MultiVOIP®
Voice/Fax over IP Gateways
MVP210/410/810
MVP210/410/810-SS
MVP210/410/810-FX
User Guide
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CONTENTS
Chapter 1 – Description and Specifications...................................................................................................... 6
Introduction............................................................................................................................................................6
Feature Comparison Matrix ..............................................................................................................................................6
Interface............................................................................................................................................................................7
Front Panel LEDs .............................................................................................................................................................7
Computer Requirements .......................................................................................................................................7
Specifications ........................................................................................................................................................8
Chapter 2 – Installing and Cabling the MultiVOIP............................................................................................. 9
Introduction............................................................................................................................................................9
Safety Warnings ....................................................................................................................................................9
Unpacking Your MultiVOIP....................................................................................................................................9
Rack Mounting Instructions for MVP410 & MVP810...........................................................................................10
Cabling Procedure for MVP210...........................................................................................................................11
Cabling Procedure for MVP-410/810 ..................................................................................................................13
Chapter 3 – Software Installation...................................................................................................................... 16
Introduction..........................................................................................................................................................16
Loading MultiVOIP Software onto the PC...........................................................................................................16
Setup Overview ...................................................................................................................................................19
Ethernet/IP......................................................................................................................................................................20
Voice/Fax........................................................................................................................................................................21
Interface..........................................................................................................................................................................23
Call Signaling..................................................................................................................................................................25
Regional..........................................................................................................................................................................27
Phone Book ....................................................................................................................................................................28
Save & Reboot................................................................................................................................................................29
Chapter 4 – Configuring Your MultiVOIP......................................................................................................... 30
Introduction..........................................................................................................................................................30
Software Categories Covered in This Chapter....................................................................................................30
How to Navigate Through the Software ..............................................................................................................31
Web Browser Interface........................................................................................................................................31
Configuration Information Checklist ....................................................................................................................31
Ethernet/IP......................................................................................................................................................................32
Voice/Fax........................................................................................................................................................................35
Configurable Payload Type.......................................................................................................................................39
Interface..........................................................................................................................................................................40
FXS Loop Start Parameters......................................................................................................................................41
Message Waiting.......................................................................................................................................................43
FXO Parameters .......................................................................................................................................................44
E&M Parameters.......................................................................................................................................................49
DID Parameters ........................................................................................................................................................52
Call Signaling..................................................................................................................................................................53
H.323 ........................................................................................................................................................................53
SIP ............................................................................................................................................................................55
SPP...........................................................................................................................................................................59
SNMP .............................................................................................................................................................................61
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Regional..........................................................................................................................................................................62
SMTP..............................................................................................................................................................................65
RADIUS ..........................................................................................................................................................................68
Logs/Traces....................................................................................................................................................................70
NAT Traversal.................................................................................................................................................................71
Supplementary Services.................................................................................................................................................72
Save Settings..................................................................................................................................................................75
Save & Reboot..........................................................................................................................................................75
Connection......................................................................................................................................................................75
Settings.....................................................................................................................................................................75
Troubleshooting Software Issues..............................................................................................................................76
Chapter 5 – Phone Book Configuration........................................................................................................... 77
Introduction..........................................................................................................................................................77
Identify Remote VOIP Site to Call .......................................................................................................................77
Identify VOIP Protocol to be Used.......................................................................................................................77
Phonebook Starter Configuration........................................................................................................................78
Outbound Phonebook.....................................................................................................................................................78
Inbound Phonebook........................................................................................................................................................80
Phone Book Descriptions....................................................................................................................................81
Outbound Phone Book/List Entries.................................................................................................................................81
Add/Edit Outbound Phone Book ...............................................................................................................................82
Inbound Phone Book/List Entries....................................................................................................................................86
Add/Edit Inbound Phone Book..................................................................................................................................87
Phone Book Save and Reboot........................................................................................................................................89
Phonebook Examples .........................................................................................................................................90
North America.................................................................................................................................................................90
Europe ............................................................................................................................................................................93
Variations of Caller ID .........................................................................................................................................99
Chapter 6 – Using the Software ...................................................................................................................... 102
Introduction........................................................................................................................................................102
Software Categories Covered in This Chapter..................................................................................................102
Statistics Section ...............................................................................................................................................104
Call Progress ................................................................................................................................................................104
Logs..............................................................................................................................................................................106
IP Statistics...................................................................................................................................................................108
Link Management .........................................................................................................................................................110
Registered Gateway Details .........................................................................................................................................111
Servers .........................................................................................................................................................................112
H.323 GateKeepers ................................................................................................................................................112
SIP Proxies .............................................................................................................................................................113
SPP Registrars........................................................................................................................................................114
Advanced......................................................................................................................................................................115
Packetization Time..................................................................................................................................................115
MultiVOIP Program Menu Items........................................................................................................................116
Updating Firmware .......................................................................................................................................................117
Implementing a Software Upgrade ...............................................................................................................................118
Identifying Current Firmware Version......................................................................................................................118
Downloading Firmware ...........................................................................................................................................119
Downloading Factory Defaults ................................................................................................................................120
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Downloading IFM Firmware..........................................................................................................................................121
Setting and Downloading User Defaults .......................................................................................................................123
Setting a Password.......................................................................................................................................................124
Windows Interface...................................................................................................................................................124
Web Browser Interface............................................................................................................................................125
Upgrading Software......................................................................................................................................................126
FTP Server File Transfers (“Downloads”) .........................................................................................................127
Web Browser Interface......................................................................................................................................132
SysLog Server Functions ..................................................................................................................................134
Appendix A – Cable Pin-outs .......................................................................................................................... 135
Appendix B – TCP/UDP Port Assignments.................................................................................................... 136
Appendix C – Installation Instructions for MVP428 Upgrade Card............................................................. 137
Appendix D – Regulatory Information............................................................................................................ 140
Appendix E – Waste Electrical and Electronic Equipment (WEEE) Statement.......................................... 142
Appendix F – C-ROHS HT/TS Substance Concentration ............................................................................. 143
INDEX................................................................................................................................................................. 144
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Chapter 1 – Description and
Specifications
Introduction
The MultiVOIP gateways, MVP210, MVP410, and MVP810 provide toll-free voice and fax communications over
the Internet or an Intranet. By integrating voice and fax into your existing data network, you can realize substantial
savings on inter-office long distance toll charges. MultiVOIP gateways connect directly to phones, fax machines,
key systems, PSTN lines, or a PBX to provide real-time, toll-quality voice connections to any office on your VOIP
network. The –SS series models only support the SIP protocol through the FXS/FXO interface with SIP
survivability as well.
Figure 1-1: MVP-410/810 Chassis
Figure 1-2: MVP-210 Chassis
The MultiVOIP model MVP210 is a two-channel unit, the model MVP410 is a four-channel, and the MVP810 is an
eight-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 MVP810.
These MultiVOIPs inter-operate with a telephone switch or PBX, acting as a switching device that directs voice
and fax calls over an IP network. The MultiVOIPs have “phonebooks,” directories that determine to who calls may
be made and the sequences that must be used to complete calls through the MultiVOIP. The phonebooks allow
the phone user to interact with the VOIP system just as they would with an ordinary PBX or telco switch. When
the phonebooks are set, special dialing sequences are minimized or eliminated altogether. Once the call
destination is determined, the phonebook settings determine whether the destination VOIP unit must strip off or
add dialing digits to make the call appear at its destination to be a local call.
Feature Comparison Matrix
The main differences between the model versions are the line type capabilities and interface options, as detailed
in the chart below:
MultiVOIP®
MultiVOIP® -SS
MultiVOIP® -FX
H.323
SPP
SIP
SIP Survivability
DID
E&M
●
●
●
●
●
●
●
●
●
●
FXS/FXO
●
●
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Chapter 2: Quick Start
Interface
While the web interface appears differs slightly, its content and organization are essentially the same as that of
the Windows interface (except for logging). These will be addressed in the following chapters.
Front Panel LEDs
Active LEDs
On both the MVP410 and MVP810 models, there are eight sets of channel-operation LEDs. However, on the
MVP410, only the lower four sets of channel-operation LEDs are functional. On the MVP810, all eight sets are
functional.
Figure 1-3. MVP410/810 LEDs
Similarly, the MVP210 models have the general-operation indicator LEDs and two sets of channel-operation
LEDs.
Figure 1-4. MVP210 LEDs
Front Panel LED Definitions
LED
Description
General Operation LEDs (one set on each MultiVOIP model)
Power
Boot
Indicates presence of power
After power up, the Boot LED will be on briefly while the MultiVOIP is booting. It lights whenever the
MultiVOIP is booting or downloading a setup configuration data set
FDX. LED indicates whether Ethernet connection is half-duplex or full-duplex (FDX) and, in half-
duplex mode, indicates occurrence of data collisions. LED is on constantly for full-duplex mode; LED
is off constantly for half-duplex mode. When operating in half-duplex mode, the LED will flash during
data collisions.
Ethernet
LNK. Link/Activity LED. This LED is lit if Ethernet connection has been made. It is off when the link
is down (i.e., when no Ethernet connection exists). While link is up, this LED will flash off to indicate
data activity.
Channel-Operation LEDs (one set for each channel)
Transmit. This indicator blinks when voice packets are being transmitted to the local area network.
XMT
RCV
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.
XSG
RSG
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.
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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Chapter 2: Quick Start
Specifications
MVP210 models
MVP410 models
MVP810 or MVP410 + 428
External transformer:
3A @5V
100-240 VAC
1.2 - 0.6 A
100-240 VAC
1.2 - 0.6 A
Operating Voltage/Current
Mains Frequencies
Power Consumption
50/60 Hz
19 watts
50/60 Hz
29 watts
50/60 Hz
46 watts
1.4” H
1.75” H x
1.75” H x
6.2” W x
9” D x
17.4” W x
8.5” D
17.4” W x
8.5” D
Mechanical Dimensions
----------------
3.6cm H
15.8cm W x
22.9cm D x
-----------------
4.5cm H x
44.2 cm W x
21.6 cm D
-----------------
4.5cm H x
44.2 cm W x
21.6 cm D
1.8lbs (.82kg)
2.6lbs (1.17kg)
with transformer
7.1 lbs
(3.2 kg)
7.7 lbs.
(3.5 kg)
Weight
Maximum: 40 degrees Celsius (104 degrees Fahrenheit) @ 20-90% non-
Ambient temperature range condensing relative humidity.
Minimum: 0 degrees Celsius (32 degrees Fahrenheit).
Warranty
2 years
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Chapter 2 – Installing and Cabling the
MultiVOIP
Introduction
The MVP210 MultiVOIP models are tabletop units that can be handled easily by one person. However, the
MVP410 and MVP810 MultiVOIPs 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 replacement.
Warning: There is danger of explosion if the battery is incorrectly replaced.
Safety Warnings Telecom
1. Never install telephone wiring during a lightning storm.
2. Never install a telephone jack in wet locations unless the jack is specifically designed for wet locations.
3. This product is to be used with UL and UL listed computers.
4. Never touch un-insulated telephone wires or terminals unless the telephone line has been disconnected at the
network interface.
5. Use caution when installing or modifying telephone lines.
6. Avoid using a telephone (other than a cordless type) during an electrical storm. There may be a remote risk of
electrical shock from lightning.
7. Do not use a telephone in the vicinity of a gas leak.
8. To reduce the risk of fire, use only a UL-listed 26 AWG or larger telecommunication line cord.
Unpacking Your MultiVOIP
When unpacking your MultiVOIP, check to see that all of the items are included in the box. For the various
MultiVOIP models, the contents of the box will be different. If any box contents are missing, contact Multi-Tech
Tech Support at 1-800-972-2439.
MVP210 models content list:
•
•
•
•
•
•
MVP210
DB9 to RJ45 cable
Power transformer
Power cord
Printed Cabling Guide
Product CD
MVP410/810 models content list:
•
•
•
•
•
•
MVP410 or MVP810
DB9 to DB25 cable
Mounting brackets and screws
Power cord
Printed Cabling Guide
Product CD
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Chapter 2: Installing and Cabling the MultiVOIP
Rack Mounting Instructions for MVP410 & MVP810
The MultiVOIPs can be mounted in an industry-standard EIA 19-inch rack enclosure.
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.
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).
This equipment should only be installed by properly qualified service personnel. Only connect like circuits -
connect SELV (Secondary Extra Low Voltage) circuits to SELV circuits and TN (Telecommunications Network)
circuits to TN circuits.
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 2-1, and then secure unit to rack rails by the brackets, as shown in Figure 2-2. Because equipment
racks vary, screws for rack-rail mounting are not provided. Follow the instructions of the rack manufacturer and
use screws that fit.
1. Position the right rack-mounting bracket on the MultiVOIP using the two vertical mounting screw holes.
2. Secure the bracket to the MultiVOIP using the two screws provided.
3. Position the left rack-mounting bracket on the MultiVOIP using the two vertical mounting screw holes.
4. Secure the bracket to the MultiVOIP using the two screws provided.
5. Remove feet (4) from the MultiVOIP unit.
6. Mount the MultiVOIP in the rack enclosure per the rack manufacture’s mounting procedure.
Figure 2-1: Bracket Attachment for Rack Mounting (MVP410 & MVP810)
Figure 2-2: Attaching MultiVOIP to Rack Rail (MVP410 & MVP810)
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Chapter 2: Installing and Cabling the MultiVOIP
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 to a live AC outlet as shown in the figure below. The –SS and –FX models do not have the
E&M jacks as shown.
Figure 2-3: Cabling for MVP210
2. Connect the MultiVOIP to a PC by using a RJ-45 (male) to DB-9 (female) cable. Plug the RJ-45 end of
the cable into the Command port of the MultiVOIP and the other end into the PC serial port.
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.
a. For an FXS or FXO connection (-SS and -FX series).
(FXS Examples: analog phone, fax machine |
FXO Examples: PBX extension, POTS line from telco central office)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the back of
the MultiVOIP. Connect the other end to the device or phone jack.
b. For an E&M connection.
(E&M Example: trunk line from telephone switch)
Connect one end of an RJ-45 phone cord to the Channel 1 E&M connector on the back of the
MultiVOIP. Connect the other end to the trunk line.
Verify that the E&M Type in the E&M Options group of the Interface dialog box is the same as the
E&M trunk type supported by the telephone switch. See Appendix B for an E&M cabling pin-out.
c. For a DID connection.
(DID Example: DID fax system or DID voice phone lines)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the back of
the MultiVOIP. Connect the other end to the DID jack.
NOTE: DID lines are polarity sensitive. If, during testing, the DID line rings busy consistently, you will need
to reverse the polarity of one end of the connector (swap the wires to the two middle pins of one RJ-11
connector).
4. Repeat the above step to connect the remaining telephone equipment to the second channel on your
MultiVOIP.
5. Ensure that the unit is properly connected to earth ground by verifying that it is reliably grounded when
mounted within a rack. This can be accomplished by connecting a grounding wire between the chassis
and a metallic object that will provide an electrical ground.
6. 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.
7. Proceed to the Software Installation chapter to load the MultiVOIP software.
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Chapter 2: Installing and Cabling the MultiVOIP
For DID channels only
For any channel on which you are using the DID interface type, you must change the jumper on the MultiVOIP
circuit card. DID is not supported on the –SS or –FX models.
1. Disconnect power. Unplug the AC power cord from the wall outlet or from the receptacle on the
MultiVOIP unit.
2. Using a #1 Phillips driver, remove the screw (at bottom of unit, near the back-cover end) that attaches the
main circuit card to the chassis of the MVP210.
3. Pull the main circuit card out about half way.
4. Identify the channels on which the DID interface will be used.
L
E
D
1
4
L
E
D1
2
L
E
D
7
L
ED
1
3
L
E
D11
L
E
D
10
LE D 9
LE
D
8
L
ED6
LE
D
5
LE
D
4
LE D 3
L
E
D
2
L
E
D1
R 113
R7
2
R
74
R114
R58
R57
R56
R5
5
R2
05
R2
MVP210 Circuit Board
Ch1
Ch2
as configured
for DID Interface
JP4
Ch 1 Jumper
Block
P7
JP7
as shipped,
for non-DID interfaces
JP8
Ch 2 Jumper
Block
JP1
F
B
3
J3
J
7
J5
J9
J
11
J1
S
1
0
J
15
as configured
for DID Interface
Figure 2-4: MVP210 Channel Jumper Settings
5. Position the jumper for each DID channel so that it does not connect the two jumper posts. For DID
operation of a VOIP channel, the MultiVOIP will work properly if you simply remove the jumper altogether,
but that is inadvisable because the jumper might be needed later if a different telephony interface is used
for that VOIP channel.
6. Slide the main circuit card back into the MultiVOIP chassis and replace the screw at the bottom of the
unit.
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Chapter 2: Installing and Cabling the MultiVOIP
Cabling Procedure for MVP-410/810
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 the figure below. The E&M jacks are not present on the
–SS and –FX models.
Command Modem connector
for remote configuration
ETH ERN ET
COMMAND
E&M FXS/FXO
E& M FXS/FXO
E&M FXS/FXO
E&M FXSF/ XO
E&M FXS/FXO
E &M FXS /FXO
E&M FXS/FXO
E&M FX S/FXO
COMMAND
MODEM
10 BASET
Voice/Fax Channel Connections
Channels 1-4 Bottom MVP410/810
Channels 5-8 Top MVP810 Only
E&M FXS/FXO
Ethernet Connection
FXS
E&M
FXO
CommandPort Connection
PSTN
Figure 2-5: Cabling for MVP-410/810
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
2-5.
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.
a. For an FXS or FXO connection (-SS and -FX series).
(FXS Examples: analog phone, fax machine |
FXO Examples: PBX extension, POTS line from central office.)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the back of the
MultiVOIP. Connect the other end to the device or phone jack.
b. For an E&M connection.
(E&M Example: trunk line from telephone switch.)
Connect one end of an RJ-45 phone cord to the Channel 1 E&M connector on the back of the
MultiVOIP. Connect the other end to the trunk line.
Verify that the E&M Type in the E&M Options group of the Interface dialog box is the same as the
E&M trunk type supported by the telephone switch. See Appendix B for an E&M cabling pin-out.
c. For a DID connection.
(DID Examples: DID fax system or DID voice phone lines.)
Connect one end of an RJ-11 phone cord to the Channel 1 FXS/FXO connector on the back of the
MultiVOIP. Connect the other end to the DID jack.
NOTE: DID lines are polarity sensitive. If, during testing, the DID line rings busy consistently, you will need to
reverse the polarity of one end of the connector (swap the connections of the wires to the two middle pins of one
RJ-11 connector).
4. Repeat step 3 to connect the remaining telephone equipment to each channel on your MultiVOIP.
Although a MultiVOIP’s channels are often all configured identically, each channel is individually
configurable. So, for example, some channels of a MultiVOIP might use the FXO interface and others the
FXS; some might use the DID interface and others E&M, etc.
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Chapter 2: Installing and Cabling the MultiVOIP
5. If you intend to configure the MultiVOIP remotely using the MultiVOIP Windows interface, connect an
RJ-11 phone cable between the Command Modem connector (not available on the –SS or –FX series)
and a receptacle served by a telco POTS line. See Figure 2-6 below.
6. The Command Modem is built into the MVP410 and 810 units only. To configure the MultiVOIP remotely
using its Windows interface, you must call into the MultiVOIP’s Command Modem. Once a connection is
made, the configuration process is identical to local configuration with the Windows interface.
Command Modem connector
for remote configuration
ETHERNET
COMMAND
MODEM
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
COMMAND
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
E&M FXS/FXO
10 BASET
MVP-410/810
Rear Panel
Grounding Screw
Telco POTS Line
Figure 2-6: MVP410/810 connections for ground & modem
7. Ensure that the unit is properly connected to earth ground by verifying that it is reliably grounded when
mounted within a rack.
8. This can be accomplished by connecting a grounding wire between the chassis grounding screw (see
Figure 2-6) and a metallic object that will provide an electrical ground.
9. Turn on power to the MultiVOIP by placing the ON/OFF switch on the back panel to the ON position. Wait
for the Boot LED on the MultiVOIP to go off before proceeding. This may take a few minutes.
10. Proceed to Chapter 3 to load the MultiVOIP software.
For DID channels only
For any channel on which you are using the DID interface type, you must change the jumper on the MultiVOIP
circuit card. DID is not supported on the –SS or –FX models.
1. Disconnect power. Unplug the AC power cord from the wall outlet or from the receptacle on the
MultiVOIP unit.
2. Using a #1 Phillips driver, remove the three screws (at back of unit) that attach the main circuit card to the
chassis of the MultiVOIP.
Figure 2-7: MVP-410/810 Rear Screw Locations
3. Pull the main circuit card out about 5 inches (the power connection to the board prevents it from being
removed entirely from the chassis).
4. Identify the channels on which the DID interface will be used.
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Chapter 2: Installing and Cabling the MultiVOIP
Figure 2-8. MVP-410/810 Channel Jumper Settings
5. Position the jumper for each DID channel so that it does not connect the two jumper posts. For DID
operation of a VOIP channel, the MultiVOIP will work properly if you simply remove the jumper altogether,
but that is inadvisable because the jumper might be needed later if a different telephony interface is used
for that VOIP channel.
6. Slide the main circuit card back into the MultiVOIP chassis and replace the three screws.
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Chapter 3 – Software Installation
Introduction
Configuring software for your MultiVOIP entails three tasks:
Loading the software onto the PC (this is “Software Installation” and is discussed in this chapter).
Setting values for telephony and IP parameters that will fit your system (details are in Chapter 4).
Establishing “phonebooks” that contain the various dialing patterns for VOIP calls made to different locations (a
detailed discussion of this is found in Chapter 5).
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.
1. Be sure that your MultiVOIP has been properly cabled and that the power is turned on.
2. Insert the MultiVOIP CD into your CD-ROM drive. The CD starts automatically. It may take a few moments
for the Multi-Tech CD installation window to display.
Figure 3-1: Analog MVP splash screen
3. When the Multi-Tech Installation CD dialog box appears, click the Install Software icon.
4. A secondary screen appears. Click on the button that matches the model you have purchased. The
installation wizard will start.
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Chapter 3: Software Installation
Figure 3-2: Welcome screen
Press Enter or click Next to continue.
5. Follow the on-screen instructions to install your MultiVOIP software. The first screen asks you to choose the
destination for the MultiVOIP software.
Figure 3-3: Destination
Choose a location and click Next.
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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Chapter 3: Software Installation
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 keyboard shortcut
Ctrl + G).
Note: If the COM port setting made here conflicts with the actual COM port resources available in the
command PC, the “Error in Opencomm handle” message will appear when the MultiVOIP program is
launched. If this occurs, you must reset the COM port.
8. A completion screen will appear.
Figure 3-4: Completion
Click Finish.
9. When setup of the MultiVOIP software is complete, you will be prompted to run the MultiVOIP software to
configure the VOIP.
Figure 3-5: Configuration
Software installation is now complete.
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Chapter 3: Software Installation
Setup Overview
With the software now installed, you are ready to get your MultiVOIP set up and working. There are a few
necessary settings that need to be entered in the configuration software to achieve this and they are noted in the
action lists for the categories below. The following chapters will cover all aspects in detail, but here we will cover
the basic configuration needed to start VOIP communications. Below you will find the list of categories requiring
information to be set before VOIP communication will be ready.
⇒ Ethernet/IP
⇒ Voice/Fax
⇒ Interface
⇒ Call Signaling
⇒ Regional
⇒ Phone Book
This setup process is followed by the Save & Reboot step which is very important.
Figure 3-6: Main Screen
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Chapter 3: Software Installation
Ethernet/IP
A unique LAN IP address is required for the MultiVOIP unit as well as a subnet mask and Gateway IP for minimal
functionality. Other settings in this category pertain to specific features and protocols that can be used, but are not
necessary for basic operation. Details for all settings are provided in chapter 4.
Figure 3-7: IP settings
Actions:
•
Select Packet Prioritization if used
Set 802.1p Priority Parameters as needed
o
ƒ
ƒ
The Priority levels can be from 0 – 7, where 0 is lowest priority (details in Chapter 4)
VLAN ID identifies a virtual LAN by a number (1 to 4094)
•
•
Set the Frame Type to match the network that the MultiVOIP is attached to
TYPE II or SNAP
Enter Gateway Name
Check to enable DHCP if used
o
o
•
•
•
•
Enter IP Address for the MultiVOIP unit
Enter Subnet IP Mask for the MultiVOIP unit
Enter Gateway IP
Enable DNS if desired
o
Enter DNS Server IP Address
•
•
Enable SRV support if needed
Diff Serv Parameters are for routers that are Diff Serv compatible
o
Setting both values to 0 effectively disables Diff Serv
•
•
FTP Server Enable is only needed for firmware and software updates to the MultiVOIP
TDM Routing can be used if necessary
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Chapter 3: Software Installation
Voice/Fax
The individual channels must be set up before use. The Copy Channel button can save a lot of time during this
step if channels are to be set with the same parameters. Some options should be noted for future changes if
necessary, but the defaults are likely to work without adjustment.
Figure 3-8: Voice & Fax settings
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Chapter 3: Software Installation
Actions:
• Select Channel
o Choose channel parameters:
ƒ
Set the Fax parameters to meet your needs
•
•
•
•
Set Max Baud Rate to match fax machine (2400 to 14400 bps)
Fax Volume should not be changed as it may impair function
Jitter Value affects the time for packet reassembly
Mode: Select T.38 or FRF 11
ƒ
ƒ
ƒ
ƒ
Modem Relay Enable allows modem traffic through the VOIP system
Adjusting Voice Gain and DTMF should not be done as it may adversely affect quality
Select a Coder or allow Automatic negotiation
Advanced Features
•
•
•
Silence Compression, when enabled, will not send silence packets
Echo Cancellation removes echo to improve voice quality
Forward Error Correction allows some bad packets to be recovered
ƒ
Choose Auto Call / OffHook Alert settings
•
For automatically calling a remote VOIP without dialing (details in Chapter 4)
Change Dynamic Jitter values if necessary (details in Chapter 4)
ƒ
ƒ
ƒ
Select any Automatic Disconnection options needed to ensure lines are not left “open”
Configurable Payload Types are best left at their defaults. Not in the –SS models
o
The Copy Channel button is available for easily transferring these settings to the other channels
•
Repeat for all channels to be used
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Chapter 3: Software Installation
Interface
The Interface Parameters are the telephony settings that are to be applied to the individual MultiVOIP channels.
Configure each channel for the type of interface you are using. Channel 1 is selected by default.
Note: Feature options are enabled or unavailable depending on the selected interface type. The one option
available for all interface types is the inter digit timer option. This option defines the maximum amount of time that
the unit will wait before mapping the dialed digits to an entry in the phone book database. If too much time
elapses between digits, and the wrong numbers are mapped, you will hear rapid busy signal. If this happens,
hang up and dial again.
If the Interface Type is FXS (Loop Start), a station device such as an analog telephone, fax machine or KTS (Key
Telephone System) is connected to an analog channel. The FXS options group is active.
If the Interface Type is FXO, the Dialing Options Regeneration, Flash Hook Timer and Ring Count groups are
enabled. The FXO Ring Count allows you to set the number of rings before the unit answers the incoming call.
Check with your local in-house phone personnel to verify whether your local PBX dial signaling is pulse or tone
(DTMF). The Flash Hook Options Generation setting allows you to enter the time, in milliseconds, for the duration
of the flash hook signal.
If the Interface Type is E & M, you are connecting to an analog E & M trunk on your PBX. Check with your in-
house phone personnel to determine the signaling type (Dial Tone or Wink) and if it is 2-wire or 4-wire. The –SS
and –FX series do not support E&M or DID operation.
Figure 3-9: Interface Parameters
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Chapter 3: Software Installation
Actions:
• Select Channel
o Select Interface Type: FXS, FXO, E&M or DID (FXS/FXO only for –SS and –FX series)
o Regeneration
ƒ
Choose how signal is regenerated; as Pulse or DTMF
o Inter Digit Timer
ƒ
Time the MultiVOIP waits between digits
o Message Waiting Indication is for E&M only
ƒ
Choose Light or None
o Inter Digit Regeneration Timer
ƒ
Length of time between sent DTMF digits
• Flash Hook Options
o
o
Generation (used in conjunction with FXO/E&M)
Detection Range (used in conjunction with FXS/E&M)
• Caller ID
o
o
o
Bellcore is the only option available
CallerID Manipulation is available if needed
CID Manipulation is not available in the –SS models
• Pass Through (opens an audio path through the MultiVOIP)
• FXS Options
o
o
o
Set Ring Count (the number of rings allowed before call abandoned; default is 8)
Use Current Loss (MultiVOIP interrupts current to disconnect)
Generate Current Reversal (activates Answer/Disconnect Supervision to FXO)
•
FXO Options
o
o
o
Ring Count (set number of rings before MultiVOIP answers)
No Response Timer (set time to attempt call before abandoning)
Supervision Button (for call answering and disconnection settings)
ƒ
Answer Fields:
•
•
•
•
•
•
Current Reversal (use current reversal to answer)
Answer Delay
Answer Delay Timer (in seconds)
Tone Detection (allow tone sequence to disconnect)
Available Tones
Answer Tones (shows current selection from Available Tones)
ƒ
Disconnect Fields
•
•
•
•
•
•
•
•
•
•
•
Current Reversal (use current reversal to disconnect)
Current Loss (loss of current will trigger disconnect)
Current Loss Timer (time after current loss to disconnect; in milliseconds)
Silence Detection Enable (use silence detection to disconnect)
Silence Detection Type (one-way or two-way)
Silence Timer (time of silence needed to trigger disconnect; in seconds)
DTMF Tone (use tones to disconnect)
Disconnect Tone Sequence (select tone pairs to use for disconnecting)
Tone Detection (disconnect from termination of tone)
Available Tones
Disconnect Tones (shows current selection from Available Tones)
•
•
E&M Options (not supported by –SS and –FX series)
o
o
o
o
o
o
o
Type
Mode (2-wire or 4-wire)
Signal (Dial Tone or Wink)
Wink Timer (range is 100 to 350 milliseconds; default is 250)
No Response Timer (time, in seconds, after which an FXO call would be disconnected)
Disconnect on Call Progress Tone (allows disconnection when PBX issues call progress tone)
Pass Through Enable (creates an open audio patch; not for use with Wink signaling)
DID Options (not supported by –SS and –FX series)
o
o
Start Modes (Immediate, Wink or Delay Dial)
Wink Timer (in milliseconds)
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Chapter 3: Software Installation
Call Signaling
There are three choices for Call Signaling: H.323, SIP and SPP, the –SS models only support SIP and the –FX
models support SIP and SPP, but not H.323. It is best to select one of these as the protocol to be used, rather
than mixing them. Single Port Protocol (SPP) is a non-standard protocol created by Multi-Tech that allows
dynamic IP allocation. Generally, the default settings will work for most users and the individual parameters may
be changed if the need arises. Additional details for all settings are found in Chapter 4.
Figure 3-10: Signaling Protocols
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Chapter 3: Software Installation
Actions:
• Configure your chosen Call Signal type
H.323 (not supported by –SS and –FX series)
o
ƒ
ƒ
ƒ
ƒ
ƒ
Use Fast Start (may be needed for third-party vendor compatibility)
Signaling Port (default is 1720)
Register with Gatekeeper (needed if the VOIP is to be controlled by a gatekeeper)
Allow Incoming Calls Through Gatekeeper Only
Gatekeeper RAS Parameters
•
•
•
Enter parameters for Primary and any Alternate Gatekeepers
RAS TTL Value (“Time To Live” in seconds)
Gatekeeper Discovery Polling Interval (time between attempts connecting to
gatekeepers)
•
Use Online Alternate Gatekeeper List
H.323 Version 4 Options (detailed descriptions of these can be found in Chapter 4)
ƒ
o
SIP
ƒ
ƒ
ƒ
ƒ
Signaling Port (default is 5060)
Use SIP Proxy (enable to work with a proxy server)
Allow Incoming Calls Through SIP Proxy Only
SIP Proxy Parameters
•
•
•
•
•
•
Enter information for Primary and any Alternate Proxy servers
Append SIP Proxy Domain Name in User ID
Enter User Name and Password
Re-Registration Time (in seconds)
Proxy Polling Interval (time between proxy server connect attempts)
TTL Value (in seconds)
o
SPP (not supported by –SS series)
ƒ
ƒ
ƒ
ƒ
ƒ
Mode (Direct, Client or Registrar)
Signaling Port (must be unique for any VOIP unit behind same firewall)
Retransmission (time before retransmission of lost packets)
Max Retransmission (number of retransmission attempts)
Client Options
•
•
Enter information for the Primary and Alternate Registrars
Polling Interval (time between connect attempts)
Keep Alive (time out for client un-registering)
Behind Proxy/NAT device
ƒ
ƒ
•
Enter Public IP of Proxy/NAT server
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Chapter 3: Software Installation
Regional
Select the country or region that the MultiVOIP unit will operate in, or use the custom option if the available
settings are not adequate.
Figure 3-11: Regional Parameters
Actions:
•
Select the choice that matches the location of the MultiVOIP from the Country/Region field
o
If there is not a selection to fit your needs, you may select Custom and set the tones manually
User Defined tones can be created for use in conjunction with FXO Supervision with the Add
button
o
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Chapter 3: Software Installation
Phone Book
Without a populated phone book, the VOIP unit is unable to translate call traffic. You will need the information for
both a local and any remote sites that are to be used. Detailed descriptions and examples are available in chapter
5.
Figure 3-12: Phone Book screens
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Chapter 3: Software Installation
Actions:
•
Select Outbound Phone Book
o
o
o
Select Add Entry
Accept Any Number may be selected to allow unmatched destinations an alternative
Enter the number necessary to get out from the PBX system followed by the calling code of the
destination in the Destination Pattern field
o
Enter the PBX access digit (same number as needed to get out of the PBX system) in the
Remove Prefix field
o
o
o
Any digits that need to be added should be put in the Add Prefix field
Enter the IP address of the call destination (add a Description if you like)
Select a Protocol type (–SS models use SIP only, -FX models do not support H.323)
ƒ
ƒ
ƒ
For H.323:
•
For SIP:
•
Enter Gateway settings
Select Transport Protocol, Proxy and URL if needed
For SPP:
Enter Registrar settings if needed
The Advanced Button will allow an Alternate IP Address to be entered for outbound traffic
•
o
•
•
Select Inbound Phone Book
o
o
o
o
o
o
o
o
Select Add Entry
Accept Any Number for inbound traffic does not work when external routing devices are used
Enter any access digits followed by the local calling code in the Remove Prefix field
Enter any digits needed to access an outside line in the Add Prefix field
Select Hunting in the Channel Number field to have the VOIP use the next available channel
Add a description if you like
Call Forward may be set up (details available in Chapter 5)
Select Registration Option
Repeat the Phone Book steps for any additional entries needed
Save & Reboot
Any time that you change settings on the VOIP unit, you must choose the Save & Reboot option; otherwise all
changes made will be lost when the MultiVOIP is reset or shutdown.
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Chapter 4 – Configuring Your MultiVOIP
Introduction
There are two methods of using your MultiVOIP; one is through a web interface, and the other is through the
Windows software interface. There are eight necessary parameters that must be set for the MultiVOIP unit to
operate properly, with some additional settings that are optional. You must know the IP address that will be used,
the IP mask, the Gateway IP, the Domain Name Server information, and the telephone interface type. The
MultiVOIP must be configured locally at first, but changes to this initial configuration can be done 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 software is used for this.
Alternatively, MultiVoipManager is a Simple Network Management Protocol (SNMP) agent program that extends
the capabilities of the MultiVOIP configuration software. MultiVoipManager allows the user to manage any number
of VOIPs on a network, whereas the MultiVOIP configuration software manages only one. The MultiVoipManager
can configure multiple VOIPs simultaneously. MultiVoipManager may reside on the same PC as the MultiVOIP
configuration software.
This chapter will explain the setup portion of the software pertaining to the list below, while Chapter 5 will cover
the Phone Book setup and Chapter 6 will discuss the Statistics options and overall maintenance of the MultiVOIP.
Software Categories Covered in This Chapter
¾ Ethernet/IP
¾ Voice/Fax
¾ Interface
¾ Call Signaling
o H.323/SIP/SPP
¾ SNMP
¾ Regional
¾ SMTP
¾ RADIUS
¾ Logs/Traces
¾ NAT Traversal
¾ Supplementary services
¾ Save Setup
¾ Connection
o Settings
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Chapter 4: Configuring your VOIP
How to Navigate Through the Software
The MultiVOIP software is launched from the Start button and is found in the All Programs area under the title of
MultiVOIP x.xx (where x represents version number). The top option is “Configuration” – choose this.
Within the software, there are several ways to arrive at the parameter that you want to use: through the left-hand
panel, from the drop-down menu, clicking a taskbar icon (if available) or a keyboard shortcut (if available). Once
the initial settings are entered, you may choose to configure the MultiVOIP through a Web browser instead.
Web Browser Interface
The MultiVOIP web browser interface gives access to the same commands and configuration parameters as are
available in the MultiVOIP Windows interface except for logging functions. When using the web browser
interface, logging can be done by email (the SMTP option).
Set up the Web Browser interface (Optional). After an IP address for the MultiVOIP unit has been established,
you can choose to configure the unit by using the MultiVOIP web browser interface. If you want to do
configuration work using the web browser interface, you must first set it up:
•
•
•
•
•
•
•
•
•
Set IP address of MultiVOIP unit using the MultiVOIP Configuration program (the Windows interface).
Save Setup in Windows interface.
Close Windows interface.
Install Java program from MultiVOIP product CD (on first use only).
Open web browser.
Browse to IP address of MultiVOIP unit.
If username and password have been established, enter them when prompted.
Set browser to allow pop-ups. The MultiVOIP Web interface makes use of pop-up windows.
The configuration screens in the web browser will have the same content as their counterparts in the
software; only the presentation differs.
Configuration Information Checklist
To assist with the organization of the information needed, below is a chart summarizing what is necessary. The
–SS and –FX models do not support E&M or DID.
Info
Info
Type of Configuration Info
Gathered:
Configuration screen where info is entered:
Obtained? Entered?
D
D
Ethernet/IP parameters
IP info for VOIP unit
• IP address
• Gateway
• DNS IP (if used)
• 802.1p Prioritization (if used)
Interface parameters
Interface Type
• E&M
• FXS/FXO*
• DID-DPO
(*In FXS/FXO systems, channels used for phone, fax,
or key system are FXS; channels used for analog
PBX extensions or analog telco lines are FXO).
Interface parameters
E&M info (only if E&M used)
• Type (1-5)
• 2 or 4 wires
• Dial Tone or Wink
Country code
Email address for VOIP (optional)
Regional parameters
SMTP parameters
Reminder: Be sure to Save Setup after entering configuration values.
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Chapter 4: Configuring your VOIP
Ethernet/IP
This section covers the Ethernet settings needed for the MultiVOIP unit. In each field, enter the values that fit the
network to which the MultiVOIP will be connected to. For many of the settings, the default values will work best –
try these settings first unless you know you definitely need to change a parameter.
Figure 4-1: Network parameters
The Ethernet/IP Parameters fields are described in the tables and text passages below. Note that both Diff Serv
parameters (Call Control PHB and VOIP Media PHB) must be set to zero if you enable Packet Prioritization
(802.1p). Nonzero Diff Serv values negate the prioritization scheme.
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Chapter 4: Configuring your VOIP
Ethernet/IP Parameter Definitions
Description
Field Name
Values
Ethernet Parameters
Packet Prioritization
Y/N
Select to activate prioritization under 802.1p protocol (described below).
Must be set to match network’s frame type. Default is Type II.
(802.1p)
Frame Type
802.1p
Type II, SNAP
A draft standard of the IEEE about data traffic prioritization on Ethernet networks. The 802.1p
draft is an extension of the 802.1D bridging standard. 802.1D determines how prioritization will
operate within a MAC-layer bridge for any kind of media. The 802.1Q draft for virtual local-area-
networks (VLANs) addresses the issue of prioritization for Ethernet networks in particular.
802.1p enacts this Quality-of-Service feature using 3 bits. This 3-bit code allows data switches to
reorder packets based on priority level. The descriptors for the 8 priority levels are given below.
802.1p PRIORITY LEVELS:
LOWEST PRIORITY
1 – Background: Bulk transfers and other activities permitted on the network, but should not
affect the use of network by other users and applications.
2 – Spare: An unused (spare) value of the user priority.
0 – Best Effort (default): Normal priority for ordinary LAN traffic.
3 – Excellent Effort: The best effort type of service that an information services organization
would deliver to its most important customers.
4 – Controlled Load: Important business applications subject to some form of “Admission
Control”, such as preplanning of Network requirement, characterized by bandwidth
reservation per flow.
5 – Video: Traffic characterized by delay < 100 ms.
6 – Voice: Traffic characterized by delay < 10 ms.
7 - Network Control: Traffic urgently needed to maintain and support network infrastructure.
HIGHEST PRIORITY
Call Control Priority
VOIP Media Priority
Others (Priorities)
VLAN ID
0-7, where 0 is
lowest priority
Sets the priority for signaling packets.
0-7, where 0 is
lowest priority
Sets the priority for media packets.
0-7, where 0 is
lowest priority
Sets the priority for SMTP, DNS, DHCP, and other packet types.
1 - 4094
The 802.1Q IEEE standard allows virtual LANs to be defined within a network.
This field identifies each virtual LAN by number.
IP Parameter fields
Gateway Name
Enable DHCP
alphanumeric
Y/N
Descriptor of current VOIP unit to distinguish it from other units in system.
Dynamic Host Configuration Protocol is a method for assigning IP address and
other IP parameters to computers on the IP network in a single message with
great flexibility. IP addresses can be static or temporary depending on the
needs of the computer.
disabled by
default
IP Address
IP Mask
Gateway
n.n.n.n
n.n.n.n
n.n.n.n
The unique LAN IP address assigned to the MultiVOIP.
Subnetwork address that allows for sharing of IP addresses within a LAN.
The IP address of the device that connects your MultiVOIP to the Internet.
Table is continued on next page…
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Chapter 4: Configuring your VOIP
Ethernet/IP Parameter Definitions (continued)
Description
Field Name
Diff Serv
Parameter
fields
Values
Diff Serv PHB (Per Hop Behavior) values pertain to a differential prioritizing system for IP packets as
handled by Diff Serv-compatible routers. There are 64 values, each with an elaborate technical
description. These descriptions are found in TCP/IP standards RFC2474, RFC2597, and, for present
purposes, in RFC3246, which describes the value 34 (34 decimal; 22 hex) for Assured Forwarding
behavior (default for Call Control PHB) and the value 46 (46 decimal; 2E hexadecimal) for Expedited
Forwarding behavior (default for VOIP Media PHB). Before using values other than these default
values of 34 and 46, consult these standards documents and/or a qualified IP telecommunications
engineer.
To disable Diff Serv, configure both fields to 0 decimal.
Call Control
PHB
0 – 63
default = 34
Value is used to prioritize call setup IP packets.
Setting this parameter to 0, in conjunction with VOIP Media PHB below will disable
Diff Serv.
VOIP Media
PHB
0 – 63
default = 46
Value is used to prioritize the RTP/RTCP audio IP packets.
Setting this parameter to 0, in conjunction with Call Control PHB above will disable
Diff Serv.
FTP Parameter fields
FTP Server
Enable
Y/N
MultiVOIP unit has an FTP Server function so that firmware and other important
operating software files can be transferred to the VOIP via the network.
Default =
disabled
See “FTP
Server File
Transfers” in
Chapter 6
DNS Parameter fields
Enable DNS
Y/N
Default =
disabled
Enables Domain Name Space/System function where computer names are resolved
using a worldwide distributed database.
Enable SRV
Y/N
Enables ‘service record’ function. Service record is a category of data in the Internet
Domain Name System specifying information on available servers for a specific
protocol and domain, as defined in RFC 2782. Newer internet protocols like SIP,
STUN, H.323, POP3, and XMPP may require SRV support from clients. Client
implementations of older protocols, like LDAP and SMTP, may have been enhanced
in some settings to support SRV.
DNS Server IP
Address
n.n.n.n
IP address of specific DNS server to be used to resolve Internet computer names.
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Chapter 4: Configuring your VOIP
Voice/Fax
Setting the Voice/FAX Parameters. The Voice/Fax section needs to be set for each channel to be used. 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.
The majority of the settings should be left at their default settings as changes often introduce problems with signal
quality. In each field, enter the values that fit your particular setup. The –SS models do not have Configurable
Payload Type available.
Figure 4-2: Voice/Fax parameters
The Voice/FAX Parameters settings are described in the tables below.
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Chapter 4: Configuring your VOIP
Voice/Fax Parameter Definitions
Field Name
Default
Select Channel
Values
Description
--
When this button is clicked, all Voice/FAX parameters are set to their default values.
Channel to be configured is selected here.
1-2 (210)
1-4 (410)
1-8 (810)
Copy Channel
--
Copies the Voice/FAX attributes of one channel to another channel. Attributes can
be copied to multiple channels or all channels at once.
Voice Gain
Input Gain
--
Signal amplification (or attenuation) in dB.
Modifies audio level entering voice channel before it is sent over the network to the
remote VOIP. The default & recommended value is 0 dB.
+31dB to
–31dB
Output Gain
DTMF Gain
+31dB to
–31dB
--
Modifies audio level being output to the device attached to the voice channel. The
default and recommended value is 0 dB.
The DTMF Gain (Dual Tone Multi-Frequency) controls the volume level of the DTMF
tones sent out for Touch-Tone dialing.
DTMF Gain,
High Tones
+3dB to
-31dB &
“mute”
Default value: -4 dB. Not to be changed except under supervision of Multi-Tech
Technical Support.
DTMF Gain, Low +3dB to
Default value: -7 dB. Not to be changed except under supervision of Multi-Tech
Tones
-31dB &
“mute”
Technical Support.
DTMF Parameters
Duration (DTMF) 60 – 3000
ms
When DTMF: Out of Band is selected, this setting determines how long each DTMF
digit ‘sounds’ or is held. Default = 100 ms.
DTMF
In/Out of Band
Out of
Band, or
Inband
When DTMF Out of Band is selected, 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.
Out of Band
Mode
RFC 2833, RFC2833 method. Uses an RTP mode defined in RFC 2833 to transmit the DTMF
SIP Info
digits.
SIP Info method. Generates dual tone multi frequency (DTMF) tones on the
telephony call leg. The SIP INFO message is sent along the signaling path of the
call.
You must set this parameter per the capabilities of the remote endpoint with which
the VOIP will communicate. The RFC2833 method is the more common of the two
methods.
FAX Parameters
Fax Enable
Y/N
Enables or disables fax capability for a particular channel.
Modem Relay
Enable
Y/N
When enabled, modem traffic can be carried on VOIP system. When disabled,
modem traffic will bypass the VOIP system (Modem Bypass mode).
Max Baud Rate
(Fax)
2400,
4800,
Set to match baud rate of fax machine connected to channel (see Fax machine’s user
manual).
7200,
Default = 14400 bps.
9600,
12000,
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
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 real time faxing of Group 3 faxes over IP networks. It
uses T.30 fax standards and includes special provisions to preclude FAX timeouts
during IP transmissions.
Table is continued on next page…
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Chapter 4: Configuring your VOIP
Voice/Fax Parameter Definitions (continued)
Coder Parameters
Coder
Manual or
Automatic
Determines whether selection of 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 law 64 kbps;
(SS models only) G.726, @ 16/24/32/40 kbps;
Select from a range of coders with 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.727, @ nine bps rates;
G.723.1 @ 5.3 kbps, 6.3 kbps; Default = G.723.1 @ 6.3 kbps, as required for H.323. Here 64K
G.729, 8kbps;
Net Coder @
6.4, 7.2, 8, 8.8, 9.6 kbps
of digital voice is compressed to 6.3K, allowing several
simultaneous conversations over the same bandwidth that would
otherwise carry only one.
To make selections from the Selected Coder drop-down list, the
Manual option must be enabled.
Selected Coder
G.711, G.729
-or-
G.729, G.711
Coder Priority has two options (G.711,G.729 or G.729, G711) on
the Selected Coder listing of the Coder group on the Voice/Fax
screen. If G.711 is the higher priority, i.e., G.711 is preferred to
G729 on the sending side, then G.711, G.729 option is selected.
Similarly, if G.729 has the higher priority, then G.729, G.711
option is selected.
It is used whenever a user wants to advertise both G.711 and
G.729 coders with higher preference to a particular coder.
It is useful when the calls are made from a particular channel on
the VOIP to two different destinations where one supports G.711
and the other supports G.729.
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.
Advanced Features
Silence
Compression
Y/N
Determines whether silence compression is enabled (checked)
for this voice channel.
With Silence Compression enabled, the MultiVOIP will not
transmit voice packets when silence is detected, thereby
reducing the amount of network bandwidth that is being used by
the voice channel (default = on).
Echo
Cancellation
Y/N
Y/N
Determines whether echo cancellation is enabled (checked) for
this voice channel.
Echo Cancellation removes echo and improves sound quality
(default = on).
Forward Error
Correction
Determines whether forward error correction is enabled
(checked) for this voice channel.
Forward Error Correction enables some of the voice packets
that were corrupted or lost to be recovered. FEC adds an
additional 50% overhead to the total network bandwidth
consumed by the voice channel (default = Off).
Table is continued on next page…
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Chapter 4: Configuring your VOIP
Voice/Fax Parameter Definitions (continued)
Field Name
Values
Description
AutoCall/Offhook Alert
Parameters
Auto Call / Offhook
Alert
AutoCall,
Offhook
Alert
The AutoCall option enables the local MultiVOIP to call a remote MultiVOIP
without the user having to dial a Phone Directory Database number. As soon as
you access the local MultiVOIP voice/fax channel, the MultiVOIP immediately
connects to the remote MultiVOIP identified in the Phone Number box of this
option.
If the “Pass Through Enable” field is checked in the Interface Parameters
screen, AutoCall must be used.
The Offhook Alert option applies only to FXS channels.
The Offhook Alert option works like this: if a phone goes off hook and yet no
number is dialed within a specific period of time (as set in the Offhook Alert
Timer field), then that phone will automatically dial the Alert phone number for
the VOIP channel. (The Alert phone number must be set in the Voice/Fax
Parameters | Phone Number field; if the VOIP system is working without a
gatekeeper unit, there must also be a matching phone number entry in the
Outbound Phonebook.). One use of this feature would be for emergency use
where a user goes off hook but does not dial, possibly indicating a crisis
situation. The Offhook Alert feature uses the Intercept Tone, as listed in the
Regional Parameters screen. This tone will be outputted on the phone that
was taken off hook but that did not dial. The other end of the connection will
hear audio from the “crisis” end as is it would during a normal phone call.
Both functions apply on a channel-by-channel basis. It would not be
appropriate for either of these functions to be applied to a channel that serves in
a pool of available channels for general phone traffic. Either function requires
an entry in the Outgoing phonebook of the local MultiVOIP and a matched
setting in the Inbound Phonebook of the remote VOIP.
Generate Local Dial
Tone
Y/N
Used for AutoCall only. If selected, dial tone will be generated locally while the
call is being established between gateways. The capability to generate dial
tone locally would be particularly useful when there is a lengthy network delay.
Offhook Alert Timer
Phone Number
0 – 3000
seconds
The length of time that must elapse before the off hook alert is triggered and a
call is automatically made to the phone number listed in the Phone Number
field.
--
Phone number used for Auto Call function or Offhook Alert Timer function. This
phone number must correspond to an entry in the Outbound Phonebook of the
local MultiVOIP and in the Inbound Phonebook of the remote MultiVOIP (unless
a gatekeeper unit is used in the VOIP system).
Table is continued on next page…
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Chapter 4: Configuring your VOIP
Voice/Fax Parameter Definitions (continued)
Field Name
Values
Description
Dynamic Jitter
Dynamic Jitter Buffer
Dynamic Jitter defines a minimum and a maximum jitter value for voice
communications. When receiving voice packets from a remote MultiVOIP,
varying delays between packets may occur due to network traffic problems.
This is called Jitter. To compensate, the MultiVOIP uses a Dynamic Jitter
Buffer. The Jitter Buffer enables the MultiVOIP to wait for delayed voice
packets by automatically adjusting the length of the Jitter Buffer between
configurable minimum and maximum values. An Optimization Factor
adjustment controls how quickly the length of the Jitter Buffer is increased
when jitter increases on the network. The length of the jitter buffer directly
affects 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 ms
Maximum Jitter Value 60 to 400
ms
The maximum dynamic jitter buffer of 400 milliseconds is the maximum delay
tolerable over a high jitter network.
Default = 300 ms
Optimization 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.
Auto Disconnect
Automatic
--
The Automatic Disconnection group provides four options which can be
Disconnection
used singly or in any combination.
Jitter Value
1-65535
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
1-65535
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 some configurations, requiring upward adjustment.
Consecutive 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
1 to 65535; Specifies how long to wait before disconnecting the call when IP network
Disconnection
Default =
30 sec.
connectivity with the remote site has been lost.
Configurable Payload Type
(Not available on the –SS series)
The Configurable Payload Type is located on the bottom of the Voice/Fax screen. The Configurable Payload
Type is used when the remote side uses a different payload type for the associated features. In previous
firmware versions, MultiVOIP’s used 101 for DTMF RFC2833. If the remote side uses some other dynamic
payload type such as 110, it will fail. To avoid these failures, the payload types are made configurable.
DTMF RFC2833 Configurable Payload Type is supported only for SIP & SPP and not for H.323.
Whenever you interoperate with older MultiVOIP products (i.e., earlier than release x.11), for backward
compatibility, make sure to configure the payload type values to default ones, which match the values of
older MultiVOIP’s.
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Chapter 4: Configuring your VOIP
Interface
The Telephony Interface parameters are set individually for each channel and include the line types as well as
some specific situational settings for those that need them. 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.). Here you will
find the various parameters grouped and organized by interface type. Note that the SS and FX models only
support FXS/FXO. In each field, enter the values that fit your particular setup. Once you have established a set
of Interface parameters for a particular channel, you can apply this entire set of Voice/FAX parameters to another
channel by using the Copy Channel button and its dialog box. To copy a set of Interface parameters to all
channels, select “Copy to All” and click Copy. The screen below shows more options available than are actually
used for clarity. Your settings will determine what fields are available. The –SS series of MultiVOIPs do not
support Caller ID Manipulation.
Figure 4-3: Telephony parameters
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Chapter 4: Configuring your VOIP
FXS Loop Start Parameters
The parameters applicable to FXS Loop Start are shown in the figure below and described in the table
that follows.
Figure 4-4: FXS Loop Start parameters
FXS Loop Start Interface: Parameter Definitions
Field Name
Values
Description
Dialing Options fields
FXS (Loop Start)
Y/N
Enables FXS Loop Start interface type.
Inter Digit Timer
1 - 10 seconds
This is the length of time that the MultiVOIP will wait between digits.
When the time expires, the MultiVOIP will look in the outbound
phonebook for the number entered and place the call accordingly.
Default = 2.
Message Waiting
Indication
--
Not applicable to –SS series MultiVOIPs.
Inter Digit
Regeneration Time
in milliseconds
The length of time between the outputting of DTMF digits.
Default = 100 ms.
FXS Options fields
FXS Ring Count,
FXS
1-10
Maximum number of rings that the MultiVOIP will issue before giving
up the attempted call.
Current Loss
Y/N
When enabled, the MultiVOIP will interrupt loop current in the FXS
circuit to initiate a disconnection. This tells the device connected to the
FXS port to hang up. The Multi-VOIP cannot drop the call; the FXS
device must go on hook.
Generate Current
Reversal
Y/N
When selected, this option implements Answer Supervision and
Disconnect Supervision to the FXO interface using current reversal to
indicate events. Applicable only when FXS and FXO interfaces are
connected back to back.
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Chapter 4: Configuring your VOIP
FXS Loop Start Interface: Parameter Definitions (continued)
Field Name
Values
Description
Flash Hook Options fields
Generation
--
Not applicable to FXS interface
Detection Range
for Min. and
Max., 50 - 1500
milliseconds
For a received flash hook to be regarded as such by the MultiVOIP, its
duration must fall between the minimum and maximum values given
here
Pass Through
Enable
Y/N
When enabled, this parameter creates an open audio path through the
MultiVOIP.
If the Pass-Through feature is enabled, the AutoCall feature must be
enabled for this VOIP channel in the Voice/Fax Parameters screen
Caller ID fields
Type
Bellcore
The MultiVOIP currently supports only one implementation of Caller
ID. That implementation is Bellcore type 1 with Caller ID placed
between the first and second rings of the call.
Enable
Y/N
Caller ID information is a description of the remote calling party
received by the called party. The description has three parts: name of
caller, phone number of caller, and time of call. The ‘time-of-call’
portion is always generated by the receiving MultiVOIP unit (on FXS
channel) based on its date and time setup.
The forms of the ‘Caller Name’ and ‘Caller Phone Number’ differ
depending on the IP transmission protocol used (H.323, SIP, or SPP)
and upon entries in the phonebook screens of the remote (CID
generating) VOIP unit. The CID Name and Number appearing on the
phone at the terminating FXS end will come either from a central office
switch (showing a PSTN phone number), or the phonebook of the
remote (CID sending) VOIP unit.
CID Manipulation
Enabled by
default with
Caller ID enable
above
This is not implemented in the –SS series VOIPs.
Caller ID Manipulation is used whenever the user wants to manipulate
the Caller ID before sending it to the remote end. Caller ID
Manipulation is activated on the Interface Screen. By enabling Caller
ID option, you can set manipulation to Transparent, User CID, Prefix,
Suffix, or Prefix and Suffix. Caller ID Manipulation is a feature, where
the Caller ID detected from the PSTN line can be changed and then
sent to the remote side over IP.
Disable
CID Mode
Transparent,
User CID,
Prefix,
The MultiVOIP is not allowed to modify the caller ID info and then
send it to the PSTN side. It only allows it to detect the caller ID
from the PSTN line, modify it and then send them via IP to the
remote end point.
Suffix
Transparent: the CID received from PSTN will be sent out as such,
without any manipulation.
User CID: the CID received from PSTN will be replaced by this User
CID value.
Prefix: the CID received from PSTN will be prefixed with this value.
Suffix: the CID received from PSTN will be suffixed with this value.
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Chapter 4: Configuring your VOIP
Message Waiting
Message Waiting Indication is a feature that displays an audible or visible indication that a message
available. A type of message waiting is sounding a special dial tone (called stutter dial tone), lighting a light,
or indicator on the phone.
When a user enables a subscription for message waiting indication, a subscription is made with the Voice
Mail Server (VMS) for that particular event. Whenever the Voice Mail Server finds a change in the state of a
corresponding mailbox or some event happens (e.g., when a new voice message is recorded or a message
is deleted, then the VMS server sends a notification to the gateway. Its indication to the user is a flashing
LED or sounding a stutter dial tone.
The message waiting feature is active when the Use SIP Proxy option is selected on the Call Signaling SIP
screen, a Primary Proxy IP address is entered in the SIP Proxy Parameters Primary Proxy field, the Voice
Mail Server Domain Name or IP Address is entered in the SIP Voice Mail Server Parameters Group, and the
Interface Type is set to FXS (Loop start). Then the FXS Options Group becomes active. The Message
Waiting Indication options are None, Light, or Stutter Dial Tone.
Figure 4-5: Message Waiting
To receive messages from the VMS (Voice Mail Server/System), the subscription needs to be enabled and
the voice mail server address has to be entered in the SIP Voice Mail Server Parameters Group.
The Voice Mail server IP Address, Port and Re-subscription time are configured on the SIP Call Signaling
screen. When this is configured, the “Subscribe with Voice Mail Server” option is activated in the inbound
phone book. Only when this option is enabled, the subscribe message will be sent to the VMS.
The following sequence needs to be done to enable all of the Message Waiting Features:
1. The "Use SIP Proxy" must be enabled, and the SIP Proxy Parameters and Voice Mail Server Parameters
in the SIP Call Signaling Menu must be set, and the Interface Type option must be set to FXS (Loop Start)
on the Interface menu's "Message Waiting Indication" options become active.
2. Then the "Message Waiting Indication" options must be set to light or stutter tone for the "Subscribe to
Voice Mail Server" option to become available in the Inbound phone book entry with that channel selected.
3. In order to send Subscriptions for Inbound Phone Book entries, all the following four conditions have to be
satisfied:
•
The user needs to enter a valid voice mail server domain name or IP address in the Voice Mail Server
Domain Name/IP Address field on the Call Signaling screen.
•
•
•
For an Inbound Phone Book entry, a subscription with Voice Mail Server checkbox is enabled on the
Add or Edit Inbound Phone Book entries screen.
The Channel type corresponding to that Inbound phone book entry has to be FXS on the Interface
screen.
The Message Waiting Indication has to be either Light or Stutter Dial Tone on the Interface
Parameters screen.
The password on the Interface screen is used for that particular channel when a “SUBSCRIBE” request is
sent (i.e., if the MultiVOIP gets a 401/407 response from a subscribe request. Then it will take the configured
password, calculate the response, and resend the “SUBSCRIBE” request.
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Chapter 4: Configuring your VOIP
FXO Parameters
The parameters applicable to the FXO telephony interface type are shown in the figure below and
described in the table that follows.
Figure 4-6: FXO parameters
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Chapter 4: Configuring your VOIP
FXO Interface: Parameter Definitions
Description
Field Name
Values
Interface Type
FXO
Enables FXO functionality
Dialing Options
Regeneration
Pulse, DTMF
Determines whether digits generated and sent out will be pulse tones
or DTMF.
Inter Digit Timer
1 to 10 seconds
This is the length of time that the MultiVOIP will wait between digits.
When the time expires, the MultiVOIP will look in the phonebook for
the number entered.
Default = 2.
Message Waiting
Indication
--
Not applicable to FXO interface
Inter Digit
Regeneration Time milliseconds
50 to 20,000
The length of time between the outputting of DTMF digits.
Default = 100 ms.
FXO Options
FXO Ring Count
1-99
Number of rings required before the MultiVOIP answers the incoming
call.
No Response
Timer
1 – 65535
(in seconds)
Length of time before call connection attempt is abandoned.
Flash Hook Options fields
Generation
50 - 1500
milliseconds
Length of flash hook that will be generated and sent out when the
remote end initiates a flash hook and it is regenerated locally. Default
= 600 ms.
Detection Range
Caller ID Type
--
Not applicable to FXO.
Caller ID fields
Bellcore
The MultiVOIP currently supports only one implementation of Caller
ID. That implementation is Bellcore type 1 with caller ID placed
between the first and second rings of the call.
Caller ID enable
Y/N
Caller ID information is a description of the remote calling party
received by the called party. The description has three parts: name of
caller, phone number of caller, and time of call. The ‘time-of-call’
portion is always generated by the receiving MultiVOIP unit (on FXS
channel) based on its date and time setup. The forms of the ‘Caller
Name’ and ‘Caller Phone Number’ differ depending on the IP
transmission protocol used (H.323, SIP, or SPP) and upon entries in
the phonebook screens of the remote (CID generating) VOIP unit.
The CID Name and Number appearing on the phone at the terminating
FXS end will come either from a central office switch (showing a PSTN
phone number), or the phonebook of the remote (CID sending) VOIP
unit.
CID Manipulation
Enabled by
default with
Caller ID enable
above
This is not implemented in the –SS series VOIPs.
Caller ID Manipulation is used whenever the user wants to manipulate
the Caller ID before sending it to the remote end. Caller ID
Manipulation is activated on the Interface Screen. By enabling Caller
ID option, you can set manipulation to Transparent, User CID, Prefix,
Suffix, or Prefix and Suffix. Caller ID Manipulation is a feature, where
the Caller ID detected from the PSTN line can be changed and then
sent to the remote side over IP.
Disable
CID Mode
Transparent,
User CID,
Prefix,
The MultiVOIP is not allowed to modify the caller ID info and then
send it to the PSTN side. It only allows it to detect the caller ID
from the PSTN line, modify it and then send them via IP to the
remote end point.
Suffix
Transparent: the CID received from PSTN will be sent out as such,
without any manipulation.
User CID: the CID received from PSTN will be replaced by this User
CID value.
Prefix: the CID received from PSTN will be prefixed with this value.
Suffix: the CID received from PSTN will be suffixed with this value.
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Chapter 4: Configuring your VOIP
FXO Supervision
When the selected Interface type is FXO, the Supervision button is active. Click on this button to access
call answering supervision parameters and call disconnection parameters that relate to the FXO interface
type.
Figure 4-7: FXO Supervision
The table below describes the settings for FXO Supervision.
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FXO Supervision Parameter Definitions
Values Description
Field Name
Answer Supervision fields
Current Reversal Y/N
When this option is selected, the FXO interface sends notice to make
connection upon detecting current reversal from the PBX (which
occurs when the called extension goes off hook).
Answer Delay
Y/N
When this option is selected, the FXO interface sends the connection
notice to the calling party only when the Answer Delay Timer expires.
The connection notice is sent regardless of whether or not the called
extension has gone off hook.
Answer Delay
Timer
1 – 65535
(in seconds)
When Answer Delay is enabled, this value determines when the FXO
interface sends the connection notice.
Tone Detection
Y/N
When selected, call disconnection will be triggered by a tone
sequence
Available Tones
dial tone,
List from which tones can be chosen to signal call answer.
ring tone,
busy tone,
unobtainable tone
(fast busy),
survivability tone,
re-order tone
Answer Tones
any tone from
Currently chosen call-answer supervision tone.
Available Tones list
Disconnect Supervision fields
There are four possible criteria for disconnection under FXO: current
reversal, current loss, tone detection, and silence detection.
Disconnection can be triggered by more than one of the three criteria.
Current Reversal
Current Loss
Y/N
Y/N
Disconnection to be triggered by reversal of current from the PBX.
Disconnection to be triggered by loss of current. That is, when Current
Loss is enabled (“Y”), the MultiVOIP will hang up the call at a specified
interval after it detects a loss of current initiated by the attached
device.
Current Loss Timer 200 to 2000
(in milliseconds)
Y/N
Determines the interval after detection of current loss at which the call
will be disconnected.
Silence Detection
Enable
Enables/disables silence-detection method of supervising call
disconnection.
Silence Detection
Type
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
Duration of silence required to trigger disconnection.
Table is continued on next page…
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Chapter 4: Configuring your VOIP
FXO Supervision Parameter Definitions (continued)
Field Name
Disconnect Supervision fields
DTMF Tone
Values
Description
Enables supervision of call disconnection using DTMF tones.
DTMF Tone Pairs
Low Tones
1
2
3
A
697Hz
770Hz
852Hz
941Hz
4
5
6
B
C
7
*
8
0
9
#
D
High Tones
1209Hz
1336Hz
1447Hz
1633Hz
Disconnect
Tone
Sequence
1st tone pair
+
These are DTMF tone pairs.
Values for first tone pair are: *, #, 0, 1-9, and A-D.
Values for second tone pair are: none, 0, 1-9, A-D, *, and #.
The tone pairs 1-9, 0, *, and # are the standard DTMF pairs found on
phone sets. The tone pairs A-D are “extended DTMF” tones, which
are used for various PBX functions.
2nd tone pair
Tone Detection Y/N
Enables supervision of call disconnection by detecting cessation of a
pre-specified tone from the PBX.
Available
Tones
dial tone,
ring tone,
List from which tones can be chosen to signal call disconnection.
busy tone,
unobtainable tone
(fast busy),
survivability tone,
re-order tone
Disconnect
Tones
any tone from
Available Tones list
Currently chosen disconnection supervision tone.
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Chapter 4: Configuring your VOIP
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. Only the analog MVP210/410/810 models support the E&M interface,
the -SS and -FX models do not.
Figure 4-8: E&M parameters
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E&M Interface Parameter Definitions
Description
Field Name
Interface
Type
Values
E&M
Enables E&M functionality
I – V
Type of E&M interface being used – the individual types are detailed
below.
Default = Type II.
Mode
2-wire or 4-wire
Each E&M interface type can be either 2-wire or 4-wire audio.
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
100 - 350
This is the length of the wink for wink signaling. Applicable only when
milliseconds
Signal parameter is set to “Wink.”
No Response Timer 1 – 65535
(in seconds)
The value here denotes the time (in seconds) after which the call
attempt would be disconnected by the FXO Interface because there
was no answer.
Disconnect on Call
Progress Tone
Y/N
Y/N
Allows call on FXO port to be disconnected when a PBX issues a call-
progress tone denoting that the phone station on the PBX that has
been involved in the call has been hung up
When enabled (“Y”), this feature is used to create an open audio path
for 2- or 4-wire. The E&M leads are passed through the VOIP
transparently.
Pass Through
Enable
Applicable only for E&M Signaling with Dial Tone (not applicable for
Wink signaling).
Dialing Options
Inter Digit Timer
1 - 10 seconds
This is the length of time that the MultiVOIP will wait between digits.
When the time expires, the MultiVOIP will look in the phonebook for
the number entered.
Default = 2.
Message Waiting
Indication
Light or None
Allows MultiVOIP to pass mode-code sequences between Avaya
Magix PBXs to turn on and off the message-waiting light on a PBX
extension phone.
Mode codes:
*53 + PBX extension
Î turns message light on.
#53 + PBX extension
Î turns message light off.
Signals to turn message-waiting lights on/off are not sent to phones
connected directly to the MultiVOIP on FXS channels, not to other
non-Avaya Magix PBX phone stations on the VOIP network
The length of time between the outputting of DTMF digits.
Default = 100 ms.
Inter Digit
Regeneration
Timer
50 – 20000
milliseconds
Flash Hook Options fields
Generation
50 - 1500
milliseconds
Length of flash hook that will be generated and sent out when the
remote end initiates a flash hook and it is regenerated locally. Default
= 600 ms.
Detection Range
for Min. and
Max., 50 - 1500
milliseconds
For a received flash hook to be regarded as such by the MultiVOIP, its
duration must fall between the minimum and maximum values given
here.
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Chapter 4: Configuring your VOIP
E&M Interface Types
There are five different types of the E&M interface and the MVP210/410/810 models support them all; but
Type IV is largely unused and will not be detailed in this section. The figures below will show the pin
assignments for the MVP RJ48 connector when used in the E&M jacks on the back of the unit as well as
how the signals are used for types one, two, three and five. Common ground between the MultiVOIP and
PBX is required for all E&M Types except Type II. Two and four wire audio is available for all E&M Types
and is shown in figure 4-9 below.
Figure 4-9: MultiVOIP E&M Pin assignments and RJ48 Jack
Figure 4-10: E&M Line Types
Figure 4-11: Audio wiring
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Chapter 4: Configuring your VOIP
DID Parameters
The parameters applicable to the Direct Inward Dial (DID) telephony interface type are shown in the figure
below and described in the table that follows. The DID interface allows one phone line to direct incoming
calls to any one of several extensions without a switchboard operator. Of course, one DID line can
handle only one call at a time. The parameters described here pertain to the customer-premises side of
the DID connection (DID-DPO, dial-pulse originating); the network side of the DID connection (DID-DPT,
dial-pulse terminating) is not supported.
The –SS and –FX models do not support DID.
Figure 4-12: DID parameters
DID Interface Parameter Definitions
Field Name
Values
Description
Interface
DID-DPO
Enables the customer-premises side of DID functionality
DID Options
MultiVOIP’s use of DID applies only for incoming DID calls. The Start
Mode used by the MultiVOIP must match that used by the originating
telephony equipment; else DID calls cannot be completed.
Start Modes
Immediate Start,
Wink Start,
Delay Dial
For Immediate Start, the VOIP detects the off-hook condition initiated
by the telco central-office call and becomes ready to receive dial digits
immediately.
For Wink Start, the VOIP detects the off-hook condition. Then the
VOIP reverses battery polarity for a specified time (140-290 ms; a
“wink”) and then becomes ready to receive dial digits.
For Delay Dial, the VOIP detects the off-hook condition. Then the
VOIP reverses battery polarity for a specified time (reverse polarity
duration has wider acceptable range than for Wink Start) and then
becomes ready to receive dial digits.
Wink Timer
(in ms)
Integer values,
in milliseconds
This is the length of the wink for Wink Start and Delay Dial signaling
modes.
Applicable only when Start Mode parameter is set to “Wink Start” or
“Delay Dial.”
Dialing Options
Inter Digit Timer
Integer values,
in seconds
This is the length of time that the MultiVOIP will wait between digits.
When the time expires, the MultiVOIP will look in the phonebook for
the number entered.
Default = 2.
Message Waiting
Indication
--
Not applicable to DID-DPO interface.
Inter-Digit
Regeneration
Timer
Integer values,
in milliseconds
This parameter is applicable when digits are dialed onto a DID-DPO
channel after the connection has been made. The length of time
between the outputting of DTMF digits.
Default = 100 ms.
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Chapter 4: Configuring your VOIP
Call Signaling
There are three types of Call Signaling available: H.323, SIP and SPP. Each type has some individual features
that may make it more appealing to use than the others, depending on your needs. The –SS and –FX models do
not support H.323 signaling.
H.323
H.323 is an ITU-T recommended set of standards for audio and video communications. The fields for this
screen are defined in the table below.
Figure 4-13: H.323 call signaling
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H.323 Call Signaling Parameter Definitions.
Field Name
Values
Description
Use Fast Start
Y/N
Enables the H.323 Fast Start procedure. May need to be enabled/disabled for
compatibility with third-party VOIP gateways.
Signaling Port
port
Default: 1720 (H.323)
Register with
Gatekeeper
Y/N
Check this field to have traffic on current VOIP gateway controlled by a
gatekeeper.
Allow Incoming
Calls Through
Gatekeeper Only
Y/N
When selected, incoming calls are accepted only if those calls come through the
gatekeeper.
GateKeeper RAS Parameters
Primary GK
Alternate GK
1 and 2
IP Address
RAS Port
--
--
This is the preferred gatekeeper for controlling the traffic of the current VOIP.
A first and a second alternate gatekeeper can be specified for use by the current
VOIP for situations where the Primary GK is busy or otherwise unavailable.
IP address of the GateKeeper.
n.n.n.n
1719
Well-known port number for GateKeepers. Must match port number (1719).
Gatekeeper
Name
alpha-
numeric
Optional. The name of the GateKeeper with which this MultiVOIP is trying to
register. A primary gatekeeper and two alternate units are listed.
RAS TTL Value
seconds The H.323 Gatekeeper “Time to Live” value. As soon as a MultiVOIP gateway
registers with a gatekeeper a countdown timer begins. The RAS TTL Value is the
interval of the countdown timer. Before the TTL countdown expires, the MultiVOIP
gateway needs to register with the gatekeeper in order to maintain the
connection. If the MultiVOIP does not register before the TTL interval expires, the
MultiVOIP gateway’s registration with the gatekeeper will expire and the
gatekeeper will no longer permit call traffic to or from that gateway. Calls in
progress will continue to function even if the gateway becomes de-registered
Gatekeeper
integer
The interval between the VOIP gateway’s successive attempts to connect to and
Discovery Polling 60 - 300 be governed by a higher level gatekeeper. The Primary GK is the highest level
Interval
Use Online
Alternate
gatekeeper. Alternate GK1 is second; Alternate GK2 is the lowest.
When selected, VOIP will seek an alternate gatekeeper (when none of the 3 gatekeepers
shown on this screen are available) from a list. The list will reside on the Primary gatekeeper
or one of the Alternate gatekeepers. The gatekeeper holding the list would download that list
onto the VOIP gateways within the system.
Gatekeeper List
H.323 Version 4 Options
H.323
Multiplexing
Y/N
Signaling for multiple phone calls can be carried on a single port rather than
opening a separate signaling port for each. This conserves bandwidth resources.
H.245 Tunneling
(Tun)
Y/N
H.245 messages are encapsulated within the Q.931 call-signaling channel.
Among other things, the H.245 messages let the two endpoints tell each other
what their technical capabilities are and determine who, during the call, will be the
client and who the server. Tunneling is the process of transmitting these H.245
messages through the Q.931 channel. The same TCP/IP socket (or logical port)
already being used for the Call Signaling Channel is then also used by the H.245
Control Channel. This encapsulation reduces the number of logical ports
(sockets) needed and reduces call setup time.
Parallel H.245
(FS + Tun)
Y/N
Y/N
FS (Fast Start) is a Q.931 feature of H.323v2 to hasten call setup as well as ‘pre-
opening’ the media channel before the CONNECT message is sent. This pre-
opening is a requirement for certain billing activities. Under Parallel H.245 FS +
Tun, this Fast Connect feature can operate simultaneously with H.245 Tunneling.
Multiplexed UDP call signaling transport. Annex E is helpful for high-volume VOIP
system endpoints. Gateways with lesser volume can afford to use TCP to
establish calls. However, for larger volume endpoints, the call setup times and
system resource usage under TCP can become problematic. Annex E allows
endpoints to perform call-signaling functions under the UDP protocol, which
involves substantially streamlined overhead (this feature should not be used on
the public Internet due to potential problems with security and bandwidth usage).
Annex –E (AE)
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Chapter 4: Configuring your VOIP
SIP
Session Initiation Protocol is the second option available for application layer control of the MultiVOIP. The
fields are detailed in the table below.
Figure 4-14: SIP call signaling
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Chapter 4: Configuring your VOIP
SIP Call Signaling Parameter Definitions
Field Name
SIP Proxy Parameters
Signaling Port
port
Values
Description
Port number on which the MultiVOIP UserAgent software module will be waiting for
any incoming SIP requests. Default = 5060
Use SIP Proxy Y/N
Allows the MultiVOIP to work in conjunction with a proxy server.
Allow Incoming Y/N
Calls Through
When selected, incoming calls are accepted only if those calls come through the
proxy.
SIP Proxy Only
Primary Proxy
Alternate Proxy --
1 and 2
Proxy Domain
Name / IP
Address
--
This is the preferred SIP proxy server for controlling the traffic of the current VOIP.
A first and a second alternate SIP proxy server can be specified for use by the
VOIP for situations where the Primary proxy server is otherwise unavailable.
Network address of the proxy server that the VOIP is using.
n.n.n.n
Append SIP
Proxy Domain
Name in User
ID
Y/N
When checked, the domain name of the SIP Proxy serving the MultiVOIP gateway
will be included as part of the User ID for that gateway. If unchecked, the SIP
Proxy’s IP address will be included as part of the User ID instead of the SIP
Proxy’s domain name.
Port Number
port
Logical port number for proxy communications. Default = 5060
This is not implemented in the –SS series VOIPs.
Default
Subscriber
This is used as the default end point register with a Proxy.
Default
Username
name
If the Username is not populated in the Phone Book, this is the Username that will
be used. This works the same for the password as well.
Password
password Password for proxy server function. See “Default Username” description above.
Re-Registration 10–65535 This is the timeout interval for registration of the MultiVOIP with a SIP proxy server.
Time
seconds
The time interval begins the moment the MultiVOIP gateway registers with the SIP
proxy server and ends at the time specified by the user in the Re-Registration Time
field (this field). When/if registration lapses, call traffic routed to/from the MultiVOIP
through the SIP proxy server will cease. However, calls in progress will continue to
function until they end.
Proxy Polling
Interval
60 - 300
The interval between the VOIP gateway’s successive attempts to connect to and
be governed by a higher level SIP proxy server. The Primary Proxy is the highest
level gatekeeper. Alternate Proxy 1 is second; Alternate Proxy 2 is the lowest
order SIP proxy server.
TTL Value
SIP proxy As soon as a MultiVOIP gateway registers with a SIP proxy server (allowing the
“Time to
Live”
value.
(in
proxy server to control its call traffic) a countdown timer begins. The TTL Value is
the interval of the countdown timer. Before the TTL countdown expires, the
MultiVOIP gateway needs to register with the gatekeeper in order to maintain the
connection. If the MultiVOIP does not register before the TTL interval expires, the
MultiVOIP gateway’s registration with the proxy server will expire and the proxy
server will no longer permit call traffic to or from that gateway. Calls in progress
will continue to function even if the gateway becomes de-registered.
seconds)
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Chapter 4: Configuring your VOIP
SIP Server Configuration
The MultiVOIP 210/410/810-SS models have the additional capability of SIP survivability. The settings for
SIP server mode are detailed below.
Figure 4-15: SIP Server configuration
SIP Server Configuration Parameter Definitions
Field Name
Operating
Mode
Values
Survivability In “Survivability” mode, the MVP-SS unit can function as a SIP server for other
-or- gateways in its network in case that network loses contact with the network’s main SIP
Description
stand-alone server (typically a PBX). When in “Survivability” mode the unit is a backup SIP server.
In “Stand-Alone” mode, the MVP-SS functions as a primary SIP server for other
gateways. In this mode, the MVP-SS operate to technical advantage with ‘smart’ SIP
phones. Such smart SIP phones can choose the SIP server under which they operate
and, consequently, can be controlled by either the SIP-based PBX or by the MVP-SS
Survivability
Status
Check
Register,
Options
One of two status-check packets is sent to the main SIP Proxy servers to which the
MVP-SS serves as a backup. This packet determines whether the MVP-SS needs to
take over SIP server functions or stay in its normal backup mode. “Options” and
“Register” are two distinct SIP request “methods.” The Options method solicits
information but does not set up a connection. The Register method conveys
information about a user’s location to the SIP server. The “Register” method may
entail more data overhead than the “Options” method. If both of these methods are
supported by your SIP server, it is OK to use either one. If only one is supported, use
the supported method.
Registrar Options
Allow Y/N
Undefined
If undefined registrations are allowed, then gateways other than those listed in the
Predefined Endpoints list can register with the MVP-SS unit as it functions in its SIP
server mode. If undefined registrations are not allowed, then incoming registrations
will be allowed if they originate from endpoints at accepted domains or IP addresses.
Registrations
Accept
any/specific Defines if registrations to the MVP-SS SIP server will be accepted from any domain or
Registrations domains
for:
only from specified domains. Multiple domains can be listed, separated by semicolons.
The “any domains” option is intended for private networks not accessible via Internet.
Domain
Names
name
Endpoints (separated by semicolon) from which the MVP-SS will accept registrations.
Accept
Registrations -or-
for:
n.n.n.n
Determines whether registrations to the MVP-SS SIP server will be accepted from any
IP address or only from specified IP addresses. Multiple IP addresses can be listed
(separated by semicolon). The “any IP addresses” option is intended for private
networks not accessible via Internet or PSTN.
any IP
addresses
IP
n.n.n.n
List of IP addresses (separated by semicolon) of endpoints from which the MVP-SS
will accept registrations.
Addresses
Re-
in seconds; The time after which the UserAgent Client is supposed to register with the proxy
Registration
Time
(default is
3600)
server. Expiration of the registration means that the gateway has lost contact with the
main SIP server and that the MVP-SS unit will enter ‘survivability’ mode. In
survivability mode, the MVP-SS unit will complete calls acting as a backup to the main
SIP server. Normally, the MVP-SS will initiate re-registration before the interval lapses.
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Chapter 4: Configuring your VOIP
SIP Server: Predefined Endpoint Parameters.
In this screen you will specify the VOIP gateways that will depend on the MVP-SS unit either as their
primary SIP server (if the MVP-SS is used in “Stand-Alone” mode, as set in the SIP Server | Configuration
screen) or as their backup SIP server (if the MVP-SS is used in “Survivability” mode, as set in the SIP
Server |Configuration screen).
The main screen for Predefined Endpoints is a list. If you click on function buttons to Add or Edit entries
in this list of endpoints, a secondary screen will appear and allow you to add new endpoints or edit
existing endpoint entries. When your work with the list is complete, click Save.
Figure 4-16: Endpoint parameters
SIP Server Predefined Endpoints Parameter Definitions
Field Name
Values
Description
Endpoint Name
name
Identifier for gateway within SIP VOIP system. Max. length is 33 characters.
Password
password
This password is for authentication of gateway to SIP server.
Registration Type
Static,
Dynamic
Static registrations are fixed and the contact information for them is configured
by the user and not subject to removal from the registration list due to timeouts.
Dynamic registrations are registered from an external endpoint with the contact
information. Dynamic entries must re-register before the re-registration interval
expires else they will be removed from the list. Endpoints removed from this list
can neither make nor receive calls.
Re-Registration
Interval
integer
The time after which the MultiVOIP UserAgent Client is supposed to register with
the proxy server.
values; in
seconds;
default is
3600
Expiration of the registration interval means that the gateway has lost contact
with the main SIP server and that the MVP-SS unit will enter its ‘survivability’
mode. In survivability mode, the MVP-SS unit will complete calls acting as a
backup to the main SIP server. Normally, however, the MVP-SS will initiate re-
registration with some small margin of time before the interval lapses.
Contact Information
Address
n.n.n.n
The IP address at which this endpoint can be reached.
Digital time slot on which SIP calls will be made. Default is 5060
See “Re-Registration Interval” entry above.
Port
0 – 64000
Re-Registration Time
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Chapter 4: Configuring your VOIP
SPP
Single Port Protocol was developed by Multi-Tech to allow for dynamic IP addressing when it is set to
Registrar/Client mode. The other choice, Direct mode, has IP addresses assigned to the gateways. The
table below describes all fields in the general SPP Call Signaling screen. The –SS models do not support
SPP.
Figure 4-17: SPP call signaling
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Chapter 4: Configuring your VOIP
SPP Call Signaling Parameter Definitions
Description
Field Name
Values
Mode
Direct,
Client, or
Registrar
In direct mode, all VOIP gateways have static IP addresses assigned to them.
In registrar/client mode, one VOIP gateway serves as registrar and all other
gateways, being its clients, point to that registrar. The registrar assigns IP
addresses dynamically.
General Options
Port
port
The UDP port on which data transmission will occur. Each client VOIP has its
own port. If two client VOIPs are both behind the same firewall, then they must
have different ports assigned to them.
If there are two clients and each is behind a different firewall, then the clients
could have different port numbers or the same port number.
(Default port number = 10000.)
Re-transmission
50 -
5000ms
If packets are lost (as indicated by absence of an acknowledgment) then the
endpoint will retransmit the lost packets after this designated time duration has
elapsed. (Default value = 2000 milliseconds.)
Max Re-
transmission
0 - 20
Number of times the VOIP will re-transmit a lost packet (if no acknowledgment
has been received). (Default value = 3)
Client Option fields are active only in registrar/client mode and only for client VOIP
units.
This is the preferred SPP registrar gateway for controlling the traffic of the current
VOIP.
Client Options
Primary Registrar --
Alternate
Registrar 1 and 2
--
A first and a second alternate SPP Registrar gateway can be specified for use by
the current VOIP for situations where the Primary Registrar gateway is busy or
otherwise unavailable.
Registrar IP
Address
n.n.n.n
This is the IP address of the registrar VOIP to which this client is assigned.
(Default value = 0.0.0.0; effectively, there is no useful default value.)
Registrar Port
10000 or
other
This is the port number of the registrar VOIP to which this client is assigned.
(Default port number = 10000.)
Polling Interval
integer
60 - 300
The interval between the VOIP gateway’s successive attempts to connect to and
be governed by a higher level SPP registrar gateway. The Primary Registrar is
the highest level registrar gateway. Alternate Registrar 1 is second; Alternate
Registrar 2 is the lowest order SPP registrar gateway.
Registrar Option fields are active only in registrar/client mode and only for
registrar VOIP units.
Registrar Options
Keep Alive
30 – 300
Time-out duration before a registrar will un-register a client that does not send its
“I’m here” signal. Client normally sends its “I’m here” signal every 20 seconds.
Timeout default = 60 seconds.
(seconds)
Proxy/NAT Device Parameters
Behind
Proxy/NAT
device
Y/N
Enables MultiVOIP (running in SPP Registrar mode) to operate ‘behind’ a
proxy/NAT device (NAT = Network Address Translation).
Proxy/NAT
n.n.n.n
The public IP address of the proxy/NAT device which the MultiVOIP is behind.
Device
Parameters –
Public IP Address
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Chapter 4: Configuring your VOIP
SNMP
If you intend to manage your MultiVOIP remotely using the MultiVoipManager software, you will need to set the
Simple Network Management Protocol parameters. To make the MultiVOIP controllable by a remote PC running
the MultiVoipManager software, check the “Enable SNMP Agent” box on the SNMP Parameters screen.
The –SS and –FX series MultiVOIPs only have limited SNMP functionality available. If this is something you wish
to use on those models, please contact Multi-Tech Support for assistance.
Figure 4-18: SNMP parameters screen
The SNMP Parameter fields are described in the table below.
SNMP Parameter Definitions
Field Name
Enable SNMP
Agent
Values
Y/N
Description
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
n.n.n.n
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.
Port Number
162
The default port number of the SNMP manager receiving the traps is the
standard port 162.
Community
Name 1
Length = 19
First community grouping.
characters (max.)
Case sensitive.
Read-Only,
Read/Write
Length = 19
characters (max.)
Case sensitive.
Read-Only,
Permissions
If this community needs to change MultiVOIP settings, select Read/Write.
Otherwise, select Read-Only to view settings.
Second community grouping
Community
Name 2
Permissions
If this community needs to change MultiVOIP settings, select Read/Write.
Otherwise, select Read-Only to view settings.
Read/Write
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Chapter 4: Configuring your VOIP
Regional
The Regional Parameters are used to set the phone signaling tones and cadences. For the country selected, the
standard set of frequency pairs will be listed for dial tone, busy tone, ‘unobtainable’ tone (fast busy or trunk busy),
ring tone, and other, more specialized tones. If you need settings that are not available, the Custom selection will
let you set the tones to what is necessary. The Regional Parameters fields are described in the table below.
Figure 4-19: Regional parameters
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Chapter 4: Configuring your VOIP
“Regional Parameter” Definitions
Description
Field Name
Country/Region
Values
USA,
Japan,
UK,
Name of a country or region that uses a certain set of tone pairs for dial tone, ring
tone, busy tone, unobtainable tone (fast busy tone), survivability tone (tone
heard briefly, 2 seconds, after going off hook denoting survivable mode of VOIP
unit), re-order tone (a tone pattern indicating the need for the user to hang up the
phone), and intercept tone (a tone that warns an a party that has gone off hook but
has not begun dialing, within a prescribed time, that an automatic emergency or
attendant number will be called; the automatic call can be used to direct an
attendant’s attention to a disabled or distressed caller, allowing an appropriate
response to be made).
Custom
In some cases, the tone-pair scheme denoted by a country name may also be used
outside of that country. The “Custom” option (button) assures that any tone-pairing
scheme worldwide can be accommodated.
Note 1: Intercept tone is applicable only when the FXS telephony interface has
been chosen in the Interface screen and when the AutoCall / OffHook Alert field is
set to OffHook Alert in the Voice/Fax Parameters screen. The time allowed for
dialing before the automatic calling process begins is set in the OffHook Alert Timer
field of the Voice/Fax Parameters screen.
Note 2: “Survivability” tone indicates a special type of call-routing redundancy &
applies to MultiVantage VOIP units only
Advisory screen
This message screen appears whenever the
Country field is changed. It informs the operator
that, upon change of the Country field value, all
User Defined Tones will be deleted.
Standard Tones fields
Type column
dial tone,
ring tone,
busy tone,
Type of telephony tone-pair for which frequency, gain, and cadence are
being presented.
unobtainable tone
(fast busy),
survivability tone,
re-order tone
Frequency 1
Frequency 2
Gain 1
freq. in Hertz
freq. in Hertz
gain in dB
+3dB to –31dB
and “mute” setting
Lower frequency of pair.
Higher frequency of pair.
Amplification factor of lower frequency of pair.
This applies to the dial, ring, busy and ‘unobtainable’ tones that the
MultiVOIP outputs as audio to the FXS, FXS, or E&M port.
Default: -16dB
Gain 2
gain in dB
+3dB to –31dB
and “mute” setting
Amplification factor of higher frequency of pair.
This applies to the dial, ring, busy, and ‘unobtainable’ (fast busy) tones
that the MultiVOIP outputs as audio to the FXS, FXO, or E&M port.
Default: -16dB
Cadence
(ms) On/Off
n/n/n/n
four integer time
values in
milliseconds; zero
value for dial-tone
indicates continuous
tone
On/off pattern of tone durations used to denote phone ringing, phone
busy, connection unobtainable (fast busy), dial tone (“0” indicates
continuous tone), survivability, and re-order. Default values differ for
different countries/regions. Although most cadences have only two parts
(an “on” duration and an “off” duration), some telephony cadences have
four parts. Most cadences, then, are expressed as two iterations of a
two-part sequence. Although this is redundant, it is necessary to allow
for expression of 4-part cadences.
Custom (button)
--
Click on the “Custom” button to bring up the Custom Tone Pair
Settings screen. (The “Custom” button is active only when “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.
Table is continued on next page…
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Chapter 4: Configuring your VOIP
“Regional Parameter” Definitions (continued)
Field Name
Values
Description
Country Selection for
Built-In Modem
(not applicable to
MVP210)
country name
MultiVOIP units operating with the X.06 software release (and above)
include a built-in modem. The administrator can dial into this modem to
configure the MultiVOIP unit remotely. The country name values in this
field set telephony parameters that allow the modem to work in the listed
country. This value may be different than the Country/Region value. For
example, a user may need to choose “Europe” as the Country/Region
value but “Denmark” as the Country-Selection-for-Built-In-Modem value.
User Defined Tones fields
Type column
alphanumeric
name
Name of supervisory tone pair. Cannot be same as name of any
standard tone pair.
Frequency 1
Frequency 2
Gain 1
Freq. in Hertz
Freq. in Hertz
+3dB to –31dB
and “mute”
setting
Lower frequency of pair.
Higher frequency of pair.
Amplification factor of lower frequency of pair.
This applies to any supervisory tones that the MultiVOIP outputs as
audio to the FXS, FXS, or E&M port. Default: “Mute”
Gain 2
+3dB to –31dB
and “mute”
setting
Amplification factor of higher frequency of pair.
This applies to any supervisory tones that the MultiVOIP outputs as
audio to the FXS, FXO, or E&M port. Default: “Mute”
Cadence
(ms) On/Off
n/n/n/n
four integer time
values in
milliseconds;
(zero value
indicates
On/off pattern of tone durations used to denote supervisory tones
specified by user. Supervisory tones relate to answering and
disconnection of calls. Although most cadences have only two parts (an
“on” duration and an “off” duration), some telephony cadences have four
parts. Most cadences, then, are expressed as two iterations of a two-
part sequence. Although this is redundant, it is necessary to allow for
expression of 4-part cadences.
continuous tone)
Setting 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 or “re-order” tones or “survivability” tones 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.
Custom Tone-Pair Settings Definitions
Field Name
Values
Description
Tone Pair
dial tone, busy tone
Identifies the type of telephony signaling tone for which frequencies are
ring tone, ‘unobtainable’ being specified.
tone, survivability tone,
re-order tone
Tone Pair Values
About Defaults: US telephony values are used as defaults on this screen.
Frequency 1 Frequency in Hertz
Frequency of lower tone of pair.
This outbound tone pair enters the MultiVOIP at the input port.
Frequency 2 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 1
+3dB to –31dB
and “mute” setting
Gain 2
+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
Cadence 1
integer time value in
On/off pattern of tone durations used to denote phone ringing, phone busy,
milliseconds; zero value dial tone (“0” indicates continuous tone) survivability and re-order.
for dial-tone indicates
continuous tone
Cadence 1 is duration of first period of tone being “on” in the cadence of
the telephony signal.
Cadence 2
Cadence 3
Cadence 4
duration in milliseconds
duration in milliseconds
duration in milliseconds
Cadence 2 is duration of first “off” period in signaling cadence.
Cadence 3 is duration of second “on” period in signaling cadence.
Cadence 4 is duration of second “off” period in the signaling cadence.
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Chapter 4: Configuring your VOIP
SMTP
Setting the 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.)
Email Address for VOIP (for email call log reporting)
This is needed only if log reports of VOIP call traffic are to be sent by email.
Ask Mail Server administrator to set up email account (with password) for the MultiVOIP unit itself. Be sure to
give a unique identifier to each individual MultiVOIP unit. Get the IP address of the mail server computer, as well.
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.
The SMTP Parameters screen is shown below:
Figure 4-20: SMTP parameters
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“SMTP Parameters” Definitions
Description
Field Name
Values
Enable SMTP
Y/N
In order to send log reports by email, this box must be checked. However,
to enable SMTP functionality, you must also select “SMTP” in the Logs
screen.
Requires
Authentication
Y/N
If this checkbox is checked, the MultiVOIP will send Authentication
information to the SMTP server. The authentication information indicates
whether or not the email sender has permission to use the SMTP server.
This is the User Name for the MultiVOIP unit’s email account.
Login password for MultiVOIP unit’s email account.
Login Name
Password
Mail Server IP
Address
alpha-numeric
alpha-numeric
n.n.n.n
This is the mail server’s IP address. This mail server must be accessible on
the IP network to which the MultiVOIP is connected.
Port Number
Mail Type
Subject
25
25 is a standard port number for SMTP.
Mail type in which log reports will be sent.
User specified. Subject line that will appear for all emailed log reports for
this MultiVOIP unit.
text or html
text
Reply-To Address email address
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).
Email address where VOIP administrator will receive log reports.
Criteria for sending log summary by email. 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, whichever comes first.
This is the number of log records that must accumulate to trigger the
sending of a log-summary email.
Recipient Address email address
Mail Criteria
Number of Records integer
Number of Days
integer
This is the number of days that must pass before triggering the sending of a
log-summary email.
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Chapter 4: Configuring your VOIP
The SMTP Parameters dialog box has a secondary dialog box, accessed by the Select Fields button, that allows
you to customize email logging. 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.
Start Date,
Time
Date and time the phone call began.
Channel
Number
Duration
Data channel carrying call.
Call Mode
Voice or fax.
Length of call.
Packets
Received
Bytes
Received
Coder
Total packets received in call.
Total bytes received in call.
Packets Sent Total packets sent in call.
Bytes Sent Total bytes sent in call.
Packets Lost Packets lost in call.
Voice Coder /Compression Rate used
for call will be listed in log.
When selected, the phonebook prefix
matched in processing the call will be
listed in log.
Prefix
Matched
Outbound
Digits
Received
The DTMF dialing digits received by this
Call Type
Indicates the Call Signaling protocol
used for the call (H.323, SIP, or SPP).
gateway from the remote gateway
presuming that DTMF is set to "Out of
Band."
Call Status
Successful or unsuccessful.
DTMF
Capability
Indicates whether the DTMF dialing
digits are carried "Inband" or "Out of
Band." The corresponding field values
differ for the 3 different VOIP
Call Direction Indicates call’s originating party.
Server
Details
The IP address of the traffic control
server (if any) being used (whether an
H.323 gatekeeper, a SIP proxy, or an
SPP registrar gateway) will be displayed
here if the call is handled through that
server.
protocols.
For H.323, this field can display "Out
of Band" or "Inband". For SIP it can
display either "Out of Band RFC2833"
or "Out of Band SIP INFO" to indicate
the out-of-band condition or "Inband"
to indicate the in-band condition. For
SPP it can display "Out of Band
RFC2833" or "Inband".
Disconnect
Reason
Indicates whether the call was
disconnected simply because the
desired conversation was done or some
other irregular cause occasioned
disconnection (e.g., a technical error or
failure). Values are "Normal" and
"Local" disconnection.
Outbound
Digits Sent
The dialing digits sent by this gateway
to the remote gateway presuming that
DTMF is set to "Out of Band."
From Details
To Details
Gateway
Number
Originating gateway
Gateway
Name
Completing or answering gateway
IP Address
IP address where call originated.
IP Address
IP address where call was completed
or answered.
Descript
Options
Identifier of site where call originated.
Descript
Options
Identifier of site where call was
completed or answered.
When selected, log will not use Silence
Compression and Forward Error
Correction by party answering call.
When selected, log will not Silence
Compression and Forward Error
Correction by call originator.
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Chapter 4: Configuring your VOIP
RADIUS
In general, RADIUS is concerned with authentication, authorization, and accounting. The MultiVOIP supports the
accounting and authentication functions. The accounting function is well suited for billing of VOIP telephony
services. In the Select Attributes secondary screen (accessed by clicking on Select Attributes button), the VOIP
administrator can select the parameters to be tallied by the RADIUS server.
Figure 4-21: RADIUS settings
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Chapter 4: Configuring your VOIP
The fields of the RADIUS screen are described in the table below.
RADIUS Screen Field Definitions
Field Name
Values
Description
Enable Accounting
Y/N
When checked, the MultiVOIP will access the accounting functionality of the
RADIUS server.
Server Address
Accounting Port
n.n.n.n
IP address of the RADIUS server that handles accounting (billing) for the
current MultiVOIP unit.
1 - 65535
TDM time slot at which RADIUS accounting information will be transmitted
and received.
Retransmission
Interval
If the MultiVOIP sends out a packet to the RADIUS server and doesn't
receive a response in the retransmit interval, it will retransmit that packet
again and wait the retransmit interval again for a response. How many times
it does this is determined by the setting in the Number of Retransmissions
field.
Number of
Retransmissions
0 - 255
Shared Secret
alpha-numeric
--
Client encryption key for the current VOIP unit.
Select Attributes
(button)
Gives access to RADIUS Attributes screen. On Attributes screen, one can
specify the parameters to be tallied by the RADIUS server for accounting
(usually billing) purposes.
The RADIUS dialog box has a secondary dialog box, RADIUS Attributes, that allows you to customize
accounting information sent to the RADIUS server by the MultiVOIP. The MultiVOIP software logs data about
many aspects of the call traffic going through the MultiVOIP. The RADIUS Attributes screen lets you pick which
aspects will be included in the accounting reports sent to the RADIUS server.
“RADIUS Attributes” Definitions
Field
Description
Field
Description
Select All
Log report to include all fields
shown.
Start Date, Time
Date and time the phone call began.
Channel
Data channel carrying call.
Call Mode
Voice or fax.
Number
Duration
Packets Sent
Bytes Sent
Length of call.
Total packets sent in call.
Total bytes sent in call.
Packets Received
Bytes Received
Coder
Total packets received in call.
Total bytes received in call.
Voice Coder /Compression Rate used for
call will be listed in log.
Packets Lost
Packets lost in call.
Prefix Matched
Call Status
When selected, the phonebook prefix
matched in processing the call will be listed
in log.
Outbound
Digits Sent
DTMF digits received by this
gateway from remote gateway
(if that DTMF set to "Out of
Band").
Successful or unsuccessful.
Server Details The IP address of the traffic control server being used will be displayed here if the call is handled
through that server. The Options field refers to non-mandatory server features that might be
activated. For example, with H.323, various H.323 Version 4 options might be listed.
From Details
To Details
Gateway
Number
Originating gateway
Gateway
Name
Completing or answering gateway
IP Address
Descript
Options
IP address where call originated.
Identifier of where call originated.
When selected, log will not use
Silence Compression and Forward
Error Correction by call originator.
IP Address IP address where call was completed/answered.
Descript
Options
Identifier of where call was completed/answered.
When selected, log will not use Silence
Compression and Forward Error Correction by
party answering call.
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Chapter 4: Configuring your VOIP
Logs/Traces
The Logs/Traces 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:
•
•
•
in the MultiVOIP program (interface),
via email (SMTP), or
at the MultiVoipManager remote VOIP system management program (SNMP).
Figure 4-22: Logs and Filters screens
If you enable console messages, you can customize the types of messages to be included/excluded in log reports
by clicking on the Filters button and using the Console Messages Filter Settings screen. If you use the logging
function, select the logging option that applies to your VOIP system design. If you intend to use a SysLog Server
program for logging, click in that Enable check box. The common SysLog logical port number is 514. If you intend
to use the MultiVOIP web browser interface for configuration and control of MultiVOIP units, be aware that the
web browser interface does not support logs directly. However, when the web browser interface is used, log files
can still be sent to the VOIP administrator via email (which requires using the SMTP logging option).
“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 IT support
personnel.
Filters (button)
Click to access secondary screen on where console messages can be
included/excluded by category and on a per-channel basis.
Turn Off Logs
Logs Buttons
GUI
Y/N
Check to disable log-reporting function.
Only one of these three log reporting methods, GUI, SMTP, or SNMP, may be chosen.
User must view logs at the MultiVOIP configuration program.
Log messages will be delivered to the MultiVoipManager application program.
Log messages will be sent to user-specified email address.
•
•
•
SNMP
SMTP
SysLog Server
Enable
Y/N
This box must be checked if logging is to be done in conjunction with a SysLog Server
program.
IP Address
Port
n.n.n.n IP address of computer, in VOIP network, on which SysLog Server program is running.
514 Logical port for SysLog Server. 514 is commonly used.
Online Statistics
Updation Interval
integer Set the interval (in seconds) at which logging information will be updated.
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Chapter 4: Configuring your VOIP
NAT Traversal
Setting the NAT Traversal parameters. NAT (Network Address Translation) parameters are applicable only when
the MultiVOIP is operating in SIP mode. STUN (Simple Traversal of UDP through NATs (Network Address
Translation)) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. This is not
available on the –SS models.
Figure 4-23: NAT Traversal
Descriptions for NAT Traversal screen fields are presented in the table below.
NAT Traversal Definitions
Field Name
Values
Description
Enable (STUN)
Y/N
Enables STUN client functionality in the MultiVOIP.
STUN (Simple Traversal of UDP through NATs (Network Address Translation))
is a protocol that allows a server to assist client gateways behind a NAT firewall
or router with their packet routing.
Name/IP (Server)
n.n.n.n
IP address of the STUN server.
Port (Server;
NAT/STUN)
port;
default=
3478
The data port (TDM time slot) at which STUN info will be transmitted and
received.
Keep Alive (Timers;
NAT/STUN)
60 – 3600
(seconds)
The interval at which the STUN client sends indicator (“Keep Alive”) packets to
the STUN server to determine whether or not the STUN server is available.
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Chapter 4: Configuring your VOIP
Supplementary Services
Supplementary Services features derive from the H.450 standard, which brings to the VOIP telephony
functionality once only available with PSTN or PBX telephony. Even though the H.450 standard refers only to
H.323, Supplementary Services are still applicable to the SIP and SPP VOIP protocols.
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 used by a
programmable phone keypad sequence (for example, #7).
Call Hold. Call Hold allows one party to maintain an idle (non-talking) connection with another party while
receiving another call (Call Waiting), while initiating another call (Call Transfer), or while performing some other
call management function. Feature is used by a programmable phone keypad sequence (for example, #7).
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. Feature is used by a
programmable phone keypad sequence (for example, #7).
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 VOIP identifying
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.
Figure 4-24: Supplementary Services
The Supplementary Services fields are described in the tables below.
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Chapter 4: Configuring your VOIP
Supplementary Services Parameter Definitions
Field Name
Values
Description
Select Channel
1-2 (210);
1-4 (410);
1-8 (810)
The channel to be configured is selected here.
Call Transfer Enable Y/N
Select to enable the Call Transfer function in the VOIP unit.
This is a “blind” transfer and the sequence of events is as follows:
Callers A and B are having a conversation.
Caller A wants to put B into contact with C.
Caller A dials call transfer sequence.
Caller A hears dial tone and dials number for caller C.
Caller A gets disconnected while Caller B gets connected to caller C.
A brief musical jingle is played for the caller on hold.
Transfer Sequence
Any phone keypad
character
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 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*#.
Call Hold Enable
Hold Sequence
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.
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 Enable
Retrieve Sequence
Y/N
Select to enable Call Waiting function in VOIP unit.
Phone keypad
characters, two
characters in length
The numbers and/or symbols that the caller must press on the phone
keypad to initiate retrieval of a waiting call.
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.
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 interface (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 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.)
Table is continued on next page…
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Chapter 4: Configuring your VOIP
Supplementary Services Definitions (continued)
Field Name
Calling Party,
Allowed Name
Type (CNI)
Description
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.
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.
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.
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.
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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Chapter 4: Configuring your VOIP
Save Settings
Save & Reboot
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, then Save & Reboot.
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.
Connection
Settings
This is also accessible from the Start menu in the MultiVOIP software folder.
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 manager of your Windows operating system. If
COM1 is not available, you must change the COM port setting to a COM port that you have confirmed as
being available on your PC.
Figure 4-25: COM port setup
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Chapter 4: Configuring your VOIP
Troubleshooting Software Issues
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. If the message
displayed is “MultiVOIP Not Found!” please try the resolutions below.
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.
Figure 4-26: Error pop-up
To change the COM port setting, use the COM Port Setup dialog box, by going to the Connection pull-
down menu and choosing “Settings” or use the left side control panel. 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).
Fixing a Cabling Problem
If the MultiVOIP cannot be located by the computer, three error messages will appear (saying “Multi-VOIP
Not Found”, “Phone Database Not Read” and “Password Phone Database Not Read).
Figure 4-27: Cabling errors
In this case, the MultiVOIP is simply disconnected from the network. For instructions on MultiVOIP cable
connections, see the Cabling section of Chapter 3.
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Chapter 5 – Phone Book Configuration
Introduction
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. The
phone numbers are not literally “listed” individually, but are, instead, described by rule.
Identify Remote VOIP Site to Call
When you’re done installing the MultiVOIP, you’ll want to confirm that it is configured and operating properly. To
do so, it’s good to have another VOIP that you can call for testing purposes. You’ll want to confirm end-to-end
connectivity. You’ll need IP and telephone information about that remote site.
If this is the very first VOIP in the system, you’ll want to coordinate the installation of this MultiVOIP with an
installation of another unit at a remote site.
Identify VOIP Protocol to be Used
Will you use H.323, SIP, or SPP? Each has advantages and disadvantages. Although it is possible to mix
protocols in a single VOIP system, it is highly desirable to use the same VOIP protocol for all VOIP units in the
system. SPP is a non-standard protocol developed by Multi-Tech. SPP is not compatible with the “Proprietary”
protocol used in Multi-Tech’s earlier generation of VOIP gateways. The –SS series of MultiVOIPs only support the
SIP protocol. The –FX models do not support H.323.
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Chapter 5: Phonebook Configuration
Phonebook Starter Configuration
This section will walk you through the phone book setup with examples that will aid in entering the correct
numbers needed to have the MultiVOIP working correctly. To do this part of the setup, you need access to
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. Once this is
accomplished, you can easily add other VOIP sites to the network.
Outbound Phonebook
1. Open the MultiVOIP program. (Start | MultiVOIP xxx | Configuration)
2. Go to Phone Book | Outbound Phonebook | Add Entry.
3. On a sheet of paper, write down the calling code of the remote VOIP (area code, country code, city code,
etc.) that you’ll be calling.
Follow the example that best fits your situation:
North America,
Long-Distance Example
Euro, National Call Example
Euro, International 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, 31; city 010) to set up one VOIP
another in Birmingham (area
0121).
Technician in Rotterdam (country
there, another in Bordeaux
(country 33; area 05).
Answer: Write down 312.
Answer: write down 0121.
Answer: write down 3305.
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
Euro, International Call
Example
Seattle/Chicago system.
London/Birmingham system.
Rotterdam/Bordeaux system.
Rotterdam VOIP works with PBX
where “9” is used for all out-of-
building calls. “0” must precede
all international calls.
Seattle VOIP works with PBX
that uses “8” for all VOIP calls.
“1” must immediately precede
area code of dialed number.
London VOIP works with PBX
that uses “9” for all out-of-
building calls whether by VOIP or
by PSTN. “0” must immediately
precede area code of dialed
number.
Answer: write down 90.
Answer: write down 81.
Answer: write down 90.
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Chapter 5: Phonebook Configuration
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
Euro, International Call
Example
Rotterdam/Bordeaux system.
London/Birmingham system.
Answer: enter 81312 as
Leading zero of Birmingham area
code is dropped when combined
Answer: enter 903305 as
Destination Pattern in
Outbound Phonebook of
Rotterdam VOIP.
Destination Pat-tern in
Outbound Phone-book of with national-dialing access
Seattle VOIP.
code. (Such practices vary by
country.)
Answer: enter 90121 as
Destination Pattern in Outbound
Phonebook of London VOIP.
Not 900121.
6. In the “Remove Prefix” field, enter the initial PBX access digit (“8” or “9”).
North America,
Long-Distance Example
Seattle/Chicago system.
Euro, National Call Example
Euro, International Call
Example
Rotterdam/Bordeaux system.
London/Birmingham system.
Answer: enter 8 in “Remove Prefix”
field of Seattle Outbound
Phonebook.
Answer: enter 9 in “Remove Prefix”
field of London Outbound
Phonebook.
Answer: enter 9 in “Remove Prefix”
field of Outbound Phonebook for
Rotterdam VOIP.
Note: Some PBXs will not ‘hand off’ the “8” or “9” to the VOIP. But for those PBX units that do, it’s important to enter
the “8” or “9” in the “Remove Prefix” field in the Outbound Phonebook. This precludes the problem of having to make
two inbound phonebook entries at remote VOIPs, one to account for situations where “8” is used as the PBX access
digit and another for when “9” is used.
7. In the “Protocol Type” field group, select the VOIP protocol that you will use (H.323, SIP, or SPP). Use the
appropriate screen under Configuration | Call Signaling to configure the VOIP protocol in detail.
8. Click OK to exit from the Add/Edit Outbound Phonebook screen.
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Chapter 5: Phonebook Configuration
Inbound Phonebook
1. Open the MultiVOIP program. (Start | MultiVOIP xxx | Configuration)
2. Go to Phone Book | Inbound Phonebook | Add Entry.
3. In the “Remove Prefix” field, enter your local calling code (area code, country code, city code, etc.)
preceded by any other “access digits” that are required to reach your local site from the remote VOIP
location (think of it as though the call were being made through the PSTN – even though it will not be).
North America,
Long-Distance Example
Euro, National Call Example
Euro, International Call
Example
Seattle/Chicago system.
London/Birmingham system.
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.
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 (Seattle)
VOIP.
Answer: 0207 is prefix to be
removed by local (London)
VOIP.
Answer: 03110 is prefix to be
removed by local
(Rotterdam) VOIP.
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
Euro, International Call
Example
Seattle/Chicago system.
London/Birmingham system.
Rotterdam/Bordeaux system.
On Rotterdam PBX, “9” is used to
get an outside line.
On Seattle PBX, “9” is used to get an On London PBX, “9” is used to get
outside line.
an outside line.
Answer: 9 is prefix to be added by
Answer: 9 is prefix to be added by
Answer: 9 is prefix to be added by
local (London) VOIP.
local (Rotterdam) VOIP.
local (Seattle) VOIP.
5. In the “Channel Number” field, enter “Hunting.” A “hunting” value means the VOIP unit will assign the call
to the first available channel. If desired, specific channels can be assigned to specific incoming calls (i.e., to
any set of calls received with a particular incoming dialing pattern).
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. For this, 40 characters are the maximum.
North America,
Long-Distance Example
Euro, National Call Example
Euro, International Call
Example
Seattle/Chicago system.
London/Birmingham system.
Rotterdam/Bordeaux system.
Possible Description:
Free Seattle access, all employees
Possible Description:
Local-rate London access, all
employees
Possible Description:
Local-rate Rotterdam access, all
employees
7. Repeat steps 2-6 for each inbound phonebook entry. When all entries are complete, go to step 8.
8. Click OK to exit the inbound phonebook screen.
9. Click on Save Setup. Highlight Save and Reboot. Click OK.
Your starter inbound phonebook configuration is complete.
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Chapter 5: Phonebook Configuration
Phone Book Descriptions
Outbound Phone Book/List Entries
Fields in the “Details” section will differ depending on the protocol (H.323, SIP, or SPP) of the selected list entry to
which the details pertain.
Figure 5-1: Outbound Phone Book
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Chapter 5: Phonebook Configuration
Add/Edit Outbound Phone Book
Figure 5-2: Add/Edit screen
Enter Outbound Phone Book data for your MultiVOIP unit. Note that the Advanced button gives access to the
Alternate IP Routing feature, if needed. Alternate IP Routing can be implemented in a secondary screen (as
described after the primary screen field definitions below). The –SS will only allow SIP settings and the –FX
models will not allow H.323.
The fields of the Add/Edit Outbound Phone Book screen are described in the table below.
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Chapter 5: Phonebook Configuration
Add/Edit Outbound Phone Book: Field Definitions
Field Name
Accept Any
Number
Values
Y/N
Description
When checked, “Any Number” appears as the value in the Destination
Pattern field.
The Any Number feature works differently depending on whether or not an
external routing device is used (Gatekeeper for H323 protocol, Proxy for SIP
protocol, Registrar for SPP protocol).
When no external routing device is used. If Any Number is selected,
calls to phone numbers not matching a listed Destination Pattern will be
directed to the IP Address in the Add/Edit Outbound Phone Book screen.
“Any Number” can be used in addition to one or more Destination Patterns.
When external routing device is used. If Any Number is selected, calls to
phone numbers not matching a listed Destination Pattern will be directed to
the external routing device used (Gatekeeper for H323 protocol, Proxy for
SIP protocol, Registrar for SPP protocol). The IP Address of the external
routing device must be set in the Phone Book Configuration screen.
Destination
Pattern
prefixes,
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 PSTN and carried on Internet or other IP network.
area codes,
exchanges,
line numbers,
extensions
Total Digits
as needed
Number of digits the phone user must dial to reach specified destination. This
field not used in North America
Remove Prefix
Add Prefix
dialed digits
dialed digits
n.n.n.n
Portion of dialed number to be removed before completing call to destination.
Digits to be added before completing call to destination.
IP Address
The IP address to which the call will be directed if it begins with the
destination pattern given.
Description
alpha-numeric
Describes the facility or geographical location at which the call will be
completed.
Protocol Type
SIP or H.323
or SPP
Indicates protocol to be used in outbound transmission. Single Port Protocol
(SPP) is a non-standard protocol designed by Multi-Tech. The –SS models
only support SIP and the –FX models do not support H.323.
The –SS and –FX models do not support H.323
H.323 fields
Use Gatekeeper
Y/N
Indicates whether or not gatekeeper is used.
Gateway H.323
ID
alpha-numeric
The H.323 ID assigned to the destination MultiVOIP. Only valid if “Use
Gatekeeper” is enabled for this entry.
Gateway Prefix
numeric
1720
This number becomes registered with the GateKeeper. Call requests sent to
the gatekeeper and preceded by this prefix will be routed to the VOIP
gateway.
H.323 Port
Number
This parameter pertains to Q.931, which is the H.323 call signaling protocol
for setup and termination of calls (aka ITU-T Recommendation I.451). H.323
employs only one “well-known” port (1720) for Q.931 signaling. If Q.931
message-oriented signaling protocol is used, 1720 must be chosen as the
H.323 Port Number.
Table is continued on next page…
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Chapter 5: Phonebook Configuration
Add/Edit Outbound Phone Book: Field Definitions (continued)
Field Name
Values
SIP Fields
Y/N
Description
Use Proxy
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.
SIP Port
Number
5060 or other
The SIP Port Number is a UDP logical port number. The VOIP will “listen”
*See RFC 3087 (“Control of for SIP messages at this logical port. If SIP is used, 5060 is the default,
Service Context using SIP
Request-URI,” by the
Network Working Group).
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 URL
sip.userphone@hostserver, Looking similar to an email address, a SIP URL identifies a user's
where “userphone” is the
telephone number and
“hostserver” is the domain
name or an address on the
network
address.
In SIP communications, each caller or callee is identified by a SIP URL:
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.
SPP Fields
The –SS series of MultiVOIPs do not support SPP
Use
Registrar
Y/N
Select this checkbox to use registrar when VOIP system is operating in
the “Registrar/Client” SPP mode. In this mode, one VOIP (the registrar, as
set in Phonebook Configuration screen) has a static IP address and all
other VOIPs (clients) point to the registrar’s IP address as functionally
their own. However, if your VOIP system overall is operating in
“Registrar/Client” mode but you want to make an exception and use Direct
mode for the destination pattern of this particular Add/Edit Phonebook
entry, leave this checkbox unselected. Also do not select this if your
overall VOIP system is operating in the Direct SPP mode – in this mode
all VOIPs are peers with unique static IP addresses.
Port Number
numeric
When operating in “Registrar/Client” mode, this is the port by which the
gateway receives all SPP data and control messages from the registrar
gateway. (This ability to receive all data and messages via one port
allows the VOIP to operate behind a firewall with only one port open.)
When operating in “Direct” mode, this is the Port by which peer VOIPs
receive data and messages.
Alternate
Phone
Number
Remote
Device is
[legacy
numeric
Y/N
Phone number associated with alternate IP routing.
When checked, this MultiVOIP can operate with ‘first-generation’
MultiVOIP units in the same IP network. These include MVP-
110/120/200/400/800.
VOIP]
This is not available for the –SS series of MultiVOIPs.
Advanced
button
Gives access to secondary screen where an Alternate IP Route can be specified for backup or
redundancy of signal paths. For SIP & H.323 operation only.
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Chapter 5: Phonebook Configuration
Clicking on the Advanced button brings up the Alternate Routing secondary screen. This feature provides an
alternate path for calls if the primary IP network cannot carry the traffic. Often in cases of failure, call traffic is
temporarily diverted into the PSTN. However, this feature could also be used to divert traffic to a redundant
(backup) unit in case one VOIP unit fails. The user must specify the IP address of the alternate route for each
destination pattern entry in the Outbound Phonebook.
Figure 5-3: Advanced button
Alternate Routing Field Definitions
Description
Alternate destination for outbound data traffic in case of excessive delay in data
transmission.
Field Name
Alternate IP
Address
Values
n.n.n.n
Round Trip
Delay
Default is
300
milliseconds
The Round Trip Delay is the criterion for judging when a data pathway is
considered blocked. When the delay exceeds the threshold specified here, the
data stream will be diverted to the alternate destination specified as the Alternate IP
Address.
The Alternate Routing function facilitates PSTN Failover protection, that is, it allows you to re-route VOIP calls
automatically over the PSTN if the VOIP system fails. The MultiVOIP can be programmed to respond to excessive
delays in the transmission of voice packets, which the MultiVOIP interprets as a failure of the IP network. Upon
detecting an excessive delay in transmission of voice packets (overly high “latency” in the network) the MultiVOIP
diverts the call to another IP address, which itself is connected to the PSTN (for example, via an FXO port on the
self-same MultiVOIP could be connected to the PSTN).
PSTN Failover Feature. The MultiVOIP can be programmed to divert calls to the PSTN temporarily in case the IP
network fails. See Figure 5-4 below for example.
4. Call completed
3. Call diverts to
via PSTN.
PSTN Line
Alt IP address in voip
accessing PSTN line.
FXO
IP
VOIP
VOIP
NETWORK
PBX
FXS
2. IP network fails.
1. Call originates.
Figure 5-4: PSTN failover
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Chapter 5: Phonebook Configuration
Inbound Phone Book/List Entries
The “Details” and “Registration Options” sections will display information based on the setup and protocols
chosen. The “Subscription Options” area is used in conjunction with a Voice Mail Server.
Figure 5-5: Inbound phonebook entries
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Chapter 5: Phonebook Configuration
Enter Inbound Phone Book data for your MultiVOIP. The fields of the Add/Edit Inbound Phone Book screen are
described in the table below.
Add/Edit Inbound Phone Book: Field Definitions
Field Name
Values
Description
Accept Any
Number
Y/N
When checked, “Any Number” appears as the value in the Remove Prefix field.
The Any Number feature of the Inbound Phone Book does not work when an
external routing device is used (Gatekeeper for H.323 protocol, Proxy for SIP
protocol, Registrar for SPP protocol).
When no external routing device is used. If Any Number is selected, calls
received from phone numbers not matching a listed Prefix (shown in the Remove
Prefix column of the Inbound Phone Book) will be admitted into the VOIP on the
channel listed in the Channel Number field. “Any Number” can be used in
addition to one or more Prefixes.
Remove Prefix dialed digits
portion of dialed number to be removed before completing call to destination
(often a local PBX)
Add Prefix
dialed digits
digits to be added before completing call to destination
(often a local PBX)
Channel
Number
Description
channel, or
“Hunting”
--
Channel number to which the call will be assigned as it enters the local telephony
equipment (often a local PBX). “Hunting” directs the call to any available channel.
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.
Forward
Condition
Unconditional,
Busy,
No Response
Unconditional. When selected, all calls received will be forwarded.
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.
Forwarding can be conditioned on both “Busy” and “No Response
Phone number or IP address to which calls will be directed.
Forward
Destination
IP address,
phone number,
port number,
etc
For H.323 calls, the Forward Destination can be either a Phone Number or an IP
Address.
For SIP calls, the Forward Destination can be one of the following:
(a) phone number,
(b) IP address,
(c) IP address: port number,
(d) phone number: IP address: port number,
(e) SIP URL, or
(f) phone #: IP address.
For SPP calls, the Forward Destination can be one of the following:
(a) phone number,
(b) IP address: port, or
(c) phone number: IP address: port.
Ring Count
integer
When “No Response” is condition for forwarding calls, this determines how many
unanswered rings are needed to trigger the forwarding.
Registration
Option
Parameters
In an H.323 VOIP system, gateways can register with the system using one of these identifiers: an
E.164 identifier, a Tech Prefix identifier, or an H.323 ID identifier. This section not available for
the –FX and –SS series models.
In a SIP VOIP system, gateways can register with the SIP Proxy. This is the only area available to
the –SS series.
In an SPP VOIP system, gateways can register with the SPP Registrar VOIP unit.
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Chapter 5: Phonebook Configuration
Authorized User Name and Password for SIP
To enable the Registration Options on the Add/Edit Inbound Phone Book, you have to activate Use SIP
Proxy Option on the Call Signaling, SIP Parameters Screen. Then add the IP address for the Primary
Proxy in the SIP Proxy Parameters. This allows you to add a Username and Password to the Inbound
Phone Book entry. The –SS models will only have a password option available.
This feature is used when the MultiVOIP registers with the proxies that support authorization and need
the username, password and the endpoint name to be unique.
The VOIP sends Register request to Registrar for each entry with its configured Username and
Password. When Authentication is enabled for the endpoint, then the registrar/proxy sends “401
Unauthorized/407 Proxy Authentication Required” response when it receives a REGISTER/INVITE
request. Now, the endpoint has to send the authentication details in the Authorization header. In this
header one of the fields is “username”.
Generally proxies accept requests even if both Endpoint Name and Username are same. But some
proxies expect that the Endpoint Name and Username should be different.
To support these proxies, we have the username and password configuration for every inbound phone
book entry which gets registered with a proxy.
If the username and password are not configured in the inbound phone book, then the registration will
happen with the default username and password that are configured in the SIP Call Signaling Page.
Phone Book Save and Reboot
When your Outbound and Inbound Phonebook entries are completed, click on Save Setup in the sidebar menu to
save your configuration. You can change your configuration at any time as needed for your system.
Remember that the initial MultiVOIP setup must be done locally or via the built-in Remote
Configuration/Command Modem using the MultiVOIP program. After the initial configuration is complete, all of
the MultiVOIP units in the VOIP system can be configured, re-configured, and updated from one location using
the MultiVOIP web interface software program or the MultiVOIP program (in conjunction with the built-in modem).
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Chapter 5: Phonebook Configuration
Phonebook Examples
North America
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.
Notice first the area code situation in those two cities: Minneapolis’s local calling area consists of multiple
adjacent area codes; Baltimore’s local calling area consists of a base area code plus an overlay area code.
Company
VOIP/PBX
5
Baltimore/
SIte
Outstate MD
Overlay
443
NW
Suburbs
St. Paul
& Suburbs
651
763
Mpls
612
Company
VOIP/PBX
SIte
...
5
SW Suburbs
952
Baltimore
410
Figure 5-7: North America example
An outline of the equipment setup in both offices is shown below.
Figure 5-8: Equipment setup example
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Chapter 5: Phonebook Configuration
The screen below shows Outbound Phonebook entries for the VOIP located in the company’s Baltimore facility.
Figure 5-9: Baltimore example
The entries in the Minneapolis VOIP’s Inbound Phonebook match the Outbound Phonebook entries of the
Baltimore VOIP, as shown below.
Figure 5-10: Minneapolis example
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.
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Chapter 5: Phonebook Configuration
The simplest case is a call 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. Similarly, the Inbound Phone Book for the Baltimore VOIP (shown first below) generally matches the
Outbound Phone Book of the Minneapolis VOIP (shown second below).
Figure 5-11: Inbound Baltimore example
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.
The Outbound Phone Book 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”.
Figure 5-12: Outbound Minneapolis example
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Chapter 5: Phonebook Configuration
Europe
The most direct use of the VOIP system is making calls between the offices where the VOIPs are located.
Consider, for example, the Wren Clothing Company. This company has VOIP-equipped offices in London, Paris,
and Amsterdam, each served by its own PBX. VOIP calls between the three offices completely avoid
international long-distance charges. These calls are free. The phonebooks can be set up to allow all Wren
Clothing employees to contact each other using 3-, 4-, or 5-digit numbers, as though they were all in the same
building.
United Kingdom
Wren Clothing Co.
5 Wren Clothing Co.
VOIP/PBX Site
VOIP/PBX Site
5London
Amsterdam
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Paris
Free VOIP Calls
France
Figure 5-13: Free VOIP calls
In another 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. Generally, local calling rates apply only
within a single area code, and, for all calls outside that area code, national rates apply. There are, however,
some European cases where local calling rates extend beyond a single area code. Local rates between Inner
and Outer London are one example of this. It is also possible, in some locations, that calls within an area code
may be national calls - but this is rare.
United Kingdom
Wren Clothing Co.
5Wren Clothing Co.
VOIP/PBX Site
Bluebird Zipper Co.
VOIP/PBX Site
5London
London
Amsterdam
The
Netherlands
Wren Clothing Co.5
VOIP/PBX Site
Calls at London local rates
Paris
Local Calling Area
France
Figure 5-14: Local calling area
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Chapter 5: Phonebook Configuration
This next example will have the following features:
•
•
Employees in all cities will be able to call each other over the VOIP system using 4-digit extensions.
Calls to Outer London and Inner London, greater Amsterdam, and greater Paris will be accessible to
all company offices as local calls.
•
Vendors in Guildford, Lyon, and Rotterdam can be contacted as national calls by all company offices.
France Country Code: 33
Lille
Paris: Area 01
Reims
Rouen
Nantes
Strasbourg
Lyon
Bordeaux
Toulouse
Marseille
Figure 5-15: UK & France codes
The Netherlands
Country Code: 31
050
Groningen
058
Leeuwarden
Texel 0222
Den Helder 0223
038 Zwolle
0299 Purmerend
Beverwijk 0251
Haarlem 023
020 Amsterdam
0294 Weesp
Aalsmeer0297
053
Enschede
070
The Hague
026
Arnhem
010
Rotterdam
0118
Middelburg
040
Eindhoven
043
Maastricht
Figure 5-16: Netherlands codes
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Chapter 5: Phonebook Configuration
An outline of the equipment setup in these three offices is shown below.
Figure 5-17: Setup example
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Chapter 5: Phonebook Configuration
The screen below shows Outbound Phone Book entries for the VOIP located in the company’s London facility.
Figure 5-18: London example outbound
The Inbound Phone Book for the London VOIP is shown below.
Figure 5-19: London example inbound
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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Chapter 5: Phonebook Configuration
The screen below shows Outbound Phone Book entries for the VOIP located in the company’s Paris facility.
Figure 5-20: Paris example outbound
The Inbound Phone Book for the Paris VOIP is shown below.
Figure 5-21: Paris example inbound
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Chapter 5: Phonebook Configuration
The screen below shows Outbound Phone Book entries for the VOIP in the company’s Amsterdam facility.
Figure 5-22: Amsterdam example outbound
The Inbound Phone Book for the Amsterdam VOIP is shown below.
Figure 5-23: Amsterdam example inbound
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Chapter 5: Phonebook Configuration
Variations of Caller ID
The Caller ID feature has dependencies on both the telco central office and the MultiVOIP phone book. See the
diagram series below:
CID Flow
Call is received
here.
Call originates here
at 1:42pm, May 31.
CID
Terminating
VoIP
CID
Generating
VoIP
Central Office
with
standard telephony
Caller ID service
FXO
FXS
IP
Network
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
Clock:
5-31,
1:42pm
phone of:
Display shows:
H.323 or SPP
Melvin Jones
763-555-8794
Protocol
*
CID Number: 763-555-8794
CID Name: Melvin Jones
Time Stamp: Date: 05/31
Time:1:42pm
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
*
Figure 5-24: Caller ID example 1
Figure 5-25: VOIP Caller ID Case #1 – Call, through telco central office with standard CID, enters VOIP system.
CID Flow
Call is received
here.
Call originates here
at 4:19pm, July 10.
CID
Generating
VoIP
CID
Ch1
Terminating
VoIP
Central Office
without
FXO
FXS
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
standard telephony
Caller ID service
Ch3
Ch4
Clock:
7/10, 4:19pm
phone of:
Display shows:
H.323 Protocol
*
Wilda Jameson
763-555-4071
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 7/10
Time: 4:19pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Remove Prefix Add Prefix Forward/Addr
*
Gatekeeper RAS Param
423
748
Figure 5-25: Caller ID example 2
Figure 5-26: VOIP Caller ID Case #2 – Call, through telco central office without standard CID, enters H.323 VOIP
system.
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Chapter 5: Phonebook Configuration
CID Flow
IP
Call is received
here.
Call originates here
at 5:47pm, Sept 27.
Ch1
Generating
Term inating
VoIP
Central Office
VoIP
FXO
FXS
without
Ch2
xx xyy yz zz
z
x
xxy yy zz zz
standard telephony
Caller ID service
J. Q. P u bl ic
J. Q. Pu bl ic
Network
Ch3
Ch4
Clock:
15:26, 5-31
phone of:
Display shows:
SPP Protocol
Henry Brampton
763-555-4077
CID Number: 423
{Channel 2}
Inbound Phone Book
CID Name: Shipping Dept
Remove Prefix Add Prefix Forward/Addr
Time Stamp: Date: 0927
Time: 1747
... if “Description” field in Add/Edit
Inbound Phone Book is used
423
748
Phone Book Configuration
Anoka-Whse-VP3
Gateway Name:
OR
Add/Edit Inbound Phone Book
Use as default entry
CID Number: 423
Remove Prefix: rs
Add Prefix:
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 0927
Time: 1747
Channel Number: Channel 2
Description: Shipping Dept
... if “Description” in Add/Edit
Inbound Phone Book is blank
Figure 5-26: Caller ID example 3
Figure 5-27: VOIP Caller ID Case #3 – Call, through telco central office without standard CID, enters SPP VOIP
system.
CID Flow
Call is received
here.
Call originates here
at 4:51pm, Oct 3.
CID
Generating
VoIP
CID
Ch1 FXS
Terminating
VoIP
401
xxxyyyzzzz
J.Q. Public
FXS
IP
Network
Ch2
xxxyyyzzzz
J.Q. Public
402
403
phone of:
Nigel Thurston
763-555-9401
Ch3
Ch4
Clock:
10/03, 4:51pm
404
Display shows:
H.323 Protocol
*
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 10/03
Time: 4:51pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
Remove Prefix Add Prefix Forward/Addr
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
*
Gatekeeper RAS Param
423
748
Figure 5-27: Caller ID example 4
Figure 5-28: VOIP Caller ID Case #4 – Remote FXS call on H.323 VOIP system.
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Chapter 5: Phonebook Configuration
CID Flow
IP
Call is received
here.
Call originates here
at 6:17pm, Nov 15.
CID
Generating
VoIP
CID
Ch1
Terminating
VoIP
Central Office
DID
FXS
without
Ch2
xxxyyyzzzz
J.Q. Public
xxxyyyzzzz
J.Q. Public
standard telephony
Caller ID service
Network
Ch3
Ch4
Clock:
11/15, 6:17pm
phone of:
Display shows:
H.323 Protocol
*
Edwin Smith
763-743-5873
CID Number: 423
Phone Book Configuration
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 11/15
Time: 6:17pm
Anoka-Whse-VP3
Gateway Name:
Q.931 Parameters
{Channel 2}
Inbound Phone Book
In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Remove Prefix Add Prefix Forward/Addr
*
Gatekeeper RAS Param
423
748
Figure 5-28: Caller ID example 5
Figure 5-29: VOIP Caller ID Case #5 – Call through telco central office without standard CID enters DID channel
in H.323 VOIP system.
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Chapter 6 – Using the Software
Introduction
This chapter will primarily cover the day to day operation and maintenance sections of the MultiVOIP software.
How to update the firmware and software are also covered here should either be needed. This section will mainly
focus on the Statistics section of the configuration software, but there are references to a few of the other sections
as they are used more in the daily operations than in a setup situation.
Software Categories Covered in This Chapter
¾ System Information
¾ Call Progress
¾ Logs
¾ IP Statistics
¾ Link Management
¾ Registered Gateway Details
¾ Servers
o H.323 GateKeepers
o SIP Proxies
o SPP Registrars
¾ Advanced
o Packetization Time
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Chapter 6: Using the Software
System Information screen
This screen presents system information at a glance. It is found under the Configuration section and its primary
use is in troubleshooting. The information presented in figure 6-1 is for reference only and is not meant to be an
exact match of your system.
Figure 6-1: System information
System Information Parameter Definitions
Field Name
Boot Version
Values
nn.nn
alpha-
numeric
Description
Indicates the version of the code that is used at the startup (booting) of the VOIP.
The boot code version is independent of the software version.
Firmware Version
nn.nn.nn
alpha-
Indicates the version of the MultiVOIP firmware.
numeric
Configuration Version nn.nn.
Indicates the version of the MultiVOIP configuration software.
nn.nn
alpha-
numeric
Phone Book Version
IFM Version
nn.nn
alpha-
numeric
Indicates the version of the MultiVOIP phone book being used.
nn
alpha-
numeric
Indicates version of the IFM module, the device that performs the transformation
between telephony signals and IP signals.
Mac Address
Up Time
numeric
Denotes the number assigned as the VOIP unit’s unique Ethernet address.
Indicates how long the VOIP has been running since its last booting.
days:
hours:
mm:ss
Hardware ID
alpha-
Indicates version of the MultiVOIP circuit board assembly being used.
numeric
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Chapter 6: Using the Software
The frequency with which the System Information screen is updated is determined by a setting in the Logs/Traces
screen (which is under the Configuration section).
Figure 6-2: Logs/Traces screen
Statistics Section
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. The following screens are
examples of what can be shown and are followed by detailed descriptions of the categories involved. The model
and signaling used will affect what is available for display.
Call Progress
Figure 6-3: Call progress screen
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Chapter 6: Using the Software
Call Progress Details: Field Definitions
Description
Field Name
Values
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
H/M/S
Voice or FAX
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 Coder
G.723, G.729,
G.711, etc.
The voice coder being used on this call.
IP Call Type
H.323, SIP, or
SPP
Indicates the Call Signaling protocol used for the call (H.323, SIP, or SPP).
The –SS and –FX series only support SIP.
IP Call Direction
incoming,
outgoing
Indicates whether the call in question is an incoming call or an outgoing
call.
Packet Details
Packets Sent
Packets Rcvd
Bytes Sent
integer value
integer value
integer value
integer value
integer value
The number of data packets sent over the IP network in the course of this
call.
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.
From – To Details
Description
Gateway Name
(from)
alphanumeric
string
Identifier for the VOIP gateway that handled the origination of this call.
IP Address (from)
Options
n.n.n.n
SC, FEC
IP address from which the call was received.
Displays VOIP transmission options in use on the current call. These may
include Forward Error Correction or Silence Compression.
Identifier for the VOIP gateway that handled the completion of this call.
Gateway Name (to)
alphanumeric
string
IP Address (to)
Options
n.n.n.n
SC, FEC
IP address to which the call was sent.
Displays VOIP transmission options in use on the current call. These may
include Forward Error Correction or Silence Compression.
DTMF/Other Details
Prefix Matched
specified
dialing digits
Displays the dialed digits that were matched to a phonebook entry.
The digits transmitted by the MultiVOIP to the PBX/telco for this call.
Outbound Digits Sent 0-9, #, *
Outbound Digits
Received
0-9, #, *
Of the digits transmitted by the MultiVOIP to the PBX/telco for this call,
these are the digits that were confirmed as being received.
Server Details
n.n.n.n
and/or other
related
The IP address (etc.) of the traffic control server (if any) being used
(whether an H.323 gatekeeper, a SIP proxy, or an SPP registrar gateway)
will be displayed here if the call is handled through that server.
descriptions
DTMF Capability
inband,
Indicates whether the DTMF dialing digits are carried "Inband" or "Out of
Band." The corresponding field values differ for the 3 different VOIP
protocols.
For H.323, this field can display "Out of Band" or "Inband". For SIP it can
display either "Out of Band RFC2833" or "Out of Band SIP INFO" to
indicate the out-of-band condition or "Inband" to indicate the in-band
condition. For SPP it can display "Out of Band RFC2833" or "Inband".
out of band
Expressions
differ slightly
for different
Call Signaling
protocols
(H.323, SIP, or
SPP).
Table is continued on next page…
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Chapter 6: Using the Software
Call Progress Details: Field Definitions (continued)
Field Name
Values
Description
Supplementary Services Status
Call on Hold
alphanumeric
alphanumeric
Describes held call by its IP address source, location/gateway identifier,
and hold duration. Location/gateway identifiers come from Gateway Name
field in Phone Book Configuration screen of remote VOIP.
Call Waiting
Caller ID
Describes waiting call by its IP address source, location/gateway identifier,
and hold duration. Location/gateway identifiers come from Gateway Name
field in Phone Book Configuration screen of remote VOIP.
“Calling Party
+ identifier”;
“Alerting Party
+ identifier”;
“Busy Party
+ identifier”;
“Connected
Party +
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 identifier comes from the “Caller Id” field in the
Supplementary Services screen of the remote VOIP unit.
identifier”
Call Status fields
Call Status
hangup, active Shows condition of current call.
Call Control Status
Tun, FS + Tun, Displays the H.323 version 4 features in use for the selected call. These
AE, Mux
include tunneling (Tun), Fast Start with tunneling (FS + Tun), Annex E
multiplexed UDP call signaling transport (AE), and Q.931 Multiplexing
(Mux).
Silence Compression
SC
“SC” stands for Silence 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
Correction
FEC
“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
Logs
Figure 6-4: Log statistics screen
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Chapter 6: Using the Software
Logs Screen Details: Field Definitions
Description
Field Name
Values
Log # column
1 or higher
All calls are assigned an event number in chronological order, with the
most recent call having the highest event number.
Start Date,Time
column
dd:mm:yyyy
hh:mm:ss
The starting time of the call. The date is presented as a day and a month
of one or two digits, and a four-digit year. This is followed by a time-of-day
in a two-digit hour, a two-digit minute, and a two-digit seconds value.
This describes how long the call lasted in hours, minutes, and seconds.
Indicates the Call Signaling protocol used for the call (H.323, SIP, or SPP).
Duration column
Type
hh:mm:ss
H.323, SIP, SPP
Status column
IP Direction
success or failure Displays the status of the call (whether the call was completed or not).
incoming,
Indicates whether the call is "incoming" or "outgoing" with respect to the
gateway.
outgoing
Mode column
From column
To column
voice or FAX
gateway name
gateway name
Indicates whether the event being described was a voice call or a FAX call.
Displays the name of the voice gateway that originates the call.
Displays the name of the voice gateway that completes the call.
Special Buttons
Previous
Next
First
Last
Delete File
--
--
--
--
--
Displays log entry before currently selected one.
Displays log entry after currently selected one.
Displays first log entry
Displays last log entry.
Deletes selected log file.
Call Details
Coder protocol
Voice coder
The voice coder being used on this call.
Disconnect Reason "Normal" or
"Local"
Indicates whether the call was disconnected simply because the desired
conversation was done or some other irregular cause occasioned
disconnection (e.g., a technical error or failure).
disconnection.
DTMF Capability
inband,
Indicates whether the DTMF dialing digits are carried "Inband" or "Out of
Band." The corresponding field values differ for the 3 different VOIP
protocols.
For H.323, this field can display "Out of Band" or "Inband". For SIP it can
display either "Out of Band RFC2833" or "Out of Band SIP INFO" to
indicate the out-of-band condition or "Inband" to indicate the in-band
condition. For SPP it can display "Out of Band RFC2833" or "Inband".
The digits, sent by MultiVOIP to PBX/telco, that were acknowledged as
having been received by the remote VOIP gateway.
out of band
Expressions
differ slightly for
different Call
Signaling
protocols.
0-9, #, *
Outbound Digits
Received
Outbound Digits
Sent
0-9, #, *
The digits transmitted by the MultiVOIP to the PBX/telco for this call.
Server Details
n.n.n.n
When the MultiVOIP is operating in the non-direct mode (with Gatekeeper
in H.323 mode; with proxy in SIP mode; or in the client/server configuration
of SPP mode), this field shows the IP address of the server that is directing
IP phone traffic.
Packets sent
Packets received
integer value
integer value
Number of data packets sent over the IP network in the course of this call.
Number of data packets received over the IP network in the course of this
call.
Packets lost
integer value
Number of voice packets from this call that were lost after being received
from the IP network.
Number of bytes of data sent over the IP network in the course of this call.
Number of bytes of data received over the IP network in the course of this
call.
Bytes sent
Bytes received
integer value
integer value
FROM Details
Gateway Name
IP Address
Options
alphanumeric
n.n.n.n
FEC, SC
Identifier for the VOIP gateway that originated this call.
IP address of the VOIP gateway from which the call was received.
Displays VOIP transmission options used by the VOIP gateway originating
the call. These may include Forward Error Correction or Silence
Compression.
TO Details
Gateway Name
IP Address
Options
alphanumeric
n.n.n.n
Identifier for the VOIP gateway that completed (terminated) this call.
IP address of the VOIP gateway at which the call was completed.
Displays transmission options used by VOIP gateway terminating the call.
Supplementary Services Info
Call Transferred To phone number
Number of party called in transfer.
Number of party called in forwarding.
Call Forwarded To
phone number
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IP Statistics
Figure 6-5: IP statistics screen
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 irretrievable; 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 comes through as static).
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IP Statistics: Field Definitions
Field Name
Values
Description
IP Address
n.n.n.n
IP address of the MultiVOIP. For an IP address to be displayed here, the MultiVOIP must
have DHCP enabled. Its IP address, in such a case, is assigned by the DHCP server.
Clears packet tallies from memory.
“Clear” button
--
Total Packets
Sum of data packets of all types.
Transmitted
integer
value
Total number of packets transmitted by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
Received
integer
value
Total number of packets received by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
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.
Transmitted
integer
value
Number of UDP packets transmitted by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
Received
integer
value
Number of UDP packets received by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
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.
Transmitted
integer
value
Number of TCP packets transmitted by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
Received
integer
value
Number of TCP packets received by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
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.
Voice signals are transmitted in Realtime Transport Protocol packets. RTP packets are a
type or subset of UDP packets.
RTP Packets
Transmitted
integer
value
Number of RTP packets transmitted by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
Received
integer
value
Number of RTP packets received by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
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.
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.
RTCP Packets
Transmitted
Received
integer
value
Number of RTCP packets transmitted by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
integer
value
Number of RTCP packets received by this VOIP gateway since the last “clearing” or
resetting of the counter within the MultiVOIP software.
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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Link Management
The Link Management screen is essentially an automated utility for pinging endpoints on your VOIP network.
This utility generates pings of variable sizes at variable intervals and records the response to the pings.
Figure 6-6: Link management
Link Management screen Field Definitions
Field Name
Values
Description
Monitor Link fields
IP Address to Ping
Pings per Test
n.n.n.n
This is the IP address of the target endpoint to be pinged.
1-999
This field determines how many pings will be generated by the Start Now
command.
Response Timeout
500 – 5000
milliseconds
The duration after which a ping will be considered to have failed.
Ping Size in Bytes
32 – 128 bytes This field determines how long or large the ping will be.
Timer Interval
between Pings
0 or 30 – 6000
minutes
This field determines how long of a wait there is between one ping and the
next.
Start Now command
button
--
Initiates pinging.
Clear command
button
--
Erases ping parameters in Monitor Link field group and restores default
values.
Link Status Parameters
These fields summarize the results of pinging.
Target of ping.
IP Address column
No. of Pings Sent
n.n.n.n
as listed
as listed
Number of pings sent to target endpoint.
Number of pings received by target endpoint.
No. of Pings
Received
Round Trip Delay
(Min/Max/Avg)
as listed,
in milliseconds
Displays how long it took from time ping was sent to time ping response
was received.
Last Error
as listed
Indicates when last data error occurred.
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Registered Gateway Details
The Registered Gateway Details screen presents a real-time display of the special operating parameters of the
Single Port Protocol (SPP). These are configured in the Call Signaling screen and in the Add/Edit Outbound
Phone Book screen.
Figure 6-7: Registered endpoints
Registered Gateway Details: Field Definitions
Field Name
Column Headings
Description alphanumeric
Values
Description
This is a descriptor for a particular VOIP gateway unit. This descriptor should
generally identify the physical location of the unit (e.g., city, building, etc.) and
perhaps even its location in an equipment rack.
IP Address
Port
Register
Duration
Status
n.n.n.n
n
The RAS address for the gateway.
Port by which the gateway exchanges H.225 RAS messages with the gatekeeper.
The time remaining in seconds before the TimeToLive timer expires. If the gateway
fails to reregister within this time, the endpoint is unregistered.
The current status of the gateway either registered or unregistered.
Registered/
unregistered
No. of Entries
The number of gateways currently registered to the Registrar. This includes all SPP
clients registered and the Registrar itself.
Details
Count of
Registered
Numbers
If a registered gateway is selected (by clicking on it in the screen), The "Count of
Registered Numbers" will indicate the number of registered phone numbers for the
selected gateway. When a client registers, all of its inbound phonebook's phone
numbers become registered.
List of
Lists all of the registered phone numbers for the selected gateway.
Registered
Numbers
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Chapter 6: Using the Software
Servers
H.323 GateKeepers
The –SS and -FX series of MultiVOIPs do not support H.323.
Figure 6-8: H.323 Gatekeepers
H.323 Gatekeepers (Statistics, Servers): Field Definitions
Field Name
Values
Description
Column Headings
IP Address
Port
n.n.n.n
n
The IP address of the gatekeeper.
TDMA time slot used for communication between MultiVOIP unit and the
gatekeeper that serves it.
GK Name
Type
alpha-
numeric string
Primary,
Predefined
n
Identifier for gatekeeper
This field describes the type of gateway as which the MultiVOIP is defined with
respect to the gatekeeper
Priority level given.
Priority
Status
registered,
not registered
The current status of the gateway either registered or unregistered.
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Chapter 6: Using the Software
SIP Proxies
Figure 6-9: SIP proxies
SIP Proxies (Statistics, Servers): Field Definitions
Description
Field Name
Values
Column Headings
IP Address
Port
n.n.n.n
port
The IP address of the SIP proxy by which the MultiVOIP is governed.
TDMA time slot used for communication between MultiVOIP unit and the SIP
Proxy that governs it.
Type
Status
Primary,
Alternate
registered,
This field describes the type of gateway as which the MultiVOIP is defined with
respect to the gatekeeper.
The current status of the MultiVOIP gateway with respect to the SIP proxy either
not registered
registered or unregistered.
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Chapter 6: Using the Software
SPP Registrars
The –SS models do not support the SPP signaling protocol.
Figure 6-10: SPP registrars
SPP Registrars (Statistics, Servers): Field Definitions
Field Name
Values
Description
Column Headings
IP
n.n.n.n
The IP address of the gatekeeper.
Address
Port
port
TDMA time slot used for communication between MultiVOIP unit and the
gatekeeper that serves it.
Type
Primary,
This field describes the type of gateway as which the MultiVOIP is defined with
respect to the gatekeeper.
The current status of the gateway either registered or unregistered.
Predefined
registered, not
registered
Status
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Advanced
Packetization Time
You can use the Packetization Time screen to specify definite packetization rates for coders selected in
the Voice/FAX Parameters screen (in the “Coder Options” group of fields). The Packetization Time
screen is accessible under the “Advanced” options entry in the sidebar list of the main VOIP software
screen. In dealing with RTP parameters, the Packetization Time screen is closely related to both
Voice/FAX Parameters and to IP Statistics. It is located in the “Advanced” group for ease of use.
Figure 6-11: Packetization time
Packetization rates can be set separately for each channel.
The table below presents the ranges and increments for packetization rates. The final column represents
recommended settings (based on the most common found) when operating with third party devices.
Packetization Ranges and Increments
Recommendations
Coder Types
G711, G726, G727
G723
Range (in Kbps); {default}
Increments (in Kbps)
Setting (in ms)
5-120
{5}
5
20
30
20
20
30-120
10-120
20-120
{30}
{10}
{20}
30
10
20
G729
NetCoder
Once the packetization rate has been set for one channel, it can be copied into other channels by using
the Copy Channel button on the Packetization Time screen. Simply click the boxes next to the channels
you wish to copy the settings for.
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Chapter 6: Using the Software
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 x.xx | … ). In this section, we describe the software functions
available on this menu.
Figure 6-12: Program menu
Several basic software functions are accessible from the MultiVOIP software menu, as shown below.
MultiVOIP Program Menu
Menu Selection
Description
Configuration
Select this to enter the Configuration program where values for IP, telephony, and
other parameters are set.
Configuration Port Setup
Select this to access the COM Port Setup screen of the MultiVOIP Configuration
program.
Date and Time Setup
Download Factory Defaults
Download Firmware
Select this for access to set calendar/clock used for data logging.
Select this to return the configuration parameters to the original factory values.
Select this to download new versions of firmware as enhancements become
available.
Download IFM Firmware
Select this to download new versions of IFM firmware as enhancements become
available. The Interface Module (IFM) is the telephony interface for analog
MultiVOIP units. There is one IFM for each channel of the MultiVOIP unit. For
each channel, the IFM handles the analog signals to and from the attached
telephone, PBX or CO line.
Download User Defaults
Set Password
To be used after a full set of parameter values, values specified by the user, have
been saved (using Save Setup). This command loads the saved user defaults
into the MultiVOIP.
Select this to create a password for access to the MultiVOIP software programs
(Program group commands, Windows interface, web browser interface, & FTP
server). Only the FTP Server function requires a password for access. The FTP
Server function also requires that a username be set 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 used).
Upgrade Software
Loads firmware (including H.323 stack) and settings from the controller PC to the
MultiVOIP unit. User can choose whether to load Factory Default Settings or
Current Configuration settings.
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Chapter 6: Using the Software
“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 Multi-Tech’s website.
Although transferring updated program files from the Multi-Tech website to the user’s PC can generally be
considered a download (from the perspective of the PC), this type of download cannot be initiated from the
MultiVOIP software’s Program menu command set.
Generally, updated firmware must be downloaded from the Multi-Tech website to the PC before it can be loaded
from the PC to the MultiVOIP.
Updating Firmware
Generally, updated firmware must be downloaded from the Multi-Tech website to the user’s PC before it can be
downloaded from that PC to the MultiVOIP.
Note that the structure of the Multi-Tech website 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.
If you choose Support, you can select “MultiVOIP” in the Product Support menu and then click on Firmware to
find MultiVOIP resources.
Figure 6-13: Web locations
Once the updated firmware has been located, it can be downloaded from the website using normal PC/Windows
procedures.
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. It is usually best to click the Browse
button and select a folder that is easy to get to and remember.
C:\Acme-Inc\MVP3000-firm
Figure 6-14: Extract files
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Chapter 6: Using the Software
Implementing a Software Upgrade
MultiVOIP software can be upgraded locally using a single command at the MultiVOIP Windows interface, 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 interface, firmware and factory default settings can also be transferred from
controller PC to MultiVOIP piecemeal using separate commands.
When using the MultiVOIP web browser interface 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 software upgrade (whether from the Windows interface or web browser interface), 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.
When a new firmware version is installed, the MultiVOIP software can be upgraded in one step using the
Upgrade Software command, or piecemeal using the Download Firmware command and the
Download Factory Defaults command.
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 Multi-Tech factory.
Upgrade Software implements both the Download Firmware command and the Download Factory
Defaults command.
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Chapter 6: Using the Software
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 use the Download Factory Defaults command, go to Start | Programs | MultiVOIP x.xx |
Download Firmware.
3. If a password has been established, the Password Verification screen will appear.
Figure 6-15: Password verification
Type in the password and click OK.
4. The MultiVOIP x.xx 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.
5. The program will locate the firmware “.bin” file in the MultiVOIP directory. Highlight the correct (newest)
“.bin” file and click Open.
Figure 6-16: Firmware file
6. Progress bars will appear at the bottom of the screen during the file transfer.
Figure: 6-17: Progress bars
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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Chapter 6: Using the Software
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 use the Download Factory Defaults command, go to Start | Programs | MultiVOIP x.xx. |
Download Factory Defaults.
3. If a password has been established, the Password Verification screen will appear.
Figure 6-18: Password verify
Type in the password and click OK.
4. The MVP x.xx - 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.
Figure 6-19: Dialog screen
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.
Figure 6-20: Progress bars
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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Chapter 6: Using the Software
Downloading IFM Firmware
The Interface Module (IFM) is the telephony interface for analog MultiVOIP units. There is one IFM for each
channel of the MultiVOIP unit. For each channel, the IFM handles the analog signals to and from the attached
telephone, PBX or CO line. The IFM communicates with the main processor indicating the status of the telephone
line. For example, it might indicate that a phone is off hook (FXS) or that an incoming ring is present (FXO). The
IFM receives operating instructions from the VOIP’s main processor. For example, the IFM might be instructed to
ring the phone (FXS) or seize the line (FXO). The IFM contains a codec (coder/decoder) to convert the incoming
audio to a PCM stream (pulse code modulation) which it sends to the DSP (digital signal processor). The IFM’s
codec also converts outgoing PCM to audio.
The firmware of the IFMs will change from time to time and you may need to upgrade the firmware on your
MultiVOIP unit. To do so, follow these instructions.
1. In the System Information screen of the MultiVOIP Configuration software, check the version number of the
IFM firmware already installed on the MultiVOIP unit. Write down the version number.
2. Exit the Configuration software program. The MultiVOIP Configuration program must be off when invoking the
Download IFM Firmware command. If it is on, the command will not work.
3. To use the Download IFM Firmware command, go to Start | Programs | MultiVOIP x.xx | Download IFM
Firmware.
4. A warning window will appear: “Downloading IFM Firmware will reboot the MultiVOIP. Do you want to
continue?” Click OK.
Figure 6-21: Download IFM firmware
5. The “Boot” LED on the front panel of the MultiVOIP will come on.
6. The software will search for an IFM firmware file to use to upgrade the system; if the file found represents
firmware newer than that already installed on the MultiVOIP (or if you want to overwrite the same version of
firmware) click Open.
Figure 6-22: IFM firmware file
7. The IFM Firmware Download screen will appear. Select “Copy to All IFMs” and click OK. (Only in very
special circumstances would different IFMs in the same VOIP be loaded with different IFM firmware.)
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Chapter 6: Using the Software
Figure 6-23: IFM firmware download
8. The main MultiVOIP Configuration screen will appear. Progress bars can be seen at the bottom of the screen
while files are being copied.
9. Then a completion screen entitled IFM Test will appear.
Figure 6-24: IFM test screen
Click OK.
10. The MultiVOIP will reboot itself. When the reboot is complete, the MultiVOIP Configuration screen will close.
11. The IFM firmware downloading process is complete.
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Chapter 6: Using the Software
Setting and Downloading User Defaults
The Download User Defaults command allows you to maintain a known working configuration that is specific to
your VOIP system. You can then experiment with alterations or improvements to the configurations confident that
a working configuration can be restored if necessary.
1. Before you can use 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.
Figure 6-25: Save & Reboot
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.
A user default file will be created. The MultiVOIP unit will reboot itself.
3. To download the user defaults, go to Start | Programs | MultiVOIP x.xx | Download User Defaults.
4. A confirmation screen will appear indicating that this action will entail rebooting the MultiVOIP.
Figure 6-26: Confirmation screen
Click OK.
5. Progress bars will appear during the file transfer process.
Figure 6-27: Progress bars
6. When the file transfer process is complete, the Dialog / IP Parameters screen will appear.
Figure 6-28: Dialog screen
7. Set the IP values per your particular VOIP system. Click OK. Progress bars will appear as the MultiVOIP
reboots itself.
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Chapter 6: Using the Software
Setting a Password
Windows Interface
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 interface.
NOTE: Record your user name and password in a safe place. If the password is lost, forgotten, or
irretrievable, the user must contact Multi-Tech 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 use the Set Password command, go to Start | Programs | MultiVOIP x.xx | Set Password.
3. You will be prompted to confirm that you want to establish a password, which will entail rebooting the
MultiVOIP (which is done automatically).
Click OK to proceed with establishing a password.
4. The Password screen will appear. If you intend to use the FTP Server function that is built into the
MultiVOIP, enter a user name. (A User Name is not needed to access the local Windows interface, the
web browser interface, 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 Multi-Tech Technical Support for advice.
Figure 6-29: Password screen
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.
6. After the password has been set, the user will be required to enter the password to gain access to the
web browser interface 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.
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Chapter 6: Using the Software
Figure 6-30: Password verification
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.
Figure 6-31: Invalid password
Web Browser Interface
Setting a password is optional when using the MultiVOIP web browser interface. Only one password can
be assigned and it works for all MultiVOIP software functions (Windows interface, web browser interface,
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 interface.
NOTE: Record your user name and password in a safe place. If the password is lost, forgotten, or
irretrievable, the user must contact Multi-Tech Tech Support in order to resume use of the MultiVOIP web
browser interface.
Figure 6-32: Web interface password
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Chapter 6: Using the Software
Upgrading Software
As noted earlier the Upgrade Software command transfers, from the controller PC to the MultiVOIP unit, firmware
(including the H.323 stack) and settings. The settings can be either Factory Default Settings or Current
Configuration Settings.
Figure 6-33: Upgrade software path
NOTE: To upgrade a MultiVOIP from software version 6.04 or earlier, an ftp primer file must first be sent to
the VOIP. This file is located in the Software/ftp_Primer folder on the CD and the file name is
"FTP_Primer.bin". Before uploading this file, it must be renamed "mvpt1ftp.bin". The VOIP will only
accept files of this name. This is a safety precaution to prevent the wrong files from being uploaded to
the VOIP. Once the primer file has been uploaded, upload the FTP firmware file. If you accepted the
defaults during the software loading process, this file is located on your local drive at C:\Program
Files\Multi-Tech Systems\MultiVOIP X.NN where the X is the software number and the .NN is the
version number of the MultiVOIP software on your local drive. Of course the firmware file is named
‘mvpt1ftp.bin’.
Important: You cannot go back to 6.04 or earlier versions using FTP. You must use ‘upgrade software’ via
the serial port.
Important: These ftp upgrade instructions do not apply to software release 6.05 and above.
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Chapter 6: Using the Software
FTP Server File Transfers (“Downloads”)
Multi-Tech has built an FTP server into the MultiVOIP unit. Therefore, file transfers from the controller PC to the
VOIP unit can be done using an FTP client program or even using a browser (e.g., Internet Explorer, Netscape, or
Firefox, used in conjunction with Windows Explorer).
The terminology of “downloads” and “uploads” gets a bit confusing in this context. File transfers from a client to a
server are typically considered “uploads.” File transfers from a large repository of data to machines with less data
capacity are considered “downloads.” In this case, these metaphors are contradictory: the FTP server is actually
housed in the MultiVOIP unit, and the controller PC, which is actually the repository of the info to be transferred,
uses an FTP client program. In this situation, we have chosen to call the transfer of files from the PC to the VOIP
“downloads.” (Be aware that some FTP client programs may use the opposite terminology, i.e., they may refer to
the file transfer as an “upload “)
You can download firmware, CAS telephony protocols, default configuration parameters, and phonebook data for
the MultiVOIP unit with this FTP functionality. These downloads are done over a network, not by a local serial
port connection. Consequently, VOIPs at distant locations can be updated from a central control point.
The phonebook downloading feature greatly reduces the data-entry required to establish inbound and outbound
phonebooks for the VOIP units within a system. Although each MultiVOIP unit will require some unique
phonebook entries, most will be common to the entire VOIP system. After the phonebooks for the first few VOIP
units have been compiled, phonebooks for additional VOIPs become much simpler: you copy the common
material by downloading and then do data entry for the few phonebook items that are unique to that particular
VOIP unit or VOIP site.
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.
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.)
Figure 6-34: Change password
As shown above, the user name and password can be set in the web interface as well as in the Windows
interface.
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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 Multi-Tech 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.
4. Enable FTP Functionality. Go to the IP Parameters screen and click on the “FTP Server: Enable” box.
Figure 6-35: Enable FTP server
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5. Identify Files to be Updated. Determine which files you want to update. Six types of files can be updated
using the FTP feature. In some cases, the file to be transferred will have “Ftp” as the part of its filename just
before the suffix (or extension). So, for example, the file “mvpt1Ftp.bin” can be transferred to update the bin
file (firmware) residing in the MultiVOIP. Similarly, the file “fxo_loopFtp.cas” could be transferred to enable
use of the FXO Loop Start telephony interface in one of the analog VOIP units and the file “r2_brazilFtp.cas”
could be transferred to enable a particular telephony protocol used in Brazil. Note, however, that before
any CAS file can be used as an update, it must be renamed to CASFILE.CAS so that it overwrites and
replaces the default CAS file.
File Type
File Names
Description
firmware “bin” file
mvpt1Ftp.bin
This is the MultiVOIP firmware file. Only one file of this type will
be in the directory.
factory defaults
CAS file
fdefFtp.cnf
This file contains factory default settings for user-changeable
configuration parameters. Only one file of this type will be in the
directory.
fxo_loopFtp.cas, These telephony files are for Channel Associated Signaling.
em_winkFtp.cas, The directory contains many CAS files, some labeled for
r2_brazilFtp.cas
r2_chinaFtp.cas
specific functionality, others for countries or regions where
certain attributes are standard. Any CAS file used must first be
renamed to “CASFILE.CAS.”
inbound phonebook
outbound phonebook
InPhBk.tmr
This file updates the inbound phonebook in the MultiVOIP unit.
This file updates the outbound phonebook in the MultiVOIP unit.
OutPhBk.tmr
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 interface within the
MultiVOIP unit).
Figure 6-36: FTP address
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 (shown below) or with an FTP
client program (varies).
Figure 6-37: FTP log in
8. Use Download. Downloading can be done with a web browser or with an FTP client program.
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Chapter 6: Using the Software
Download with Web Browser:
• 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).
• Drag-and-drop files from the local Windows browser (e.g., Windows Explorer) to the web browser.
Figure 6-38: Drag and drop file
• You may be asked to confirm the overwriting of files on the MultiVOIP. Do so.
Figure 6-39: Overwrite confirmation
• File transfer between PC and VOIP will look like transfer within VOIP directories.
Figure 6-40: Copy screen
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Download with FTP Client Program:
• 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).
• 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 interface 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.
9. Verify Transfer. The files transferred will appear in the directory of the MultiVOIP.
Figure 6-41: Verify transfer
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 interface.
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Web Browser Interface
Figure 6-42: Web interface main page
You can control the MultiVOIP unit with a graphic user interface (interface) based on the common web browser
platform. Qualifying browsers are Internet Explorer 6+, Netscape 6+, and Mozilla Firefox 1.0+.
MultiVOIP Web Browser interface Overview
Function
Configuration Prerequisite
Browser Version Requirement
Remote configuration and control of MultiVOIP units.
Local Windows interface must be used to assign IP address to MultiVOIP.
Internet Explorer 6.0 or higher; or
Netscape 6.0 or higher; or
Mozilla Firefox 1.0 or higher.
Java Requirement
Java Runtime Environment
version 1.4.0_01 or higher
(this application program is included with MultiVOIP)
The initial configuration step of assigning the VOIP unit an IP address must still be done locally using the
Windows interface. However, all additional configurations can be done via the web interface.
The content and organization of the web interface is directly parallel to the Windows interface. For each screen in
the Windows interface, there is a corresponding screen in the web interface. The fields on each screen are the
same, as well.
The Windows interface gives access to commands via icons and pull-down menus whereas the web interface
does not. The web interface, however, cannot perform logging in the same direct mode done in the Windows
interface. However, when the web interface is used, logging can be done by email (SMTP).
The graphic layout of the web interface is also somewhat larger-scale than that of the Windows interface. For
that reason, it’s helpful to use as large of a video monitor as possible.
The primary advantage of the web interface 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 interface, 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 interface.
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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Figure 6-43: Java install screen
During the installation, you must specify which browser you’ll use in the Select Browsers screen.
Figure 6-44: Browser choice
When installation is complete, the Java program runs automatically in the background as a plug-in supporting the
MultiVOIP web interface. No user actions are required.
After the Java program has been installed, you can access the MultiVOIP using the web browser interface. Close
the MultiVOIP Windows interface. 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
interface or for the MultiVOIP’s FTP Server function. See “Setting a Password -- Web Browser interface” earlier
in this chapter.) The web browser interface offers essentially the same control over the VOIP as can be achieved
using the Windows interface. As noted earlier, logging functions cannot be handled via the web interface. 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 interface than with the Windows interface.
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Chapter 6: Using the Software
SysLog Server Functions
Multi-Tech has built SysLog server functionality into the software of the MultiVOIP units. SysLog is a de facto
standard for logging events in network communication systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement this functionality, you will need a SysLog
client program (sometimes referred to as a “daemon”). SysLog client programs, both paid and freeware can be
obtained from Kiwi Enterprises (search the Internet for kiwi syslog daemon), among other firms. Read the End-
User License Agreement carefully and observe license requirements. SysLog client programs essentially give
you a means of structuring console messages for convenience and ease of use.
Multi-Tech Systems does not endorse any particular SysLog client program. SysLog client programs by qualified
providers should suffice for use with MultiVOIP units.
Before a SysLog client program is used, the SysLog functionality must be enabled within the MultiVOIP in the
Logs menu under Configuration.
Figure 6-45: Enable SysLog
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.
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.).
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Appendix A – Cable Pin-outs
Command Cable
RJ-45 Connector
End-to-End Pin Info
1 2 3 4 5 6 7 8
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
1
2
Circuit Signal Name
TD+ Data Transmit Positive
TD- Data Transmit Negative
1
2
3
4
5
6
7
8
3
6
RD+ Data Receive Positive
RD- Data Receive Negative
Voice/Fax Channel Connectors
Figure B-1: RJ-48 & RJ-11 Connectors
Pin Functions (E&M Interface)
Pin Description
Function
1
2
3
4
5
6
7
8
M
E
T1
R
T
R1
SG
SB
Input
Output
4-Wire Output
4-Wire Input, 2-Wire Input
4-Wire Input, 2-Wire Input
4-Wire Output
Signal Ground (Output)
Signal Battery (Output)
Pin Functions (FXS/FXO Interface)
FXS Pin Description
FXO Pin Description
2
3
4
5
N/C
Ring
Tip
2
3
4
5
N/C
Tip
Ring
N/C
N/C
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Appendix B – TCP/UDP Port
Assignments
Well Known Port Numbers
The following description of port number assignments for Internet Protocol (IP) communication is taken from the
Internet Assigned Numbers Authority (IANA) web site (www.iana.org).
“The Well Known Ports are assigned by the IANA and on most systems can only be used
by system (or root) processes or by programs executed by privileged users. Ports are
used in the TCP [RFC793] to name the ends of logical connections which carry long term
conversations. For the purpose of providing services to unknown callers, a service
contact port is defined. This list specifies the port used by the server process as its
contact port. The contact port is sometimes called the "well-known port". To the extent
possible, these same port assignments are used with the UDP [RFC768]. The range for
assigned ports managed by the IANA is 0-1023.”
Well-known port numbers especially pertinent to MultiVOIP operation are listed below.
Port Number Assignment List
Function
telnet
Port Number
23
tftp
69
snmp
161
snmp tray
162
gatekeeper registration
H.323
SIP
1719
1720
5060
514
SysLog
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Appendix C – Installation Instructions
for MVP428 Upgrade Card
Installing the MVP428 Upgrade Card
In this procedure, you will install an additional circuit board into the MVP410, improving 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 hereꢀ
(2 places).
(C)
Attach upgrade card
(screws into standoffs
-- 4 places).
ꢀ
ꢀ
(B)
Mate 60-pin
connectors.
Figure C-1: MVP 248 installation
Procedure in Detail
1. Power down and unplug the MVP410 unit.
2. Using a Phillips driver, remove the blank cover plate at the rear of the MVP410 chassis. Save the screws.
screws on blank cover plate (2)
Figure C-2: Remove screws from cover plate
3. Using a Phillips driver, remove the three screws that secure the main circuit board and back panel
assembly to the chassis.
Important: Follow standard ESD precautions to protect the circuit board from static electricity damage.
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Appendix C: MVP428 Upgrade Card
back panel screws (3)
Figure C-3: Remove screws from back panel
4. Slide the main circuit board out of the chassis far enough to unplug the power connector.
power connector
Figure C-4: Accessing the power connector
5. Unplug the power connector from the main circuit board.
6. Slide the main circuit board completely out of the chassis and place on a non-conductive, static-safe
tabletop surface.
7. Remove mounting hardware (2 screws, 2 nuts, and 4 standoffs) from its package.
8. On the phone-jack side of the circuit card, three screws attach the circuit card to the back panel. Two of
these screws are adjacent to the four phone-jack pairs. Remove these two screws.
Screw locations (2)
at phone-jack edge
of board.
Figure C-5: Screws replaced with standoffs
9. Replace these two screws with standoffs.
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Appendix C: MVP428 Upgrade Card
10. There are two copper-plated holes at the LED edge of the circuit card. Place a nut beneath each hole
(lock washer side should be in contact with board) and attach a standoff to each location).
Standoff locations (2) at LED edge
of board (top view).
Standoff/nut attachment
(rear bottom view)
Figure C-6: Standoffs at LED edge of board
11. Locate the male 60-pin vertical connector near the LED edge of the main circuit card. Check that pins are
straight and evenly spaced. If not, then correct for straightness and spacing. Locate the 60-pin female
connector on the upgrade circuit card.
12. Set the upgrade circuit card on top of the main circuit card. Align the upgrade card’s 4 pairs of phone-
jacks with the 4 pairs of holes in the backplane of the main card. Slide the phone jacks into the holes.
13. Mate the upgrade card’s 60-pin female connector with the main card’s 60-pin male connector.
*
*
These screws (4 places)
*
attach upgrade card
to main card.
*
*
60-pin connectors
Figure C-7: Attaching upgrade card to main circuit card
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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Appendix D – Regulatory Information
EMC, Safety, and R&TTE Directive Compliance
The CE mark is affixed to this product to confirm compliance with the following European Community Directives:
Council Directive 89/336/EEC of 3 May 1989 on the approximation of the laws of Member States relating to electromagnetic
compatibility,
and
Council Directive 73/23/EEC of 19 February 1973 on the harmonization of the laws of Member States relating to electrical
equipment designed for use within certain voltage limits,
and
Council Directive 1999/5/EC of 9 March 1999 on radio equipment and telecommunications terminal equipment and the mutual
recognition of their conformity.
FCC Part 15 Class A Statement
This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to 47 CFR Part 15
regulations. The stated limits in this regulation are designed to provide reasonable protection against harmful interference 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. However, there is no
guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio
or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct
the interference by one or more of the following measures:
•
•
•
•
Reorient or relocate the receiving antenna.
Increase the separation between the equipment and receiver.
Plug the equipment into an outlet on a circuit different from that to which the receiver is connected.
Consult the dealer or an experienced radio/TV technician for help.
This device complies with Part 15 of the CFR 47 rules. Operation of this device is subject to the following conditions: (1) This
device may not cause harmful interference, and (2) this device must accept any interference that may cause undesired
operation.
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.
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 D: Regulatory Information
FCC Part 68 Telecom
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.
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.
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.
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.
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.
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 may request you to remove the equipment form t network until the
problem is resolved.
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.
Manufacturer:
Trade name:
Multi-Tech Systems, Inc.
MultiVOIP®
Model number:
MVP-210/410/810
US: AU7DDNAN46050
RJ-48C
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, MN 55112
Tel: (763) 785-3500
FAX: (763) 785-9874
FCC registration number:
Modular jack (USOC):
Service center in USA:
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Appendix E – Waste Electrical and
Electronic Equipment (WEEE)
Statement
July, 2005
The WEEE directive places an obligation on EU-based manufacturers, distributors, retailers and importers to
take-back electronics products at the end of their useful life. A sister Directive, ROHS (Restriction of Hazardous
Substances) complements the WEEE Directive by banning the presence of specific hazardous substances in the
products at the design phase. The WEEE Directive covers all Multi-Tech products imported into the EU as of
August 13, 2005. EU-based manufacturers, distributors, retailers and importers are obliged to finance the costs of
recovery from municipal collection points, reuse, and recycling of specified percentages per the WEEE
requirements.
Instructions for Disposal of WEEE by Users in the European Union
The symbol shown below is on the product or on its packaging, which indicates that this product must not be
disposed of with other waste. Instead, it is the user’s responsibility to dispose of their waste equipment by handing
it over to a designated collection point for the recycling of waste electrical and electronic equipment. The separate
collection and recycling of your waste equipment at the time of disposal will help to conserve natural resources
and ensure that it is recycled in a manner that protects human health and the environment. For more information
about where you can drop off your waste equipment for recycling, please contact your local city office, your
household waste disposal service or where you purchased the product.
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Appendix F – C-ROHS HT/TS Substance
Concentration
依照中国标准的有毒有害物质信息
根据中华人民共和国信息产业部(MII) 制定的电子信息产品(EIP)
标准－中华人民共和国《电子信息产品污染控制管理办法》（第39 号），也称作中国
RoHS，下表列出了Multi-Tech Systems Inc. 产品中可能含有的有毒物质(TS) 或有害物质(HS)
的名称及含量水平方面的信息。
有害/有毒物质/元素
铅
成分名称
汞
镉
(CD)
六价铬
(CR6+)
多溴联苯
(PBB)
多溴二苯醚
(PBDE)
(PB)
(Hg)
O
X
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
印刷电路板
电阻器
电容器
X
O
O
O
O
X
铁氧体磁环
继电器/光学部件
IC
二极管/晶体管
振荡器和晶振
调节器
O
O
电压传感器
变压器
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
扬声器
连接器
LED
O
X
O
O
O
O
O
O
O
O
O
O
螺丝、螺母以及其它五金件
交流-直流电源
O
O
O
O
O
O
O
O
O
O
O
O
软件/文档 CD
手册和纸页
O
O
O
O
O
O
O
O
O
O
O
O
底盘
X
表示所有使用类似材料的设备中有害/有毒物质的含量水平高于SJ/Txxx-2006 限量要求。
O
表示不含该物质或者该物质的含量水平在上述限量要求之内。
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INDEX
IP Statistics fields, 109
A
C
L
Auto Disconnect, 39
AutoCall/Offhook, 38
LED descriptions, 7
Link Management fields, 110
Logs (Statistics) field definitions, 107
Cabling: 210, 11; 410/810, 13
Call Hold, 72
N
Call Name Identification, 72
Call Progress fields, 105
Call Transfer, 72
NAT Traversal screen fields, 71
P
Call Waiting, 72
Coder Parameters fields, 37
Creating a User Default Configuration, 75
Custom Tones and Cadences, 64
Packet Prioritization 802.1p, 33
Packetization rates, 115
R
D
RADIUS Screen field definitions, 69
Regional parameter definitions, 62
DID Interface Parameters, 52
DID-DPO Interface parameter definitions, 52
Diff Serv PHB value, 34
S
DTMF inband, 36
DTMF out of band, 36
Dynamic Jitter, 39
Saving the MultiVOIP Configuration, 75
Set Baud Rate, 75
Set Log Reporting Method, 70
Set SNMP parameters, 61
Set Telephony Interface parameters, 40
Setting Ethernet/IP parameters, 32
Setting password, 124
E
E&M parameter definitions, 50
E&M Parameters, 49
Email log reports, 65
Setting user defaults, 123
Error message: Comm. Port Unavailable, 76; MultiVOIP Not
Found, 76; Phone Database not Read, 76
Expansion card (4-to-8 channel) installation, 137
SIP Call Signaling parameter definitions, 56
SMTP parameters definitions, 66
Specifications, 8
SPP Call Signaling parameter definitions, 59
STUN clients and servers, 71
Supervisory signaling, 40
Supplementary Services parameter definitions, 72
Survivable SIP, 57
F
FRF11, 36
FTP Server function, 127
FTP Server, logging out, 131
FXO Interface parameter definitions, 45
FXO Parameters, 44
FXO Supervision parameter definitions, 47
FXS Loop Start parameters, 41
SysLog Server function: enabling, 134
T
T.38, 36
H
U
H.323 Call Signaling parameter definitions, 54
Updating firmware, 117
I
V
Identifying current firmware version, 118
IFM firmware, 121
Voice/FAX parameter definitions, 35
Multi-Tech Systems, Inc.
144
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