP parameters, Supplementary  
Services parameters, Logs, and System Information. In the MultiVOIP  
software, these seven types of parameters are grouped together under  
“Configuration” and each has its own dialog box for entering values.  
Generally, you can reach the dialog box for these parameter groups in  
one of four ways: pulldown menu, toolbar icon, keyboard shortcut, or  
sidebar. ..  
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6. Set IP Parameters. This dialog box can be reached by pulldown  
menu, toolbar icon, keyboard shortcut, or sidebar.  
Accessing “IP Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt + I  
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In each field, enter the values that fit your particular network.  
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The IP Parameters fields are described in the table below.  
IP Parameter Definitions  
Field Name  
Values  
Description  
Enable  
Diffserv  
Y/ N  
Diffserv is used for QoS  
(quality of service).  
When enabled, we  
configure the TOS (Type  
of Service) bits in the IP  
header so routers  
supporting Diffserv can  
give priority to the  
VOIP’s IP packets.  
Disabled by default.  
Frame Type  
IP Address  
IP Mask  
Type II, SNAP  
4-places, 0-255  
4-places, 0-255  
Must be set to match  
network’s frame type.  
Default is Type II.  
The unique LAN IP  
address assigned to the  
MultiVOIP.  
Subnetwork address that  
allows for sharing of IP  
addresses within a LAN.  
Gateway  
4-places, 0-255.  
(feature not yet  
implemented; for  
future use)  
The IP address of the  
device that connects your  
MultiVOIP to the  
Internet.  
Enable DNS  
Y/ N.  
Enables Domain Name  
Space/ System function  
where computer names  
are resolved using a  
worldwide distributed  
database.  
(feature not yet  
implemented; for  
future use)  
DNS Server IP  
Address  
4-places, 0-255  
IP address of specific  
DNS server to be used to  
resolve Internet  
computer names.  
FTP Server  
Enable  
Y/ N  
See “FTP Server  
MultiVOIP unit has an  
FTP Server function so  
File Transfers” in that firmware and other  
Operation &  
Maintenance  
chapter.  
important operating  
software files can be  
transferred to the voip  
via the network.  
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7. Enable Web Browser GUI (Optional). After an IP address for the  
MultiVOIP unit has been established, you can choose to do any further  
configuration of the unit (a) by using the MultiVOIP web browser GUI,  
or (b) by continuing to use the MultiVOIP Windows GUI. If you want  
to do configuration work using the web browser GUI, you must first  
enable it. To do so, follow the steps below.  
A. Set IP address of MultiVOIP unit using the MultiVOIP  
Configuration program (the Windows GUI).  
B. Save Setup in Windows GUI.  
C. Close Windows GUI.  
D. Install Java program from MultiVOIP product CD (on first use  
only).  
E. Open web browser.  
F. Browse to IP address of MultiVOIP unit.  
G. If username and password have been established, enter them  
when when prompted.  
H. Use web browser GUI to configure or operate MultiVOIP unit.  
The configuration screens in the web browser GUI will have the same  
content as their counterparts in the Windows GUI; only the graphic  
presentation will be different.  
For more details on enabling the MultiVOIP web GUI, see the “Web  
Browser Interface” section of the Operation & Maintenance chapter of  
this manual.  
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8. Set Voice/FAX Parameters. This dialog box can be reached by  
pulldown menu, toolbar icon, keyboard shortcut, or sidebar.  
Accessing “Voice/FAX Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + H  
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In each field, enter the values that fit your particular network.  
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Note that Voice/ FAX parameters are applied on a channel-by-channel  
basis. However, once you have established a set of Voice/ FAX  
parameters for a particular channel, you can apply this entire set of  
Voice/ FAX parameters to another channel by using the Copy Channel  
button and its dialog box. To copy a set of Voice/ FAX parameters to all  
channels, select “Copy to All” and click Copy.  
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The Voice/FAX Parameters fields are described in the tables below.  
Voice/Fax Parameter Definitions  
Field Name Values  
Description  
Default  
--  
When this button is clicked, all  
Voice/ FAX parameters are set to their  
default values.  
Select  
Channel  
1-2 (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.  
Signal amplification (or attenuation)  
in dB.  
Voice Gain  
Input Gain  
--  
+31dB  
to  
–31dB  
Modifies audio level entering voice  
channel before it is sent over the  
network to the remote VOIP. The  
default & recommended value is 0 dB.  
Modifies audio level being output to  
the device attached to the voice  
channel. The default and  
Output Gain +31dB  
to  
–31dB  
recommended value is 0 dB.  
DTMF Parameters  
DTMF Gain  
--  
The DTMF Gain (Dual Tone Multi-  
Frequency) controls the volume level  
of the digital tones sent out for Touch-  
Tone dialing.  
DTMF Gain, +3dB to  
High Tones  
Default value: -4 dB. Not to be  
-31dB & changed except under supervision of  
“mute”  
DTMF Gain, +3dB to  
Low Tones  
MultiTech’s Technical Support.  
Default value: -7 dB. Not to be  
-31dB & changed except under supervision of  
“mute” MultiTech’s Technical Support.  
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Field Name  
Technical Configuration (Analog)  
Voice/Fax Parameter Definitions (cont’d)  
Values  
DTMF Parameters  
Description  
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.  
When DTMF Out of Band is  
selected (checked), the MultiVOIP  
detects DTMF tones at its input and  
regenerates them at its output.  
When DTMF Inband is selected,  
the DTMF digits are passed through  
the MultiVOIP unit as they are  
received.  
DTMF  
In/ Out of  
Band  
Out of  
Band, or  
Inband  
FAX Parameters  
Fax Enable  
Y/ N  
Enables or disables fax capability for a  
particular channel.  
Max Baud  
Rate  
(Fax)  
2400, 4800,  
7200, 9600,  
12000,  
Set to match baud rate of fax machine  
connected to channel (see Fax machine’s  
user manual).  
14400 bps  
Default = 14400 bps.  
Fax Volume  
(Default =  
-9.5 dB )  
-18.5 dB  
to –3.5 dB  
Controls output level of fax tones. To  
be changed only under the direction of  
Multi-Tech’s Technical Support.  
Jitter Value  
(Fax)  
Default =  
400 ms  
Defines the inter-arrival packet  
deviation (in milliseconds) for the fax  
transmission. A higher value will  
increase the delay, allowing a higher  
percentage of packets to be  
reassembled. A lower value will  
decrease the delay allowing fewer  
packets to be reassembled.  
Mode (Fax)  
FRF 11;  
T.38  
(T.38 not  
currently  
sup-  
FRF11 is frame-relay FAX standard  
using these coders: G.711, G.728, G.729,  
G.723.1.  
T.38 is an ITU-T standard for storing  
and forwarding FAXes via email using  
X.25 packets. It uses T.30 fax standards  
and includes special provisions to  
preclude FAX timeouts during IP  
transmissions.  
ported)  
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Technical Configuration (Analog)  
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  
G.723.1 @ digital voice is compressed to 6.3K,  
5.3 kbps,  
6.3 kbps;  
G.729,  
8kbps;  
Net Coder  
@
allowing several simultaneous  
conversations over the same  
bandwidth that would otherwise  
carry only one.  
To make selections from the  
6.4, 7.2, 8,  
8.8, 9.6  
kbps  
Selected Coder drop-down list, the  
Manual option must be enabled.  
Max  
bandwidth  
(coder)  
11 – 128  
kbps  
This drop-down list enables you to  
select the maximum bandwidth  
allowed for this channel. The Max  
Bandwidth drop-down list is  
enabled only if the Coder is set to  
Automatic.  
If coder is to be selected  
automatically (“Auto” setting), then  
enter a value for maximum  
bandwidth.  
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Technical Configuration (Analog)  
Voice/Fax Parameter Definitions (cont’d)  
Field Name Values  
Advanced Features  
Description  
Silence  
Compression  
Y/ N  
Determines whether silence  
compression is enabled (checked) for  
this voice channel.  
With Silence Compression enabled, the  
MultiVOIP will not transmit voice  
packets when silence is detected,  
thereby reducing the amount of  
network bandwidth that is being used  
by the voice channel.  
Default = off.  
Echo  
Cancellation  
Y/ N  
Y/ N  
Determines whether echo cancellation is  
enabled (checked) for this voice  
channel.  
Echo Cancellation removes echo and  
improves sound quality. Default = on.  
Forward  
Error  
Correction  
Determines whether forward error  
correction is enabled (checked) for this  
voice channel.  
Forward Error Correction enables  
some of the voice packets that were  
corrupted or lost to be recovered. FEC  
adds an additional 50% overhead to the  
total network bandwidth consumed by  
the voice channel.  
Default = Off  
Auto Call  
Enable  
Y/ N  
The Auto Call 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.  
Phone No.  
(Auto Call)  
--  
Phone number used for Auto Call  
function. A corresponding phone  
number must be listed in the Outbound  
Phonebook.  
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Technical Configuration (Analog)  
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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Technical Configuration (Analog)  
Voice/Fax Parameter Definitions (cont’d)  
Field Name Values  
Dynamic Jitter  
Maximum 60 to 400  
Description  
The maximum dynamic jitter buffer  
of 400 milliseconds is the maximum  
delay tolerable over a high jitter  
network.  
Jitter Value  
ms  
Default = 300 msec  
Optimizat-  
ion Factor  
0 to 12  
The Optimization Factor  
determines how quickly the length  
of the Dynamic Jitter Buffer is  
changed based on actual jitter  
encountered on the network.  
Selecting the minimum value of 0  
means low voice delay is desired,  
but increases the possibility of jitter-  
induced voice quality problems.  
Selecting the maximum value of 12  
means highest voice quality under  
jitter conditions is desired at the  
cost of increased voice delay.  
Default = 7.  
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Technical Configuration (Analog)  
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  
seconds;  
Default =  
30 sec.  
Specifies how long to wait before  
disconnecting the call when IP  
network connectivity with the  
remote site has been lost.  
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Technical Configuration (Analog)  
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 + I  
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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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Technical Configuration (Analog)  
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.  
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.  
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Technical Configuration (Analog)  
FXS Loop Start Interface: Parameter Definitions  
Field Name  
Values  
Description  
Message  
Y/ N  
Applicable only when  
Waiting Light  
MultiVOIP is used with Avaya  
Magix PBX units equipped with  
Merlin Messaging Centralized  
mail. When enabled, the  
Message Waiting Light feature  
allows the PBX to send mode-  
codes and message-waiting  
indications to another Avaya  
Magix PBX, which in turn will  
turn on the message waiting  
light on a phone station. It also  
allows Direct Inward Dialing,  
such that no additional dial  
tone is needed on voip call.  
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  
Ring Count,  
FXS  
integer values  
Y/ N  
FXS Options,  
Current Loss  
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.  
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Technical Configuration (Analog)  
FXS Ground Start Parameters (not supported). The parameters  
applicable to FXS Ground Start are shown in the figure below and  
described in the table that follows.  
FXS Ground Start Interface: Parameter Definitions  
Field Name  
Values  
Description  
FXS Ground  
Start  
Y/ N  
Enables FXS Loop Start  
interface type.  
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  
Y/ N  
Applicable only when  
Waiting Light  
MultiVOIP is used with Avaya  
Magix PBX units equipped with  
Merlin Messaging Centralized  
mail. When enabled, the  
Message Waiting Light feature  
allows the PBX to send mode-  
codes and message-waiting  
indications to another Avaya  
Magix PBX, which in turn will  
turn on the message waiting  
light on a phone station. It also  
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Technical Configuration (Analog)  
allows Direct Inward Dialing,  
such that no additional dial  
tone is needed on voip call.  
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  
Ring Count,  
FXS  
integer values  
Y/ N  
FXS Options,  
Current Loss  
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.  
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Technical Configuration (Analog)  
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, FXO  
Y/ N  
Enables FXO functionality  
Dialing Options  
Regeneration  
Pulse, DTMF  
Determines whether digits  
generated and sent out will be  
pulse tones or DTMF.  
Inter Digit  
Timer  
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.  
Flash Hook  
Timer  
integer values,  
in 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.  
Message  
Y/ N  
Applicable only when  
Waiting Light  
MultiVOIP is used with Avaya  
Magix PBX units equipped with  
Merlin Messaging Centralized  
mail. When enabled, the  
Message Waiting Light feature  
allows the PBX to send mode-  
codes and message-waiting  
indications to another Avaya  
Magix PBX, which in turn will  
turn on the message waiting  
light on a phone station. It also  
allows Direct Inward Dialing,  
such that no additional dial  
tone is needed on voip call.  
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Technical Configuration (Analog)  
FXO Interface: Parameter Definitions (cont’d)  
Field Name Values Description  
Dialing Options (cont’d)  
Inter Digit milliseconds  
The length of time between the  
outputting of DTMF digits.  
Default = 100 ms.  
Regeneration  
Time  
FXO Disconnect On  
There are three possible criteria  
for disconnection under FXO:  
current loss, tone detection, and  
silence detection.  
Disconnection can be triggered  
by more than one of the three  
criteria.  
Current Loss  
Y/ N  
Disconnection to be triggered  
by loss of current. That is,  
when Current Loss is enabled  
(“Y”), the MultiVOIP will hang  
up the call when it detects a loss  
of current initiated by the  
attached device.  
FXO Current  
Detect Timer  
integer values  
The minimum time required for  
(in milliseconds ) detecting the current loss signal  
on the FXO interface. In other  
words, this is the minimum  
length of time the current must  
be absent to validate ‘current  
loss’ as a disconnection  
criterion. Default = 500 ms.  
Tone Detection Y/ N  
Disconnection to be triggered  
by a tone sequence.  
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FXO Interface: Parameter Definitions (cont’d)  
Field Name  
Values  
Description  
FXO Disconnect On (cont’d)  
Disconnect  
Tone Sequence  
1st tone pair  
+
These are DTMF tone pairs.  
Values for first tone pair are:  
*, #, 0, 1-9, and A-D.  
2nd tone pair  
Values for second tone pair are:  
none, 0, 1-9, A-D, *, and #.  
The tone pairs 1-9, 0, *, and #  
are the standard DTMF pairs  
found on phone sets. The tone  
pairs A-D are “extended  
DTMF” tones, which are used  
for various PBX functions.  
DTMF Tone Pairs  
Low Tones  
697Hz  
2
5
8
0
3
6
9
#
A
B
C
D
1
4
7
*
770Hz  
852Hz  
941Hz  
High Tones 1209Hz 1336Hz 1447Hz 1633Hz  
Silence  
Detection  
One-Way or  
Two-Way  
Disconnection to be triggered  
by silence in one direction only  
or in both directions  
simultaneously.  
Silence Timer  
in seconds  
integer value  
Y/ N  
Duration of silence required to  
trigger disconnection.  
Disconnect on  
Call Progress  
Tone  
Allows call on FXO port to be  
disconnected when a PBX  
issues a call-progress tone  
denoting that the phone station  
on the PBXthat has been  
involved in the call has been  
hung up.  
Ring Count,  
FXO  
integer value  
Number of rings required  
before the MultiVOIP answers  
the incoming call.  
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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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Technical Configuration (Analog)  
E&M Interface Parameter Definitions  
Field Name  
Interface  
Type  
Values  
Description  
E&M  
enables E&M functionality  
Types 1-5. Each  
type can be 2-  
wire or 4-wire.  
Refers to the type of E&M  
interface being used.  
Signal  
Dial Tone or  
Wink  
When Dial Tone is selected, no  
wink is required on the E lead  
or M lead in the call initiation or  
setup.  
When Wink is selected, a wink  
is required during call setup.  
Wink Timer  
(in ms)  
integer values,  
in milliseconds  
This is the length of the wink  
for wink signaling.  
Applicable only when Signal  
parameter is set to “Wink.”  
Pass Through  
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 actually unused.  
Applicable only for E&M  
Signaling with Dial Tone.  
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10. Set SNMP Parameters (Remote Voip Management). This dialog  
box can be reached by pulldown menu, keyboard shortcut, or  
sidebar. To make the MultiVOIP controllable by a remote PC  
running the MultiVoipManager software, check the “Enable SNMP  
Agent” box on the SNMP Parameters screen.  
Accessing “SNMP Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + M  
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In each field, enter the values that fit your particular system.  
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The SNMP Parameter fields are described in the table below.  
SNMP Parameter Definitions  
Field Name  
Values  
Y/ N  
Description  
Enable SNMP  
Agent  
Enables the SNMP code in the  
firmware of the MultiVOIP. This  
must be enabled for the MultiVOIP  
to communicate with and be  
controllable by the  
MultiVoipManager software.  
Default: disabled  
Trap Manager Parameters  
Address  
4 places; n.n.n.n  
n = 0-255  
IP address of MultiVoipManager  
PC.  
Community  
Name  
--  
A “community” is a group of VOIP  
endpoints that can communicate  
with each other. Often “public” is  
used to designate a grouping where  
all end users have access to entire  
VOIP network. However, calling  
permissions can be configured to  
restrict access as needed.  
The default port number of the  
SNMP manager receiving the traps  
is the standard port 162.  
Port Number  
162  
Community  
Name 1  
Length = 19  
characters (max.)  
Case sensitive.  
First community grouping.  
Permissions  
Read-Only,  
Read/ Write  
If this community needs to change  
MultiVOIP settings, select  
Read/ Write. Otherwise, select  
Read-Only to view settings.  
Community  
Name 2  
Length = 19  
characters (max.)  
Case sensitive.  
Second community grouping  
If this community needs to change  
MultiVOIP settings, select  
Read/ Write. Otherwise, select  
Read-Only to view settings.  
Permissions  
Read-Only,  
Read/ Write  
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Technical Configuration (Analog)  
11. Set Regional Parameters (Phone Signaling Tones & Cadences). ).  
This dialog box can be reached by pulldown menu, keyboard  
shortcut, or sidebar.  
Accessing “Regional Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + R  
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The Regional Parameters screen will appear. For the country selected,  
the standard set of frequency pairs will be listed for dial tone, busy  
tone, ‘unobtainable’ tone (fast busy or trunk busy), and ring tone.  
In each field, enter the values that fit your particular system.  
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The Regional Parameters fields are described in the table below.  
“Regional Parameter” Definitions  
Field Name  
Values  
Description  
Country/  
Region  
USA, Japan, UK,  
Custom  
Name of a country or region that  
uses a certain set of tone pairs for  
dial tone, ring tone, busy tone, and  
‘unobtainable’ tone (fast busy  
tone). In some cases, the tone-pair  
scheme denoted by a country name  
may also be used outside of that  
country. The “Custom” option  
(button) assures that any tone-  
pairing scheme worldwide can be  
accommodated.  
Type column  
dial tone,  
ring tone,  
busy tone,  
Type of telephony tone-pair for  
which frequency, gain, and  
cadence are being presented.  
unobtainable  
tone (fast busy)  
Frequency 1  
Frequency 2  
Gain 1  
frequency in  
Hertz  
Lower frequency of pair.  
Higher frequency of pair.  
frequency in  
Hertz  
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 th  
eFXS, FXS, or E&M port. Default:  
-16dB  
Gain 2  
gain in dB  
+3dB to –31dB  
and “mute”  
setting  
Amplification factor of higher  
frequency of pair.  
This applies to the dial, ring, busy,  
and ‘unobtainable’ (fast busy) tones  
that the MultiVOIP outputs as  
audio to the FXS, FXO, or E&M  
port. Default: -16dB  
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“Regional Parameter” Definitions (cont’d)  
Field Name  
Values  
Description  
Cadence  
n/ n/ n/ n  
On/ off pattern of tone durations  
used to denote phone ringing,  
phone busy, connection  
(msec) On/ Off four integer time  
values in  
milli-seconds;  
zero value for  
dial-tone  
unobtainable (fast busy), and dial  
tone (“0” indicates continuous  
tone). Default values differ for  
different countries/ regions.  
indicates  
continuous tone  
Although most cadences have only  
two parts (an “on” duration and  
an “off” duration), some telephony  
cadences have four parts. Most  
cadences, then, are expressed as  
two iterations of a two-part  
sequence. Although this is  
redundant, it is necessary to allow  
for expression of 4-part cadences.  
Ratio of “make” duration versus  
“break” duration when a tone  
pulse is generated. 60/ 40 applies  
to US telephony; 67/ 33 applies  
internationally (note, however, that  
US telephony standards are used in  
certain regions/ nations outside the  
US).  
Pulse  
Generation  
Ratio  
pair of integer  
values in  
milliseconds;  
60/ 40 or 67/ 33  
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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Technical Configuration (Analog)  
12. Set Custom Tones and Cadences (optional). . The Regional  
Parameters dialog box has a secondary dialog box that allows you to  
customize DTMF tone pairs to create unique ring-tonesdial-tones,  
busy-tones or “unobtainable” tones (fast busy signal) 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.  
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, and dial tone (“0”  
indicates continuous tone).  
Cadence 1 is duration of first  
period of tone being “on” in the  
cadence of the telephony signal  
(which could be ring-tone, busy-  
tone, unobtainable-tone, or dial-  
tone).  
indicates  
continuous tone  
Cadence 2  
duration in  
milliseconds  
Cadence 2 is duration of first  
“off” period in signaling  
cadence.  
Cadence 3  
Cadence 4  
duration in  
milliseconds  
Cadence 3 is duration of second  
“on” period in signaling cadence.  
Cadence 4 is duration of second  
“off” period in the signaling  
cadence, after which the 4-part  
cadence pattern of the telephony  
signal repeats.  
duration in  
milliseconds  
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Technical Configuration (Analog)  
13. Set SMTP Parameters (Log Reports by Email). The SMTP  
Parameters screen is applicable when the VOIP administrator has  
chosen to receive log reports by email (this is done by selecting the  
“SMTP” checkbox in the Others screen and selecting “Enable SMTP”  
in the SMTP Parameters screen.). The SMTP Parameters screen can  
be reached by pulldown menu, keyboard shortcut, or sidebar.  
Accessing “SMTP Parameters”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt + S  
MultiVOIP as Email Sender. When SMTP is used, the MultiVOIP will  
actually be given its own email account (with Login Name and  
Password) on some mail server connected to the IP network. Using this  
account, the MultiVOIP will then send out email messages containing  
log report information. The “Recipient” of the log report email is  
ordinarily the VoIP administrator. Because the MultiVOIP cannot  
receive email, a “Reply-To” address must also be set up. Ordinarily,  
the “Reply-To” address is that of a technician who has access to the  
mail server or MultiVOIP or both, and the VoIP administrator might  
also be designated as the “Reply-To” party. The main function of the  
Reply-To address is to receive error or failure messages regarding the  
emailed reports.  
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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.  
Login Name  
Password  
alpha-  
numeric, per  
email domain  
This is the User Name for the  
MultiVOIP unit’s email account.  
alpha-  
numeric  
Login password for MultiVOIP  
unit’s email account.  
This 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  
25  
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......  
“SMTP Parameters” Definitions (cont’d)  
Field Name  
Values  
Description  
Mail Type  
text or html  
Mail type in which log reports will  
be sent.  
Subject  
text  
User specified. Subject line that will  
appear for all emailed log reports for  
this MultiVOIP unit.  
User specified. This email address  
functions as a source email identifier  
for the MultiVOIP, which, of course,  
cannot usefully receive email  
messages. The Reply-To address  
provides a destination for returned  
messages indicating the status of  
messages sent by the MultiVOIP  
(esp. to indicate when log report  
email was undeliverable or when an  
error has occurred).  
Reply-To  
Address  
email address  
User specified. Email address at  
which VOIP administrator will  
receive log reports.  
Recipient  
Address  
email address  
Criteria for sending log summary by  
email.  
Mail Criteria  
The log summary email will be sent  
out either when the user-specified  
number of log messages has  
accumulated, or once every day or  
multiple days, which ever comes first.  
This is the number of log records  
that must accumulate to trigger the  
sending of a log-summary email.  
This is the number of days that must  
pass before triggering the sending of  
a log-summary email.  
Number of  
Records  
integer  
integer  
Number of  
Days  
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Technical Configuration (Analog)  
The SMTP Parameters dialog box has a secondary dialog box, Custom  
Fields, that allows you to customize email log messages for the  
MultiVOIP. The MultiVOIP software logs data about many aspects of  
the call traffic going through the MultiVOIP. The Custom Fields screen  
lets you pick which aspects will be included in the email log reports.  
“Custom Fields” Definitions  
Field  
Description  
Field  
Description  
Select All Log report to  
include all fields  
shown.  
Channel  
Number  
Data channel  
carrying call.  
Start  
Date,  
Time  
Call  
Date and time the  
phone call began.  
Duration  
Length of call.  
Voice or fax.  
Mode  
Total packets  
received in call.  
Packets  
Sent  
Total packets sent  
in call.  
Packets  
Received  
Bytes  
Bytes  
Sent  
Total bytes sent in  
call.  
Total bytes received  
in call.  
Received  
Packets  
Lost  
Packets lost in  
call.  
Coder  
Voice Coder  
/ Compression Rate  
used for call will be  
listed in log.  
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“Custom Fields” Definitions (cont’d)  
Field  
Description  
Field  
Description  
Outbound  
Digits  
Digits put out by  
MultiVOIP onto  
the phone line.  
Prefix  
Matched  
When selected, the  
phonebook prefix  
matched in  
processing the call  
will be listed in log.  
Call  
Status  
Successful or  
unsuccessful.  
To Details  
From Details  
Completing or  
answering gateway  
Gateway  
Number  
IP Addr  
Originating  
gateway  
Gatew N.  
IP Addr  
IP address where call  
was completed or  
answered.  
IP address where  
call originated.  
Identifier of site  
where call was  
completed or  
Descript  
Options  
Identifier of site  
where call  
originated.  
Descript  
Options  
answered.  
When selected, log  
will not use/ non-use  
of Silence  
Compression and  
Forward Error  
Correction by party  
answering call.  
When selected, log  
will not use/ non-  
use of Silence  
Compression and  
Forward Error  
Correction by call  
originator.  
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14. Set Log Reporting Method. The Logs screen lets you choose how  
the VoIP administrator will receive log reports about the MultiVOIP’s  
performance and the phone call traffic that is passing through it. Log  
reports can be received in one of three ways:  
A. in the MultiVOIP program (GUI),  
B. via email (SMTP), or  
C. at the MultiVoipManager remote voip system  
management program (SNMP).  
Accessing “Logs” Screen  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt + O  
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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.  
Turn Off Logs  
Logs Buttons  
Y/ N  
Check to disable log reporting function.  
Only one of these three log reporting  
methods, GUI, SMTP, or SNMP, may be  
chosen.  
GUI  
Y/ N  
Y/ N  
Y/ N  
Y/ N  
User must view logs at the MultiVOIP  
configuration program.  
SNMP  
SMTP  
Log messages will be delivered to the  
MultiVoipManager application program.  
Log messages will be sent to user-  
specified email address.  
SysLog Server  
Enable  
This box must be checked if logging is to  
be done in conjunction with a SysLog  
Server program. For more on SysLog  
Server, see Operation & Maintenance  
chapter.  
IP Address  
Port  
n.n.n.n  
for n=  
0-255  
IP address of computer, connected to  
voip network, on which SysLog Server  
program is running.  
514  
Logical port for SysLog Server. 514 is  
commonly used.  
Online  
integer  
Set the interval (in seconds) at which  
logging information will be updated.  
Statistics  
Updation  
Interval  
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Technical Configuration (Analog)  
15. Set Supplementary Services Parameters. This dialog box can be  
reached by pulldown menu, keyboard shortcut, or sidebar.  
Accessing “Supplementary Services” Parameters  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt +H  
Supplementary Services features derive from the H.450 standard,  
which brings to voip telephony functionality once only available with  
PSTN or PBX telephony. Supplementary Services features can be used  
under H.323 only and not under SIP.  
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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 program-mable 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  
1-2 (210); The channel to be configured is  
1-4 (410); selected here.  
1-8 (810)  
Call  
Transfer  
Enable  
Y/ N  
Select to enable the Call Transfer  
function in the voip unit.  
This is a “blind” transfer and the  
sequence of events is as follows:  
Callers A and B are having a  
conversation.  
Caller A wants to put B into contact  
with C.  
Caller A dials call transfer sequence.  
Caller A hears dial tone and dials  
number for caller C.  
Caller A gets disconnected while  
Caller B gets connected to caller C.  
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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Technical Configuration (Analog)  
Supplementary Services Definitions (cont’d)  
Field Name Values  
Description  
Caller ID  
This is the identifier of a specific  
channel of the ‘home’ voip unit. The  
Caller Id field typically describes a  
person, office, or location, for  
example, “Harry Smith,” or “Bursar’s  
Office,” or “Barnesville Factory.”  
Default  
--  
--  
When this button is clicked, all  
Supplementary Service parameters  
are set to their default values.  
Copy  
Channel  
Copies the Supplementary Service  
attributes of one channel to another  
channel. Attributes can be copied to  
multiple channels or all channels at  
once.  
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Technical Configuration (Analog)  
16. Set Baud Rate. The Connection option in the sidebar menu has a  
“Settings” item that includes the baud-rate setting for the COM port  
of the computer running the MultiVOIP software.  
First, it is important to note that the default COM port established by  
the MultiVOIP program is COM1. Do not accept the default value  
until you have checked the COM port allocation on your PC. To do  
this, check for COM port assignments in the system resource dialog  
box(es) of your Windows operating system. If COM1 is not available,  
you must change the COM port setting to COM2 or some other COM  
port that you have confirmed as being available on your PC.  
The default baud rate is 115,200 bps.  
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Technical Configuration (Analog)  
17. View System Information screen and set updating interval (optional).  
This dialog box can be reached by pulldown menu, keyboard shortcut,  
or sidebar.  
Accessing “System Information” Screen  
Pulldown  
Icon  
Shortcut  
Sidebar  
Ctrl + Alt +Y  
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Technical Configuration (Analog)  
This screen presents vital system information at a glance. It’s primary  
use is in troubleshooting.  
System Information Parameter Definitions  
Field Name Values  
Description  
Boot Code  
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.  
Mac  
Address  
alpha-  
numeric  
Denotes the number assigned as the  
voip unit’s unique Ethernet address.  
Up Time  
hours:  
mm:ss  
Indicates how long the voip has been  
running since its last booting.  
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Technical Configuration (Analog)  
The frequency with which the System Information screen is updated is  
determined by a setting in the Logs screen  
18. Saving the MultiVOIP Configuration. When values have been set  
for all of the MultiVOIP’s various operating parameters, click on Save  
Setup in the sidebar.  
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Technical Configuration (Analog)  
19. Creating a User Default Configuration. When a “Setup” (complete  
grouping of parameters) is being saved, you will be prompted about  
designating that setup as a “User Default” setup. A User Default  
setup may be useful as a baseline of site-specific values to which you  
can easily revert. Establishing a User Default Setup is optional.  
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Chapter 7: T1 Phonebook  
Configuration  
(North American Telephony Standards)  
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Configuring the MVP2400/2410  
MultiVOIP Phonebooks  
When a VoIP serves a PBX system, it’s important that the operation of  
the VoIP be transparent to the telephone end user. That is, the VoIP  
should not entail the dialing of extra digits to reach users elsewhere on  
the network that the VoIP serves. On the contrary, VOIP service more  
commonly reduces dialed digits by allowing users (served by PBXs in  
facilities in distant cities) to dial their co-workers with 3-, 4-, or 5-digit  
extensions as if they were in the same facility.  
Furthermore, the setup of the VoIP generally should allow users to  
make calls on a non-toll basis to any numbers accessible without toll by  
users at all other locations on the VoIP system. Consider, for example,  
a company with VOIP-equipped offices in New York, Miami, and Los  
Angeles, each served by its own PBX. When the VOIP phone books are  
set correctly, personnel in the Miami office should be able to make calls  
without toll not only to the company’s offices in New York and Los  
Angeles, but also to any number that’s local in those two cities.  
To achieve transparency of the VoIP telephony system and to give full  
access to all types of non-toll calls made possible by the VOIP system,  
the VoIP administrator must properly configure the “Outbound” and  
“Inbound” phone-books of each VoIP in the system.  
The “Outbound” phonebook for a particular VoIP unit describes the  
dialing sequences required for a call to originate locally (typically in a  
PBX in a particular facility) and reach any of its possible destinations at  
remote VoIP sites, including non-toll calls completed in the PSTN at the  
remote site.  
The “Inbound” phonebook for a particular VoIP unit describes the  
dialing sequences required for a call to originate remotely from any  
other VOIP sites in the system, and to terminate on that particular  
VOIP.  
Briefly stated, the MultiVOIP’s Outbound phonebook lists the phone stations  
it can call; its Inbound phonebook describes the dialing sequences that can be  
used to call that MultiVOIP and how those calls will be directed. (Of course,  
the phone numbers are not literally “listed” individually, but are,  
instead, described by rule.)  
Consider two types of calls in the three-city system described above:  
(1) calls originating from the Miami office and terminating in the New  
York (Manhattan) office, and (2) calls originating from the Miami office  
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T1 PhoneBook Configuration  
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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E1 Phonebook Configuration  
MultiVOIP User Guide  
Phonebook configuration screens can be accessed using icons or the  
sidebar menu.  
Phonebook Icons  
Description  
Phonebook Configuration  
Inbound Phonebook  
Entries List  
Add Inbound Phonebook  
Entry  
Edit selected Inbound  
Phonebook Entry  
Outbound Phonebook  
Entries List  
Add Outbound  
Phonebook Entry  
Edit selected Outbound  
Phonebook Entry  
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T1 PhoneBook Configuration  
Phonebook Sidebar Menu  
1. Go to the PhoneBook Configuration screen (using either the sidebar  
or drop-down menu).  
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MultiVOIP User Guide  
In consultation with your VOIP administrator, enter the Gateway  
Name and values for Q.931 parameters and Gatekeeper RAS  
parameters. Determine whether your voip system will operate with a  
proxy server. Determine which H.323 version 4 functions you will  
implement. (They are not always applicable. See field description for  
each parameter.)  
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T1 PhoneBook Configuration  
The table below describes all fields in the PhoneBook Configuration  
screen.  
PhoneBook Configuration Parameter Definitions  
Field Name  
Values  
Description  
Gateway  
Name  
Y/ N  
This field allows you to specify  
a name for this MultiVOIP.  
When placing a call, this name  
is sent to the remote  
MultiVOIP for display in Call  
Progress listings, Logs, etc.  
Q.931 Parameters  
Use Fast Start  
Y/ N  
Enables the H.323 Fast Start  
procedure. May need to be  
enabled/ disabled for  
compatibility with third-party  
VOIP gateways.  
Call Signalling  
Port  
port  
number  
Default: 1720 (H.323)  
GateKeeper RAS Parameters  
IP address of the GateKeeper.  
IP Address  
Port Number  
Well-known port number for  
GateKeepers.  
Must match port number of  
GateKeeper, 1719.  
Gateway  
Prefix  
This number becomes  
registered with the  
GateKeeper. Call requests sent  
to the gatekeeper and  
preceded by this prefix will be  
routed to the VOIP gateway.  
Gatekeeper  
Name  
alpha-  
numeric  
string  
Optional. The name of the  
GateKeeper with which this  
MultiVOIP is trying to register.  
The H.323 ID is used to  
register this particular  
MultiVOIP with the  
H.323 ID  
GateKeeper.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
PhoneBook Configuration Parameter Definitions  
(cont’d)  
Values Description  
Proxy Server Parameters  
Field Name  
Enable Proxy  
Y/ N  
Allows the MultiVOIP to work  
in conjunction with a proxy  
server.  
Proxy Server  
IP Address  
n.n.n.n  
where  
Network address of the proxy  
server that the voip is using.  
n=0-255  
Port Number  
Logical port number for proxy  
communications.  
H.323 Version 4 Parameters  
Q.931  
Multiplexing  
(Mux)  
Y/ N  
Y/ N  
Signalling for multiple phone  
calls can be carried on a single  
port rather than opening a  
separate signalling port for  
each call. This conserves  
bandwidth resources.  
H.245  
H.245 messages are  
Tunneling  
(Tun)  
encapsulated within the Q.931  
call signalling channel.  
Among other things, the H.245  
messages let the two endpoints  
tell each other what their  
technical capabilities are and  
determine who, during the  
call, will be the client and who  
the server. Tunneling is the  
process of transmitting these  
H.245 messages through the  
Q.931 channel. The same  
TCP/ IP socket (or logical port)  
already being used for the Call  
Signalling Channel is then also  
used by the H.245 Control  
Channel. This encapsulation  
reduces the number of logical  
ports (sockets) needed and  
reduces call setup time.  
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T1 PhoneBook Configuration  
PhoneBook Configuration Parameter Definitions  
(cont’d)  
Field Name  
Values  
Description  
H.323 Version 4 Parameters  
FS (Fast Start or Fast Connect)  
is a Q.931 feature of H.323v2 to  
hasten call setup as well as  
‘pre-opening’ the media  
Parallel H.245  
(FS + Tun)  
Y/ N  
channel before the CONNECT  
message is sent. This pre-  
opening is a requirement for  
certain billing activities.  
Under Parallel H.245 FS + Tun,  
this Fast Connect feature can  
operate simultaneously with  
H.245 Tunneling (see  
description above).  
Multiplexed UDP call  
Annex –E (AE) Y/ N  
signalling transport. Annex E  
is helpful for high-volume voip  
system endpoints. Gateways  
with lesser volume can afford  
to use TCP to establish calls.  
However, for larger volume  
endpoints, the call setup times  
and system resource usage  
under TCP can become  
problematic. Annex E allows  
endpoints to perform call  
signalling functions under the  
UDP protocol, which involves  
substantially streamlined  
overhead. (This feature should  
not be used on the public  
Internet because of potential  
problems with security and  
bandwidth usage.)  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
2. Select PhoneBook Modify and then select  
Outbound Phone Book/ List Entries.  
Click Add.  
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T1 PhoneBook Configuration  
3. The Add/Edit Outbound PhoneBook screen appears.  
Enter Outbound PhoneBook data for your MVP2400/ 2410. Note that  
the Advanced button gives access to the Alternate IP Routing feature, if  
needed. Alternate IP Routing can be implemented in a secondary  
screen (as described after the primary screen field definitions below).  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
The fields of the Add/Edit Outbound Phone Book screen are described  
in the table below.  
Add/Edit Outbound Phone Book: Field Definitions  
Field Name  
Values  
Description  
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  
for  
the IP address to which the  
call will be directed if it  
begins with the destination  
pattern given  
n = 0-255  
Description  
alpha-  
numeric  
Describes the facility or  
geographical location at  
which the call will be  
completed.  
Indicates protocol to be used in  
outbound transmission.  
Protocol Type  
SIP or H.323  
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T1 PhoneBook Configuration  
Add/Edit Outbound Phone Book: Field Definitions  
(cont’d)  
Field Name  
Values  
Description  
H.323 fields  
Indicates whether or not  
gatekeeper is used.  
Use Gatekeepr  
Y/ N  
The H.323 ID assigned to the  
destination MultiVOIP. Only  
valid if “Use Gatekeeper” is  
enabled for this entry.  
H.323 ID  
Gateway  
Prefix  
This number becomes  
registered with the  
GateKeeper. Call requests  
sent to the gatekeeper and  
preceded by this prefix will  
be routed to the VOIP  
gateway.  
Q.931 Port  
Number  
1720  
Q.931 is the call signalling  
protocol for setup and  
termination of calls (aka ITU-  
T Recommendation I.451).  
H.323 employs only one  
“well-known” port (1720) for  
Q.931 signalling. If Q.931  
message-oriented signalling  
protocol is used, the port  
number 1720 must be chosen.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
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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MultiVOIP User Guide  
T1 PhoneBook Configuration  
Advanced  
--  
Gives access to secondary  
button  
screen where an Alternate IP  
Route can be specified for  
backup or redundancy of  
signal paths. See discussion  
on next page.  
Clicking on the Advanced button brings up the Alternate Routing secondary screen.  
This feature provides an alternate path for calls if the primary IP network cannot carry  
the traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN.  
However, this feature could also be used to divert traffic to a redundant (backup) unit  
in case one voip unit fails. The user must specify the IP address of the alternate route  
for each destination pattern entry in the Outbound Phonebook.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Alternate Routing Field Definitions  
Field  
Values  
Description  
Name  
Alternate n.n.n.n  
Alternate destination for outbound data traffic  
in case of excessive delay in data transmission.  
IP  
where  
Address  
n= 0-255  
Round  
Trip  
Delay  
milliseconds The Round Trip Delay is the criterion for  
judging when a data pathway is considered  
blocked. When the delay exceeds the  
threshold specified here, the data stream will  
be diverted to the alternate destination  
specified as the Alternate IP Address.  
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T1 PhoneBook Configuration  
4. Select PhoneBook Modify and then select Inbound PhoneBook/List Entries.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
5. The Add/Edit Inbound PhoneBook screen appears.  
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  
Remove Prefix  
dialed digits portion of dialed number to  
be removed before  
completing call to destination  
(often a local PBX)  
Add Prefix  
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.  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
Add/Edit Inbound Phone Book: Field Definitions  
(cont’d)  
Field Name  
Values  
Description  
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.  
Forward  
Condition  
Uncondit.;  
Busy  
No Resp.  
Busy. When selected, calls  
will be forwarded when  
station is busy.  
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.  
Forward  
Address/  
Number  
IP addr. or  
phone  
number  
Phone number or IP  
address to which calls will  
be directed.  
Ring Count  
integer  
When No Response is  
condition for forwarding  
calls, this determines how  
many unanswered rings  
are needed to trigger the  
forwarding.  
6. When your Outbound and Inbound PhoneBook entries are  
completed, click on Save Setup in the sidebar menu to save your  
configuration.  
You can change your configuration at any time as needed for your  
system.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Remember that the initial MVP2400/ 2410 setup must be done locally  
using the MultiVOIP program. However, after the initial configuration  
is complete, all of the MVP2400/ 2410 units in the VOIP system can be  
configured, re-configured, and updated from one location using the  
MultiVoipManager software program.  
T1 Phonebook Examples  
The following example demonstrates how Outbound and Inbound  
PhoneBook entries work in a situation of multiple area codes. Consider  
a company with offices in Minneapolis and Baltimore.  
3 Sites, All-T1 Example  
Notice first the area code situation in those two cities: Minneapolis’s  
local calling area consists of multiple adjacent area codes; Baltimore’s  
local calling area consists of a base area code plus an overlay area code.  
Company  
VOIP/PBXꢀ  
Baltimore/  
SIte  
Outstate MD  
Overlay  
443  
NW  
Suburbs  
St. Paul  
& Suburbs  
651  
763  
Mpls  
612  
Company  
VOIP/PBX  
SIte  
...  
ꢀ
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  
ꢀ
R
o
u
t
e
r
T1  
Digital  
VoIP  
PBX  
-7003  
200.2.9.7  
-7002  
325-7001  
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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 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.  
257  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Similarly, the Inbound PhoneBook for the Baltimore VOIP (shown first  
below) generally matches the Outbound PhoneBook of the Minneapolis  
VOIP (shown second below).  
Notice the extended prefix to be removed: 14103257. This entry allows  
Minneapolis users to contact Baltimore co-workers as though they were  
in the Minneapolis facility, using numbers in the range 7000 to 7999.  
Note also that a comma (as in the entry 9,443) denotes a delay in  
dialing. A one-second delay is commonly used to allow a second dial-  
tone to be generated for calls going outside of the facility’s PBX system.  
258  
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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” .  
259  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Configuring Mixed Digital/Analog VOIP Systems  
The MVP2400/ 2410 digital MultiVOIP unit is compatible with analog  
VOIPs. In many cases, digital and analog VOIP units will appear in the  
same telephony/ IP system. In addition to MVP-210/ 410/ 810  
MultiVOIP units (Series II units), legacy analog VOIP units (Series I  
units made by MultiTech) may be included in the system, as well.  
When legacy VOIP units are included, the VOIP administrator must  
handle two styles of phonebooks in the same VOIP network. The  
diagram below shows a small-scale system of this kind: one digital  
VOIP (the MVP2400) operates with two Series II analog VOIPs (an  
MVP210 and an MVP410), and two Series I legacy VOIPs (two MVP200  
units).  
EXAMPLE:  
Digital & An alog VOIPs  
in Sam e System  
Site D:  
Pierre, SD  
Area Code 615  
PSTN  
PBX  
T1  
200.2.9.9  
Digital  
VoIP  
MVP2400  
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  
260  
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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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E1 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).  
262  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
Note 1. The “x” is a wildcard character.  
Note 2. By specifying “Channel 0,” we instruct the  
MVP2400/ 2410 to choose any available data  
channel to carry the call.  
Note 3. Note that Site F key system has only 30 extensions  
(x7400-7429). This destination pattern (140226374)  
actually directs calls to 402-263-7430 through  
402-263-7499 into the key system, as well.  
This means that such calls, which belong on the  
PSTN, cannot be completed. In some cases, this  
might be inconsequential because an entire  
exchange (fully used or not) might have been  
reserved for the company or it might be  
unnecessary to reach those numbers. However, to  
specify only the 30 lines actually used by the key  
system, the destination pattern 140226374 would  
have to be replaced by three other destination  
patterns, namely 1402263740, 1402263741, and  
1402263742. In this way, calls to 402-263-7430  
through 402-263-7499 would be properly directed  
to the PSTN. In the Site D outbound phonebook,  
the 30 lines are defined exactly, that is, without  
making any adjacent phone numbers unreachable  
through the voip system.  
263  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Outbound Phone Book for MVP2400 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  
200.2.9.5 Calls to extensions  
(thirty) of key  
740  
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.  
264  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
Inbound Phonebook for MVP2400/2410 Digital VOIP  
(Site D)  
Remove Add  
Channel  
Comment  
Prefix  
Prefix  
Number  
1615  
9,  
0
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.  
265  
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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  
(Bismarck).  
1507  
1507  
101#  
Note 3.  
200.2.9.8 To originate calls  
to any PSTN  
phone in  
Rochester area  
using the FXO  
channel (channel  
#1) of the Site B  
VOIP.  
102  
200.2.9.8 To originate calls  
to phone  
connected to FXS  
port (channel #2)  
of the Site B VOIP  
(Rochester).  
421  
200.2.9.6 Calls to Site E  
(Cheyenne).  
1615  
200.2.9.9 Calls to Pierre area  
PSTN via Site D  
PBX.  
31  
1615  
492  
200.2.9.9 Calls to Pierre PBX  
extensions with  
four digits.  
Note 3. The pound sign (“#”) is a delimiter separating the  
VOIP number from the standard telephony phone number.  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
Inbound Phonebook for MVP410 Analog VOIP (Site F)  
Remove Add  
Channel  
Number  
4
Comment  
Prefix  
Prefix  
1402  
Access to Lincoln local PSTN by  
users at remote VOIP locations  
via FXO port at Site F.  
1402  
263740  
1402  
263741  
1402  
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.  
267  
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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.  
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.  
268  
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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  
269  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
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.  
270  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
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 MVP2400/ 2410 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 MVP2400/ 2410: 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.  
271  
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E1 Phonebook Configuration  
Site D calling Site F  
MultiVOIP User Guide  
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 intial “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).  
272  
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MultiVOIP User Guide  
Site F calling Site D  
T1 PhoneBook Configuration  
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 MVP2400/ 2410 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 MVP2400/ 2410 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 MVP2400/ 2410 can be completely  
transparent to phone users within the company.  
273  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Chapter 8: E1 Phonebook  
Configuration  
(European Telephony Standards)  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
MVP3010 Inbound and Outbound MultiVOIP  
Phonebooks  
Important  
Definition:  
The MultiVOIP’s Outbound phonebook  
lists the phone stations it can call;  
its Inbound phonebook describes the  
dialing sequences that can be used to  
call that MultiVOIP and how those calls  
will be directed.  
When a VOIP serves a PBX system, the operation of the VOIP should be  
transparent to the telephone end user and savings in long-distance  
calling charges should be enjoyed. Use of the VOIP should not require  
the dialing of extra digits to reach users elsewhere on the VOIP  
network. On the contrary, VOIP service more commonly reduces  
dialed digits by allowing users (served by PBXs in facilities in distant  
cities) to dial their co-workers with 3-, 4-, or 5-digit extensions -- as if  
they were in the same facility. More importantly, the VOIP system  
should be configured to maximize savings in long-distance calling  
charges. To achieve both of these objectives, ease of use and maximized  
savings, the VOIP phonebooks must be set correctly.  
NOTE: VOIPs are commonly used for  
another reason, as well: VOIPs  
allow an organization to  
integrate phone and data traffic  
onto a single network. Typically  
these are private networks.  
275  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
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.  
Wren Clothing Co.  
VOIP/PBX Site  
Amsterdam  
VOIP/PBX Site  
London  
ꢀ
ꢀ
The  
Netherlands  
Wren Clothing Co.  
VOIP/PBX Site  
Paris  
ꢀ
Free VOIP Calls  
France  
276  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
Local Rate Calls: Within Local Calling Area of Remote  
VOIP  
In the second use of the VOIP system, the local calling area of each  
VOIP location becomes accessible to all of the VOIP system’s users. As  
a result, international calls can be made at local calling rates. For  
example, suppose that Wren Clothing buys its zippers from The  
Bluebird Zipper Company in the western part of metropolitan London.  
In that case, Wren Clothing personnel in both Paris and Amsterdam  
could call the Bluebird Zipper Company without paying international  
long-distance rates. Only London local phone rates would be charged.  
This applies to calls completed anywhere in London’s local calling area  
(which includes both Inner London and Outer London). Generally,  
local calling rates apply only within a single area code, and, for all calls  
outside that area code, national rates apply. There are, however, some  
European cases where local calling rates extend beyond a single area  
code. Local rates between Inner and Outer London are one example of  
this. (It is also possible, in some locations, that calls within an area code  
may be national calls. But this is rare.)  
United Kingdom  
Wren Clothing Co.  
Wren Clothing Co.  
VOIP/PBX Site  
Amsterdam  
VOIP/PBX Site  
London  
Bluebird Zipper Co.  
London  
ꢀ
ꢀ
The  
Netherlands  
Wren Clothing Co.  
VOIP/PBX Site  
Paris  
ꢀ
Calls at London local rates  
Local Calling Area  
France  
277  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Similarly, the VOIP system allows Wren Clothing employees in London  
and Amsterdam to call anywhere in Paris at local rates; it allows Wren  
Clothing employees in Paris and London to call anywhere in  
Amsterdam at local rates.  
United Kingdom  
Wren Clothing Co.  
VOIP/PBX Site  
Wren Clothing Co.  
VOIP/PBX Site  
London  
Amsterdam  
ꢀ
ꢀ
The  
Netherlands  
Wren Clothing Co.ꢀ  
VOIP/PBX Site  
Paris  
Calls at Amsterdam local rates  
Calls at Paris local rates  
Local Calling Areas  
France  
278  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
National Rate Calls: Within Nation of Remote VOIP Site  
In the third use of the VOIP system, the national calling area of each  
VOIP location becomes accessible to all of the VOIP system’s users. As  
a result, international calls can be made at national calling rates. Again,  
significant savings are possible. For example, suppose that the Wren  
Clothing Company buys its buttons from the Chickadee Button  
Company in the Dutch city of Rotterdam. In that case, Wren Clothing  
personnel in both London and Paris could call the Chickadee Button  
Company without paying international long-distance rates; only Dutch  
national calling rates would be charged. This applies to calls completed  
anywhere in The Netherlands.  
United Kingdom  
The  
Wren Clothing Co.  
VOIP/PBX Site  
Netherlands  
London  
ꢀ Wren Clothing Co.  
VOIP/PBX Site  
Amsterdam  
ꢀ
Chickadee Button Co.  
Rotterdam  
Wren Clothing Co.  
VOIP/PBX Site  
ꢀ
Paris  
Calls at Dutch  
National Rates  
France  
279  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Similarly, the VOIP system allows Wren Clothing employees in London  
and Amsterdam to call anywhere in France at French national rates; it  
allows Wren Clothing employees in Paris and Amsterdam to call  
anywhere in the United Kingdom at its national rates.  
United Kingdom  
Wren Clothing Co.  
VOIP/PBX Site  
Wren Clothing Co.  
London  
VOIP/PBX Site  
ꢀ
Amsterdam  
ꢀ
The  
Netherlands  
Wren Clothing Co.  
VOIP/PBX Site  
Paris  
ꢀ
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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T1 PhoneBook Configuration  
In order for any VOIP phone call to be made, there must be both an  
Inbound Phonebook entry and an Outbound Phonebook entry that  
describe the end-to-end connection. The phone station originating the  
call must be connected to the VOIP system. The Outbound Phonebook  
for that VOIP unit must have a destination pattern entry that includes  
the ‘called’ phone (that is, the phone completing the call). The Inbound  
Phonebook of the VOIP where the call is completed must have a  
destination pattern entry that includes the digit sequence dialed by the  
originating phone station.  
The PhoneBook Configuration procedure below is brief, but it is  
followed by an example case. For many people, the example case may  
be easier to grasp than the procedure steps. Configuration is not  
difficult, but all phone number sequences, destination patterns, and  
other information must be entered exactly; otherwise connections will  
not be made.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Phonebook configuration screens can be accessed using icons or the  
sidebar menu.  
Phonebook Icons  
Description  
Phonebook Configuration  
Inbound Phonebook  
Entries List  
Add Inbound Phonebook  
Entry  
Edit selected Inbound  
Phonebook Entry  
Outbound Phonebook  
Entries List  
Add Outbound  
Phonebook Entry  
Edit selected Outbound  
Phonebook Entry  
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T1 PhoneBook Configuration  
Phonebook Sidebar Menu  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Phonebook Configuration Procedure  
1. Go to the PhoneBook Configuration screen (using either the sidebar  
menu, drop-down menu, or icon).  
In consultation with your VOIP administrator, enter the Gateway  
Name and values for Q.931 parameters and Gatekeeper RAS  
parameters. Determine whether your voip system will operate with a  
proxy server. Determine which H.323 version 4 functions you will  
implement. (They are not always applicable. See field description for  
each parameter.)  
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T1 PhoneBook Configuration  
The table below describes all fields in the PhoneBook Configuration  
screen.  
PhoneBook Configuration Parameter Definitions  
Field Name  
Values  
Description  
Gateway  
Name  
Y/ N  
This field allows you to specify a name for  
this MultiVOIP. When placing a call, this  
name is sent to the remote MVP3000 for  
display in Call Progress listings, Logs, etc.  
Q.931 Parameters  
Use Fast Start  
Y/ N  
Enables the H.323 Fast Start procedure.  
May need to be enabled/ disabled for  
compatibility with third-party VOIP  
gateways.  
Call Signalling port  
Port number  
Default: 1720 (H.323)  
GateKeeper RAS  
Parameters  
IP address of the GateKeeper.  
IP Address  
Port Number  
Well-known port number for GateKeepers.  
Must match port number of GateKeeper,  
1719.  
Gateway  
Prefix  
This number becomes registered with the  
GateKeeper. Call requests sent to the  
gatekeeper and preceded by this prefix will  
be routed to the VOIP gateway.  
Gatekeeper  
Name  
alpha-  
numeric  
string  
Optional. The name of the GateKeeper with  
which this MultiVOIP is trying to register.  
H.323 ID  
The H.323 ID is used to register this  
particular MultiVOIP with the GateKeeper.  
H.323 ID is an alias entry sent to the  
GateKeeper, made of alpha-numeric  
characters. For NetMeeting endpoints,  
numbers are preferred over letters. The  
H.323 ID identifies the IP calling sequence  
that the GateKeeper must ‘dial’ to contact  
the remote VOIP.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
PhoneBook Configuration Parameter Definitions  
(cont’d)  
Values Description  
Proxy Server Parameters  
Field Name  
Enable Proxy  
Y/ N  
Proxy Server  
IP Address  
n.n.n.n  
where  
Network address of the proxy  
server that the voip is using.  
n=0-255  
Port Number  
Logical port number for proxy  
communications.  
H.323 Version 4 Parameters  
Q.931  
Multiplexing  
(Mux)  
Y/ N  
Y/ N  
Signalling for multiple phone  
calls can be carried on a single  
port rather than opening a  
separate signalling port for  
each call. This conserves  
bandwidth resources.  
H.245  
H.245 messages are  
Tunneling  
(Tun)  
encapsulated within the Q.931  
call signalling channel.  
Among other things, the H.245  
messages let the two endpoints  
tell each other what their  
technical capabilities are and  
determine who, during the  
call, will be the client and who  
the server. Tunneling is the  
process of transmitting these  
H.245 messages through the  
Q.931 channel. The same  
TCP/ IP socket (or logical port)  
already being used for the Call  
Signalling Channel is then also  
used by the H.245 Control  
Channel. This encapsulation  
reduces the number of logical  
ports (sockets) needed and  
reduces call setup time.  
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T1 PhoneBook Configuration  
PhoneBook Configuration Parameter Definitions  
(cont’d)  
Field Name  
Values  
Description  
H.323 Version 4 Parameters  
FS (Fast Start or Fast Connect)  
is a Q.931 feature of H.323v2 to  
hasten call setup as well as  
‘pre-opening’ the media  
Parallel H.245  
(FS + Tun)  
Y/ N  
channel before the CONNECT  
message is sent. This pre-  
opening is a requirement for  
certain billing activities.  
Under Parallel H.245 FS + Tun,  
this Fast Connect feature can  
operate simultaneously with  
H.245 Tunneling (see  
description above).  
Multiplexed UDP call  
Annex –E (AE) Y/ N  
signalling transport. Annex E  
is helpful for high-volume voip  
system endpoints. Gateways  
with lesser volume can afford  
to use TCP to establish calls.  
However, for larger volume  
endpoints, the call setup times  
and system resource usage  
under TCP can become  
problematic. Annex E allows  
endpoints to perform call  
signalling functions under the  
UDP protocol, which involves  
substantially streamlined  
overhead. (This feature should  
not be used on the public  
Internet because of potential  
problems with security and  
bandwidth usage.)  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
2. Select PhoneBook Modify and then select  
Outbound Phone Book/List Entries.  
Click Add.  
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T1 PhoneBook Configuration  
3. The Add/Edit Outbound PhoneBook screen appears.  
Enter Outbound PhoneBook data for your MVP3010. Note that the  
Advanced button gives access to the Alternate IP Routing feature, if  
needed. Alternate IP Routing can be implemented in a secondary  
screen (as described after the primary screen field definitions below).  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
The fields of the Add/Edit Outbound Phone Book screen are described  
in the table below.  
Add/Edit Outbound Phone Book: Field Definitions  
Field Name  
Values  
Description  
Destination  
Pattern  
prefixes,  
area codes,  
exchanges,  
line  
numbers,  
extensions  
Defines the beginning of  
dialing sequences for calls  
that will be connected to  
another VOIP in the system.  
Numbers beginning with  
these sequences are diverted  
from the PTSN and carried  
on Internet or other IP  
network.  
Total Digits  
as needed  
number of digits the phone  
user must dial to reach  
specified destination  
Remove Prefix  
dialed digits portion of dialed number to  
be removed before  
completing call to destination  
Add Prefix  
IP Address  
dialed digits digits to be added before  
completing call to destination  
n.n.n.n  
the IP address to which the  
call will be directed if it  
begins with the destination  
pattern given  
for = 0-255  
Description  
alpha-  
numeric  
Describes the facility or  
geographical location at  
which the call will be  
completed.  
Indicates protocol to be used in  
outbound transmission.  
Protocol Type  
SIP or H.323  
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T1 PhoneBook Configuration  
Add/Edit Outbound Phone Book: Field Definitions  
(cont’d)  
Field Name  
Values  
Description  
H.323 fields  
Indicates whether or not  
gatekeeper is used.  
Use Gatekeepr  
Y/ N  
The H.323 ID assigned to the  
destination MultiVOIP. Only  
valid if “Use Gatekeeper” is  
enabled for this entry.  
H.323 ID  
Gateway  
Prefix  
This number becomes  
registered with the  
GateKeeper. Call requests  
sent to the gatekeeper and  
preceded by this prefix will  
be routed to the VOIP  
gateway.  
Q.931 Port  
Number Q.931  
Port Number  
1720  
Q.931 is the call signalling  
protocol for setup and  
termination of calls (aka ITU-  
T Recommendation I.451).  
H.323 employs only one  
“well-known” port (1720) for  
Q.931 signalling. If Q.931  
message-oriented signalling  
protocol is used, the port  
number 1720 must be chosen.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Add/Edit Outbound Phone Book: Field Definitions  
(cont’d)  
Description  
Field Name  
SIP Fields  
Use Proxy  
Values  
Y/ N  
Select if proxy server is used.  
Transport  
Protocol  
TCP or  
UDP  
Voip administrator must choose  
between UDP and TCP  
transmission protocols. UDP is a  
high-speed, low-overhead  
connectionless protocol where  
data is transmitted without  
acknowledgment, guaranteed  
delivery, or guaranteed packet  
sequence integrity. TCP is slower  
connection-oriented protocol  
with greater overhead, but  
having acknowledgment and  
guarantees delivery and packet  
sequence integrity.  
The SIP Port Number is a  
UDP logical port number.  
The voip will “listen” for SIP  
messages at this logical port.  
If SIP is used, 5060 is the  
default, standard, or “well  
known” port number to be  
used. If 5060 is not used,  
then the port number used is  
that specified in the SIP  
Request URI (Universal  
Resource Identifier).  
SIP Port  
Number  
5060 or other  
*See RFC3087  
(“Control of  
Service  
Context using  
SIP Request-  
URI,” by the  
Network  
Working  
Group).  
Looking similar to an email  
address, a SIP URL  
SIP URL  
sip.userphone  
@
identifies a user's address.  
In SIP communications, each  
caller or callee is identified  
by a SIP url:  
hostserver,  
where  
“userphone”  
is the  
sip:user_name@host_name.  
The format of a sip url is very  
similar to an email address,  
except that the “sip:“ prefix is  
used.  
telephone  
number and  
“hostserver”is  
the domain  
name or an  
address on the  
network  
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T1 PhoneBook Configuration  
Advanced  
--  
Gives access to secondary  
button  
screen where an Alternate IP  
Route can be specified for  
backup or redundancy of  
signal paths. See discussion  
on next page.  
Clicking on the Advanced button brings up the Alternate Routing secondary screen.  
This feature provides an alternate path for calls if the primary IP network cannot carry  
the traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN.  
However, this feature could also be used to divert traffic to a redundant (backup) unit  
in case one voip unit fails. The user must specify the IP address of the alternate route  
for each destination pattern entry in the Outbound Phonebook.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Alternate Routing Field Definitions  
Field  
Values  
Description  
Name  
Alternate n.n.n.n  
Alternate destination for outbound data traffic  
in case of excessive delay in data transmission.  
IP  
where  
Address  
n= 0-255  
Round  
Trip  
Delay  
milliseconds The Round Trip Delay is the criterion for  
judging when a data pathway is considered  
blocked. When the delay exceeds the  
threshold specified here, the data stream will  
be diverted to the alternate destination  
specified as the Alternate IP Address.  
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T1 PhoneBook Configuration  
4. Select PhoneBook Modify and then select Inbound PhoneBook/List  
Entries.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
5. The Add/Edit Inbound PhoneBook screen appears.  
Enter Inbound PhoneBook data for your MVP3010. The fields of the  
Add/ Edit Inbound PhoneBook screen are described in the table below.  
Add/Edit Inbound Phone Book: Field Definitions  
Field  
Values  
Description  
Name  
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)  
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.  
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T1 PhoneBook Configuration  
Add/Edit Inbound Phone Book: Field Definitions  
(cont’d)  
Field Name  
Values  
Description  
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.  
Forward  
Condition  
Uncondit.;  
Busy  
No Resp.  
Busy. When selected, calls  
will be forwarded when  
station is busy.  
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.  
Forward  
Address/  
Number  
IP addr. or  
phone  
number  
Phone number or IP  
address to which calls will  
be directed.  
Ring Count  
integer  
When No Response is  
condition for forwarding  
calls, this determines how  
many unanswered rings  
are needed to trigger the  
forwarding.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
6. When your Outbound and Inbound PhoneBook entries are  
completed, click on Save Setup in the sidebar menu to save your  
configuration.  
You can change your configuration at any time as needed for your  
system.  
Remember that the initial MVP3010 setup must be done locally using  
the MultiVOIP program. However, after the initial configuration is  
complete, all of the MVP3010 units in the VOIP system can be  
configured, re-configured, and updated from one location using the  
MultiVoipManager software program.  
E1 Phonebook Examples  
To demonstrate how Outbound and Inbound PhoneBook entries work  
in an international VOIP system, we will re-visit our previous example  
in greater detail. It’s an international company with offices in London,  
Paris, and Amsterdam. In each office, a MVP3010 has been connected  
to the PBX system.  
3 Sites, All-E1 Example  
The VOIP system will have the following features:  
1. Employees in all cities will be able  
to call each other over the VOIP  
system using 4-digit extensions.  
2. Calls to Outer London and Inner  
London, greater Amsterdam, and  
greater Paris will be accessible to all  
company offices as local calls.  
3. Vendors in Guildford, Lyon, and  
Rotterdam can be contacted as  
national calls by all company offices.  
Note that the phonebook entries for Series II analog MultiVOIP used in  
Euro-type telephony settings will be the same in format as entries for  
the MVP3010.  
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T1 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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T1 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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T1 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  
T1 PhoneBook Configuration  
Configuring Digital & Analog VOIPs in Same System  
The MVP3010 digital MultiVOIP unit is compatible with analog VOIPs.  
In many cases, digital and analog VOIP units will appear in the same  
telephony/ IP system. In addition to MVP-210/ 410/ 810 MultiVOIP  
units (Series II units), legacy analog VOIP units (Series I units made by  
MultiTech) may be included in the system, as well. When legacy VOIP  
units are included, the VOIP administrator must handle two styles of  
phonebooks in the same VOIP network. The diagram below shows a  
small-scale system of this kind: one digital VOIP (the MVP3010)  
operates with two Series II analog VOIPs (an MVP210 and an  
MVP410), and two Series I legacy VOIPs (two MVP200 units).  
EXAMPLE:  
Digital & An alog VOIPs  
in Sam e 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  
CO Port  
(Holds phonebook for both  
Series #1 analog VOIPs.)  
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.  
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T1 PhoneBook Configuration  
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  
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.  
0207  
xxx  
xxxx  
0
Gives remote voip users  
(Note 2.) access to phone numbers  
in 0207 area code (Inner  
London) in which Site D  
is located.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
0208  
xxx  
xxxx  
200.2.9.9  
0
Gives remote voip users  
(Note 2.) access to phone numbers  
in 0208 area code (Outer  
London) for which calls  
are local from Site D  
(Inner London).  
Note 1. The “x” is a wildcard character.  
Note 2. By specifying “Channel 0,” we instruct the MVP3010 to  
choose any available data channel to carry the call.  
Note 3. Note that Site F key system has only 30 extensions  
(x7400-7429). This destination pattern (018226374) actually  
directs calls to 402-263-7430 through  
402-263-7499 into the key system, as well.  
This means that such calls, which belong on the PSTN, cannot be  
completed. In some cases, this might be inconsequential because  
an entire exchange (fully used or not) might have been reserved  
for the company or it might be unnecessary to reach those  
numbers. However, to specify only the 30 lines actually used by  
the key system, the destination pattern 018226374 would have to  
be replaced by three other destination patterns, namely  
0182263740, 0182263741, and 0182263742. In this way, calls to  
0182-263-7430 through 0182-263-7499 would be properly directed  
to the PSTN. In the Site D outbound phonebook, the 30 lines are  
defined exactly, that is, without making any adjacent phone  
numbers unreachable through the voip system.  
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MultiVOIP User Guide  
T1 PhoneBook Configuration  
The Outbound PhoneBook of the MVP3010 is shown below.  
Outbound Phone Book for MVP3010 Digital VOIP (Site D)  
Destin.  
Pattern  
Remov  
e
Add  
Prefix  
IP  
Comment  
Address  
Prefix  
201  
200.2.9.7 To originate calls to Site A  
(Birmingham).  
901189  
901189  
101#  
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 (Reading, UK).  
421  
90182  
--  
--  
200.2.9.6 Calls to Site E (Carlisle).  
Calls to Tavistock local  
PSTN (Site F) could be  
arranged by operator or  
possibly by auto-attendant.  
200.2.9.5 Calls to extensions of key  
phone system at Tavistock  
office.  
90182  
263  
740  
90182  
263  
741  
9
9
9
--  
--  
--  
200.2.9.5  
90182  
263  
200.2.9.5  
742  
102  
200.2.9.8 To originate calls to phone  
connected to FXS port  
(channel #2) of the Site B  
VOIP (Reading).  
Note 3. The pound sign (“#”) is a delimiter separating the VOIP  
number from the standard telephony phone number.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
The Inbound PhoneBook of the MVP3010 is shown below.  
Inbound Phone Book for MVP3010 Digital VOIP (Site D)  
Remove  
Prefix  
Add  
Prefix  
Channel  
Number  
Comments  
0207  
9,7  
Note 4.  
Note 5.  
0
Allows phone users at remote voip sites  
to call local numbers (those within the  
Site D area code, 0207, Inner London)  
over the VOIP network.  
0208  
9,8  
0
0
Allows phone users at remote voip sites  
to call local numbers (those in Outer  
London) over the VOIP network.  
Allows phone users at remote voip sites  
to call extensions of the Site D PBX  
using three digits, beginning with “3” .  
Note 4.  
Note 5.  
3
0207  
39883  
Note 4. “9” gives PBX station users access to outside line.  
Note 5. The comma represents a one-second pause, the time  
required for the user to receive a dial tone on the outside line  
(PSTN). Commas can be used in the Inbound Phonebook, but not  
in the Outbound Phonebook.  
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T1 PhoneBook Configuration  
Outbound Phone Book for MVP410 Analog VOIP  
(Site F)  
IP  
Destin.  
Pattern  
201  
Remove Add  
Prefix  
Comment  
Prefix  
Address  
200.2.9.7 To originate calls  
to Site A  
(Birmingham).  
200.2.9.8 To originate calls  
to any PSTN  
01189  
0118  
101#  
Note 3.  
phone in Reading  
area using the  
FXO channel  
(channel #1) of the  
Site B VOIP.  
102  
200.2.9.8 To originate calls  
to phone  
connected to FXS  
port (channel #2)  
of the Site B VOIP  
(Reading).  
421  
200.2.9.6 Calls to Site E  
(Carlisle).  
0207  
200.2.9.9 Calls to Inner  
London area  
PSTN via Site D  
PBX.  
0208  
3
200.2.9.9 Calls to Inner  
London area  
PSTN via Site D  
PBX.  
200.2.9.9 Calls to Inner  
London PBX  
--  
0207  
398  
8
extensions with  
three digits.  
Note 3. The pound sign (“#”) is a delimiter separating the  
VOIP number from the standard telephony phone number.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Inbound Phonebook for MVP410 Analog VOIP (Site F)  
Remove Add  
Channel  
Number  
4
Comment  
Prefix  
Prefix  
01822  
2
Calls to Tavistock local  
PSTN through FXO port  
(Port #4) at Site F.  
0182  
263  
740  
0182  
263  
741  
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  
0182  
263  
complete these calls.  
742  
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T1 PhoneBook Configuration  
Outbound Phone Book for MVP210 Analog VOIP  
(Site E)  
IP  
Destin.  
Pattern  
201  
Remove Add  
Prefix  
Comment  
Prefix  
Address  
200.2.9.7 To originate calls  
to Site A  
(Birmingham).  
01189  
0118  
101#  
Note 3.  
200.2.9.8 To originate calls  
to any PSTN  
phone in Reading  
area using the  
FXO channel  
(channel #1) of the  
Site B VOIP.  
102  
200.2.9.8 To originate calls  
to phone  
connected to FXS  
port (channel #2)  
of the Site B VOIP  
(Reading).  
01822  
01822  
--  
200.2.9.5 Calls to Tavistock  
area PSTN (via  
FXO channel of  
the Site F VOIP).  
200.2.9.5 Calls to Tavistock  
key system  
0182  
26374  
operator or auto-  
attendant.  
0207  
8
0207  
200.2.9.9 Calls to London  
area PSTN via Site  
D PBX.  
200.2.9.9 Calls to London  
PBX extensions  
0207  
398  
with four digits.  
Note 3. The pound sign (“#”) is a delimiter separating the  
VOIP number from the standard telephony phone number.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
Inbound Phonebook for MVP210 Analog VOIP (Site E)  
Remove Add  
Channel  
Number  
1
Comment  
Prefix  
Prefix  
421  
Call Completion Summaries  
Site A calling Site C, Method 1  
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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T1 PhoneBook Configuration  
Site D calling Site C  
1. Dial 901189435632.  
2. “9” gets outside line. On some PBXs, an “8” may be used to  
direct calls to the VOIP, while “9” directs calls to the PSTN.  
However, some PBX units can be programmed to identify the  
destination patterns of all calls to be directed to the VOIP.  
3. PBX at Site D is programmed to divert all calls made to the 118  
area code and exchange 943 into the VOIP network. (It would  
also be possible to divert all calls to all phones in area code 118  
into the VOIP network, but it may not be desirable to do so.)  
4. The MVP3010 removes the prefix “0118” and adds the prefix  
“101#” for compatibility with the analog MultiVOIP’s  
phonebook scheme. The “#” is a delimiter separating the analog  
VOIP’s phone number from the digits that the analog VOIP  
must dial onto its local PSTN to complete the call. The digits  
“101#9435632” are forwarded to the Site B analog VOIP.  
5. The call passes through the IP network (in this case, the Internet).  
6. The call arrives at the Site B VOIP. This analog VOIP receives  
this dialing string from the MVP3010: 101#9435632. The analog  
VOIP, seeing the “101” prefix, uses its own channel #1 (an FXO  
port) to connect the call to the PSTN. Then the analog VOIP  
dials its local phone number 9435632 to complete the call.  
NOTE: In the case of Reading, Berkshire,,  
England, both “1189” and “1183” are  
considered local area codes. This is, in a  
sense however, a matter of terminology.  
It simply means that numbers of the  
form 9xx-xxxx and  
3xx-xxxx are both local calls for users at  
other sites in the VOIP network.  
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E1 Phonebook Configuration  
Site D calling Site F  
MultiVOIP User Guide  
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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Site F calling Site D  
T1 PhoneBook Configuration  
A voip call from a Tavistock key extension to extension 3117 on the PBX in Inner  
London.  
A. The required entry in the Tavistock Outbound Phonebook to facilitate  
origination of the call, would be “3”. The string 02073988 is added, preceding  
the “3”. The call would be directed to the Inner London voip’s IP address,  
200.2.9.9.  
B. The corresponding entry in the Inner-London Inbound Phonebook to  
facilitate completion of the call would be 020739883.  
1. The caller in Tavistock picks up the phone receiver, presses a  
button on the key phone set. This button has been assigned to a  
particular voip channel.  
2. The caller in Tavistock hears dial tone from the Tavistock voip.  
3. The caller in Tavistock dials 02073983117.  
4. The Tavistock voip sends the entire dialed string to the Inner-  
London voip  
at IP address 200.2.9.9.  
5. The Inner-London voip matches the called digits 02073983117to its  
Inbound Phonebook entry “020739883, ” which it removes. Then it  
adds back the “3” as a prefix.  
6. The Inner-London PBX dials extension 3117 in the office in Inner  
London.  
Variations in PBX Characteristics  
The exact dialing strings needed in the Outbound and Inbound  
Phonebooks of the MVP3010 will depend on the capabilities of the PBX.  
Some PBXs require trunk access codes (like an “8” or “9” to access an  
outside line or to access the VOIP network). Other PBXs can  
automatically distinguish between intra-PBX calls, PSTN calls, and  
VOIP calls.  
Some PBX units can also insert digits automatically when they receive  
certain dialing strings from a phone station. For example, a PBX may  
be programmable to insert automatically the three-digit VOIP identifier  
strings into calls to be directed to analog VOIPs.  
The MVP3010 offers complete flexibility for inter-operation with PBX  
units so that a coherent dialing scheme can be established to connect a  
company’s multiple sites together in a way that is convenient and  
intuitive for phone users. When working together with modern PBX  
units, the presence of the MVP3010 can be completely transparent to  
phone users within the company.  
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E1 Phonebook Configuration  
MultiVOIP User Guide  
International Telephony Numbering Plan Resources  
Due to the expansion of telephone number capacity to accommodate  
pagers, fax machines, wireless telephony, and other new phone  
technologies, numbering plans have been changing worldwide. Many  
new area codes have been established; new service categories have been  
established (for example, to accommodate GSM, personal numbering,  
corporate numbering, etc.). Below we list several web sites that present  
up-to-date information on the telephony numbering plans used around  
the world. While we find these to be generally good resources, we  
would note that URLs may change or become nonfunctional, and we  
cannot guarantee the quality of information on these sites.  
URL  
Description  
http:/ / phonebooth.interocitor.net  
/ wtng  
The World Telephone  
Numbering Guide  
presents excellent  
international  
numbering info that  
is both broad and  
detailed. This  
includes info on re-  
numbering plans  
carried out world-  
wide 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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URL  
T1 PhoneBook Configuration  
Description  
http:/ / kropla.com/ phones.htm  
Guide to  
international use of  
modems.  
http:/ / www.numberplan.org/  
National and  
international  
numbering plans  
based on direct input  
from regulators  
worldwide. Includes  
lists of telecom  
carriers per country.  
http:/ / www.eto.dk/  
European  
Telecommunications  
Office. Primarily  
concerned with  
mobile/ wireless  
radiotelephony,  
GSM, etc.  
http:/ / www.eto.dk/ ETNS.htm  
European Telephony  
Numbering Space.  
Resources for pan-  
European telephony  
services, standards,  
etc. Part of ETO site.  
http:/ / www.regtp.de/ en/ reg_tele/ start List of European  
/ fs_05.html  
telecom regulatory  
agencies by country  
(from German  
telecom authority).  
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Chapter 9: Analog Phonebook  
Configuration  
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MultiVOIP User Guide  
Analog Phonebook Configuration  
Phonebooks for Series II analog MultiVOIP units (MVP210, MVP410,  
and MVP810) are, in principle, configured the same as phonebooks for  
digital MultiVOIP products that would operate in the same  
environment (under either North American or European telephony  
standards, T1 or E1).  
Therefore, if you are operating an analog MultiVOIP unit in a North  
American telephony environment, you will find useful phonebook  
instructions and examples in Chapter 7: T1 Phonebook Configuration. If  
you are operating an analog MultiVOIP unit in a European telephony  
environment, you will find useful phonebook instructions and  
examples in Chapter 8: E1 Phonebook Configuration.  
Most of the examples in Chapters 7 and 8 describe systems containing  
both digital and analog MultiVOIP units.  
You will also find useful information in Chapter 2: Quick Start Guide.  
See especially these sections:  
Phonebook Starter Configuration  
Phonebook Tips  
Phonebook Example (One Common Situation)  
Chapter 2 also contains a “Phonebook Worksheet” section. You may  
want to print out several worksheet copies. Paper copies can be very  
helpful in comparing phonebooks at multiple sites at a glance. This will  
assist you in making the phonebooks clear and consistent and will  
reduce ‘surfing’ between screens on the configuration program.  
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Operation and Maintenance  
MultiVOIP User Guide  
Chapter 10: Operation and  
Maintenance  
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MultiVOIP User Guide  
Operation and Maintenance  
Operation and Maintenance  
Although most Operation and Maintenance functions of the software  
are in the Statistics group of screens, an important summary appears in  
the System Information of the Configuration screen group.  
System Information screen  
This screen presents vital system information at a glance. Its primary  
use is in troubleshooting. This screen is accessible via the  
Configuration pulldown menu, the Configuration sidebar menu, or by  
the keyboard shortcut Ctrl + Alt + Y.  
System Information Parameter Definitions  
Field Name Values  
Description  
Boot Code  
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.  
Mac  
Address  
alpha-  
numeric  
Denotes the number assigned as the  
voip unit’s unique Ethernet address.  
Up Time  
hours:  
mm:ss  
Indicates how long the voip has been  
running since its last booting.  
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Operation and Maintenance  
MultiVOIP User Guide  
The frequency with which the System Information screen is updated is  
determined by a setting in the Logs screen  
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Operation and Maintenance  
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.  
Pulldown  
Icon  
Shortcut  
Sidebar  
Alt + A  
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Operation and Maintenance  
MultiVOIP User Guide  
The Call Progress Details Screen  
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Operation and Maintenance  
Call Progress Details: Field Definitions  
Field Name  
Values  
Description  
Channel  
1-n  
Number of data channel or time  
slot on which the call is carried.  
This is the channel for which call-  
progress details are being viewed.  
Call Details  
Duration  
Mode  
Hours:  
Minutes:  
Seconds  
The length of the call in hours,  
minutes, and seconds (hh:mm:ss).  
Indicates whether the call being  
described was a voice call or a  
FAX call.  
Voice or FAX  
Voice Coder  
Packets Sent  
G.723, G.729,  
G.711, etc.  
The voice coder being used on  
this call.  
integer value  
integer value  
integer value  
integer value  
integer value  
0-9, #, *  
The number of data packets sent  
over the IP network in the course  
of this call.  
Packets Rcvd  
Bytes Sent  
The number of data packets  
received over the IP network in  
the course of this call.  
The number of bytes of data sent  
over the IP network in the course  
of this call.  
Bytes Rcvd  
Packets Lost  
The number of bytes of data  
received over the IP network in  
the course of this call.  
The number of voice packets from  
this call that were lost after being  
received from the IP network.  
Outbound  
Digits  
The digits transmitted by the  
MultiVOIP to the PBX/ telco for  
this call.  
Prefix  
Matched  
Displays the dialed digits that  
were matched to a phonebook  
entry.  
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MultiVOIP User Guide  
Call Progress Details: Field Definitions (cont’d)  
From – To Details  
Description  
Gateway  
Name  
alphanumeric  
string  
Identifier for the VOIP gateway  
that handled this call.  
IP Address  
x.x.x.x,  
IP address from which the call  
was received.  
where x has a  
range of 0 to  
255  
Options  
SC, FEC  
Displays VOIP transmission  
options in use on the current call.  
These may include Forward Error  
Correction or Silence  
Compression.  
Silence  
SC  
“SC” stands for Silence  
Compression  
Compression. With Silence  
Compression enabled, the  
MultiVOIP will not transmit voice  
packets when silence is detected,  
thereby reducing the amount of  
network bandwidth that is being  
used by the voice channel.  
Forward Error FEC  
Correction  
“FEC” stands for Forward Error  
Correction. Forward Error  
Correction enables some of the  
voice packets that were corrupted  
or lost to be recovered. FEC adds  
an additional 50% overhead to the  
total network bandwidth  
consumed by the voice channel.  
Default = Off  
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Operation and Maintenance  
Call Progress Details: Field Definitions (cont’d)  
Field Name Values Description  
Supplementary Services  
Status  
Call on Hold  
alphanumeric  
Describes held call by its IP  
address source, location/ gateway  
identifier, and hold duration.  
Location/ gateway identifiers  
comes from Gateway Name field  
in Phone Book Configuration  
screen of remote voip.  
Call Waiting  
alphanumeric  
Describes waiting call by its IP  
address source, location/ gateway  
identifier, and hold duration.  
Location/ gateway identifiers  
comes from Gateway Name field  
in Phone Book Configuration  
screen of remote voip.  
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MultiVOIP User Guide  
Call Progress Details: Field Definitions (cont’d)  
Field Name Values Description  
Supplementary Services  
Status  
Caller ID  
There are four  
values:  
“Calling Party  
+ identifier”;  
“Alerting  
This field shows the identifier and  
status of a remote voip (which has  
Call Name Identification enabled)  
with which this voip unit is  
currently engaged in some voip  
transmission. The status of the  
engagement (Connected, Alerting,  
Busy, or Calling) is followed by  
the identifier of a specific channel  
of a remote voip unit. This  
Party +  
identifier”;  
“Busy Party  
+ identifier”;  
and  
identifier comes from the “Caller  
Id” field in the Supplementary  
Services screen of the remote  
voip unit.  
“Connected  
Party +  
identifier”  
Status  
hangup, active  
Shows condition of current call.  
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 signalling transport  
(AE), and Q.931 Multiplexing  
(Mux). See Phonebook  
Configuration Parameters (in T1  
or E1 chapters) for more on  
H.323v4 features.  
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Operation and Maintenance  
About Logs  
Accessing “Statistics: Logs”  
Pulldown  
Icon  
Shortcut  
Sidebar  
Alt + L  
The Logs Screen  
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Logs Screen Details: Field Definitions  
Field Name  
Values  
Description  
Event # column  
1 or higher  
All calls are assigned an event  
number in chronological order,  
with the most recent call having  
the highest event number.  
Start Date,Time  
column  
dd:mm:yyyy  
hh:mm:ss  
The starting time of the call  
(event). The date is presented as  
a day expression of one or two  
digits, a month expression of one  
or two digits, and a four-digit  
year. This is followed by a time-  
of-day expression presented as a  
two-digit hour, a two-digit  
minute, and a two-digit seconds  
value. (statistics, logs) field  
This describes how long the call  
(event) lasted in hours, minutes, and  
seconds.  
Duration column hh:mm:ss  
Status column  
Mode column  
success or  
failure  
Displays the status of the call,  
i.e., whether the call was  
completed successfully or not.  
Indicates whether the (event)  
being described was a voice call  
or a FAX call.  
voice or FAX  
From column  
To column  
gateway name  
gateway name  
Displays the name of the voice  
gateway that originates the call.  
Displays the name of the voice  
gateway that completes the call.  
Special Buttons  
Last  
Displays last log entry.  
Deletes selected log file.  
Delete File  
Call Details  
Packets sent  
integer value  
The number of data packets sent  
over the IP network in the course  
of this call.  
Bytes sent  
integer value  
The number of bytes of data sent  
over the IP network in the course  
of this call.  
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Logs Screen Details: Field Definitions (cont’d)  
Field Name Values Description  
Call Details (cont’d)  
Packets loss  
(lost)  
integer value  
The number of voice packets  
from this call that were lost  
after being received from the IP  
network.  
Voice coder  
G.723, G.729,  
G.711, etc.  
The voice coder being used on  
this call.  
Packets received  
integer value  
The number of data packets  
received over the IP network in  
the course of this call.  
Bytes received  
integer value  
0-9, #, *  
The number of bytes of data  
received over the IP network in  
the course of this call.  
Outbound digits  
The digits transmitted by the  
MultiVOIP to the PBX/ telco for  
this call.  
FROM Details  
Gateway Name  
alphanumeric  
string  
Identifier for the VOIP gateway  
that originated this call.  
IP address of the VOIP gateway  
from which the call was  
received.  
IP Address  
x.x.x.x,  
where x has a  
range of 0 to 255  
FEC, SC  
Options  
Displays VOIP transmission  
options used by the VOIP  
gateway originating the call.  
These may include Forward  
Error Correction or Silence  
Compression.  
TO Details  
Gateway Name  
IP Address  
Options  
alphanumeric  
string  
Identifier for the VOIP gateway  
that completed (terminated)  
this call.  
x.x.x.x,  
where x has a  
range of 0 to 255  
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.  
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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.  
CT Ph#  
phone number  
string  
Call Transfer phone number.  
About Reports  
This feature not implemented as of this writing.  
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About IP Statistics  
Accessing IP Statistics  
Pulldown  
Icon  
Shortcut  
Sidebar  
Alt + I  
IP Statistics Screen  
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IP Statistics: Field Definitions  
Field  
Values  
Description  
Name  
UDP versus TCP. (User Datagram  
Protocol versus Transmission Control  
Protocol). UDP provides  
unguaranteed, connectionless  
transmission of data across an IP  
network. By contrast, TCP provides  
reliable, connection-oriented  
transmission of data.  
Both TCP and UDP split data into  
packets called “datagrams.” However,  
TCP includes extra headers in the  
datagram to enable retransmission of  
lost packets and reassembly of packets  
into their correct order if they arrive out  
of order. UDP does not provide this.  
Lost UDP packets are unretrievable;  
that is, out-of-order UDP packets  
cannot be reconstituted in their proper  
order..  
Despite these obvious disadvantages,  
UDP packets can be transmitted much  
faster than TCP packets -- as much as  
three times faster. In certain  
applications, like audio and video data  
transmission, the need for high speed  
outweighs the need for verified data  
integrity. Sound or pictures often  
remain intelligible despite a certain  
amount of lost or disordered data  
packets (which appear as static).  
Clears packet tallies from memory.  
“Clear”  
button  
--  
Total Packets  
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  
Transmit  
integer  
value  
ted  
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  
with  
Errors  
integer  
value  
Total number of error-laden packets  
received by this VOIP gateway since the  
last “clearing” or resetting of the  
counter within the MultiVOIP software.  
UDP Packets  
User Datagram Protocol packets.  
Transmit  
ted  
integer  
value  
Number of UDP packets transmitted by  
this VOIP gateway since the last  
“clearing” or resetting of the counter  
within the MultiVOIP software.  
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  
with  
Errors  
integer  
value  
Number of error-laden UDP packets  
received by this VOIP gateway since the  
last “clearing” or resetting of the  
counter within the MultiVOIP software.  
TCP Packets  
Transmission Control Protocol packets.  
Transmit  
ted  
integer  
value  
Number of TCP packets transmitted by  
this VOIP gateway since the last  
“clearing” or resetting of the counter  
within the MultiVOIP software.  
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  
with  
Errors  
integer  
value  
Number of error-laden TCP packets  
received by this VOIP gateway since the  
last “clearing” or resetting of the  
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  
ted  
integer  
value  
Number of RTP packets transmitted by  
this VOIP gateway since the last  
“clearing” or resetting of the counter  
within the MultiVOIP software.  
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  
with  
Errors  
integer  
value  
Number of error-laden RTP packets  
received by this VOIP gateway since the  
last “clearing” or resetting of the  
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  
ted  
integer  
value  
Number of RTCP packets transmitted  
by this VOIP gateway since the last  
“clearing” or resetting of the counter  
within the MultiVOIP software.  
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  
with  
Errors  
integer  
value  
Number of error-laden RTCP packets  
received by this VOIP gateway since the  
last “clearing” or resetting of the  
counter within the MultiVOIP software.  
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About T1/E1 Statistics  
Accessing T1 Statistics  
Pulldown  
Icon  
Shortcut  
Sidebar  
Alt + T  
The T1 and E1 Statistics screens are only accessible and applicable for the  
MVP2400, MVP2410, and MVP3010.  
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T1 Statistics Screen  
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T1 Statistics: Field Definitions  
Field  
Values  
Description  
Name  
Red Alarm  
Integer tally of The alarm condition declared when a  
alarms  
counted since  
last reset.  
device receives no signal or cannot  
synchronize to the signal being received. A  
Red Alarm is generated if the incoming  
data stream has no transitions for 176  
consecutive pulse positions.  
Blue Alarm  
Tally since last  
reset.  
Alarm signal consisting of all 1’s (including  
framing bit positions) which indicates  
disconnection or failure of attached equipment.  
Loss of  
Frame  
Tally since last  
reset.  
Loss of data frame synchronization.  
Alignment  
Excessive  
Zeroes  
Tally since last  
reset.  
Displayed value will increment if  
consecutive zeroes beyond a set threshold  
are detected. I.e., tally increments if more  
than 7 consecutive zeroes in the received  
data stream are detected under B8ZS line  
coding, or if 15 consecutive zeroes are  
detected under AMI line coding.  
Signaling has been frozen at the most  
recent values due to loss of frame  
alignment, loss of multiframe  
Status  
Freeze  
Signalling  
Active  
alignment or due to a receive slip.  
Line loopback deactivation signal has  
been detected in the receive bit stream.  
Line  
Loopback  
Deactivation  
Signal  
A short exists between the transmit pair for  
at least 32 consecutive pulses.  
Transmit  
Line Short  
For use by MTS Technical Support  
personnel.  
Transmit  
Data  
Overflow  
The frequency of the transmit clock is less  
than the frequency of the transmit system  
interface working clock. A frame is  
repeated.  
Transmit  
Slip Positive  
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T1 Statistics: Field Definitions (cont’d)  
Field  
Values  
Description  
Name  
Yellow  
Alarm  
Tally since last  
reset.  
The alarm signal sent by a remote T1/ E1  
device to indicate that it sees no receive  
signal or cannot synchronize on the  
receive signal.  
[To be supplied.]  
Frame  
Search  
Restart Flag  
Loss of  
MultiFrame  
Alignment  
Tally since last  
reset.  
In D4 or ESF mode, displayed value will  
increment if multiframe alignment has  
been lost or if loss of frame alignment has  
been detected.  
Transmit  
Slip  
Tally since last  
reset.  
Slip in transmitted data stream. Slips  
indicate a clocking mismatch (or lack of  
synchronization) between T1/ E1 devices.  
When slips occur, data may be lost or  
repeated.  
Pulse  
Density  
Violation  
The pulse density of the received data  
stream is below the requirement defined  
by ANSI T1.403 or more than 15  
consecutive zeros are detected.  
Line  
The line loopback activation signal  
has been detected in the received bit  
stream.  
Loopback  
Activation  
Signal  
Transmit  
Line Open  
At least 32 consecutive zeros were  
transmitted.  
Transmit  
Data  
For use by MTS Technical Support  
Personnel.  
Underrun  
Transmit  
Slip  
Negative  
The frequency of the transmit clock is  
greater than the frequency of the  
transmit system interface working  
clock. A frame is skipped.  
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T1 Statistics: Field Definitions (cont’d)  
Field  
Values  
Description  
Name  
Two successive pulses of the same  
polarity have been received and these  
pulses are not part of zero substitution.  
On an AMI-encoded line, this represents a  
line error. On a B8ZS line, this may  
represent the substitution for a string of 8  
zeroes.  
Bipolar  
Violation  
Integer tally of  
violation count  
since last reset.  
Receive Slip  
Tally since last  
reset.  
A receive slip (positive or negative) has  
occurred. Slips indicate a clocking  
mismatch (or lack of synchronization)  
between T1/ E1 devices. When slips occur,  
data may be lost or repeated.  
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E1 Statistics Screen  
MultiVOIP User Guide  
E1 Statistics: Field Definitions  
Description  
Field  
Values  
Name  
Red Alarm  
Integer tally of The alarm condition declared when a  
alarms  
counted since  
last reset.  
device receives no signal or cannot  
synchronize to the signal being received. A  
Red Alarm is generated if the incoming  
data stream has no transitions for 176  
consecutive pulse positions.  
Blue Alarm  
Tally since last  
reset.  
Alarm signal consisting of all 1’s (including  
framing bit positions) which indicates  
disconnection or failure of attached  
equipment.  
Loss of  
Frame  
Tally since last  
reset.  
Loss of data frame synchronization.  
Alignment  
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E1 Statistics: Field Definitions  
(cont’d)  
Field  
Values  
Description  
Name  
Detected alarm indication signal in  
timeslot 16 according to ITU-T G.775.  
Indicates the incoming time slot 16  
contains less than 4 zeros in each of  
two consecutive time slot 16  
Receive  
Timeslot 16  
Alarm  
Indication  
Signal  
multiframe periods.  
Transmit  
Line Short  
A short exists between the transmit  
pair for at least 32 consecutive pulses.  
Transmit  
Data  
For use by MTS personnel.  
Overflow  
Transmit  
Slip Positive  
The frequency of the transmit clock is  
less than the frequency of the transmit  
system interface working clock. A  
frame is repeated.  
Yellow  
Alarm  
Tally since last  
reset.  
The alarm signal sent by a remote T1/ E1  
device to indicate that it sees no receive  
signal or cannot synchronize on the  
receive signal.  
Status  
Freeze  
Signalling  
Active  
Signaling has been frozen at the most  
recen tvalues due to loss of frame  
alignment, loss of multiframe alignment  
or due to a receive slip.  
Loss of  
MultiFrame  
Alignment  
Tally since last  
reset.  
In D4 or ESF mode, displayed value will  
increment if multiframe alignment has  
been lost or if loss of frame alignment has  
been detected.  
Receive  
Timeslot 16  
Loss of  
The time slot 16 data stream contains all  
zeros for at least 16 contiguously received  
time slots.  
Signal  
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E1 Statistics: Field Definitions  
(cont’d)  
Field  
Values  
Description  
Name  
Receive  
The framing pattern '0000' in 2  
Timeslot 16  
Loss of  
MultiFrame  
Alignment  
consecutive CAS multiframes were not  
found or in all time slot 16 of the previous  
multiframe all bits were reset.  
At least 32 consecutive zeroes were  
transmitted.  
Transmit  
Line Open  
For use by MTS Technical Support  
Personnel.  
Transmit  
Data  
Underrun  
The frequency of the transmit clock is  
greater than the frequency of the transmit  
system interface working clock. A frame  
is skipped.  
Transmit  
Slip  
Negative  
Bipolar Violation (or BPV) refers to two  
successive pulses of the same polarity on  
the E1 line. On an AMI-encoded line, this  
represents a line error. On a B8ZS line,  
this may represent the substitution for a  
string of 8 zeroes.  
Bipolar  
Violation  
Integer tally of  
violation count  
since last reset.  
Excessive  
Zeroes  
Tally since last  
reset.  
Displayed value will increment if  
consecutive zeroes beyond a set threshold  
are detected. I.e., tally increments if more  
than 7 consecutive zeroes in the received  
data stream are detected under B8ZS line  
coding, or if 15 consecutive zeroes are  
detected under AMI line coding.  
Transmit  
Slip  
Tally since last  
reset.  
Slip in transmitted data stream. Slips  
indicate a clocking mismatch (or lack of  
synchronization) between T1/ E1 devices.  
When slips occur, data may be lost or  
repeated.  
Receive Slip  
Tally since last  
reset.  
Slip in received data stream. Slips indicate a  
clocking mismatch (or lack of synchronization)  
between T1/ E1 devices. When slips occur, data  
may be lost or repeated.  
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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.  
Date and Time Setup  
Select this for access to set  
calendar/ clock used for data logging.  
Download CAS Protocol  
Telephony CAS files are for Channel  
Associated Signalling. There are many  
CAS files, some labeled for specific  
functionality, others for countries or  
regions where certain telephony attributes  
are standard.  
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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 User Defaults  
To be used after a full set of parameter  
values, values specified by the user,  
have been saved (using Save Setup).  
This command loads the saved user  
defaults into the MultiVOIP.  
Set Password  
Select this to create a password for  
access to the MultiVOIP software  
programs (Program group commands,  
Windows GUI, web browser GUI, &  
FTP server). Only the FTP Server  
function requires a password for access.  
The FTP Server function also requires  
that a UserName be established along  
with the password.  
Uninstall  
Select this to uninstall the MultiVOIP  
software (most, but not all components  
are removed from computer when this  
command is invoked).  
Upgrade Software  
Loads firmware (including H.323  
stack) and factory default settings from  
the controller PC to the MultiVOIP  
unit.  
“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.  
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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.  
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  
Beginning with the 4.03/ 6.03 software release, 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 commmand.  
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Download Firmware transfers the firmware (including the H.323  
protocol stack) in the PC’s MultiVOIP directory into the nonvolatile  
flash memory of the MultiVOIP.  
Download Factory Defaults sets all configuration parameters to the  
standard default values that are loaded at the MultiTech factory.  
Upgrade Software implements both the Download Firmware  
command and the Download Factory Defaults command.  
Downloading Firmware  
1. The MultiVoip Configuration program must be off when invoking  
the Download Firmware command. If it is on, the command will  
not work.  
2. To invoke the Download Factory Defaults command, go to Start |  
Programs | MVP____ x.xx | Download Firmware.  
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3. If a password has been established, the Password Verification screen  
will appear.  
Type in the password and click OK.  
4. The MultiVOIP ___- Firmware screen appears saying  
“MultiVOIP [model number] is up. Reboot to Download Firmware?”  
Click OK to download the firmware.  
The “Boot” LED on the MultiVOIP will light up and remain lit during  
the file transfer process.  
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5. The program will locate the firmware “.bin” file in the MultiVOIP  
directory. Highlight the correct (newest) “.bin” file and click Open.  
6. Progress bars will appear at the bottom of the screen during the file  
transfer.  
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.  
7. The Download Firmware procedure is complete.  
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Downloading CAS Protocol  
1. The MultiVoip Configuration program must be off when invoking  
the Download CAS Protocol command. If it is on, the command  
will not work.  
2. To invoke the Download H.323 PDL command, go to Start |  
Programs | MVP____ x.xx | Download H.323 PDL.  
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3. If a password has been established, the Password Verification screen  
will appear.  
Type in password and click OK.  
4. The MultiVOIP ____- Firmware screen appears saying “MultiVOIP  
[model number] is up. Reboot to Download Firmware?”  
Click OK to download the CAS Protocol file(s) to the MultiVOIP.  
The “Boot” LED on the MultiVOIP will light up and remain lit during  
the file transfer process.  
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5. The program will locate the CAS protocol file in the MultiVOIP  
directory. Highlight the correct (newest) file and click Open.  
6. Progress bars will appear at the bottom of the screen during the file  
transfer.  
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.  
7. The Download CAS Protocol 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.  
3. If a password has been established, the Password Verification screen  
will appear.  
Type in the password and click OK.  
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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.  
5. After the PC gets a response from the MultiVOIP, the Dialog – IP  
Parameters screen will appear.  
The user should verify that the correct IP parameter values are listed  
on the screen and revise them if necessary. Then click OK.  
6. Progress bars will appear at the bottom of the screen during the data  
transfer.  
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.  
7. The Download Factory Defaults procedure is complete.  
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Setting and Downloading User Defaults  
The Download User Defaults commandallows you to maintain a known  
working configuration that is specific to your VOIP system. You can then  
experiment with alterations or improvements to the configurations confident  
that a working configuration can be restored if necessary.  
1. Before you can invoke the Download User Defaults command, you  
must first save a set of configuration parameters by using the Save  
Setup command in the sidebar menu of the MultiVOIP software.  
2. Before the setup configuration is saved, you will be prompted to save  
the setup as the User Default Configuration. Select the checkbox and  
click OK.  
Save Current Setup as User Default Configuration  
MultiVOIP _____ will be brought down.  
OK  
Cancel  
Help  
A user default file will be created.  
3. The MVP____- Firmware screen appears saying “MultiVOIP [model  
number] is up. Reboot to Download Firmware?”  
Click OK to download the factory defaults. The “Boot” LED on the  
MultiVOIP will light up and remain lit during the file transfer  
process.  
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4. Progress bars will appear during the file transfer process.  
5. When the file transfer process is complete, the Dialog-- IP  
Parameters screen will appear.  
6. Set the IP values per your particular VOIP system. Click OK.  
Progress bars will appear as the MultiVOIP reboots itself.  
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Setting a Password (Windows GUI)  
After a user name has been designated and a password has been set,  
that password is required to gain access to any functionality of the  
MultiVOIP software. Only one user name and password can be  
assigned to a voip unit. The user name will be required when  
communicating with the MultiVOIP via the web browser GUI.  
NOTE: Record your user name and password in a safe place. If  
the password is lost, forgotten, or unretrievable, the user  
must contact MultiTech Tech Support in order to resume  
use of the MultiVOIP unit.  
1. The MultiVoip configuration program must be off when invoking  
the Set Password command. If it is on, the command will not work.  
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 H323 stack) and factory  
default configuration settings. As such, Upgrade Software implements  
the functions of both Download Firmware and Download Factory  
Defaults in a single command.  
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FTP Server File Transfers (“Downloads”)  
With the 4.03/ 6.03 software release, MultiTech has built an FTP server  
into the MultiVOIP unit. Therefore, file transfers from the controller PC  
to the voip unit can be done using an FTP client program or even using  
a browser (e.g., Internet Explorer or Netscape, used in conjunction with  
Windows Explorer).  
The terminology of “downloads” and “uploads” gets a bit confusing in  
this context. File transfers from a client to a server are typically  
considered “uploads.” File transfers from a large respository 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.  
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,  
These telephony files are  
em_winkFtp.cas, for Channel Associated  
r2_brazilFtp.cas  
r2_chinaFtp.cas  
Signalling. The directory  
contains many CAS files,  
some labeled for specific  
functionality, others for  
countries or regions where  
certain attributes are  
standard.  
H323 PDL file  
This file is specific to  
the particular version of  
the H.323 standard  
being used. This file  
rarely needs to be  
updated.  
inbound  
phonebook  
InPhBk.tmr  
This file updates the  
inbound phonebook in  
the MultiVOIP unit.  
outbound  
phonebook  
OutPhBk.tmr  
This file updates the  
outbound phonebook in  
the MultiVOIP unit.  
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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  
Beginning with the 4.03/ 6.03 software release, you can control the  
MultiVOIP unit with a graphic user interface (GUI) based on the  
common web browser platform. Qualifying browsers are  
InternetExplorer6 and Netscape6.  
MultiVOIP Web Browser GUI Overview  
Function  
Remote configuration and control  
of MultiVOIP units.  
Configuration  
Prerequisite  
Local Windows GUI must be used  
to assign IP address to MultiVOIP.  
Browser Version  
Requirement  
Internet Explorer 6.0 or higher  
Netscape 6.0 or higher  
Java Requirement  
Java Runtime 1.0 or higher  
(application program included with  
MultiVOIP)  
Video Useability  
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.  
The web GUI, however, cannot perform logging in the same direct  
mode done in the Windows GUI. However, when the web GUI is used,  
logging can be done by email (SMTP).  
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The graphic layout of the web GUI is also somewhat larger-scale than  
that of the Windows GUI. For that reason, it’s helpful to use as large of  
a video monitor as possible.  
The primary advantage of the web GUI is remote access for control and  
configuration. The controller PC and the MultiVOIP unit itself must  
both be connected to the same IP network and their IP addresses must  
be known.  
In order to use the web GUI, you must also install a Java application  
program on the controller PC. This Java program is included on the  
MultiVOIP product CD. ). Java is needed to support drop-down menus  
and multiple windows in the web GUI.  
To install the Java program, go to the Java directory on the MultiVOIP  
product CD. Double-click on the EXE file to begin the installation.  
Follow the instructions on the Install Shield screens.  
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During the installation, you must specify which browser you’ll use in  
the Select Browsers screen.  
When installation is complete, the Java program becomes accessible in  
your Start | Programs menu (Java resources are readily available via  
the web). However, the Java program runs automatically in the  
background as a plug-in supporting the MultiVOIP web GUI. No overt  
user actions are required.  
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After the Java program has been installed, you can access the  
MultiVOIP using the web browser GUI. Close the MultiVOIP  
Windows GUI. Start the web browser. Enter the IP address of the  
MultiVOIP unit. Enter a password when prompted. (A password is  
needed here only if password has been set for the local Windows GUI  
or for the MultiVOIP’s FTP Server function. See “Setting a Password --  
Web Browser GUI” earlier in this chapter.) The web browser GUI  
offers essentially the same control over the voip as can be achieved  
using the Windows GUI. As noted earlier, logging functions cannot be  
handled via the web GUI. And, because network communications will  
be slower than direct communications over a serial PC cable, command  
execution will be somewhat slower over the web browser GUI than  
with the Windows GUI.  
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SysLog Server Functions  
Beginning with the 4.03/ 6.03 software release, we have built SysLog  
server functionality into the software of the MultiVOIP units. SysLog is  
a de facto standard for logging events in network communication  
systems.  
The SysLog Server resides in the MultiVOIP unit itself. To implement  
this functionality, you will need a SysLog client program (sometimes  
referred to as a “daemon”). SysLog client programs, both paid and  
freeware, can be obtained from Kiwi Enterprises, among other firms.  
Read the End-User License Agreement carefully and observe license  
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 customisable  
options available.”  
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Before a SysLog client program is used, the SysLog functionality must  
be enabled within the MultiVOIP in the Logs menu under  
Configuration.  
The IP Address used will be that of the MultiVOIP itself.  
In the Port field, entered by default, is the standard (‘well-known’)  
logical port, 514.  
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Configuring the SysLog Client Program. Configure the SysLog client  
program for your own needs. In various SysLog client programs, you  
can define where log messages will be saved/ archived, opt for  
interaction with an SNMP system (like MultiVoipManager), set the  
content and format of log messages, determine disk space allocation  
limits for log messages, and establish a hierarchy for the seriousness of  
messages (normal, alert, critical, emergency, etc.). A sample  
presentation of SysLog info in the Kiwi daemon is shown below.  
SysLog programs will vary in features and presentation.  
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Chapter 11: Warranty, Service, and  
Tech Support  
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Limited Warranty  
Multi-Tech Systems, Inc. (“MTS”) warrants that its products will be free  
from defects in material or workmanship for a period of two years from  
the date of purchase, or if proof of purchase is not provided, two years  
from date of shipment. MTS MAKES NO OTHER WARRANTY,  
EXPRESSED OR IMPLIED, AND ALL IMPLIED WARRANTIES OF  
MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE  
ARE HEREBY DISCLAIMED. This warranty does not apply to any  
products which have been damaged by lightning storms, water, or  
power surges or which have been neglected, altered, abused, used for a  
purpose other than the one for which they were manufactured, repaired  
by the customer or any party without MTS’s written authorization, or  
used in any manner inconsistent with MTS’s instructions.  
MTS’s entire obligation under this warranty shall be limited (at MTS’s  
option) to repair or replacement of any products which prove to be  
defective within the warranty period, or, at MTS’s option, issuance of a  
refund of the purchase price. Defective products must be returned by  
Customer to MTS’s factory—transportation prepaid.  
MTS WILL NOT BE LIABLE FOR CONSEQUENTIAL DAMAGES  
AND UNDER NO CIRCUMSTANCES WILL ITS LIABILITY EXCEED  
THE PURCHASE PRICE FOR DEFECTIVE PRODUCTS.  
Repair Procedures for U.S. and Canadian  
Customers  
In the event that service is required, products may be shipped, freight  
prepaid, to our Mounds View, Minnesota factory:  
Multi-Tech Systems, Inc.  
2205 Woodale Drive  
Mounds View, MN 55112  
Attn: Repairs, Serial # ________________  
A Returned Materials Authorization (RMA) is not required. Return  
shipping charges (surface) will be paid by MTS.  
Please include, inside the shipping box, a description of the problem, a  
return shipping address (it must be a street address, not a P.O. Box  
number), your telephone number, and if the product is out of warranty,  
a check or purchase order for repair charges.  
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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  
tsupport@multitech.com. Please direct your questions regarding repair  
expediting, receiving, shipping, billing, etc., to our Repair Accounting  
department at (800) 328-9717 or (763) 717-5631, or email  
mtsrepair@multitech.com.  
Repairs for damages caused by lightning storms, water, power surges,  
incorrect installation, physical abuse, or used-caused damages are  
billed on a time-plus-materials basis.  
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Warranty, Service, & Tech Support  
MultiVOIP User Guide  
Technical Support  
Multi-Tech Systems has an excellent staff of technical support personnel  
available to help you get the most out of your Multi-Tech product. If  
you have any questions about the operation of this unit, or experience  
difficulty during installation you can contact Tech Support via the  
following:  
Contacting Technical Support  
Country By E-mail  
By telephone  
France  
India  
U.K.  
support@multitech.fr  
(33) 1-64 61 09  
81  
support@  
multitechindia.com  
(91) 124-340778  
(44) 118 959 7774  
(800) 972-2439  
(763) 785-3500  
support@  
multitech.co.uk  
U.S. &  
Canada  
tsupport@  
multitech.com  
Rest of  
World  
support@  
multitech.com  
Internet: http:/ / www.multitech.com/ _forms/email_tech_support.htm  
Please have your product information available, including model and  
serial number.  
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Chapter 12: Regulatory Information  
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Regulatory Information  
MultiVOIP User Guide  
EMC, Safety, and R&TTE Directive Compliance  
The CE mark is affixed to this product to confirm compliance with the  
following European Community Directives:  
Council Directive 89/ 336/ EEC of 3 May 1989 on the approximation of the  
laws of Member States relating to electromagnetic compatibility,  
and  
Council Directive 73/ 23/ EEC of 19 February 1973 on the harmonization of  
the laws of Member States relating to electrical equipment designed for use  
within certain voltage limits,  
and  
Council Directive 1999/ 5/ EC of 9 March 1999 on radio equipment and  
telecommunications terminal equipment and the mutual recognition of their  
conformity.  
FCC Declaration  
NOTE: This equipment has been tested and found to comply with the  
limits for a Class A digital device, pursuant to Part 15 of the FCC Rules.  
These limits are designed to provide reasonable protection against  
harmful interference when the equipment is operated in a commercial  
environment. This equipment generates, uses and can radiate radio  
frequency energy, and if not installed and used in accordance with the  
instructions, may cause harmful interference to radio communications.  
Operation of this equipment in a residential area is likely to cause  
harmful interference in which case the user will be required to correct  
the interference at his own expense.  
This device complies with Part 15 of the FCC rules.  
Operation is subject to the following two conditions:  
(1) This device may not cause harmful interference.  
(2) This device must accept any interference that may cause  
undesired operation.  
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MultiVOIP User Guide  
Regulatory Information  
Warning: Changes or modifications to this unit not expressly approved  
by the party responsible for compliance could void the user’s authority  
to operate the equipment.  
Industry Canada  
This Class A digital apparatus meets all requirements of the Canadian  
Interference-Causing Equipment Regulations.  
Cet appareil numérique de la classe A  
respecte toutes les exigences du  
Reglement Canadien sur le matériel brouilleur.  
FCC Part 68 Telecom  
1. This equipment complies with part 68 of the Federal  
Communications Commission Rules. On the outside surface of this  
equipment is a label that contains, among other information, the FCC  
registration number. This information must be provided to the  
telephone company.  
2. As indicated below, the suitable jack (Universal Service Order Code  
connecting arrangement) for this equipment is shown. If applicable,  
the facility interface codes (FIC) and service order codes (SOC) are  
shown.  
3. An FCC compliant telephone cord and modular plug is provided  
with this equipment. This equipment is designed to be connected to  
the telephone network or premises wiring using a compatible  
modular jack that is Part 68 compliant. See installation instructions  
for details.  
4. If this equipment causes harm to the telephone network, the  
telephone company will notify you in advance that temporary  
discontinuance of service may be required. If advance notice is not  
practical, the telephone company will notify the customer as soon as  
possible.  
5. The telephone company may make changes in its facilities,  
equipment, operation, or procedures that could affect the operation of  
the equipment. If this happens, the telephone company will provide  
advance notice to allow you to make necessary modifications to  
maintain uninterrupted service.  
6. If trouble is experienced with this equipment (the model of which is  
indicated below), please contact Multi-Tech Systems, Inc. at the  
address shown below for details of how to have repairs made. If the  
equipment is causing harm to the network, the telephone company  
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Regulatory Information  
MultiVOIP User Guide  
may request you to remove the equipment form t network until the  
problem is resolved.  
7. No repairs are to be made by you. Repairs are to be made only by  
Multi-Tech Systems or its licensees. Unauthorized repairs void  
registration and warranty.  
8. Manufacturer:  
Multi-Tech Systems, Inc.  
MultiVOIP  
Trade name:  
Model number:  
MVP2400  
FCC registration number:  
Modular jack (USOC):  
Service center in USA:  
US: AU7DDNAN46050  
RJ-48C  
Multi-Tech Systems, Inc.  
2205 Woodale Drive  
Mounds View, MN 55112  
Tel: (763) 785-3500  
FAX: (763) 785-9874  
Canadian Limitations Notice  
Notice: The Industry Canada label identifies certified equipment. This  
certification means that the equipment meets certain  
telecommunications network protective, operational and safety  
requirements. The Department does not guarantee the equipment will  
operate to the user’s satisfaction.  
Before installing this equipment, users should ensure that it is  
permissible to be connected to the facilities of the local  
telecommunications company. The equipment must also be installed  
using an acceptable method of connection. The customer should be  
aware that compliance with the above conditions may not prevent  
degradation of service in some situations.  
Repairs to certified equipment should be made by an authorized  
Canadian maintenance facility designated by the supplier. Any repairs  
or alterations made by the user to this equipment, or equipment  
malfunctions, may give the telecommunications company cause to  
request the user to disconnect the equipment.  
Users should ensure for their own protection that the electrical ground  
connections of the power utility, telephone lines and internal metallic  
water pipe system, if present, are connected together. This precaution  
may be particularly important in rural areas.  
Caution: Users should not attempt to make such connections  
themselves, but should contact the appropriate electric inspection  
authority, or electrician, as appropriate.  
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Appendix A: Expansion Card  
Installation  
(MVP24-48 & MVP30-60)  
397  
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T1/E1 Expansion Cards  
MultiVOIP User Guide  
Installation  
Both the MVP2410 and the MVP3010 use the same mechanical chassis.  
This chassis accommodates a second MultiVOIP circuit card or  
motherboard module. The add-on module for the MVP2410 is the  
MVP24-48 product; the add-on module for the MVP3010 is the MVP30-  
60 product.  
To install an expansion card into an MVP2410 or MVP3010, you must:  
1. Power down and unplug the MVP2410/ 3010 unit.  
2. Using a Phillips or star-bit screwdriver, remove the blank plate at the  
rear of the MVP2410/ 3010 chassis (see Figure A-1). Save the screw.  
Figure A-1: Remove Plate Covering Expansion Slot  
3. A power cable for the expansion card (+5V) is already present within  
the MVP2410/ 3010 unit. This power cable has a two-pin “molex”  
connector. When the rear cover plate has been removed, the cable is  
accessible from the rear at the right side of the expansion slot. Locate  
this connector within the MVP2410/ 3010. See Figure A-2.  
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MultiVOIP User Guide  
T1/E1 Expansion Cards  
Power Cable  
Molex Connector  
Figure A-2: MVP2410/3010 Chassis (top/rear view)  
4. While keeping the power cable out of the way, fit the MVP24-48 or  
MVP30-60 card into the grooves of the expansion slot. Push it in far  
enough to allow connection of the power cable to the receptacle on the  
vertical plate of the expansion card. (See Figure A-2.) Connect the  
power cable.  
5. Push the expansion card fully into the chassis. See Figure A-3.  
Figure A-3: Sliding Expansion Card into Chassis  
Secure the vertical plate of the expansion card to the chassis with a  
screw.  
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T1/E1 Expansion Cards  
MultiVOIP User Guide  
Operation  
The MVP2410/ 3010 front panel has two sets of identical LEDs. In the  
MVP2410/ 3010 without an expansion card, only the left-hand set of  
LEDs is functional. However, when the MultiVOIP unit has been  
upgraded with an MVP24-48 or MVP30-60 expansion card, the right-  
hand set of LEDs will also become active.  
Remember that the expansion card must be configured as though it  
were simply another complete MultiVOIP unit: it requires its own  
T1/ E1 line; it requires its own connection to a computer running the  
MultiVOIP configuration software. All of the procedures and  
operations that apply to the original motherboard of the  
MVP2410/ 3010 will also apply to the expansion card. See applicable  
User Guide chapters for details.  
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Appendix B: Cable Pinouts  
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Cable Pinouts  
MultiVOIP User Guide  
Appendix B: 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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Cable Pinouts  
MultiVOIP User Guide  
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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TCP/UDP Port Assignments  
Appendix C: 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 (dubyadubyadubya.iana.org).  
“The Well Known Ports are assigned by  
the IANA and on most systems can only  
be used by system (or root) processes or  
by programs executed by privileged  
users. Ports are used in the TCP  
[RFC793] to name the ends of logical  
connections which carry long term  
conversations. For the purpose of  
providing services to unknown callers, a  
service contact port is defined. This list  
specifies the port used by the server  
process as its contact port. The contact  
port is sometimes called the "well-  
known port". To the extent possible,  
these same port assignments are used  
with the UDP [RFC768]. The range for  
assigned ports managed by the IANA is  
0-1023.”  
Well-known port numbers especially pertinent to MultiVOIP operation are listed  
below.  
Port Number Assignment List  
Well-Known Port Numbers  
Function  
telnet  
Port Number  
23  
tftp  
69  
snmp  
snmp tray  
gatekeeper registration  
H.323  
SIP  
161  
162  
1719  
1720  
5060  
514  
SysLog  
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Appendix D: Installation  
Instructions for MVP428  
Upgrade Card  
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MVP428 Upgrade Installation  
MultiVOIP User Guide  
Installation Instructions for MVP428 Upgrade Card  
In this procedure, you will install an additional circuit board into the MVP410,  
converting it from a 4-channel voip to an 8-channel voip.  
Summary: (A) Attach four standoffs to main circuit card.  
(B) Mate the 60-pin connectors (male connector  
on main circuit card; female on upgrade card).  
(C) Attach upgrade card to main circuit card (4  
screws).  
*
*
(A)  
Replace main card screws  
with standoffs here  
*
(2 places).  
Add standoffs here ꢀ  
(2 places).  
(C)  
Attach upgrade card  
(screws into standoffs  
-- 4 places).  
ꢀ
ꢀ
(B)  
Mate 60-pin  
connectors.  
Figure D-1. Installation Summary  
Procedure in Detail  
1. Power down and unplug the MVP410 unit.  
2. Using a Phillips driver, remove the blank cover plate at the rear of the  
MVP410 chassis. Save the screws.  
screws on blank cover plate (2)  
Figure D-2: Removing screws from blank cover plate  
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MultiVOIP User Guide  
MVP428 Upgrade Installation  
3. Using a Phillips driver, remove the three screws that secure the main circuit  
board and back panel assembly to the chassis.  
NOTE:  
Follow standard ESD  
precautions to protect the  
circuit board from static  
electricity damage.  
back panel screws (3)  
Figure D-3: Removing screws from back panel  
4. Slide the main circuit board out of the chassis far enough to unplug the  
power connector.  
power connector  
Figure D-4: Accessing power connector  
5. Unplug the power connector from the main circuit board.  
6. Slide the main circuit board completely out of the chassis and place on a  
non-conductive, static-safe table-top surface.  
7. Remove mounting hardware (2 screws, 2 nuts, and 4 standoffs) from its  
package.  
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MVP428 Upgrade Installation  
MultiVOIP User Guide  
8. On the phone-jack side of the circuit card, three screws attach the circuit  
card to the back panel. Two of these screws are adjacent to the four phone-  
jack pairs. Remove these two screws.  
Screw locations (2)  
at phone-jack edge  
of board.  
Figure D-5: Screws to be removed and replaced with standoffs  
(phone-jack edge of board; top view)  
9. Replace these two screws with standoffs.  
10. There are two copper-plated holes at the LED edge of the circuit card.  
Place a nut beneath each hole (lockwasher side should be in contact with  
board) and attach a standoff to each location\.  
Standoff locations (2) at LED edge  
of board (top view).  
Standoff/nut attachment  
(rear bottom view)  
Figure D-6: Standoffs at LED edge of board (top view)  
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MultiVOIP User Guide  
MVP428 Upgrade Installation  
11. Locate the male 60-pin vertical connector near the LED edge of the main  
circuit card. Check that pins are straight and evenly spaced. If not, then  
correct for straightness and spacing. Locate the 60-pin female connector  
on the upgrade circuit card.  
12. Set the upgrade circuit card on top of the main circuit card. Align the  
upgrade card’s 4 pairs of phone-jacks with the 4 pairs of holes in the  
backplane of the main card. Slide the phone jacks into the holes.  
13. Mate the upgrade card’s 60-pin female connector with the main card’s 60-  
pin male connector.  
*
*
These screws (4 places)  
*
attach upgrade card  
to main card.  
*
*
60-pin connectors  
Figure D-7. Attaching upgrade card to main circuit card  
(secure 4 Phillips screws; mate 60-pin connectors)  
14. There are four copper-plated attachment holes, two each at the front and  
rear edges of the upgrade card. Attach the upgrade card to the main card  
using 4 Phillips screws. The upgrade card should now be firmly attached  
to the main card.  
15. Slide the main circuit card back into the chassis far enough to allow re-  
connection of power cable.  
16. Re-connect power cable.  
17. Slide the main circuit card fully into the chassis.  
18. Re-attach the backplane of the main circuit card to the chassis with 3  
screws.  
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Index  
MultiVOIP User Guide  
Index  
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MultiVOIP User Guide  
Index  
INDEX  
T1/E1 ........................................126  
accessing Supplementary Services  
screen  
A
abbreviated dialing, inter-office  
E1.............................................. 276  
T1.............................................. 237  
access codes, PBX .......................... 42  
access codes, types  
PBX ............................................ 47  
PSTN .......................................... 47  
special ......................................... 47  
access digits, PBX42. See phonebook  
digits, types used  
analog........................................219  
T1/E1 ........................................145  
accessing System Information screen  
analog........................................231  
T1/E1 ........................................157  
accessing T1 Statistics screen ....339  
accessing T1/E1/ISDN Parameters  
screen ........................................113  
accessing Voice/FAX Parameters  
screen ................................103, 178  
Add Inbound Phonebook Entry icons  
E1..............................................282  
T1..............................................238  
Add Outbound Phonebook Entry icon  
E1..............................................282  
T1..............................................238  
Add Prefix (inbound) field  
E1..............................................296  
T1..............................................252  
Add Prefix (outbound) field  
E1..............................................290  
T1..............................................246  
Add/Edit Inbound Phonebook field  
definitions  
access to network  
analog........................................ 202  
T1/E1 ........................................ 128  
access to remote PSTN  
E1................................................ 14  
T1................................................ 10  
accessing Statistics, Logs screen  
.................................................. 331  
accessing Call Progress (Statistics)  
screen........................................ 325  
accessing configuration parameter  
groups  
analog........................................ 173  
T1/E1 .......................................... 98  
accessing interface parameters...... 188  
accessing IP Parameters screen  
analog........................................ 174  
T1/E1 .......................................... 99  
accessing IP Statistics screen........ 335  
accessing Logs (Statistics) screen  
.................................................. 331  
accessing logs screen  
E1......................................296, 297  
T1......................................252, 253  
Add/Edit Inbound Phonebook screen  
E1..............................................296  
T1..............................................252  
Add/Edit Inbound Phonebook screen  
fields (E1)  
Add Prefix.................................296  
Channel Number .......................296  
Description (callee location) .....297  
Enable (Call Forwarding)..........297  
Forward Address/Number.........297  
Forward Condition....................297  
Remove Prefix...........................296  
Ring Count................................297  
Add/Edit Inbound Phonebook screen  
fields (T1)  
analog........................................ 216  
T1/E1 ........................................ 142  
accessing Regional Parameters  
analog........................................ 203  
T1/E1 ........................................ 129  
accessing SMTP parameters  
analog........................................ 210  
T1/E1 ........................................ 136  
accessing SNMP parameters  
analog........................................ 200  
413  
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Index  
MultiVOIP User Guide  
Add Prefix................................. 252  
Add/Edit Outbound Phonebook screen  
E1..............................................289  
T1..............................................245  
add-on module (analog, 4-to-8  
channel), installation .................408  
add-on module (T1/E1)  
operation ...................................400  
add-on module (T1/E1), installation  
..................................................398  
Address (SNMP) field  
Channel Number....................... 252  
Description (callee location)..... 253  
Enable (Call Forwarding) ......... 253  
Forward Address/Number......... 253  
Forward Condition.................... 253  
Remove Prefix .......................... 252  
Ring Count................................ 253  
Add/Edit Outbound Phonebook field  
definitions  
E1.............................. 290, 291, 292  
T1.............................. 246, 247, 248  
Add/Edit Outbound Phonebook fields  
(E1)  
analog........................................202  
T1/E1 ........................................128  
Advanced button, Outbound  
Phonebook  
Add Prefix................................. 290  
Advanced button....................... 292  
Description................................ 290  
destination pattern..................... 290  
Gateway Prefix ......................... 291  
H.323 ID................................... 291  
IP Address................................. 290  
Protocol Type............................ 290  
Q.931 Port Number................... 291  
Remove Prefix .......................... 290  
SIP Port Number....................... 292  
SIP URL.................................... 292  
Total Digits............................... 290  
Transport Protocol (SIP)........... 292  
Use Gatekeeper......................... 291  
Use Proxy (SIP)........................ 292  
Add/Edit Outbound Phonebook fields  
(T1)  
Add Prefix................................. 246  
Advanced button....................... 248  
Description................................ 246  
destination pattern..................... 246  
Gateway Prefix ......................... 247  
H.323 ID................................... 247  
IP Address................................. 246  
Protocol Type............................ 246  
Q.931 Port Number................... 247  
Remove Prefix .......................... 246  
SIP Port Number....................... 248  
SIP URL.................................... 248  
Total Digits............................... 246  
Transport Protocol (SIP)........... 248  
Use Gatekeeper......................... 247  
Use Proxy (SIP)........................ 248  
E1..............................................293  
T1..............................................249  
Advanced Features field group  
analog........................................184  
T1/E1 ........................................109  
airflow.............................................66  
Alerting Party  
Supplementary Services (analog)  
..............................226, 227, 228  
Supplementary Services (T1/E1)  
..............................152, 153, 154  
Allowed Name Type (analog)  
Alerting Party............226, 227, 228  
Calling Party .............................225  
Allowed Name Type (T1/E1)  
Alerting Party............152, 153, 154  
Calling Party .............................151  
Allowed Name Types, Call Name ID  
(analog)  
Alerting Party............................226  
Busy Party.................................227  
Calling Party .............................225  
Connected Party........................228  
Allowed Name Types, Call Name ID  
(T1/E1)  
Alerting Party............................152  
Busy Party.................................153  
Calling Party .............................151  
Connected Party........................154  
Alternate IP Address field  
E1..............................................294  
T1..............................................250  
Alternate IP Routing  
E1..............................................289  
414  
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MultiVOIP User Guide  
Index  
T1.............................................. 245  
Alternate Routing field definitions  
E1.............................................. 294  
T1.............................................. 250  
Alternate Routing field definitions  
(E1)  
Alternate IP Address................. 294  
Round Trip Delay ..................... 294  
Alternate Routing field definitions  
(T1)  
analog........................................182  
T1/E1 ........................................107  
baud rate, setting  
analog........................................230  
T1/E1 ........................................156  
Bipolar Violation (E1 stats) field ..346  
Bipolar Violation (T1 stats) field ..343  
Blue Alarm (E1 stats) field ...........344  
Blue Alarm (T1 stats) field ...........341  
Boot Code Version  
Alternate IP Address................. 250  
Round Trip Delay ..................... 250  
analog phonebook......................... 321  
using T1 & E1 examples for..... 321  
analog phonebook examples......... 162  
analog telephony interface parameters  
.................................................. 166  
Annex E field  
E1.............................................. 287  
T1.............................................. 243  
area codes ....................................... 46  
Auto Call Enable field  
System Info (analog).................232  
System Info (T1/E1)..........158, 323  
Boot LED  
analog models .............................21  
MVP-210.....................................73  
MVP-410/810 .............................71  
on MVP-2400..............................70  
on MVP-2410/3010.....................69  
booting time  
analog..........................................21  
E1................................................16  
T1................................................13  
box contents  
verifying......................................61  
busy tone, custom  
analog........................................ 184  
T1/E1 ........................................ 109  
Auto Disconnect field group  
analog........................................ 187  
T1/E1 ........................................ 112  
Automatic Disconnection field  
analog........................................ 187  
T1/E1 ........................................ 112  
Avaya Magix PBX (FXO)  
and Message Waiting Light ...... 195  
Avaya Magix PBX (FXS Ground  
Start)  
analog................................207, 208  
T1/E1 ................................133, 134  
busy-tones  
analog........................................207  
T1/E1 ........................................133  
Bytes Received (call progress) field  
..................................................327  
Bytes Received (SMTP logs) field  
analog........................................213  
T1/E1 ........................................139  
Bytes received (statistics, logs) field  
..................................................333  
Bytes Sent (call progress) field .....327  
Bytes Sent (SMTP logs) field  
and Message Waiting Light ...... 192  
Avaya Magix PBX (FXS Loop Start)  
and Message Waiting Light ...... 191  
B
bandwidth  
analog........................................213  
T1/E1 ........................................139  
Bytes sent (statistics, logs) field....332  
coder (analog)........................... 183  
coder (T1/E1)............................ 108  
battery caution ................................ 60  
baud rate, default (MultiVOIP  
software connection)  
T1/E1 .................................... 156  
analog.................................... 230  
baud rate, fax  
C
cable length, maximum span  
E1..............................................121  
T1..............................................116  
cabling diagram, quick  
415  
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Index  
MultiVOIP User Guide  
analog models ....................... 33, 34  
Call Forward Parameters (inbound  
phonebook)  
E1..............................................297  
T1..............................................253  
Call Forwarded To  
logs (statistics) field ..................334  
Call Hold (analog).........................220  
Call Hold (T1/E1) .........................146  
Call Hold Enable  
analog........................................223  
T1/E1 ........................................149  
Call Mode (SMTP logs) field  
analog........................................213  
T1/E1 ........................................139  
Call Name Identification (analog)  
Alerting Party............226, 227, 228  
Calling Party .............................225  
Call Name Identification (T1/E1)  
Alerting Party............152, 153, 154  
Calling Party .............................151  
Call Name Identification (analog).220  
Call Name Identification (T1/E1) .146  
Call On Hold  
Call Progress Details (statistics)  
field ...............................327, 329  
Call on Hold (call progress) field..329  
Call Progress (Statistics) ...............325  
Call Progress Details (statistics)  
screen field  
E1 models ................................... 33  
MVP210...................................... 34  
MVP2400.................................... 34  
MVP2410.................................... 33  
MVP3010.................................... 33  
MVP410...................................... 33  
MVP810...................................... 33  
T1 models ............................. 33, 34  
cabling problem, fixing  
analog models ........................... 173  
T1/E1 models.............................. 98  
cabling procedure  
MVP2400.................................... 69  
MVP2410.................................... 68  
MVP3010.................................... 68  
MVP410................................ 70, 72  
MVP810...................................... 70  
Cadence 1 (custom) field  
analog........................................ 209  
T1/E1 ........................................ 135  
Cadence 2 (custom) field  
analog........................................ 209  
T1/E1 ........................................ 135  
Cadence 3 (custom) field  
analog........................................ 209  
T1/E1 ........................................ 135  
Cadence 4 (custom) field  
analog........................................ 209  
T1/E1 ........................................ 135  
Cadence field  
analog........................................ 206  
T1/E1 ........................................ 132  
cadences, custom  
analog........................................ 207  
T1.E1 ................................ 135, 209  
T1/E1 ........................................ 133  
cadences, signaling  
analog........................................ 203  
T1/E1 ........................................ 129  
Call Control Status  
Call Progress Details (statistics)  
field....................................... 330  
Call Control Status (call progress)  
field........................................... 330  
Call Duration field  
Call On Hold.........................327  
Call Waiting..........................327  
Caller ID................................327  
Call On Hold.........................329  
Call Waiting..........................329  
Caller ID................................330  
Call Progress Details (statistics)  
screen fields  
Channel .................................327  
Duration ................................327  
Mode .....................................327  
Voice Coder ..........................327  
Packets Sent ..........................327  
Packets Received...................327  
Bytes Sent .............................327  
Bytes Received......................327  
Packets Lost ..........................327  
Outbound Digits....................327  
Prefix Matched......................327  
analog........................................ 187  
T1/E1 ........................................ 112  
416  
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Index  
Gateway Name...................... 328  
IP Address............................. 328  
Options.................................. 328  
Silence Compression............. 328  
Forward Error Correction ..... 328  
Status .................................... 330  
Call Control Status................ 330  
Call Signalling Port field  
Canadian Class A requirements ....395  
Canadian Limitations Notice  
(regulatory)................................396  
CAS Protocol field  
E1..............................................122  
T1..............................................117  
CAS Protocols, downloading........357  
CAS vs. CCS  
E1.............................................. 285  
T1.............................................. 241  
Call Status (SMTP logs) field  
analog........................................ 214  
T1/E1 ........................................ 140  
Call Transfer (analog)................... 220  
Call Transfer (T1/E1) ................... 146  
Call Transfer Enable  
analog........................................ 222  
T1/E1 ........................................ 148  
Call Transferred To  
logs (statistics) field.................. 334  
Call Waiting  
Call Progress Details (statistics)  
field............................... 327, 329  
Call Waiting (analog).................... 220  
Call Waiting (call progress) field.. 329  
Call Waiting (T1/E1).................... 146  
Call Waiting Enable  
T1......................................117, 122  
CCS vs. CAS  
T1......................................117, 122  
CD  
MultiVOIP ..................................26  
Channel (call progress) field .........327  
channel capacity................................8  
analog..........................................18  
E1................................................14  
T1................................................10  
Channel Number (inbound) field  
E1..............................................296  
T1..............................................252  
Channel Number (SMTP logs) field  
analog........................................213  
T1/E1 ........................................139  
city codes ........................................46  
Clear (IP Statistics) button ............336  
Clocking field  
analog........................................ 223  
T1/E1 ........................................ 149  
Caller ID  
E1..............................................124  
T1..............................................119  
coder (analog)  
Call Progress Details (statistics)  
field............................... 327, 330  
Caller ID (analog)......................... 220  
Caller ID (call progress) field ....... 330  
Caller ID (Supplementary Services)  
field  
analog........................................ 229  
T1/E1 ........................................ 155  
Caller ID (T1/E1).......................... 147  
Caller Name Identification Enable  
analog........................................ 224  
T1/E1 ........................................ 150  
calling area codes............................ 46  
Calling Party  
bandwidth, max.........................183  
G.711.........................................183  
G.723.1......................................183  
G.726.........................................183  
G.727.........................................183  
G.729.........................................183  
Net Coder..................................183  
Coder (SMTP logs) field  
analog........................................213  
T1/E1 ........................................139  
coder (T1/E1)  
bandwidth, max.........................108  
G.711.........................................108  
G.723.1......................................108  
G.726.........................................108  
G.727.........................................108  
G.729.........................................108  
Net Coder..................................108  
Supplementary Services (analog)  
.............................................. 225  
Supplementary Services (T1/E1)  
.............................................. 151  
417  
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MultiVOIP User Guide  
Coder field  
T1/E1 ........................................128  
Community Name 1 (SNMP) field  
analog........................................202  
T1/E1 ........................................128  
compatibility, Fast Start  
E1..............................................285  
T1..............................................241  
compatibility, H.450 with H.323, not  
with SIP  
analog........................................ 183  
T1/E1 ........................................ 108  
Coder Parameters field group  
analog........................................ 183  
T1/E1 ........................................ 108  
COL LED  
analog models ............................. 21  
COM port  
on command PC.......................... 80  
COM port (analog models)  
conflict, resolving ..................... 172  
error message............................ 172  
COM port (T1/E1 models)  
analog..................................19, 219  
E1................................................15  
T1................................................11  
T1/E1 ........................................145  
compression standard  
conflict, resolving ....................... 97  
error message.............................. 97  
COM port allocation  
E1..............................................124  
T1..............................................119  
compression, silence  
analog........................................ 230  
T1/E1 ........................................ 156  
COM port assignments  
analog........................................ 230  
T1/E1 ........................................ 156  
COM port conflict  
error message.............................. 80  
COM Port Setup screen .................. 80  
COM Port Setup screen (analog  
models) ..................................... 172  
COM Port Setup screen (T1/E1  
models) ....................................... 97  
comma  
meaning/use in phonebook ......... 48  
comma use  
and second dial tone.................... 48  
command cable pinout.................. 402  
command PC  
COM port assignment (detailed). 80  
COM port requirement................ 32  
demands upon ............................. 32  
non-dedicated use ....................... 32  
operating system ......................... 32  
settings........................................ 32  
specifications .............................. 32  
Command PC  
COM port requirement................ 22  
non-dedicated use of................... 22  
operating system ......................... 22  
community (voip) defined  
analog........................................184  
T1/E1 ........................................109  
Compression, Silence (SMTP logs)  
analog........................................214  
T1/E1 ........................................140  
computer requirements....................22  
configuration of voip (analog)  
local versus remote....................163  
configuration of voip (T1/E1)  
local versus remote................87, 88  
Configuration option (MultiVOIP  
program menu)..........................347  
Configuration Parameter Groups,  
accessing  
analog........................................173  
T1/E1 ..........................................98  
configuration procedure, local  
detailed, analog .........................169  
detailed, T1/E1............................94  
summary, analog .......................168  
summary, T1/E1..........................93  
configuration, local  
analog........................................165  
T1/E1 ..........................................89  
configuration, phonebook  
E1..............................................281  
starter...........................................39  
T1..............................................237  
configuration, saving  
analog........................................233  
T1/E1 ........................................159  
analog........................................ 202  
418  
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MultiVOIP User Guide  
Index  
user............................................ 362  
configuration, starter  
phone/IP...................................... 36  
configuration, technical  
country  
ISDN type and...........................125  
switch type and ISDN ...............125  
Country (ISDN) field  
saving.......................................... 38  
configuration, user default  
analog........................................ 234  
T1/E1 ........................................ 160  
Configuring MultiVOIP phonebooks,  
general  
E1.............................................. 275  
T1.............................................. 236  
confirming connectivity.................. 57  
conflicts  
COM port.................................... 80  
Connection Problems, Solving  
analog........................................ 172  
T1/E1 .......................................... 97  
connectivity  
confirmation of ........................... 57  
confirming with remote voip 31, 39  
pinging and ................................. 58  
connectivity test.............................. 54  
Consecutive Packets Lost field  
analog........................................ 187  
T1/E1 ........................................ 112  
console messages.......... 38, 54, 56, 57  
console messages, enabling  
E1/ISDN....................................123  
country codes ..................................46  
Country field (ISDN)  
T1/ISDN....................................118  
Country/Region (tone schemes) field  
analog........................................205  
T1/E1 ........................................131  
CRC and ESF frame format (T1)..116  
CRC Check field  
T1..............................................116  
Creating a User Default Configuration  
analog........................................234  
T1/E1 ........................................160  
CT Ph#  
logs (statistics) field ..................334  
Current Loss (FXO disconnect  
criteria) field..............................196  
Current Loss field  
FXS Ground Start......................193  
FXS Loop Start .........................191  
Custom (tones, Regional)field  
analog........................................206  
T1/E1 ........................................132  
custom cadences  
analog........................................ 217  
T1/E1 ........................................ 143  
contacting technical support ......... 392  
coordinated phonebook entries  
E1.............................................. 281  
T1.............................................. 237  
Copy Channel command  
analog........................................ 180  
T1/E1 ........................................ 105  
Copy Channel field  
analog........................................209  
T1/E1 ........................................135  
custom DTMF  
analog................................207, 208  
T1/E1 ................................133, 134  
Custom Fields (SMTP) definitions  
analog................................213, 214  
T1/E1 ................................139, 140  
Custom Fields, SMTP log email  
(analog)  
analog........................................ 181  
T1/E1 ........................................ 106  
Copy Channel, Supplementary  
Services command  
analog........................................ 221  
T1/E1 ........................................ 147  
Copy Channel, Supplementary  
Services field  
Bytes Received..........................213  
Bytes Sent .................................213  
Call Mode..................................213  
Call Status .................................214  
Channel Number .......................213  
Coder.........................................213  
Options......................................214  
Options......................................214  
Description (callee)...................214  
Description (caller) ...................214  
analog........................................ 229  
T1/E1 ........................................ 155  
419  
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MultiVOIP User Guide  
Duration.................................... 213  
Custom Tone-Pair Settings definitions  
analog................................208, 209  
T1/E1 ................................134, 135  
Custom Tone-Pair Settings fields  
(analog)  
Frequency 1...............................208  
Frequency 2...............................208  
Gain 1........................................208  
Gain 2........................................208  
Tone Pair...................................208  
Custom Tone-Pair Settings fields  
(T1/E1)  
Frequency 1...............................134  
Frequency 2...............................134  
Gain 1........................................134  
Gain 2........................................134  
Tone Pair...................................134  
custom tones, setting  
analog........................................207  
T1/E1 ........................................133  
customized log email  
From Gateway Number ............ 214  
From IP Address....................... 214  
Outbound Digits........................ 214  
Packets Lost.............................. 213  
Packets Received ...................... 213  
Packets Sent.............................. 213  
Prefix Matched.......................... 214  
Select All .................................. 213  
Start Date, Time........................ 213  
To Gateway Number................. 214  
To IP Address ........................... 214  
Custom Fields, SMTP log email  
(T1/E1)  
Bytes Received ......................... 139  
Bytes Sent................................. 139  
Call Mode ................................. 139  
Call Status................................. 140  
Channel Number....................... 139  
Coder ........................................ 139  
Options...................................... 140  
Options...................................... 140  
Description (callee)................... 140  
Description (caller) ................... 140  
Duration.................................... 139  
From Gateway Number ............ 140  
From IP Address....................... 140  
Outbound Digits........................ 140  
Packets Lost.............................. 139  
Packets Received ...................... 139  
Packets Sent.............................. 139  
Prefix Matched.......................... 140  
Select All .................................. 139  
Start Date, Time........................ 139  
To Gateway Number................. 140  
To IP Address ........................... 140  
Custom Tone-Pair Settings (analog)  
fields  
analog................................213, 214  
T1/E1 ................................139, 140  
D
data capacity......................................8  
analog..........................................18  
E1................................................14  
T1................................................10  
Date & Time Setup (program menu  
option), command .....................349  
Date and Time Setup option  
(MultiVOIP program menu) .....347  
debugging messages  
analog........................................218  
T1/E1 ........................................143  
Default (Supplementary Services)  
field  
Cadence 1.................................. 209  
Cadence 2.................................. 209  
Cadence 3.................................. 209  
Cadence 4.................................. 209  
Custom Tone-Pair Settings (T1/E1)  
fields  
Cadence 1.................................. 135  
Cadence 2.................................. 135  
Cadence 3.................................. 135  
Cadence 4.................................. 135  
analog........................................229  
T1/E1 ........................................155  
Default (Voice/FAX) field  
analog........................................181  
T1/E1 ........................................106  
default baud rate (MultiVOIP  
software connection)  
analog........................................230  
T1/E1 ........................................156  
default configuration, user  
420  
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Index  
analog........................................ 234  
T1/E1 ........................................ 160  
default values, software ................ 360  
delay, packets  
analog........................................ 185  
T1/E1 ........................................ 110  
delay, versus voice quality  
analog........................................ 186  
T1/E1 ........................................ 111  
Delete File button  
Logs (Statistics) screen............. 332  
Description (callee location)  
E1.............................................. 297  
T1.............................................. 253  
Description (callee, outbound  
phonebook)  
T1/E1 ........................................133  
DID  
and FXO interface.....................195  
FXS Ground Start......................192  
FXS Loop Start .........................191  
digits in phonebook  
specialized codes.........................47  
types............................................46  
dimensions  
analog models .............................25  
E1 models....................................24  
T1 models....................................23  
Direct Inward Dialing  
FXS Ground Start..............191, 192  
Disconnect on Call Progress Tone  
(FXO) field................................197  
Disconnect Tone Sequence (FXO)  
field ...........................................197  
disconnection criteria, FXO ..196, 197  
DNS Server IP Address  
E1.............................................. 290  
T1.............................................. 246  
Description, From Details (SMTP  
logs) field  
analog........................................ 214  
T1/E1 ........................................ 140  
Description, To Details (SMTP logs)  
field  
analog........................................ 214  
T1/E1 ........................................ 140  
Destination Pattern (outbound) field  
E1.............................................. 290  
T1.............................................. 246  
destination patterns  
digits used................................... 46  
tips about..................................... 46  
destination patterns, discussion  
E1.............................................. 280  
T1.............................................. 236  
dial tone, custom  
analog................................ 207, 208  
T1/E1 ................................ 133, 134  
dial tone, second  
and comma use............................ 48  
pausing for .................................. 48  
Dialing Options (FXO) fields195, 196  
dialing patterns  
digits used................................... 46  
inbound/outbound matching ....... 48  
tips about..................................... 46  
dial-tones  
analog........................................176  
T1/E1 ........................................101  
Download CAS Protocol (program  
menu option) , command...........357  
Download CAS Protocol option  
(MultiVOIP program menu) .....347  
Download Factory Defaults (program  
menu option) , command...........360  
Download Factory Defaults option  
(MultiVOIP program menu) .....348  
Download Firmware (program menu  
option), command .............353, 354  
Download Firmware option  
description (MultiVOIP program  
menu) ........................................348  
Download User Defaults (program  
menu option) , command...........362  
Download User Defaults option  
description (MultiVOIP program  
menu) ........................................348  
downloading firmware, machine  
perspective ........................348, 371  
downloading user defaults.............362  
downloads vs. uploads (FTP)........371  
dropping digits, in phonebook.........47  
DTMF  
extended ....................................197  
standard.....................................197  
analog........................................ 207  
421  
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MultiVOIP User Guide  
DTMF frequency chart ................. 197  
DTMF Gain (High Tones) field  
analog........................................ 181  
T1/E1 ........................................ 106  
DTMF Gain (Low Tones) field  
analog........................................ 181  
T1/E1 ........................................ 106  
DTMF Gain field  
analog........................................ 181  
T1/E1 ........................................ 106  
DTMF In/Out of Band field  
analog........................................ 182  
T1/E1 ........................................ 107  
DTMF inband  
analog........................................ 182  
T1/E1 ........................................ 107  
DTMF out of band  
analog........................................ 182  
T1/E1 ........................................ 107  
DTMF Parameters  
T1/E1 ........................................ 106  
DTMF, custom tone pairs  
analog................................ 207, 208  
T1/E1 ................................ 133, 134  
Duration (call progress) field........ 327  
Duration (DTMF) field  
uses of .........................................71  
E&M Interface Parameter fields  
Interface ....................................199  
Pass Through.............................199  
Signal ........................................199  
Type ..........................................199  
Wink Timer...............................199  
E&M Parameter definitions ..........199  
E&M Parameters...........................198  
E1 Parameter definitions121, 122, 124  
Clocking....................................124  
Line Build-Out..........................124  
Line Coding...............................124  
PCM Law..................................124  
Pulse Shape Level .....................124  
E1 Parameter fields  
CAS Protocol ........................122  
CRC Check ...............................122  
Frame Format............................122  
Long-Haul Mode.......................122  
E1 Parameters screen ....................120  
E1 Statistics field definitions344, 345,  
346  
E1 Statistics fields  
Bipolar Variation.......................346  
Blue Alarm................................344  
Excessive Zeroes.......................346  
Loss of Frame Alignment..........344  
Loss of MultiFrame Alignment.345  
Receive Slip ..............................346  
Receive Timeslot 16 Alarm  
analog........................................ 182  
T1/E1 ........................................ 107  
Duration (SMTP logs) field  
analog........................................ 213  
T1/E1 ........................................ 139  
Duration (statistics, logs) field...... 332  
Dynamic Jitter Buffer field  
Indication Signal ...................345  
Receive Timeslot 16 Loss of  
analog........................................ 185  
T1/E1 ........................................ 110  
Dynamic Jitter field group  
analog........................................ 185  
T1/E1 ........................................ 110  
Dynamic Jitter fields  
MultiFrame Alignment..........346  
Receive Timeslot 16 Loss of Signal  
..............................................345  
Red Alarm.................................344  
Status Freeze Signalling Active 345  
Transmit Data Overflow............345  
Transmit Data Underrun ...........346  
Transmit Line Open ..................346  
Transmit Line Short ..................345  
Transmit Slip.............................346  
Transmit Slip Negative .............346  
Transmit Slip Positive...............345  
Yellow Alarm............................345  
E1 telephony parameters.................91  
E1/ISDN Parameter definitions.....123  
analog........................................ 186  
T1/E1 ........................................ 111  
E
E&M interface (MVP210)  
matching telco trunk line ............ 73  
uses of......................................... 73  
E&M interface (MVP-410/810)  
matching telco trunk line ............ 71  
422  
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Index  
E1/ISDN Parameter fields  
T1/E1 ........................................148  
Enable Call Waiting  
Country ..................................... 123  
Enable ISDN-PRI ..................... 123  
Operator.................................... 123  
Terminal Network..................... 123  
Echo Cancellation field  
analog........................................ 184  
T1/E1 ........................................ 109  
echo, removing  
analog........................................ 184  
T1/E1 ........................................ 109  
Edit selected Inbound Phonebook  
Entry icon  
E1.............................................. 282  
T1.............................................. 238  
Edit selected Outbound Phonebook  
Entry icon  
analog........................................223  
T1/E1 ........................................149  
Enable Caller Name Identification  
analog........................................224  
T1/E1 ........................................150  
Enable Console Messages field  
analog........................................218  
T1/E1 ........................................143  
Enable Diffserv field  
analog........................................176  
T1/E1 ........................................101  
Enable DNS field  
analog........................................176  
T1/E1 ........................................101  
Enable ISDN-PRI field  
E1.............................................. 282  
T1.............................................. 238  
email account for voip unit  
E1/ISDN....................................123  
T1/ISDN....................................118  
Enable Proxy field  
analog........................................ 211  
T1/E1 ........................................ 137  
email address for voip  
E1..............................................286  
T1..............................................242  
Enable SMTP field  
analog................................ 167, 210  
quick ........................................... 31  
T1/E1 .................................. 92, 136  
email log reports  
analog........................................211  
T1/E1 ........................................137  
Enable SNMP Agent.............126, 200  
Enable SNMP Agent field  
analog........................................ 210  
quick ........................................... 37  
recipient ...................................... 37  
reply-to address........................... 37  
subject line.................................. 37  
T1/E1 ........................................ 136  
email logs, illustration  
analog........................................ 215  
T1/E1 ........................................ 141  
EMC, Safety, R&TTE Directive  
Compliance............................... 394  
emergency phone numbers  
analog........................................202  
T1/E1 ........................................128  
enabling SMTP  
analog........................................210  
T1/E1 ........................................136  
enabling web browser GUI  
analog..................................36, 177  
T1/E1 ........................................102  
Error Correction (SMTP logs)  
analog........................................214  
T1/E1 ........................................140  
error correction, forward  
caution about............................... 48  
Enable (Call Fwdg)  
E1.............................................. 297  
T1.............................................. 253  
Enable Call Hold  
analog........................................ 223  
T1/E1 ........................................ 149  
Enable Call Transfer  
analog........................................184  
T1/E1 ........................................109  
error message  
COM port conflict.......................80  
COM port conflict (T1/E1) .......172  
error message (analog models)  
MultiVOIP Not Found ..............173  
Phone Database Not Read.........173  
error message (T1/E1 models)  
analog........................................ 222  
423  
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MultiVOIP User Guide  
MultiVOIP Not Found................ 98  
analog........................................182  
T1/E1 ........................................107  
fax machine  
Phone Database Not Read........... 98  
ESF and CRC frame format (T1).. 116  
ethernet cable pinout..................... 402  
Ethernet interface  
analog.......................................... 18  
Ethernet LEDs (analog)  
connecting to analog voip  
(MVP210) ...............................73  
connecting to analog voip (MVP-  
410/810)..................................71  
FAX Parameters  
analog........................................182  
T1/E1 ........................................107  
fax tones, output level  
analog........................................182  
T1/E1 ........................................107  
Fax Volume field  
analog........................................182  
T1/E1 ........................................107  
FCC Declaration ...........................394  
FCC Part 68 Telecom rules...........395  
FCC registration number...............396  
FCC rules, Part 15.........................394  
firmware upgrade, implementing..353  
firmware version, identifying........353  
firmware, downloading .................354  
firmware, obtaining updated .........349  
Flash Hook Timer field.................195  
forgotten password................364, 367  
Forward Address/Number  
E1..............................................297  
T1..............................................253  
Forward Condition (Call Fwdg)  
E1..............................................297  
T1..............................................253  
Forward Error Correction (call  
progress) field ...........................328  
Forward Error Correction (SMTP  
logs)  
COL ............................................ 21  
LNK............................................ 21  
RCV............................................ 21  
XMT ........................................... 21  
European Community Directives.. 394  
Event # (statistics, logs) field........ 332  
Excessive Zeroes (E1 stats) field .. 346  
Excessive Zeroes (T1 stats) field .. 341  
exchanges, phone  
dedicated..................................... 47  
institutional ................................. 47  
local ............................................ 47  
non-local ..................................... 47  
organizational ............................. 47  
expansion card (analog, 4-to-8  
channel) installation.................. 408  
expansion card (T1/E1) installation  
.................................................. 398  
expansion card (T1/E1)operation.. 400  
F
factory default software settings ... 360  
factory defaults, downloading....... 360  
factory repair for customers U.S. &  
Canada ...................................... 390  
fast busy (unobtainable) tones  
analog................................ 133, 207  
Fast ConnectSee Fast Start. See Fast  
Start  
analog........................................214  
T1/E1 ........................................140  
Forward Error Correction field  
analog........................................184  
T1/E1 ........................................109  
forward on busy  
T1......................................253, 297  
Forward upon No Response  
E1..............................................297  
T1..............................................253  
Frame Format field  
E1.............................................. 287  
T1.............................................. 243  
Fast Start compatibility  
E1.............................................. 285  
T1.............................................. 241  
Fast Start plus H.245 Tunneling field  
E1.............................................. 287  
T1.............................................. 243  
fax baud rate, default  
analog........................................ 182  
T1/E1 ........................................ 107  
Fax Enable field  
E1..............................................121  
424  
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Index  
T1.............................................. 116  
frame relay, and fax  
FTP Server Enable field  
analog........................................176  
T1/E1 ........................................101  
FTP Server function  
as added feature.........................371  
enabling.....................................373  
FTP Server, contacting..................375  
FTP Server, invoking  
analog........................................ 182  
T1/E1 ........................................ 107  
Frame Search Restart Flag (T1 stats)  
field........................................... 342  
Frame Type field  
analog........................................ 176  
T1/E1 ........................................ 101  
free calls  
E1.............................................. 276  
T1.............................................. 236  
frequencies, touch tone ................. 197  
Frequency 1 (custom tone) field  
analog........................................ 208  
T1/E1 ........................................ 134  
Frequency 1 (tone pair scheme)  
analog........................................ 205  
T1/E1 ........................................ 131  
Frequency 2 (custom tone) field  
analog........................................ 208  
T1/E1 ........................................ 134  
Frequency 2 (tone pair scheme)  
analog........................................ 205  
T1/E1 ........................................ 131  
frequency, power  
analog models ............................. 25  
E1 models ................................... 24  
T1 models ................................... 23  
FRF11  
analog........................................ 182  
T1/E1 ........................................ 107  
From (gateway, statistics, logs) field  
.................................................. 332  
front panel  
analog models ............................. 21  
E1................................................ 16  
MVP2400.................................... 13  
MVP2410.................................... 13  
MVP3010.................................... 16  
T1................................................ 13  
FTP client program....................... 371  
FTP client program, obtaining...... 373  
FTP client programs  
download/transfer  
using FTP client program..........379  
using web browser ....................377  
FTP Server, logging in..................376  
FTP Server, logging out................380  
FTP transfers  
file types............................371, 374  
phonebooks ...............................371  
server location...........................371  
FXO Current Detect Timer field...196  
FXO Disconnect On fields....196, 197  
FXO disconnection criteria ...........196  
FXO disconnection, triggering of196,  
197  
FXO interface (MVP210)  
uses of .........................................73  
FXO Interface Parameter definitions  
..........................................195, 196  
FXO Interface Parameter Definitions  
..................................................197  
FXO Interface Parameter fields  
Disconnect on Call Progress Tone  
..............................................197  
Disconnect Tone Sequence .......197  
Ring Count................................197  
Silence Detection ......................197  
Silence Timer............................197  
FXO interface(MVP-410/810)  
uses of .........................................71  
FXO Parameter fields  
Current Loss..............................196  
Flash Hook................................196  
FXO Current Detect Timer .......196  
Inter Digit Regeneration Timer.196  
Inter Digit Timer (dialing) ........195  
Message Waiting Light .............196  
Regeneration (dialing)...............195  
Tone Detection..........................196  
FXO Parameters............................194  
graphic vs. textual orientation... 380  
FTP file transfers  
using FTP client program.......... 373  
using web browser .................... 373  
425  
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Index  
MultiVOIP User Guide  
FXS Ground Start Interface parameter  
definitions ................................. 192  
FXS Ground Start Parameter fields  
Inter Digit Timer....................... 192  
Message Waiting Light............. 192  
FXS Ground Start Parameters....... 192  
FXS interface(MVP210)  
Gateway Name (call progress) field  
..................................................328  
Gateway Name (callee, statistics,  
logs) field ..................................333  
Gateway Name (caller, statistics, logs)  
field ...........................................333  
Gateway Name field  
uses of......................................... 73  
FXS interface(MVP-410/810)  
E1..............................................285  
T1..............................................241  
Gateway Number, From Details  
(SMTP logs) field  
analog........................................214  
T1/E1 ........................................140  
Gateway Number, To Details (SMTP  
logs) field  
analog........................................214  
T1/E1 ........................................140  
Gateway Prefix (Gatekeeper) field  
E1..............................................285  
T1..............................................241  
Gateway Prefix (outbound  
uses of......................................... 71  
FXS Loop Start Interface parameter  
definitions ................................. 190  
FXS Loop Start Parameter fields  
Current Loss.............................. 191  
Inter Digit Timer....................... 190  
Message Waiting Light............. 190  
Ring Count................................ 191  
FXS Loop Start Parameters .......... 190  
FXS/FXO connector  
MVP-210 .................................... 72  
MVP-410/810 ............................. 71  
phonebook) field  
G
E1..............................................291  
T1..............................................247  
grounding  
in rack installations .....................66  
MVP210......................................73  
MVP410......................................71  
MVP810......................................71  
grounding screw, diagrams  
Gain 1 (custom tone) field  
analog........................................ 208  
T1/E1 ........................................ 134  
Gain 1 (tone pair scheme)  
analog........................................ 205  
T1/E1 ........................................ 131  
Gain 2 (custom tone) field  
(MVP-410/810/2410/3010).........33  
GUI (log reporting type) button  
analog........................................218  
T1/E1 ........................................143  
analog........................................ 208  
T1/E1 ........................................ 134  
Gain 2 (tone pair scheme)  
analog........................................ 205  
T1/E1 ........................................ 131  
gatekeeper interaction  
H
analog models ............................. 19  
E1 models ................................... 15  
T1 models ............................. 11, 19  
Gatekeeper Name field  
H.245 Tunneling field  
E1..............................................286  
T1..............................................242  
H.323  
E1.............................................. 285  
T1.............................................. 241  
GateKeeper RAS Parameters  
T1.............................................. 241  
Gateway (IP Parameters) field  
analog........................................ 176  
T1/E1 ........................................ 101  
compatibility (analog models).....18  
compatibility (E1 models)...........14  
compatibility (T1 models)...........10  
H.323 Annex E field  
E1..............................................287  
T1..............................................243  
H.323 coder  
analog........................................183  
426  
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Index  
T1/E1 ........................................ 108  
H.323 fields (Outbound Phonebook)  
E1.............................................. 291  
T1.............................................. 247  
H.323 ID (Gatekeeper) field  
E1.............................................. 285  
T1.............................................. 241  
H.323 ID (Outbound Phonebook)  
field  
T1...................................... 247, 291  
H.323 version 4 features  
E1................................................ 14  
T1.......................................... 10, 19  
H.323 Version 4 Parameters  
E1...................................... 286, 287  
T1...................................... 242, 243  
H.450 features, incompatible with SIP  
analog.................................. 19, 219  
E1................................................ 15  
T1................................................ 11  
T1/E1 ........................................ 145  
H.450 functionality  
in band, DTMF  
analog........................................182  
T1/E1 ........................................107  
inbound phonebook  
example.......................................49  
Inbound Phonebook Entries List icon  
E1..............................................282  
T1..............................................238  
Inbound Phonebook entries, list  
E1..............................................295  
T1..............................................251  
inbound phonebook example  
quick............................................43  
inbound vs. outbound phonebooks  
E1..............................................280  
T1..............................................236  
Industry Canada requirements.......395  
info sources  
analog telephony details......30, 166  
E1 details.....................................30  
E1 telephony details....................91  
IP details......................................29  
IP details (analog system) .........165  
IP details (T1/E1 system)............89  
SMTP details...............................31  
T1 details.....................................29  
T1 telephony details....................90  
voip email account ......................31  
info sources (analog models)  
logs for...................................... 334  
Hold Sequence  
analog........................................ 223  
T1/E1 ........................................ 149  
Hold Sequence (analog)................ 220  
Hold Sequence (T1/E1) ................ 146  
hookup  
MVP210...................................... 34  
MVP2400.................................... 34  
MVP2410.................................... 33  
MVP3010.................................... 33  
MVP410...................................... 33  
MVP810...................................... 33  
HyperTerminal program  
SMTP details.............................167  
voip email account ....................167  
info sources (T1/E1 models)  
SMTP details...............................92  
voip email account ......................92  
Input Gain field  
analog........................................181  
T1/E1 ........................................106  
installation  
airflow.........................................66  
analog prerequisites...........165, 166  
E1 prerequisites.....................30, 91  
expansion card (analog, 4-to-8  
channel).................................408  
expansion card (T1/E1).............398  
full summary ...............................28  
in a nutshell.................................26  
in rack..........................................65  
IP prerequisites......................29, 30  
and connectivity testing ........ 54, 55  
I
IANA ............................................ 406  
icon  
variable version....................... 9, 77  
icons, phonebook  
E1.............................................. 282  
T1.............................................. 238  
identifying current firmware version  
.................................................. 353  
implementing firmware upgrade... 353  
427  
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MultiVOIP User Guide  
log reports by email (analog  
models) ................................. 167  
log reports by email (T1/E1  
IP Address (callee, statistics, logs)  
field ...........................................333  
IP Address (caller, statistics, logs)  
field ...........................................333  
IP Address (Gatekeeper) field  
E1..............................................285  
T1..............................................241  
IP Address (outbound phonebook)  
E1..............................................290  
T1..............................................246  
IP Address field  
analog........................................176  
T1/E1 ........................................101  
IP Address, From Details (SMTP  
logs) field  
analog........................................214  
T1/E1 ........................................140  
IP address, SysLog Server  
models) ................................... 92  
software (detailed) ...................... 75  
T1 prerequisites..................... 29, 90  
T1/E1 prerequisites..................... 89  
upgrade card (analog, 4-to-8  
channel)................................. 408  
upgrade card (T1/E1)................ 398  
voip email account(analog models)  
.............................................. 167  
voip email account(T1/E1 models)  
................................................ 92  
installation preparations (optional)  
log reports by email .................... 31  
voip email account...................... 31  
installation, mechanical  
analog models ............................. 18  
E1 models ................................... 14  
T1 models ................................... 10  
installation, quick  
analog........................................218  
T1/E1 ........................................144  
IP Address, To Details (SMTP logs)  
field  
log reports by email .................... 31  
voip email account...................... 31  
installing Java vis-a-vis web GUI. 383  
integrated phone/data networks .... 275  
Inter Digit Regeneration Time field  
.................................................. 196  
Inter Digit Timer (dialing) field  
FXO .......................................... 195  
FXS Ground Start ..................... 192  
FXS Loop Start......................... 190  
Interface field (E&M)................... 199  
interface parameters, accessing..... 188  
interface parameters, setting ......... 188  
interfaces  
analog telephony......................... 33  
inter-office dialing  
E1.............................................. 276  
T1.............................................. 237  
inter-operation (analog)  
with T1/E1 voips......................... 18  
inter-operation with phone system  
analog models ............................. 18  
E1 models ................................... 14  
T1 models ................................... 10  
IP Address (call progress) field..... 328  
analog........................................214  
T1/E1 ........................................140  
IP Mask field  
analog........................................176  
T1/E1 ........................................101  
IP parameter definitions  
analog........................................176  
T1/E1 ........................................101  
IP Parameter fields (analog)  
DNS Server IP Address.............176  
Enable Diffserv .........................176  
Enable DNS...............................176  
Frame Type ...............................176  
FTP Server Enable ....................176  
Gateway ....................................176  
IP Address.................................176  
IP Mask.....................................176  
IP Parameter fields (T1/E1)  
DNS Server IP Address.............101  
Enable Diffserv .........................101  
Enable DNS...............................101  
Frame Type ...............................101  
FTP Server Enable ....................101  
Gateway ....................................101  
IP Address.................................101  
IP Mask.....................................101  
428  
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IP Parameters screen, accessing  
analog........................................ 174  
T1/E1 .......................................... 99  
IP startup configuration .................. 36  
IP Statistics field definitions. 335, 337  
IP Statistics fields  
Clear.......................................... 335  
Received (RTCP Packets)......... 338  
Received (RTP Packets) ........... 338  
Received (TCP Packets) ........... 337  
Received (Total Packets) .......... 335  
Received (UDP Packets)........... 337  
Received with errors (RTCP  
T1/E1 ........................................110  
K
key system  
connecting to analog voip  
(MVP210) ...............................73  
connecting to analog voip (MVP-  
410/810)..................................71  
L
lab voip network  
use in setup..................................48  
Last button  
Packets)................................. 338  
Received with errors (RTP Packets)  
.............................................. 338  
Received with errors (TCP Packets)  
.............................................. 337  
Received with errors (Total  
Logs (Statistics) screen .............332  
LED definitions  
analog models .............................21  
E1................................................16  
MVP2400....................................13  
MVP2410....................................13  
MVP3010....................................16  
T1................................................13  
LED definitions (analog)  
Boot.............................................21  
COL.............................................21  
Ethernet.......................................21  
LNK ............................................21  
Power ..........................................21  
RCV (channel)...........................21  
RCV (Ethernet) ...........................21  
RSG.............................................21  
XMT (channel) ..........................21  
XMT (Ethernet)...........................21  
XSG.............................................21  
LED definitions (E1)  
Boot.............................................16  
COL.............................................16  
E1................................................17  
IC ................................................17  
LC ...............................................17  
LNK ............................................16  
LS................................................17  
ONL ............................................17  
Power ..........................................16  
PRI ..............................................17  
RCV ............................................16  
XMT............................................16  
LED definitions (T1)  
Packets)................................. 337  
Received with errors (UDP  
Packets)................................. 337  
Transmitted (RTCP Packets) .... 338  
Transmitted (RTP Packets)....... 338  
Transmitted (TCP Packets)....... 337  
Transmitted (Total Packets)...... 335  
Transmitted (UDP Packets) ...... 337  
IP Statistics function..................... 335  
ISDN parameters, setting.............. 125  
ISDN-PRI  
types supported......................... 125  
ISDN-PRI implementations.......... 125  
J
Java  
installing ................................... 383  
web GUI and............................. 383  
jitter buffer  
analog........................................ 185  
T1/E1 ........................................ 110  
Jitter Value (Fax) field  
analog........................................ 182  
T1/E1 ........................................ 107  
Jitter Value field  
analog........................................ 187  
T1/E1 ........................................ 112  
jitter, dynamic  
analog........................................ 185  
Boot.............................................13  
429  
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MultiVOIP User Guide  
COL ............................................ 13  
detailed, T1/E1............................94  
summary, analog .......................168  
summary, T1/E1..........................93  
local exchange numbers..................47  
local voip configuration (analog) ..163  
local voip configuration (T1/E1).....87  
local Windows GUI vs. web GUI  
comparison................................382  
local-rate access (E1)  
to remote PSTN...........................14  
local-rate calls to remote voip sites  
E1..............................................277  
log report email, customizing  
analog................................213, 214  
T1/E1 ................................139, 140  
log report email, triggering  
analog.......................................212  
T1/ E1 .......................................138  
log reporting method, setting  
IC ................................................ 13  
LC ............................................... 13  
LNK............................................ 13  
LS................................................ 13  
ONL............................................ 13  
Power.......................................... 13  
PRI.............................................. 13  
RCV............................................ 13  
T1................................................ 13  
XMT ........................................... 13  
LED indicators  
E1................................................ 16  
T1................................................ 12  
LED indicators (analog)  
channel operation........................ 20  
general operation......................... 20  
LED indicators, active  
analog.......................................... 20  
E1................................................ 16  
T1................................................ 12  
LED sets (T1/E1), left and right ... 400  
LED types  
analog........................................216  
T1/E1 ........................................142  
log reports  
analog models ...........................167  
T1/E1 models..............................92  
log reports & SMTP  
analog models ............................. 20  
lifting  
precaution about.......................... 60  
limitations notice (regulatory),  
Canadian ................................... 396  
limited warranty............................ 390  
Line Build Out field  
analog........................................210  
T1/E1 ........................................136  
log reports and SMTP  
quick............................................37  
log reports by email  
E1.............................................. 124  
T1.............................................. 119  
Line Coding field  
analog........................................210  
quick............................................37  
T1/E1 ........................................136  
log reports, quick.............................31  
logging options  
analog........................................217  
T1/E1 ........................................143  
logging update interval  
analog........................................217  
T1/E1 ........................................143  
logging, web GUI and...................382  
Login Name (SMTP) field  
analog........................................211  
T1/E1 ........................................137  
Logs (Statistics) fields  
E1.............................................. 124  
T1.............................................. 119  
Line Loopback Activation Signal (T1  
stats) field.................................. 342  
Line Loopback Deactivation Signal  
(T1 stats) field........................... 341  
lithium battery caution.................... 60  
LNK LED  
analog models ............................. 21  
loading of weight in rack ................ 66  
local configuration  
analog........................................ 165  
T1/E1 .......................................... 89  
local configuration procedure  
detailed, analog......................... 169  
Bytes received...........................333  
Bytes Sent .................................332  
Call Forwarded to......................334  
430  
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Index  
Call Transferred to.................... 334  
CT Ph#...................................... 334  
Duration.................................... 332  
Event #...................................... 332  
From (gateway)......................... 332  
Gateway Name (callee)............. 333  
Gateway Name (caller) ............. 333  
H.450 functionality................... 334  
IP Address (callee).................... 333  
IP Address (caller) .................... 333  
Mode......................................... 332  
Options (caller) ......................... 333  
Options callee ........................... 333  
Outbound digits......................... 333  
Packets Lost.............................. 333  
Packets received........................ 333  
Packets Sent.............................. 332  
Start Date, Time........................ 332  
Status ........................................ 332  
Supplementary Services info .... 334  
To (gateway)............................. 332  
Voice coder............................... 333  
Logs (Statistics) function ........... 331  
Logs (Statistics) screen  
Delete File button...................... 332  
Last button ................................ 332  
logs and web browser GUI  
analog........................................ 217  
T1/E1 ........................................ 143  
logs by email, illustration  
analog........................................ 215  
T1/E1 ........................................ 141  
Logs screen definitions  
analog........................................ 217  
T1/E1 ........................................ 143  
Logs screen field definitions  
Turn Off Logs ...........................218  
Logs screen parameters (T1/E1)  
Enable Console Messages.........143  
GUI ...........................................143  
IP Address (SysLog Server)......143  
Online Statistics Updation Interval  
..............................................143  
Port (SysLog Server).................143  
SMTP........................................143  
SNMP........................................143  
SysLog Server Enable...............143  
Turn Off Logs ...........................143  
logs screen, accessing  
analog........................................216  
T1/E1 ........................................142  
long distance call savings  
T1..............................................236  
long-distance call savings  
E1..............................................275  
Long-Haul Mode field  
E1..............................................121  
T1..............................................116  
Loss of Frame Alignment (E1 stats)  
field ...........................................344  
Loss of Frame Alignment (T1 stats)  
field ...........................................341  
Loss of MultiFrame Alignment (E1  
stats) field..................................345  
Loss of MultiFrame Alignment (T1  
stats) field..................................342  
lost packets, consecutive  
analog........................................187  
T1/E1 ........................................112  
lost password.........................364, 367  
M
analog........................................ 218  
T1/E1 ........................................ 143  
Logs screen parameters (analog)  
Enable Console Messages......... 218  
GUI........................................... 218  
IP Address (SysLog Server)...... 218  
Online Statistics Updation Interval  
.............................................. 218  
Port (SysLog Server) ................ 218  
SMTP........................................ 218  
SNMP ....................................... 218  
SysLog Server Enable............... 218  
Mac Address  
System Info (analog).................232  
System Info (T1/E1)..........158, 323  
mail criteria (SMTP), records  
analog........................................212  
T1/E1 ........................................138  
Mail Server IP Address (SMTP) field  
analog........................................211  
T1/E1 ........................................137  
Mail Type (SMTP logs) field  
analog........................................212  
T1/E1 ........................................138  
431  
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MultiVOIP User Guide  
mains frequency  
T1/E1 ........................................110  
Mode (call progress) field .............327  
Mode (Fax) field  
analog........................................182  
T1/E1 ........................................107  
Mode (statistics, logs) field...........332  
model descriptions  
analog models ............................. 25  
E1 models ................................... 24  
T1 models ................................... 23  
management (analog)  
local ............................................ 19  
remote (SNMP)........................... 19  
remote (web browser GUI)......... 19  
management (E1 models)  
E1................................................14  
mounting  
local ............................................ 15  
remote (SNMP)........................... 15  
remote (web browser GUI)......... 15  
management (T1 models)  
local ............................................ 11  
remote (SNMP)........................... 11  
remote (web browser GUI)......... 11  
management of voips, remote  
analog........................................ 200  
T1/E1 ........................................ 126  
Max bandwidth (coder)  
analog........................................ 183  
T1/E1 ........................................ 108  
Max Baud Rate field  
analog........................................ 182  
T1/E1 ........................................ 107  
maximum cable span  
E1.............................................. 121  
T1.............................................. 116  
Maximum Jitter Value field  
analog........................................ 186  
T1/E1 ........................................ 111  
Message Waiting Light (FXO)  
and Avaya Magix PBX............. 195  
and DID .................................... 195  
Message Waiting Light (FXS Ground  
Start)  
and Avaya Magix PBX............. 192  
and DID .................................... 192  
Message Waiting Light (FXS Loop  
Start)  
and Avaya Magix PBX............. 191  
and DID .................................... 191  
Message Waiting Light field  
analog models .............................18  
E1 models....................................14  
T1 models....................................10  
mounting in rack .............................65  
procedure for...............................67  
safety.....................................60, 66  
mounting options...............................8  
mounting, physical  
quick............................................32  
Multiplexed UDP field  
E1..............................................287  
T1..............................................243  
MultiVOIP configuration software .35  
analog..........................................19  
E1 models....................................15  
T1 models....................................11  
MultiVOIP Program Menu items..347  
MultiVOIP Program Menu options  
Configuration ............................347  
Date & Time Setup ...................347  
Download CAS Protocol...........347  
Download Factory Defaults ......348  
Download Firmware..................348  
Set Password .............................348  
Uninstall....................................348  
Upgrade Software .....................348  
MultiVOIP program menu, option  
descriptions .......................347, 348  
MultiVOIP software  
installing......................................75  
location of files............................78  
program icon location .................79  
uninstalling..........................82, 368  
MultiVOIP software (analog)  
FXO .......................................... 195  
FXS Ground Start ..................... 192  
FXS Loop Start......................... 191  
Minimum Jitter Value field  
moving around in ......................173  
MultiVOIP software (T1/E1)  
moving around in ........................98  
MultiVoipManager............................9  
analog........................................163  
analog........................................ 185  
432  
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Index  
T1/E1 .......................................... 88  
MultiVoipManager software  
analog.......................................... 19  
E1 models ................................... 15  
T1 models ................................... 11  
MVP210  
grounding.................................... 73  
unpacking.................................... 64  
MVP2400  
cabling procedure........................ 69  
unpacking.................................... 62  
MVP2410  
cabling procedure........................ 68  
unpacking.................................... 61  
MVP3010  
cabling procedure........................ 68  
unpacking.................................... 61  
MVP410  
cabling procedure.................. 70, 72  
grounding.................................... 71  
unpacking.................................... 63  
MVP810  
cabling procedure........................ 70  
grounding.................................... 71  
unpacking.................................... 63  
O
obtaining updated firmware ..........349  
official phone numbers  
caution about...............................48  
Online Statistics Updation Interval  
field (Logs)  
analog........................................218  
T1/E1 ........................................144  
operating system ..........................22  
operating temperature .....................66  
operating voltage  
analog models .............................25  
T1 models..............................23, 24  
operation  
expansion card (T1/E1).............400  
Operator (ISDN) field  
E1/ISDN....................................123  
T1/ISDN....................................118  
Optimization Factor field  
analog........................................186  
T1/E1 ........................................111  
Options (call progress) field..........328  
Options (callee, statistics, logs) field  
..................................................333  
Options, From Details (SMTP logs)  
field  
N
analog........................................214  
T1/E1 ........................................140  
Options, To Details (SMTP logs) field  
analog........................................214  
T1/E1 ........................................140  
out of band, DTMF  
analog........................................182  
T1/E1 ........................................107  
Outbound Digits (call progress) field  
..................................................327  
Outbound Digits (SMTP logs) field  
analog........................................214  
T1/E1 ........................................140  
Outbound digits (statistics, logs) field  
..................................................333  
outbound phonebook  
national-rate calls to foreign voip sites  
E1.............................................. 279  
network access  
analog........................................ 202  
T1/E1 ........................................ 128  
Network Disconnection field  
analog........................................ 187  
T1/E1 ........................................ 112  
network/terminal settings, voip and  
PBX  
E1/ISDN ................................... 123  
T1/ISDN ................................... 118  
Number of Days (email log criteria)  
analog........................................ 212  
T1/E1 ........................................ 138  
Number of Records (email log  
criteria)  
analog........................................ 212  
T1/E1 ........................................ 138  
numbering plan resources ............. 318  
example.......................................49  
Outbound Phonebook Entries List  
icon  
E1..............................................282  
T1..............................................238  
Outbound Phonebook entries, list  
433  
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Index  
MultiVOIP User Guide  
E1.............................................. 288  
T1.............................................. 244  
outbound phonebook example  
tips about.....................................46  
PBX characteristics, variations in  
E1..............................................317  
T1..............................................273  
PBX interaction  
analog models .............................18  
E1 models....................................14  
T1 models....................................10  
PC, command  
COM port assignment (detailed).80  
COM port requirement................32  
demands upon .............................32  
non-dedicated use........................32  
operating system..........................32  
settings ........................................32  
specifications...............................32  
PCM Law field  
E1..............................................124  
T1..............................................119  
Permissions (SNMP) field  
analog........................................202  
T1/E1 ........................................128  
personnel requirement  
for rack installation .....................66  
to lift during installation..............67  
to lift unit during installation.......60  
phone exchanges  
dedicated .....................................47  
institutional..................................47  
local.............................................47  
non-local .....................................47  
organizational..............................47  
Phone Number (Auto Call) field  
analog........................................184  
Phone Number (Auto Call)field  
T1/E1 ........................................109  
Phone Signaling Tones & Cadences  
analog........................................203  
T1/E1 ........................................129  
phone startup configuration.............36  
phone switch types  
quick ........................................... 39  
outbound vs. inbound phonebooks  
E1.............................................. 280  
T1.............................................. 236  
Output Gain field  
analog........................................ 181  
T1/E1 ........................................ 106  
output level, fax tones  
analog........................................ 182  
T1/E1 ........................................ 107  
outside line, access to................ 47, 49  
P
Packets Lost (call progress) field.. 327  
Packets Lost (SMTP logs) field  
analog........................................ 213  
T1/E1 ........................................ 139  
Packets lost (statistics, logs) field . 333  
Packets Received (call progress) field  
.................................................. 327  
Packets Received (SMTP logs) field  
analog........................................ 213  
T1/E1 ........................................ 139  
Packets received (statistics, logs) field  
.................................................. 333  
Packets Sent (call progress) field.. 327  
Packets Sent (SMTP logs) field  
analog........................................ 213  
T1/E1 ........................................ 139  
Packets sent (statistics, logs) field 332  
packets, consecutive lost  
analog........................................ 187  
T1/E1 ........................................ 112  
Parallel H.245 field  
E1.............................................. 287  
T1.............................................. 243  
Pass Through (E&M) field ........... 199  
Password (SMTP) field  
ISDN implementations in..........125  
phone/IP details  
importance of writing down........28  
importance of writing down  
(analog) .................................165  
importance of writing down  
analog........................................ 211  
T1/E1 ........................................ 137  
password, lost/forgotten........ 364, 367  
password, setting........................... 364  
web browser GUI...................... 367  
patents ............................................. 2  
patterns, destination  
(T1/E1)....................................89  
434  
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MultiVOIP User Guide  
Index  
phonebook  
FTP remote file transfers .......... 371  
phonebook configuration  
Gateway Name..........................241  
Gateway Prefix..........................241  
H.245 Tunneling .......................242  
H.323 ID ...................................241  
IP Address (Gatekeeper) Name.241  
Parallel H.245 (Tunneling with Fast  
Start)......................................243  
Port Number (Gatekeeper) ........241  
Port Number (proxy server) ......242  
Proxy Server IP Address...........242  
Q.931 Multiplexing...................242  
Use Fast Start ............................241  
phonebook destination patterns.......46  
phonebook dialing patterns .............46  
phonebook digits  
dropping ......................................47  
leading.........................................47  
non-PSTN type............................47  
specialized codes.........................47  
types used....................................46  
phonebook entries, coordinating  
E1..............................................281  
T1..............................................237  
phonebook examples  
starter .......................................... 39  
phonebook configuration (analog)162,  
321  
phonebook configuration (remote) 371  
phonebook configuration (T1/E1) .. 87  
Phonebook Configuration icon  
E1.............................................. 282  
T1.............................................. 238  
Phonebook Configuration Parameter  
definitions  
E1.............................. 285, 286, 287  
T1.............................. 241, 242, 243  
Phonebook Configuration procedure  
T1.............................................. 237  
Phonebook Configuration Procedure  
E1.............................................. 281  
Phonebook Configuration screen  
E1.............................................. 284  
T1.............................................. 237  
Phonebook Configuration screen  
fields (E1)  
Annex E (H.323, UDP  
analog........................................162  
mixed digital/analog....................49  
phonebook icons  
E1..............................................282  
T1..............................................238  
phonebook objectives &  
considerations  
E1..............................................280  
phonebook sidebar menu  
E1..............................................283  
T1..............................................239  
phonebook tips................................46  
phonebook worksheet ...............52, 53  
phonebook, analog voips...............321  
phonebook, inbound  
example.......................................49  
example, quick ............................43  
phonebook, outbound  
example.......................................49  
example, quick ............................39  
phonebooks, inbound vs. outbound  
E1..............................................280  
T1..............................................236  
multiplexing)......................... 287  
Call Signalling Port................... 285  
Enable Proxy............................. 286  
Gatekeeper Name...................... 285  
Gateway Name.......................... 285  
Gateway Prefix ......................... 285  
H.245 Tunneling....................... 286  
H.323 ID................................... 285  
IP Address (Gatekeeper) Name 285  
Parallel H.245 (Tunneling with Fast  
Start) ..................................... 287  
Port Number (Gatekeeper)........ 285  
Port Number (proxy server)...... 286  
Proxy Server IP Address........... 286  
Q.931 Multiplexing................... 286  
Use Fast Start............................ 285  
Phonebook Configuration screen  
fields (T1)  
Annex E (H.323, UDP  
multiplexing)......................... 243  
Call Signalling Port................... 241  
Enable Proxy............................. 242  
Gatekeeper Name...................... 241  
435  
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MultiVOIP User Guide  
phonebooks, objectives &  
considerations  
T1.............................................. 236  
Phonebooks, objectives &  
considerations  
for technical configuration (T1/E1)  
................................................89  
prerequisites for installation  
E1 info.........................................30  
IP info..........................................29  
T1 info.........................................29  
PRI  
ISDN implementations..............125  
product CD......................................26  
use in software installation....35, 75  
Product CD  
analog..........................................19  
E1 models....................................15  
T1 models....................................11  
product family...............................8, 9  
product groups ..................................8  
Program Menu items.....................347  
Protocol Type (outbound phonebook)  
E1..............................................290  
T1..............................................246  
Proxy Server IP Address  
E1..............................................286  
T1..............................................242  
Proxy Server Parameters  
E1..............................................286  
T1..............................................242  
Pulse Density Violation (T1 stats)  
field ...........................................342  
Pulse Generation Ratio (analog) field  
..................................................206  
Pulse Shape Level field  
E1.............................................. 275  
phonebooks, sample........................ 51  
physical mounting, quick................ 32  
pinging and connectivity................. 58  
pinout  
command cable ......................... 402  
ethernet cable............................ 402  
T1/E1 connector........................ 403  
Voice/FAX connector............... 403  
placement (physical), quick ............ 32  
Port field, SysLog Server  
analog........................................ 218  
T1/E1 ........................................ 144  
Port Number (Gatekeeper) field  
E1.............................................. 285  
T1.............................................. 241  
Port Number (proxy server)  
E1.............................................. 286  
T1.............................................. 242  
Port Number (SMTP) field  
analog........................................ 211  
T1/E1 ........................................ 137  
port number (SNMP) field  
analog........................................ 202  
T1/E1 ........................................ 128  
power consumption  
analog models ............................. 25  
E1 models ................................... 24  
T1 models ................................... 23  
power frequency  
E1..............................................124  
T1..............................................119  
Q
analog models ............................. 25  
E1 models ................................... 24  
T1 models ................................... 23  
Power LED  
Q.931 Multiplexing field  
E1..............................................286  
T1..............................................242  
Q.931 Parameters  
analog models ............................. 21  
Prefix Matched (call progress) field  
.................................................. 327  
Prefix Matched (SMTP logs) field  
analog........................................ 214  
T1/E1 ........................................ 140  
prerequisites  
T1..............................................241  
Q.931 Port Number (outbound  
phonebook) field  
E1..............................................291  
T1..............................................247  
R
for technical configuration (analog)  
.............................................. 165  
rack mounting  
grounding ....................................66  
436  
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Index  
safety..................................... 60, 66  
rack mounting instructions.............. 65  
rack mounting procedure ................ 67  
rack, equipment  
weight capacity of....................... 66  
rack-mountable voip models........... 60  
RCV (channel) LED  
Regeneration (dialing, FXO) field 195  
Regional Parameter definitions  
analog................................205, 206  
T1/E1 ................................131, 132  
Regional Parameter fields (analog)  
Cadence.....................................206  
Custom (tones)..........................206  
Pulse Generation Ratio..............206  
Regional Parameter fields (T1/E1)  
Cadence.....................................132  
Country/Region (tone schemes) 131  
Custom (tones)..........................132  
Frequency 1...............................131  
Frequency 2...............................131  
Gain 1........................................131  
Gain 2........................................131  
type (of tone).............................131  
regional parameters, setting  
analog........................................203  
T1/E1 ........................................129  
remote control/configuration  
web GUI and .............................383  
remote phonebook configuration ..371  
remote voip  
using to confirm configuration31,  
39  
remote voip configuration (analog)  
..................................................163  
remote voip configuration (T1/E1) .87  
Remote Voip Management  
analog........................................200  
T1/E1 ........................................126  
Remove Prefix (inbound) field  
E1..............................................296  
T1..............................................252  
Remove Prefix (outbound) field  
E1..............................................290  
T1..............................................246  
repair procedures for customers U.S.  
& Canada ..................................390  
Reply-To Address (email logs) field  
T1/E1 ........................................138  
Reply-To Address (email logs)field  
analog........................................212  
Reports function............................334  
Retrieve Sequence  
analog models........................... 21  
RCV (Ethernet) LED  
analog models ............................. 21  
Receive Slip (E1 Stats) field......... 346  
Receive Slip (T1 Stats) field......... 343  
Receive Timeslot 16 Alarm Indication  
Signal (E1 stats) field................ 345  
Receive Timeslot 16 Loss of  
MultiFrame Alignment (E1 stats)  
field........................................... 346  
Receive Timeslot 16 Loss of Signal  
(E1 stats) field........................... 345  
Received (RTCP Packets, IP Stats)  
field........................................... 338  
Received (RTP Packets, IP Stats) field  
.................................................. 338  
Received (TCP Packets, IP Stats) field  
.................................................. 337  
Received (Total Packets, IP Stats)  
field........................................... 336  
Received (UDP Packets, IP Stats)  
field........................................... 337  
Received with Errors (RTCP Packets,  
IP Stats) field ............................ 338  
Received with Errors (RTP Packets,  
IP Stats) field ............................ 338  
Received with Errors (TCP Packets,  
IP Stats) field ............................ 337  
Received with Errors (Total Packets,  
IP Stats) field ............................ 337  
Received with Errors (UDP Packets,  
IP Stats) field ............................ 337  
Recipient Address (email logs) field  
T1/E1 ........................................ 138  
Recipient Address (email logs)field  
analog........................................ 212  
recovering voice packets  
analog........................................ 184  
T1/E1 ........................................ 109  
Red Alarm (E1 stats) field ............ 344  
Red Alarm (T1 stats) field ............ 341  
analog........................................223  
T1/E1 ........................................149  
437  
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MultiVOIP User Guide  
Retrieve Sequence (analog) .......... 220  
Retrieve Sequence (T1/E1)........... 146  
RFC768......................................... 406  
RFC793......................................... 406  
ring cadences, custom  
analog................................ 207, 209  
T1/E1 ................................ 133, 135  
Ring Count (FXO) field................ 197  
Ring Count field  
FXS Ground Start ..................... 193  
FXS Loop Start......................... 191  
Ring Count forwarding condition  
E1.............................................. 297  
T1.............................................. 253  
ring tone, custom  
analog................................ 207, 208  
T1/E1 ................................ 133, 134  
ring-tones  
T1................................................10  
second dial tone  
and comma use............................48  
Select All (SMTP logs) field  
analog........................................213  
T1/E1 ........................................139  
Select Channel field  
analog........................................181  
T1/E1 ........................................106  
Select Channel, Supplementary  
Services field  
analog........................................222  
T1/E1 ........................................148  
Selected Coder field  
analog........................................183  
T1/E1 ........................................108  
Set Baud Rate  
analog........................................230  
T1/E1 ........................................156  
Set Custom Tones & Cadences  
analog........................................207  
T1/E1 ........................................133  
Set ISDN Parameters ....................125  
Set Log Reporting Method  
analog........................................ 207  
T1/E1 ........................................ 133  
Round Trip Delay field  
E1.............................................. 294  
T1.............................................. 250  
RSG LED  
analog........................................216  
T1/E1 ........................................142  
Set Password (program menu option) ,  
command...................................364  
Set Password (web browser GUI) ,  
command...................................367  
Set Password option description  
(MultiVOIP program menu) .....348  
Set Regional Parameters  
analog........................................203  
T1/E1 ........................................129  
Set SMTP Parameters  
analog........................................210  
T1/E1 ........................................136  
Set SNMP Parameters  
analog models........................... 21  
S
Safety Recommendations for Rack  
Installations................................. 66  
safety warnings ............................... 60  
Safety Warnings Telecom.......... 60  
sample phonebooks......................... 51  
Save Setup command  
analog........................................ 233  
T1/E1 ........................................ 159  
saving configuration  
analog........................................ 233  
T1/E1 ........................................ 159  
user............................................ 362  
saving tech configuration................ 38  
Saving the MultiVOIP Configuration  
analog........................................ 233  
T1/E1 ........................................ 159  
savings on toll calls  
analog........................................200  
T1/E1 ........................................126  
Set Supplementary Services  
Parameters  
analog........................................219  
T1/E1 ........................................145  
Set T1/E1/ISDN Parameters .........113  
Set Telephony Interface Parameters  
..................................................188  
E1.............................................. 275  
T1.............................................. 236  
scale-ability  
E1................................................ 14  
438  
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Set Voice/FAX Parameters  
SIP incompatibility with H.450  
analog........................................ 178  
T1/E1 ........................................ 103  
setting IP parameters  
analog........................................ 174  
T1/E1 .......................................... 99  
setting password............................ 364  
web browser GUI...................... 367  
setting user defaults ...................... 362  
setup, saving  
Supplementary Services  
analog..................................19, 219  
E1................................................15  
T1................................................11  
T1/E1 ........................................145  
SIP Port Number field  
E1..............................................292  
T1..............................................248  
SIP port number, standard  
analog........................................ 233  
T1/E1 ........................................ 159  
user............................................ 362  
setup, saving user values............... 362  
Signal (type, E&M) field .............. 199  
signaling cadences  
E1..............................................292  
T1..............................................248  
SIP URL field  
E1..............................................292  
T1..............................................248  
SMTP  
analog........................................ 203  
T1/E1 ........................................ 129  
signaling parameters (analog  
telephony) ................................. 188  
signaling tones  
analog........................................ 203  
T1/E1 ........................................ 129  
signaling types  
quick setup ..................................37  
SMTP (log reporting type) button  
analog........................................218  
T1/E1 ........................................144  
SMTP logs by email, illustration  
analog........................................215  
T1/E1 ........................................141  
SMTP Parameters definitions  
analog telephony......................... 33  
analog telephony (MVP210)....... 73  
analog telephony (MVP-410/810)  
................................................ 71  
Silence Compression (call progress)  
field........................................... 328  
Silence Compression (SMTP logs)  
analog........................................ 214  
T1/E1 ........................................ 140  
Silence Compression field  
analog........................................ 184  
T1/E1 ........................................ 109  
Silence Detection (FXO) field ...... 197  
Silence Timer (FXO) field............ 197  
simulated voip network  
use in startup............................... 48  
SIP  
compatibility  
analog models ......................... 19  
E1 models ............................... 15  
T1 models ............................... 11  
SIP Fields (Outbound Phonebook)  
E1.............................................. 292  
T1.............................................. 248  
analog........................................211  
T1/E1 ........................................137  
SMTP Parameters fields (analog)  
Enable SMTP............................211  
Login Name...............................211  
Mail Server IP Address.............211  
Mail Type..................................212  
Number of Days........................212  
Number of Records ...................212  
Password ...................................211  
Port Number..............................211  
Recipient Address .....................212  
Reply-To Address .....................212  
Subject.......................................212  
SMTP Parameters fields (T1/E1)  
Enable SMTP............................137  
Login Name...............................137  
Mail Server IP Address.............137  
Mail Type..................................138  
Number of Days........................138  
Number of Records ...................138  
Password ...................................137  
Port Number..............................137  
Recipient Address .....................138  
439  
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MultiVOIP User Guide  
Reply-To Address..................... 138  
software  
Subject ...................................... 138  
SMTP parameters, accessing  
analog........................................ 210  
T1/E1 ........................................ 136  
SMTP parameters,setting  
control .........................................35  
uninstalling (detailed) .................82  
updates (analog) ........................163  
updates (T1/E1)...........................88  
software (MultiVOIP)  
analog........................................ 210  
T1/E1 ........................................ 136  
SMTP port, standard  
analog ...................................... 211  
T1/ E1....................................... 137  
SMTP prerequisites  
analog models ........................... 167  
quick ........................................... 31  
T1/E1 models.............................. 92  
SMTP, enabling  
analog........................................ 210  
T1/E1 ........................................ 136  
SNMP (log reporting type) button  
analog........................................ 218  
T1/E1 ........................................ 144  
SNMP agent program  
uninstalling................................368  
software configuration  
summary......................................75  
software installation  
detailed........................................75  
quick............................................35  
software loading..............................75  
software loading, quick...................35  
software version numbers ...............77  
software, MultiVOIP (analog)  
screen-surfing in........................173  
software, MultiVOIP (T1/E1)  
moving around in ........................98  
screen-surfing in..........................98  
software, MultiVOIP(analog)  
moving around in ......................173  
software, on command PC ..............35  
Solving Common Connection  
analog........................................ 163  
T1/E1 .......................................... 88  
SNMP agent, enabling  
Problems  
analog........................................172  
T1/E1 ..........................................97  
sound quality, improving  
analog........................................ 200  
T1/E1 ........................................ 126  
SNMP Parameter Definitions  
analog........................................184  
T1/E1 ........................................109  
specialized codes, in dialing............47  
specifications  
analog models .............................25  
E1 models....................................24  
T1 models....................................23  
Start Date, Time (SMTP logs) field  
analog........................................213  
T1/E1 ........................................139  
Start Date,Time (statistics, logs) field  
..................................................332  
starter configuration  
inbound phonebook.....................43  
outbound phonebook...................39  
phone/IP ......................................36  
startup tasks.....................................28  
Options (caller...............................333  
Status (call progress) field.............330  
Status (statistics, logs) field...........332  
T1/E1 ........................................ 128  
SNMP Parameter fields (analog)  
Address ..................................... 202  
Community Name (2)............... 202  
Community Name 1.................. 202  
Enable SNMP Agent................. 202  
Permissions (1) ......................... 202  
Permissions (2) ......................... 202  
Port Number.............................. 202  
SNMP Parameter fields (T1/E1)  
Address ..................................... 128  
Community Name (2)............... 128  
Community Name 1.................. 128  
Enable SNMP Agent................. 128  
Permissions (1) ......................... 128  
Permissions (2) ......................... 128  
Port Number.............................. 128  
SNMP Parameters, setting  
analog........................................ 200  
T1/E1 ........................................ 126  
440  
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Index  
Status Freeze Signalling Active (E1  
stats) field.................................. 345  
Status Freeze Signalling Active (T1  
stats) field.................................. 341  
Subject (email logs) field  
analog........................................ 212  
T1/E1 ........................................ 138  
supervisory signaling  
analog telephony......................... 33  
supervisory signaling (analog)...... 189  
supervisory signaling parameters  
(analog telephony) .................... 188  
supervisory signaling types  
Enable Call Transfer .................148  
Enable Call Waiting..................149  
Enable Caller Name Identification  
..............................................150  
Hold Sequence ..........................149  
Retrieve Sequence.....................149  
Select Channel...........................148  
Transfer Sequence.....................148  
Supplementary Services Info  
logs for ......................................334  
Supplementary Services Parameter  
buttons (analog)  
Copy Channel............................229  
Default.......................................229  
Supplementary Services Parameter  
buttons (T1/E1)  
Copy Channel............................155  
Default.......................................155  
Supplementary Services Parameter  
Definitions  
MVP210...................................... 73  
MVP-410/810 ............................. 71  
Supplementary Services (analog)  
Alerting Party............ 226, 227, 228  
Call Hold................................... 220  
Call Hold Enable....................... 223  
Call Name Identification........... 220  
Call Transfer ............................. 220  
Call Waiting.............................. 220  
Call Waiting Enable.................. 223  
Caller Name Identification Enable  
.............................................. 224  
Calling Party............................. 225  
Enable Call Hold....................... 223  
Enable Call Transfer ................. 222  
Enable Call Waiting.................. 223  
Enable Caller Name Identification  
.............................................. 224  
Hold Sequence.......................... 223  
Retrieve Sequence..................... 223  
Select Channel .......................... 222  
Transfer Sequence..................... 222  
Supplementary Services (T1/E1)  
Alerting Party............ 152, 153, 154  
Call Hold................................... 146  
Call Hold Enable....................... 149  
Call Name Identification........... 147  
Call Transfer ............................. 146  
Call Transfer Enable ......... 148, 222  
Call Waiting.............................. 146  
Call Waiting Enable.................. 149  
Caller Name Identification Enable  
.............................................. 150  
Calling Party............................. 151  
Enable Call Hold....................... 149  
analog222, 223, 224, 225, 226, 227,  
228, 229  
T1/E1148, 149, 150, 151, 152, 153,  
154, 155  
Supplementary Services Parameter  
fields (analog)  
Call Transfer Enable .................222  
Call Waiting Enable..................223  
Hold Sequence ..........................223  
Retrieve Sequence.....................223  
Transfer Sequence.....................222  
Supplementary Services Parameter  
fields (analog)  
Alerting Party............................226  
Allowed Name Types225, 226,  
227, 228  
Busy Party.................................227  
Call Hold Enable.......................223  
Call Name Identification Enable224  
Caller ID....................................229  
Calling Party .............................225  
Connected Party........................228  
Select Channel...........................222  
Supplementary Services Parameter  
fields (T1/E1)  
Call Transfer Enable .................148  
Call Waiting Enable..................149  
Hold Sequence ..........................149  
441  
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Index  
MultiVOIP User Guide  
Retrieve Sequence..................... 149  
System Information screen  
Transfer Sequence..................... 148  
Supplementary Services Parameter  
fields (T1/E1)  
Alerting Party............................ 152  
Allowed Name Types151, 152,  
153, 154  
for op & maint...........................323  
System Information screen, accessing  
analog........................................231  
T1/E1 ........................................157  
System Information update interval,  
setting  
Busy Party................................. 153  
Call Hold Enable....................... 149  
Call Name Identification Enable150  
Caller ID ................................... 155  
Calling Party............................. 151  
Connected Party........................ 154  
Select Channel .......................... 148  
Supplementary Services Parameters  
screen, accessing  
analog........................................ 219  
T1/E1 ........................................ 145  
Supplementary Services parameters,  
setting  
analog........................................ 219  
T1/E1 ........................................ 145  
Supplementary Services, incompatible  
with SIP  
analog.................................. 19, 219  
E1................................................ 15  
T1................................................ 11  
T1/E1 ........................................ 145  
support, technical.......................... 392  
switch types (phone) and ISDN-PRI  
.................................................. 125  
SysLog client programs  
availability ................................ 386  
features & presentation types.... 388  
SysLog Server Enable field  
analog........................................ 218  
T1/E1 ........................................ 144  
SysLog Server function  
as added feature ........................ 386  
capabilities of............................ 388  
enabling..................................... 387  
location of................................. 386  
SysLog Server IP Address field  
analog........................................ 218  
T1/E1 ........................................ 144  
SysLog Server, enabling  
analog........................................231  
for op & maint...........................324  
T1/E1 ........................................157  
T
T1 model descriptions.....................10  
T1 Parameter definitions116, 117, 119  
Clocking....................................119  
Line Build-Out..........................119  
Line Coding...............................119  
PCM Law..................................119  
Pulse Shape Level .....................119  
Yellow Alarm Format ...............119  
T1 Parameter fields  
CAS Protocol ........................117  
CRC Check ...............................116  
Frame Format............................116  
Long-Haul Mode.......................116  
T1/E1/ISDN ..............................116  
T1 Parameters screen ....................115  
T1 Statistics field definitions 342, 343  
T1 Statistics fields  
Bipolar Violation.......................343  
Frame Search Restart Flag ........342  
Line Loopback Activation Signal  
..............................................342  
Loss of MultiFrame Alignment.342  
Pulse Density Violation.............342  
Receive Slip ..............................343  
Transmit Data Underrun ...........342  
Transmit Line Open ..................342  
Transmit Slip.............................342  
Transmit Slip Negative .............342  
Yellow Alarm............................342  
T1 telephony parameters.................90  
T1/E1 connector pinout.................403  
T1/ E1 Statistics function ...........339  
T1/E1/ISDN field  
E1..............................................121  
T1..............................................116  
analog........................................ 217  
T1/E1 ........................................ 143  
442  
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MultiVOIP User Guide  
Index  
T1/E1/ISDN Parameters screen,  
accessing................................... 113  
T1/E1/ISDN parameters, setting... 113  
T1/ISDN Parameter definitions .... 118  
T1/ISDN Parameter fields  
Country ..................................... 118  
Enable ISDN-PRI ..................... 118  
Operator.................................... 118  
Terminal Network..................... 118  
table-top voip models...................... 60  
TCP/UDP compared  
operating .....................................66  
Terminal Network field  
E1/ISDN....................................123  
T1/ISDN....................................118  
terminal/network settings, voip and  
PBX  
E1/ISDN....................................123  
T1/ISDN....................................118  
tips, phonebook...............................46  
To (gateway, statistics, logs) field.332  
toll-call savings  
E1.............................................. 292  
IP Statistics context................... 336  
T1.............................................. 248  
tech configuration  
saving.......................................... 38  
technical configuration  
E1..............................................275  
T1..............................................236  
toll-free access (T1)  
to remote PSTN...........................10  
within voip network ....................10  
toll-free access (within voip network)  
E1................................................14  
T1................................................10  
Tone Detection (FXO disconnect  
criteria) field..............................196  
Tone Pair (custom) field  
analog........................................208  
T1/E1 ........................................134  
tone pairs, custom  
analog........................................207  
T1/E1 ........................................133  
tones, signaling  
analog........................................203  
T1/E1 ........................................129  
Total Digits (outbound) field  
E1..............................................290  
T1..............................................246  
touch tone frequencies...................197  
Transfer Sequence  
startup ......................................... 36  
technical configuration (analog)  
prerequisites to.......................... 165  
summary ................................... 162  
technical configuration (T1/E1)  
prerequisites to............................ 89  
summary ..................................... 87  
technical configuration procedure  
detailed, analog......................... 169  
detailed, T1/E1............................ 94  
summary, analog....................... 168  
summary, T1/E1 ......................... 93  
technical support........................... 392  
telco authorities and ISDN............ 125  
telecom safety warnings ............ 60  
telephony interface parameters,  
setting........................................ 188  
telephony interfaces  
analog........................................222  
T1/E1 ........................................148  
Transfer Sequence (analog)...........220  
Transfer Sequence (T1/E1) ...........146  
Transmit Data Overflow (E1 stats)  
field ...........................................345  
Transmit Data Overflow (T1 stats)  
field ...........................................341  
Transmit Data Underrun (E1 stats)  
field ...........................................346  
Transmit Data Underrun (T1 stats)  
field ...........................................342  
uses of................................... 71, 73  
telephony interfaces, analog............ 33  
telephony signaling cadences  
analog........................................ 203  
T1/E1 ........................................ 129  
telephony signaling tones  
analog........................................ 203  
T1/E1 ........................................ 129  
telephony startup configuration ...... 36  
telephony toning schemes  
analog........................................ 207  
T1/E1 ........................................ 133  
temperature  
443  
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Index  
MultiVOIP User Guide  
Transmit Line Open (E1 stats) field  
.................................................. 346  
Transmit Line Open (T1 stats) field  
.................................................. 342  
Transmit Line Short (E1 stats) field  
.................................................. 345  
Transmit Line Short (T1 stats) field  
.................................................. 341  
Transmit Slip (E1 stats) field........ 346  
Transmit Slip (T1 stats) field........ 342  
Transmit Slip Negative (E1 stats) field  
.................................................. 346  
Transmit Slip Negative (T1 stats) field  
.................................................. 342  
Transmit Slip Positive (E1 stats) field  
.................................................. 345  
Transmit Slip Positive (T1 stats) field  
.................................................. 341  
Transmitted (RTCP Packets, IP Stats)  
field........................................... 338  
Transmitted (RTP Packets, IP Stats)  
field........................................... 338  
Transmitted (TCP Packets, IP Stats)  
field........................................... 337  
Transmitted (Total Packets, IP Stats)  
field........................................... 336  
Transmitted (UDP Packets, IP Stats)  
field........................................... 337  
Transport Protocol (SIP) field  
E1..............................................287  
T1..............................................243  
UDP/TCP compared  
E1..............................................292  
IP Statistics context...................336  
T1..............................................248  
unconditional forwarding  
E1..............................................297  
T1..............................................253  
Uninstall (program menu option) ,  
command...................................368  
Uninstall option description  
(MultiVOIP program menu) .....348  
uninstalling MultiVOIP software82,  
368  
unobtainable tone, custom  
analog................................207, 208  
T1/E1 ................................133, 134  
unobtainable tones  
analog................................133, 207  
unpacking  
MVP210......................................64  
MVP2410..............................61, 62  
MVP3010....................................61  
MVP410......................................63  
MVP810......................................63  
Up Time  
System Info (analog).................232  
System Info (T1/E1)..........158, 323  
update interval (logging)  
E1.............................................. 292  
T1.............................................. 248  
trap manager parameters (SNMP)  
T1/E1 ........................................ 128  
triggering log report email  
analog........................................217  
T1/E1 ........................................143  
updated firmware, obtaining .........349  
upgrade  
E1................................................14  
T1................................................10  
upgrade card (analog, 4-to-8 channel)  
installation.................................408  
upgrade card (T1/E1) installation..398  
Upgrade Software option description  
MultiVOIP program menu ........348  
upgrade, firmware.........................353  
uploads vs. downloads (FTP)........371  
Use Fast Start (Q.931) field  
analog ...................................... 212  
T1/ E1....................................... 138  
troubleshooting ............................... 58  
Turn Off Logs field  
analog........................................ 218  
T1/E1 ........................................ 143  
Type (E&M type) field ................. 199  
Type (of tone) field  
analog........................................ 205  
T1/E1 ........................................ 131  
E1..............................................285  
T1..............................................241  
Use Gatekeeper (Outbound  
U
UDP multiplexed (H.323 Annex E)  
field  
Phonebook) field  
444  
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MultiVOIP User Guide  
Index  
E1.............................................. 291  
T1.............................................. 247  
Use Proxy (SIP) field  
T1/E1 ........................................111  
Voice/FAX connector pinout ........403  
Voice/FAX Parameter definitions  
analog................................186, 187  
T1/E1 ................................111, 112  
Voice/FAX Parameter Definitions  
analog........ 181, 182, 183, 184, 185  
T1/E1 ........ 106, 107, 108, 109, 110  
Voice/FAX Parameter fields (analog)  
Auto Call Enable.......................184  
Automatic Disconnection..........187  
Call Duration.............................187  
Consecutive Packets Lost..........187  
Copy Channel............................181  
Default.......................................181  
DTMF Gain...............................181  
DTMF Gain (High Tones) ........181  
DTMF Gain (Low Tones).........181  
DTMF In/Out of Band ..............181  
Duration (DTMF)......................181  
Dynamic Jitter Buffer................185  
Echo Cancellation .....................184  
Fax Enable.................................182  
Fax Volume...............................182  
Forward Error Correction..........184  
Input Gain .................................181  
Jitter Value................................187  
Jitter Value (Fax) ......................182  
Max Baud Rate (Fax)................182  
Maximum Jitter Value...............186  
Minimum Jitter Value ...............185  
Mode (Fax)................................182  
Network Disconnection.............187  
Optimization Factor ..................186  
Output Gain...............................181  
Phone Number (Auto Call) .......184  
Select Channel...........................181  
Silence Compression.................184  
Voice Gain ................................181  
Voice/FAX Parameter fields (T1/E1)  
Auto Call Enable.......................109  
Automatic Disconnection..........112  
Call Duration.............................112  
Consecutive Packets Lost..........112  
Copy Channel............................106  
Default.......................................106  
DTMF Gain...............................106  
DTMF Gain (High Tones) ........106  
E1.............................................. 292  
T1.............................................. 248  
user default configuration, creating  
analog........................................ 234  
T1/E1 ........................................ 160  
user defaults, downloading ........... 362  
user defaults, setting ..................... 362  
user name  
Windows GUI........................... 364  
user values (software), saving....... 362  
V
variations in PBX characteristics  
E1.............................................. 317  
T1.............................................. 273  
version numbers................................ 9  
version numbers (software) ............ 77  
version, firmware.......................... 353  
Voice Coder (call progress) field.. 327  
Voice coder (statistics, logs) field. 333  
voice delay  
analog................................ 185, 186  
T1/E1 ................................ 110, 111  
Voice Gain field  
analog........................................ 181  
T1/E1 ........................................ 106  
voice packets (analog)  
recovering lost/corrupted .......... 184  
voice packets (T1/E1)  
recovering lost/corrupted .......... 109  
voice packets, consecutive lost  
analog........................................ 187  
T1/E1 ........................................ 112  
voice packets, delayed  
analog................................ 185, 186  
T1/E1 ................................ 110, 111  
voice packets, re-assembling  
analog........................................ 182  
voice packets, re-assembly  
T1/E1 ........................................ 107  
voice quality, improving  
analog........................................ 184  
T1/E1 ........................................ 109  
voice quality, versus delay  
analog........................................ 186  
445  
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Index  
MultiVOIP User Guide  
DTMF Gain (Low Tones)......... 106  
voip software (T1/E1)  
DTMF In/Out of Band .............. 106  
Duration (DTMF) ..................... 106  
Dynamic Jitter Buffer ............... 110  
Echo Cancellation..................... 109  
Fax Enable ................................ 107  
Fax Volume .............................. 107  
Forward Error Correction ......... 109  
Input Gain................................. 106  
Jitter Value................................ 112  
Jitter Value (Fax) ...................... 107  
Max Baud Rate ......................... 107  
Maximum Jitter Value .............. 111  
Minimum Jitter Value............... 110  
Mode (Fax) ............................... 107  
Network Disconnection ............ 112  
Optimization Factor.................. 111  
Output Gain .............................. 106  
Phone Number (Auto Call)....... 109  
Select Channel .......................... 106  
Silence Compression................. 109  
Voice Gain................................ 106  
Voice/FAX Parameters screen,  
accessing  
host PC........................................88  
voip system example, conceptual (E1)  
calls to remote PSTN ................277  
foreign calls, national rates........279  
voip site to voip site ..................276  
voip system example, digital &  
analog, with phonebook details  
E1..............................................305  
T1..............................................260  
voip system example, digital only,  
with phonebook details  
E1..............................................298  
T1..............................................254  
voip(E1)  
basic functions of ........................15  
voip(T1)  
basic functions of ........................11  
voltage, operating  
analog models .............................25  
E1 models....................................24  
T1 models....................................23  
W
analog........................................ 178  
T1/E1 ........................................ 103  
Voice/FAX parameters, setting  
analog........................................ 178  
T1/E1 ........................................ 103  
voip dialing digits  
warnings, safety ..............................60  
warranty ........................................390  
web browser GUI and logs  
analog........................................217  
T1/E1 ........................................143  
web browser GUI, enabling  
non-PSTN type ........................... 47  
types used.................................... 46  
voip email account  
analog..................................36, 177  
T1/E1 ........................................102  
web browser interface  
analog........................................ 211  
T1/E1 ........................................ 137  
voip management, remote  
analog........................................ 200  
T1/E1 ........................................ 126  
voip network, lab/simulated  
browser version requirement381,  
384  
general.......................................381  
Java requirement .......................381  
prerequisite local assigning of IP  
address...................................382  
video useability .........................381  
web GUI  
use in startup............................... 48  
voip protocols  
H.323 .......................................... 31  
SIP .............................................. 31  
uniformity in network ................. 31  
voip software  
host PC.................................. 22, 32  
voip software (analog)  
Java and.....................................383  
remote control/configuration and  
..............................................383  
web GUI vs. local Windows GUI  
comparison................................382  
web GUI, logging and...................382  
weight  
host PC...................................... 163  
446  
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MultiVOIP User Guide  
Index  
analog models ............................. 25  
E1 models ................................... 24  
T1 models ................................... 23  
weight loading  
in rack ......................................... 66  
weight of unit  
lifting precaution......................... 60  
personnel requirement................. 60  
Well Known Ports......................... 406  
well-known port number, SMTP  
analog ...................................... 211  
T1/ E1....................................... 137  
well-known port, gatekeeper  
registration  
E1.............................................. 285  
T1.............................................. 241  
well-known port, Q.931 params,  
H.323  
T1..............................................248  
well-known port, SNMP  
analog........................................202  
T1/E1 ........................................128  
wink signaling (E&M) ..................199  
Wink Timer (E&M) field..............199  
worksheet  
phonebook.............................52, 53  
X
XMT (channel) LED  
analog models...........................21  
XMT (Ethernet) LED  
analog models .............................21  
XSG LED  
analog models...........................21  
Y
E1...................................... 285, 291  
T1...................................... 241, 247  
well-known port, SIP  
Yellow Alarm (E1 stats) field .......345  
Yellow Alarm (T1 stats) field .......342  
Yellow Alarm Format field (T1)...119  
E1.............................................. 292  
447  
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S000249C  
449  
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