Multitech MultiVOIP MVP130, MultiVOIP MVP130-FXS User Manual

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MultiVOIP®

Voice/Fax over IP Gateways

MVP130 MVP130-FXS

User Guide

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User Guide

S000386C

Analog MultiVOIP Units (Models MVP130) (Models MVP130-FXS)

This publication may not be reproduced, in whole or in part, without prior expressed written permission from MultiTech Systems, Inc. All rights reserv ed.

Multi-Tech Systems, Inc. makes no representations or warranty with respect to the contents hereof and specifically disclaims any implied warranties of merchantability or fitness for any particular purpose. Furthermore, Multi-Tech Systems, Inc. reserves the right to revise this publication and to make changes from time to time in the content hereof without obligation of Multi-Tech Systems, Inc. to notify any person or organization of such revisions or changes. Check Multi-Tech’s Web site for current versions of our product documentation.

Record of Revisions

Revision Date Description

A 09/26/05 Doc re-organization. Follows S000249K. Describes 1.08 software release. B 04/25/07 Update tech support contact list & revise warranty. C 02/08/08 Format revision and software version 1.11 update.

Patents

This Product is covered by one or more of the following U.S. Patent Numbers: 6151333, 5757801, 5682386,

5.301.274; 5.309.562; 5.355.365; 5.355.653; 5.452.289; 5.453.986. Other Patents Pending.

Trademark

Registered trademarks of Multi-Tech Systems, Inc. are MultiVOIP, Multi-Tech, and the Multi-Tech logo. Windows and NetMeeting are registered trademarks of Microsoft.

World Headquarters

Multi-Tech Systems, Inc. 2205 Woodale Drive Mounds View, Minnesota 55112 Phone: 763-785-3500 or 800-328-9717 Fax: 763-785-9874 http://www.multitech.com

Technical Support

Country By Email By Phone

Europe, Middle East, Africa: support@multitech.co.uk (44) 118 959 7774 U.S., Canada, all others: support@multitech.com

(800) 972-2439 or (763) 717-5863

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CONTENTS

Chapter 1 – Description and Specifications...................................................................................................... 6

Introduction............................................................................................................................................................6

Interface............................................................................................................................................................................6

Front Panel LEDs .............................................................................................................................................................7

Computer Requirements.......................................................................................................................................7

Specifications ........................................................................................................................................................7

Chapter 2 – Installing and Cabling the MultiVOIP............................................................................................. 8

Introduction............................................................................................................................................................8

Safety Warnings....................................................................................................................................................8

Unpacking Your MultiVOIP....................................................................................................................................8

Cabling Procedure for MVP130.............................................................................................................................9

Chapter 3 – Software Installation...................................................................................................................... 10

Introduction..........................................................................................................................................................10

Loading MultiVOIP Software onto the PC...........................................................................................................10

Setup Overview...................................................................................................................................................13

Ethernet/IP......................................................................................................................................................................14

Voice/Fax........................................................................................................................................................................15

Interface..........................................................................................................................................................................17

Call Signaling..................................................................................................................................................................19

Regional..........................................................................................................................................................................21

Phone Book....................................................................................................................................................................22

Save & Reboot................................................................................................................................................................23

Chapter 4 – Configuring Your MultiVOIP......................................................................................................... 24

Introduction..........................................................................................................................................................24

Software Categories Covered in This Chapter....................................................................................................24

How to Navigate Through the Software ..............................................................................................................25

Web Browser Interface........................................................................................................................................25

Configuration Information Checklist ....................................................................................................................25

Ethernet/IP......................................................................................................................................................................26

Voice/Fax........................................................................................................................................................................29

Configurable Payload Type.......................................................................................................................................33

Interface..........................................................................................................................................................................34

FXS Loop Start Parameters......................................................................................................................................35

Message Waiting.......................................................................................................................................................37

FXO Parameters.......................................................................................................................................................38

DID Parameters ........................................................................................................................................................43

Call Signaling..................................................................................................................................................................44

H.323 ........................................................................................................................................................................44

SIP............................................................................................................................................................................46

SPP...........................................................................................................................................................................48

SNMP .............................................................................................................................................................................50

Regional..........................................................................................................................................................................51

SMTP..............................................................................................................................................................................54

RADIUS..........................................................................................................................................................................57

Logs/Traces....................................................................................................................................................................59

NAT Traversal.................................................................................................................................................................60

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Supplementary Services.................................................................................................................................................61

Save Settings..................................................................................................................................................................64

Save & Reboot..........................................................................................................................................................64

Connection......................................................................................................................................................................64

Settings.....................................................................................................................................................................64

Troubleshooting Software Issues..............................................................................................................................65

Chapter 5 – Phone Book Configuration........................................................................................................... 66

Introduction..........................................................................................................................................................66

Identify Remote VOIP Site to Call.......................................................................................................................66

Identify VOIP Protocol to be Used.......................................................................................................................66

Phonebook Starter Configuration........................................................................................................................67

Outbound Phonebook.....................................................................................................................................................67

Inbound Phonebook........................................................................................................................................................69

Phone Book Descriptions....................................................................................................................................70

Outbound Phone Book/List Entries.................................................................................................................................70

Add/Edit Outbound Phone Book ...............................................................................................................................71

Inbound Phone Book/List Entries....................................................................................................................................75

Add/Edit Inbound Phone Book..................................................................................................................................76

Phone Book Save and Reboot........................................................................................................................................78

Phonebook Examples .........................................................................................................................................79

North America.................................................................................................................................................................79

Europe............................................................................................................................................................................82

Variations of Caller ID .........................................................................................................................................88

Chapter 6 – Using the Software........................................................................................................................ 91

Introduction..........................................................................................................................................................91

Software Categories Covered in This Chapter....................................................................................................91

Statistics Section.................................................................................................................................................93

Call Progress..................................................................................................................................................................93

Logs................................................................................................................................................................................95

IP Statistics.....................................................................................................................................................................97

Link Management ...........................................................................................................................................................99

Registered Gateway Details .........................................................................................................................................100

Servers .........................................................................................................................................................................101

H.323 GateKeepers ................................................................................................................................................101

SIP Proxies .............................................................................................................................................................102

SPP Registrars........................................................................................................................................................103

Advanced......................................................................................................................................................................104

Packetization Time..................................................................................................................................................104

MultiVOIP Program Menu Items........................................................................................................................105

Updating Firmware .......................................................................................................................................................106

Implementing a Software Upgrade ...............................................................................................................................107

Identifying Current Firmware Version......................................................................................................................107

Downloading Firmware ...........................................................................................................................................108

Downloading Factory Defaults ................................................................................................................................109

Downloading IFM Firmware..........................................................................................................................................110

Setting and Downloading User Defaults .......................................................................................................................112

Setting a Password.......................................................................................................................................................113

Windows Interface...................................................................................................................................................113

Web Browser Interface............................................................................................................................................114

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Upgrading Software......................................................................................................................................................115

FTP Server File Transfers (“Downloads”) .........................................................................................................116

Web Browser Interface......................................................................................................................................121

SysLog Server Functions..................................................................................................................................123

Appendix A – Cable Pin-outs.......................................................................................................................... 124

Appendix B – TCP/UDP Port Assignments.................................................................................................... 125

Appendix C – Warranty and Repair Policies ................................................................................................. 126

Appendix D – Regulatory Information............................................................................................................ 128

Appendix E – Waste Electrical and Electronic Equipment (WEEE) Statement.......................................... 130

Appendix F – C-ROHS HT/TS Substance Concentration............................................................................. 131

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Chapter 1 – Description and

Specifications

Introduction

The MultiVOIP MVP130 is a single-channel unit and the MVP130FXS is a single-channel unit that supports the FXS telephony interface only. Both of these MultiVOIP units have a 10/100Mbps Ethernet interface and a command port for configuration. The MVP130 and MVP130FXS are table-top models.

Figure 1-1: MVP130 Chassis

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 telephone company (telco) switch. When the phonebooks are set, special dialing sequences are mini mized or eliminated altogether. Once the call destination is determined, the phonebook settings determine whether the de stination VOIP unit must strip off or add dialing digits to make the call appear at its destination to be a local call.

Interface

There are two options for accessing your MultiVOIP, one is the Windows software that is included and is necessary for the initial setup, and the other is a web-based interface that uses your web browser to access the unit. 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.

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Chapter 2: Quick Start

Front Panel LEDs

On both the MVP130 and MVP130-FXS models, there are eight LEDs. These are explained in the table below.

Figure 1-2. MVP130/130-FXS LEDs

Front Panel LED Definitions

LED Description Power

Boot

Ethernet

TX RX

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 FD. LED indicates whether Ethernet connection is half-duplex or full-duplex 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. LK. 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.

Transmit. This indicator blinks when voice packets are being transmitted to the local area network. Receive. This indicator blinks when voice packets are being received from the local area net work. Transmit Signal. This indicator lights when the FXS-configured channel is off-hook or the FXO-configured

channel (MVP130 only) is receiving a ring from the Telco or PBX. Receive Signal. This indicator lights when the FXS-configured channel is ringing or the FXO-configured

channel (MVP130 only) has taken the line off-hook.

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 monitorin g functions can be done remotely via the IP network.

Specifications

MVP130 & MVP130-FXS Operating Voltage/Current Mains Frequencies Power Consumption

Mechanical Dimensions

Weight Ambient temperature range

100-240VAC / 1.0 A

50/60 Hz

4.5 watts (9.7 watts with phone off hook)

1.0” H

4.3” W x

5.6” D x

----------------

2.5 cm H

10.9 cm W x

14.2 cm D x

8 oz. (23 g)

Maximum

: 60 degrees Celsius (140 degrees Fahrenheit) @ 20-90% non-condensing relative humidity. Minimum

: 0 degrees Celsius (32 degrees Fahrenheit).

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Chapter 2 – Installing and Cabling the

MultiVOIP

Introduction

The MVP130 MultiVOIP models are tabletop units that can be handled easily by one person. These products must be installed by qualified service personnel in a restricted-access area, in accordan ce with Articles 110-16, 10-17, and 110-18 of the National Electrical Code, ANSI/NFPA 70.

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.

MVP130 models content list:

• MVP130 or MVP130-FXS

• DB9 to RJ45 cable

• Power transformer

• Power cord

• RJ-11 phone cord

• Printed Cabling Guide

• Product CD

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Chapter 2: Installing and Cabling the MultiVOIP

Cabling Procedure for MVP130

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.

Figure 2-1: Rear connections for MVP130

2. Connect the MultiVOIP to a PC by using the 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 connector on the back of the MultiVOIP. Connect the other end of the cable to your network.

a. For an FXS or FXO connection (FXS only for the MVP130-FXS).

(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 FXS/FXO connector on the back of the MultiVOIP. Connect the other end to the device or phone jack.

b. For a DID connection. (Not supported by the MVP130-FXS)

(DID Example: DID fax system or DID voice phone lines) Connect one end of an RJ-11 phone cord to the 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. Turn on power to the MultiVOIP by placing the ON/OFF switch on the side to the ON position. Wait for the BOOT LED on the MultiVOIP to go off before proceeding. This may take a few minutes.

5. Proceed to the Software Installation chapter to load the MultiVOIP software.

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Figure 2-2: Cable connections

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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.

3. When the Multi-Tech Installation CD dialog box appears, click the Install Software icon.

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Figure 3-1: splash screen

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Chapter 3: Software Installation

4. A secondary screen appears. Click on the button that matches the model you have purchased. The installation wizard will start.

Figure 3-2: Initial 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.

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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Figure 3-3: Destination

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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

o Set 802.1p Priority Parameters as needed

 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

o TYPE II or SNAP

• Enter Gateway Name

o Check to enable DHCP if used

• 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.

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Figure 3-8: Voice & Fax settings

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Actions:

• Select Channel

o Choose ch annel parameters:

 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

 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

 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)  Select any Automatic Disconnection options needed to ensure lines are not left “open”  Configurable Payload Types are best left at their defaults.

o The Copy Channel button is available for easily transferring these settings to the other channels

• Repeat for all channels to be used

Chapter 3: Software Installation

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Chapter 3: Software Installation

Interface

The Interface Parameters are the telephony settings that are to be applied to the MultiVOIP channel. Note: Feature options are enabled or unavailable depending on the selected interface type. The one optio n

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.

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Figure 3-9: Interface Parameters

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Actions:

• Select Channel

o Select Interface Type: FXS, FXO, or DID (FXS only for the MVP130-FXS) o Regeneration

 Choose how signal is regenerated; as Pulse or DTMF

o Inter Digit Timer

 Time the MultiVOIP waits between digits

o Message Waiting Indication is available if desired o Inter Digit Regeneration Timer

 Length of time between sent DTMF digits

• Flash Hook Options

o Generation (used in conjunction with FXO) o Detection Range (used in conjunction with FXS)

• Caller ID

o Bellcore is the only option available o CallerID Manipulation is available if needed

• Pass Through (opens an audio path through the MultiVOIP)

• FXS Options

o Set Ring Count (the number of rings allowed before call abandoned; default is 8) o Use Current Loss (MultiVOIP interrupts current to disconnect) o Generate Current Reversal (activates Answer/Disconnect Supervision to FXO)

• FXO Options (not available for the MVP130-FXS)

o Ring Count (set number of rings before MultiVOIP answers) o No Response Timer (set time to attempt call before abandoning) o Supervision Button (for call answering and disconnection settings)

 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)

 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)

• DID Options (not available for the MVP130-FXS)

o Start Modes (Immediate, Wink or Delay Dial) o Wink Timer (in milliseconds)

Chapter 3: Software Installation

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Chapter 3: Software Installation

Call Signaling

There are three choices for Call Signaling: H.323, SIP and SPP. 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.

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Figure 3-10: Signaling Protocols

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Actions:

• Configure your chosen Call Signal type

o H.323

o SIP

o SPP

Chapter 3: Software Installation

 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)  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)

 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

• 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 o User Defined tones can be created for use in conjunction with FXO Supervision with the Add

button

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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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Actions:

• Select Outbound Phone Book

o Select Add Entry o Accept Any Number may be selected to allow unmatched destinations an alternative o 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 Any digits that need to be added should be put in the Add Prefix field o Enter the IP address of the call destination (add a Description if you like) o Select a Protocol type

 For H.323:

• Enter Gateway settings

 For SIP:

• Select Transport Protocol, Proxy and URL if needed

 For SPP:

• Enter Registrar settings if needed

o The Advanced Button will allow an Alternate IP Address to be entered for outbound traffic

• Select Inbound Phone Book

o Select Add Entry o Accept Any Number for inbound traffic does not work when external routing devices are used o Enter any access digits followed by the local calling code in the Remove Prefix field o Enter any digits needed to access an outside line in the Add Prefix field o Select Hunting in the Channel Number field to have the VOIP use the next available channel o Add a description if you like o Call Forward may be set up (details available in Chapter 5) o Select Registration Option

• Repeat the Phone Book steps for any additional entries needed

Chapter 3: Software Installation

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 n.nn (where n 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 para meters 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 establ ished, 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.

Type of Configuration Info

Gathered:

IP info for VOIP unit

• IP address

• Gateway

• DNS IP (if used)

• 802.1p Prioritization (if used)

Interface Type

• FXS/FXO*

• DID-DPO

DID info (only if DID used)

• Wink

• Immediate

• Delay Dial

Country code Email address for VOIP (optional)

Reminder: Be sure to Save Setup after entering configuration values.

Configuration screen where info is entered:

Ethernet/IP parameters

Interface parameters

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

Regional parameters SMTP parameters

Info

Obtained?

Info

Entered?

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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.

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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Figure 4-1: Network parameters

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Chapter 4: Configuring your VOIP

Ethernet/IP Parameter Definitions

Field Name Values Description

Ethernet Parameters

Packet Prioritization (802.1p)

Frame Type Type II, SNAP Must be set to match network’s frame type. Default is Type II.

802.1p A draft standard of the IEEE about data traffic prioritization on Ethernet networks. The 802.1p

Call Control Priority 0-7, where 0 is

VOIP Media Priority 0-7, where 0 is

Others (Priorities) 0-7, where 0 is

VLAN ID 1 - 4094 The 802.1Q IEEE standard allows virtual LANs to be defined within a network.

IP Parameter fields Gateway Name alphanumeric Descriptor of current VOIP unit to distinguish it from other units in system. Enable DHCP Y/N

IP Address IP Mask Gateway

Table is continued on next page…

Y/N Select to activate prioritization under 802.1p protocol (described below).

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-areanetworks (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

lowest priority

disabled by default

n.n.n.n n.n.n.n n.n.n.n

Sets the priority for signaling packets.

Sets the priority for media packets.

Sets the priority for SMTP, DNS, DHCP, and other packet types.

This field identifies each virtual LAN by number.

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.

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.

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Ethernet/IP Parameter Definitions (continued)

Field Name Values Description Diff Serv Parameter fields

Call Control PHB

VOIP Media PHB

FTP Parameter fields

FTP Server Enable

DNS Parameter fields

Enable DNS Y/N

Enable SRV Y/N Enables ‘service record’ function. Service record is a category of data in the Internet

DNS Server IP Address

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. 0 – 63 default = 34

0 – 63 default = 46

Y/N Default = disabled See “FTP Server File Transfers” in Chapter 6

Default = disabled

n.n.n.n

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. 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.

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.

Enables Domain Name Space/System function where computer names are resolved using a worldwide distributed database.

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.

IP address of specific DNS server to be used to resolve Internet computer names.

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Voice/Fax

Setting the Voice/FAX Parameters. The Voice/Fax section needs to be set for your system. 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.

Modem relay is not supported in MVP130 and MVP130-FXS models. Instead, modem bypass is supported automatically when modems are used for communication. It is recommended to disable the FAX relay when doing modem bypass for a higher success rate.

The Voice/FAX Parameters settings are described in the tables below.

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Figure 4-2: Voice/Fax parameters

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Voice/Fax Parameter Definitions

Field Name Values Description

Default -- When this button is clicked, all Voice/FAX parameters are set to their default values. Select Channel 1-2 (210)

1-4 (410) 1-8 (810)

Copy Channel -- Copies the Voice/FAX attributes of one channel to another channel. Attributes can be

Voice Gain -- Signal amplification (or attenuation) in dB. Input Gain +31dB to

–31dB

Output Gain +31dB to

–31dB

DTMF Gain --

DTMF Gain, High Tones

DTMF Gain, Low Tones

DTMF Parameters

Duration (DTMF) 60 – 3000

DTMF In/Out of Band

Out of Band Mode

FAX Parameters

Fax Enable Y/N Enables or disables fax capability for a particular channel. Max Baud Rate

(Fax)

Fax Volume (Default =

-9.5 dB) Jitter Value (Fax) Default =

Mode (Fax) FRF 11;

Table is continued on next page…

+3dB to

-31dB & “mute”

+3dB to

-31dB & “mute”

ms Out of

Band, or Inband

RFC 2833, SIP Info

2400, 4800, 7200, 9600, 12000, 14400 bps

-18.5 dB to –3.5 dB

400 ms

T.38

Channel to be configured is selected here.

copied to multiple channels or all channels at once.

Modifies audio level entering voice channel before it is sent over the network to the remote VOIP. The default & recommended value is 0 dB.

Modifies audio level being output to the device attached to the voice channel. The default and 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.

Default value: -4 dB. Not to be changed except under supervision of Multi-Tech Technical Support.

Default value: -7 dB. Not to be changed except under supervision of Multi-Tech Technical Support.

When DTMF: Out of Band is selected, this setting determines how long each DTMF digit ‘sounds’ or is held. Default = 100 ms.

When DTMF Out of Band is selected, 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.

RFC2833 method. Uses an RTP mode defined in RFC 2833 to transmit the DTMF digits. SIP Info method. Generates dual tone multi frequency (DTMF) tones on the telephony call leg. The SIP INFO message is sent along the signaling path of the call. You must set this parameter per the capabilities of the remote endpoint with which the VOIP will communicate. The RFC2833 method is the more common of the two methods.

Set to match baud rate of fax machine connected to channel (see Fax machine’s user manual). Default = 14400 bps.

Controls output level of fax tones. To be changed only under the direction of MultiTech’s Technical Support.

Defines the inter-arrival packet deviation (in milliseconds) for the fax transmission. A higher value will increase the delay, allowing a higher percentage of packe ts to be reassembled. A lower value will decrease the delay allowing fewer packets to be reassembled.

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.

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Coder Parameters

Coder Manual or

Automatic

Selected Coder

Selected Coder: “Coder Priority”

Max bandwidth (coder)

Silence Compression

Echo Cancellation

Forward Error Correction

Table is continued on next page…

G.711 a/u law 64 kbps; G.726, @ 16/24/32/40 kbps; G.727, @ nine bps rates; G.723.1 @ 5.3 kbps, 6.3 kbps; G.729, 8kbps; Net Coder @

6.4, 7.2, 8, 8.8, 9.6 kbps

G.711, G.729

-or- G.729, G.711

11 – 128 kbps

Advanced Features

Y/N Determines whether silence compression is enabled (checked)

Y/N Determines whether echo cancellation is enabled (checked) for

Y/N Determines whether forward error correction is enabled

Chapter 4: Configuring your VOIP

Voice/Fax Parameter Definitions (continued)

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 is negotiated.

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. Default = G.723.1 @ 6.3 kbps, as required for H.323. Here 64K 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.

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.

This drop-down list enables you to select the maximum bandwidth allowed for this channel. The Max Bandwidth dropdown 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.

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).

this voice channel. Echo Cancellation removes echo and improves sound quality (default = on).

(checked) for this voice channel. Forward Error Correction enables some of the voice packets that were corrupted or lost to be recovered. FEC adds an additional 50% overhead to the total network bandwidth consumed by the voice channel (default = Off).

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Voice/Fax Parameter Definitions (continued)

Field Name Values Description

AutoCall/Offhook Alert

Parameters

Auto Call / Offhook Alert

Generate Local Dial Tone

Offhook Alert Timer 0 – 3000

Phone Number -- Phone number used for Auto Call function or Offhook Alert Timer function. This

Table is continued on next page…

AutoCall, Offhook Alert

Y/N Used for AutoCall only. If selected, dial tone will be generated locally while the

seconds

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.

call is being established between gateways. The capability to generate dial tone locally would be particularly useful when there is a lengthy network delay.

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 must correspond to an entry in the Outbound Phonebook of the local MultiVOIP and in the Inbound Phonebook of the remote MultiVOIP (unless a gatekeeper unit is used in the VOIP system).

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Voice/Fax Parameter Definitions (continued)

Field Name Values Description

Dynamic Jitter

Dynamic Jitter Buffer

Minimum Jitter Value 60 to 400

Maximum Jitter Value 60 to 400

Optimization Factor 0 to 12

Auto Disconnect

Automatic Disconnection

Jitter Value 1-65535

Call Duration 1-65535

Consecutive Packets Lost

Network Disconnection

1-65535

1 to 65535; Default = 30 sec.

Configurable Payload Type

Chapter 4: Configuring your VOIP

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.

The minimum dynamic jitter buffer of 60 milliseconds is the minimum delay that would be acceptable over a low jitter network. Default = 150 ms

The maximum dynamic jitter buffer of 400 milliseconds is the maximum delay tolerable over a high jitter network. Default = 300 ms 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.

The Automatic Disconnection group provides four options which can be used singly or in any combination.

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 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 defines the number of consecutive packets that are lost after which the call is automatically disconnected. Inactive by default. When active, default = 30

Specifies how long to wait before disconnecting the call when IP network connectivity with the remote site has been lost.

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 but not for H.323. Whenever you interoperate with older MultiVOIP products (i.e., earlier than release n.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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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. Here you will find the various parameters grouped and organized by interface type. In each field, enter the values that fit your particular setup. The screen below shows more options available than are actually used for clarity. Your settings will determine what fields are available.

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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 i n 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.

Message Waiting Indication

Inter Digit Regeneration Time

FXS Options fields

FXS Ring Count, FXS

Current Loss Y/N When enabled, the MultiVOIP will interrupt loop current in the FXS

Generate Current Reversal

Table is continued on next page…

in milliseconds

1-99 Maximum number of rings that the MultiVOIP will issue before giving

Y/N When selected, this option implements Answer Supervision and

When the time expires, the MultiVOIP will look in the outbound phonebook for the number entered and place the call accordingly. Default = 2.

See details below.

The length of time between the outputting of DTMF digits. Default = 100 ms.

up the attempted call.

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.

Disconnect Supervision to the FXO interface using current reversal to indicate events. Applicable only when FXS and FXO interfaces are connected back to back.

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FXS Loop Start Interface: Parameter Definitions (continued)

Field Name Values Description

Flash Hook Options fields

Generation Detection Range

Pass Through Enable

Caller ID fields

Type Bellcore The MultiVOIP currently supports only one implementation of Caller

Enable Y/N Caller ID information is a description of the remote calling party

CID Manipulation Enabled by

CID Mode Transparent,

-- Not applicable to FXS interface for Min. and

Max., 50 - 1500

milliseconds Y/N When enabled, this parameter creates an open audio path through the

default with Caller ID enable above Disable

User CID, Prefix, Suffix

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

MultiVOIP. If the Pass-Through feature is enabled, the AutoCall feature must be enabled for this VOIP channel in the Voice/Fax Parameters screen

ID. That implementation is Bellcore type 1 with Caller ID placed between the first and second rings of the call.

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. 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.

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.

Transparent without any manipulation. User CID: the CID received from PSTN will be replaced by this User CID value. Prefix Suffix

: the CID received from PSTN will be sent out as such,

: the CID received from PSTN will be prefixed with this value. : the CID received from PSTN will be suffixed with this value.

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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 condition s 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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FXO Parameters

The parameters applicable to the FXO telephony interface type are shown in the figure below and described in the table that follows.

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Figure 4-6: FXO parameters

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Chapter 4: Configuring your VOIP

FXO Interface: Parameter Definitions

Field Name Values Description

Interface Type FXO Enables FXO functionality

Dialing Options

Regeneration Pulse, DTMF Determines whether digits generated and sent out will be pulse tones

Inter Digit Timer 1 to 10 seconds This is the length of time that the MultiVOIP will wait between digits.

Message Waiting Indication

Inter Digit Regeneration Time

FXO Options

FXO Ring Count 1-99 Number of rings required before the MultiVOIP answers the incoming No Response

Timer

Flash Hook Options fields

Generation 50 - 1500

Detection Range

Caller ID fields

Caller ID Type Bellcore The MultiVOIP currently supports only one implementation of Caller

Caller ID enable Y/N Caller ID information is a description of the remote calling party

CID Manipulation Enabled by

CID Mode Transparent,

50 to 20,000 milliseconds

1 – 65535 (in seconds)

milliseconds

default with Caller ID enable above Disable

User CID, Prefix, Suffix

or DTMF.

When the time expires, the MultiVOIP will look in the phonebook for the number entered.

Default = 2.

Not applicable to FXO interface

The length of time between the outputting of DTMF digits. Default = 100 ms.

call. Length of time before call connection attempt is abandoned.

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. Not applicable to FXO.

ID. That implementation is Bellcore type 1 with caller ID placed between the first and second rings of the call.

Transparent without any manipulation. User CID CID value. Prefix: Suffix

: the CID received from PSTN will be sent out as such,

: the CID received from PSTN will be replaced by this User

the CID received from PSTN will be prefixed with this value.

: the CID received from PSTN will be suffixed with this value.

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FXO Supervision

When the selected Interface type is FXO, the Supervision button is active. Click on this button to access call answering supervision parameters and call disconnection para meters that relate to the FXO interface type.

The table below describes the settings for FXO Supervision.

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Figure 4-7: FXO Supervision

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Chapter 4: Configuring your VOIP

FXO Supervision Parameter Definitions

Field Name Values Description

Answer Supervision fields

Current Reversal Y/N When this option is selected, the FXO interface sends notice to make

Answer Delay Y/N When this option is selected, the FXO interface sends the conn ection

Answer Delay Timer

Tone Detection Y/N When selected, call disconnection will be triggered by a tone Available Tones dial tone,

Answer Tones any tone from

Disconnect Supervision fields

Current Reversal Y/N Disconnection to be triggered by reversal of current from the PBX. Current Loss Y/N Disconnection to be triggered by loss of current. That is, when Current

Current Loss Timer 200 to 2000

Silence Detection Enable

Silence Detection Type

Silence Timer in seconds

Table is continued on next page…

1 – 65535 (in seconds)

ring tone, busy tone, unobtainable tone (fast busy), survivability tone, re-order tone

Available Tones list

(in milliseconds) Y/N Enables/disables silence-detection method of supervising call

One-Way or Two-Way

integer value Duration of silence required to trigger disconnection.

connection upon detecting current reversal from the PBX (which occurs when the called extension goes off hook).

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.

When Answer Delay is enabled, this value determines when the FXO interface sends the connection notice.

sequence List from which tones can be chosen to signal call answer.

Currently chosen call-answer supervision tone.

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.

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. Determines the interval after detection of current loss at which the call will be disconnected.

disconnection. Disconnection to be triggered by silence in one direction only or in

both directions simultaneously

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FXO Supervision Parameter Definitions (continued)

Field Name Values Description

Disconnect Supervision fields

DTMF Tone Enables supervision of call disconnection using DTMF tones.

DTMF Tone Pairs

Low Tones

High Tones

1 2 3 A 4 5 6 B 7 8 9 C * 0 # D

1209Hz 1336Hz 1447Hz 1633Hz

697Hz 770Hz 852Hz 941Hz

Disconnect Tone Sequence

Tone Detection Y/N Enables supervision of call disconnection by detecting cessation of a

Available Tones

Disconnect Tones

1st tone pair + 2nd tone pair

dial tone, ring tone, busy tone, unobtainable tone (fast busy), survivability tone, re-order tone

any tone from Available Tones list

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.

pre-specified tone from the PBX. List from which tones can be chosen to signal call disconnection.

Currently chosen disconnection supervision tone.

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DID Parameters

The parameters applicable to the Direct Inward Dial (DID) telephony interface ty pe 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, dialpulse terminating) is not supported. The –FXS model does not support DID.

Figure 4-8: DID parameters

DID Interface Parameter Definitions

Field Name Values Description

Interface DID-DPO Enables the customer-premises side of DID functionality

DID Options

Start Modes Immediate Start,

Wink Start, Delay Dial

Wink Timer (in ms)

Dialing Options Inter Digit Timer Integer values,

Message Waiting Indication

Inter-Digit Regeneration Timer

Integer values, in milliseconds

in seconds

-- Not applicable to DID-DPO interface.

Integer values, in milliseconds

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.

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.

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.”

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.

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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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.

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 4-9: 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. Ma y need to be enabled / disabled for

compatibility with third-party VOIP gateways. Signaling Port Register with

Gatekeeper Allow Incoming

Calls Through Gatekeeper Only

GateKeeper RAS Parameters

Primary GK -- This is the preferred gatekeeper for controlling the traffic of the current VOIP. Alternate GK 1 and 2 IP Address RAS Port 1719 Well-known port number for GateKeepers. Must match port number (1719).

Gatekeeper Name RAS TTL Value

Gatekeeper Discovery Polling Interval Use Online Alternate Gatekeeper List

H.323 Version 4 Options H.323

Multiplexing H.245 Tunneling

(Tun)

Parallel H.245 (FS + Tun)

Annex –E (AE) Y/N Multiplexed UDP call signaling transport. Annex E is helpful for high-volume VOIP

port

Y/N Check this field to have traffic on current VOIP gateway controlled by a

Y/N When selected, incoming calls are accepted only if those calls come through the

-- A first and a second alternate gatekeeper can be specified for use by the current

n.n.n.n

alphanumeric seconds

integer 60 - 300

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.

Y/N Signali ng for multiple phone calls can be carried on a single port rather than Y/N H.245 messages are encapsulated within the Q.931 call-signaling channel.

Y/N FS (Fast Start) is a Q.931 feature of H.323v2 to hasten call setup as well as ‘pre-

Default: 1720 (H.323)

gatekeeper.

VOIP for situations where the Primary GK is busy or otherwise unavailable.

IP address of the GateKeeper.

Optional. The name of the GateKeeper with which this MultiVOIP is trying to

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

The interval between the VOIP gateway’s successive attempts to connect to and

be governed by a higher level gatekeeper. The Primary GK is the highest level

gatekeeper. Alternate GK1 is second; Alternate GK2 is the lowest.

opening a separate signaling port for each. This conserves bandwidth resources.

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.

opening’ the media channel before the CONNECT message is sent. This preopening is a requirement for certain billing activities. Under Parallel H.245 FS + Tun, this Fast Connect feature can operate simultaneously with H.245 Tunneling.

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).

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SIP

Session Initiation Protocol is the second option available for application layer control of the MultiVOIP. The fields are detailed in the table below.

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Figure 4-10: SIP call signaling

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SIP Call Signaling Parameter Definitions

Field Name Values Description

SIP Proxy Parameters

Signaling Port

Use SIP Proxy Y/N Allows the MultiVOIP to work in conjunction with a proxy server. Allow Incoming

Calls Through SIP Proxy Only

Primary Proxy -- This is the preferred SIP proxy server for controlling the traffic of the current VOIP. Alternate Proxy 1 and 2 Proxy Domain Name / IP Address

Append SIP Proxy Domain Name in User ID

Port Number Default

Subscriber Default

Username Password Re-Registration

Time

Proxy Polling Interval

TTL Value SIP proxy

port

Y/N When selected, incoming calls are accepted only if those calls come through the

-- A first and a second alternate SIP proxy server can be specified for use by the

n.n.n.n

Y/N When checked, the domain name of the SIP Proxy serving the MultiVOIP gateway

port

name

password

10–65535 seconds

60 - 300 The interval be tween the VOIP gateway’s successive attempts to connect to and

“Time to Live” value. (in seconds)

Port number on which the MultiVOIP UserAgent software module will be waiting for any incoming SIP requests. Default = 5060

proxy.

VOIP for situations where the Primary proxy server is otherwise unavailable. Network address of the proxy server that the VOIP is using.

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.

Logical port number for proxy communications. Default = 5060 This is used as the default end point register with a Proxy.

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 for proxy server function. See “Default Username” description above. This is the timeout interval for registration of the MultiVOIP with a SIP proxy server.

The time interval begins the moment the MultiVOIP gateway registers with the SIP proxy server and ends at the time specified by the user in the Re-Registration Time field (this field). When/if registration lapses, call traffic routed to/from the MultiVOIP through the SIP proxy server will cease. However, calls in progress will continue to function until they end.

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. As soon as a MultiVOIP gateway registers with a SIP proxy server (allowing the proxy server to control its call traffic) a countdown timer begins. The TTL Value is the interval of the countdown timer. Before the TTL countdown expires, the MultiVOIP gateway needs to register with the gatekeeper in order to maintain the connection. If the MultiVOIP does not register before the TTL interval expires, the MultiVOIP gateway’s registration with the proxy server will expire and the proxy server will no longer permit call traffic to or from that gateway. Calls in progress will continue to function even if the gateway becomes de-registered.

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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.

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Figure 4-11: SPP call signaling

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SPP Call Signaling Parameter Definitions

Field Name Values Description

Mode Direct,

Client, or Registrar

General Options

Port

Re-transmission 50 -

Max Retransmission

Client Options

Primary Registrar -- This is the preferred SPP registrar gateway for controlling the traffic of the current Alternate

Registrar 1 and 2 Registrar IP

Address Registrar Port 100 00 or

Polling Interval integer

Registrar Options

Keep Alive 30 – 300

Proxy/NAT Device Parameters Behind Proxy/NAT device

Proxy/NAT Device Parameters – Public IP Address

port

5000ms 0 - 20 Number of times the VOIP will re-transmit a lost packet (if no acknowledgment

-- A first and a second alternate SPP Registrar gateway can be specified for use by

n.n.n.n

other

60 - 300

(seconds)

Y/N Enables MultiVOIP (running in SPP Registrar mode) to operate ‘behind’ a

n.n.n.n

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.

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.) 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.)

has been received). (Default value = 3) Client Option fields are active only in registrar/client mode and only for client VOIP

units. VOIP. the current VOIP for situations where the Primary Registrar gateway is busy or

otherwise unavailable. This is the IP address of the registrar VOIP to which this client is assigned. (Default value = 0.0.0.0; effectively, there is no useful default value.)

This is the port number of the registrar VOIP to which this client is assigned. (Default port number = 10000.)

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. Time-out duration before a registrar wills 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.

proxy/NAT device (NAT = Network Address Translation).

The public IP address of the proxy/NAT device which the MultiVOIP is behind.

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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 MVP130 MultiVOIPs only have limited SNMP functionality available. If this is something you wish to use, please contact Multi-Tech Support for assistance.

Figure 4-12: SNMP parameters screen

The SNMP Parameter fields are described in the table below.

SNMP Parameter Definitions

Field Name Values Description

Enable SNMP Agent

Trap Manager Parameters

Address Community Name

Port Number 162 The default port number of the SNMP manager receiving the traps is the Community

Name 1 Permissions Read-Only, Community

Name 2 Permissions Read-Only,

Y/N Enables the SNMP code in the firmware of the MultiVOIP. This must be

enabled for the MultiVOIP to communicate with and be controllable by the MultiVoipManager software. Default: disabled

n.n.n.n

-- A “community” is a group of VOIP endpoints that can communicate with each

Length = 19 characters (max.) Case sensitive.

Read/Write Length = 19 characters (max.) Case sensitive.

Read/Write

IP address of MultiVoipManager PC. 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.

standard port 162. First community grouping.

If this community needs to change MultiVOIP settings, select Read/Write. Otherwise, select Read-Only to view settings. Second community grouping

If this community needs to change MultiVOIP settings, select Read/Write. Otherwise, select Read-Only to view settings.

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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 4-13: Regional parameters

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“Regional Parameter” Definitions

Field Name Values Description

Country/Region USA,

Japan, UK, Custom

Advisory screen This message screen appears whenever the

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). 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:

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

Intercept tone is applicable only when the FXS telephony interface has

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, unobtainable tone (fast busy), survivability tone,

re-order tone Frequency 1 freq. in Hertz Lower frequency of pair. Frequency 2 freq. in Hertz Higher frequency of pair. Gain 1 gain in dB

+3dB to –31dB

and “mute” setting

Gain 2 gain in dB

+3dB to –31dB

and “mute” setting Cadence

(ms) On/Off

Custom (button) --

Table is continued on next page…

n/n/n/n

four integer time

values in

milliseconds; zero

value for dial-tone

indicates continuous

tone

Type of telephony tone-pair for which frequency, gain, and cadence are being presented.

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 or FXS port. Default: -16dB

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 or FXO port. Default: -16dB

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.

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.

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“Regional Parameter” Definitions (continued)

Field Name Values Description

Country Selection for Built-In Modem (not applicable to MVP130 or 130-FXS)

User Defined Tones fields

Type column

Frequency 1 Freq. in Hertz Lower frequency of pair. Frequency 2 Freq. in Hertz Higher frequency of pair. Gain 1 +3dB to –31dB

Gain 2 +3dB to –31dB

Cadence (ms) On/Off

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.

alphanumeric name

and “mute” setting

n/n/n/n four integer time values in milliseconds; (zero value indicates continuous tone)

Name of supervisory tone pair. Cannot be same as name of any standard tone pair.

Amplification factor of lower frequency of pair. This applies to any supervisory tones that the MultiVOIP outputs as audio to the FXS or FXS port. Default: “Mute”

Amplification factor of higher frequency of pair. This applies to any supervisory tones that the MultiVOIP outputs as audio to the FXS or FXO port. Default: “Mute”

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.

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 all ows 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

ring tone, ‘unobtainable’ 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. Frequency 2 Frequenc y in Hertz Frequency of higher tone of pair. Gain 1 +3dB to –31dB

and “mute” setting

Gain 2 +3dB to –31dB

and “mute” setting

Cadence 1 integer time value in

milliseconds; zero value for dial-tone indicates

continuous tone Cadence 2 duration in milliseconds Cadence 2 is duration of first “off” period in signaling cadence. Cadence 3 duration in milliseconds Cadence 3 is duration of second “on” period in signaling cadence. Cadence 4 duration in milliseconds Cadence 4 is duration of second “off” period in the signaling cadence.

Identifies the type of telephony signaling tone for which frequencies are being specified.

This outbound tone pair enters the MultiVOIP at the input port. 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 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 On/off pattern of tone durations used to denote phone ringing, phone busy, dial tone (“0” indicates continuous tone) survivability and re-order. Cadence 1 is duration of first period of tone being “on” in the cadence of the telephony signal.

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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:

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Figure 4-14: SMTP parameters

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“SMTP Parameters” Definitions

Field Name Values Description

Enable SMTP Y/N In order to send log re ports by email, this box must be checked. Ho wever, to

enable SMTP functionality, you must also select “SMTP” in the Logs

screen. Requires Authentication

Login Name Password Mail Server IP Address Port Number 25 25 is a standard port number for SMTP. Mail Type text or html Mail type in which log reports will be sent. Subject text User specified. Subject line that will appear for all emailed log reports for

Reply-To Address

Recipient Address

Mail Criteria

Number of Records integer This is the number of log records that must accumulate to trigger the Number of Days integer This is the number of days that must pass before triggering the sending of a

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.

alpha-numeric alpha-numeric n.n.n.n

email address

This is the User Name for the MultiVOIP unit’s email account.

This is the mail server’s IP address. This mail server must be accessible on the IP network to which the MultiVOIP is connected.

this MultiVOIP unit.

User specified. This email address functions as a source email identifier for

the MultiVOIP, which, of course, cannot usefully receive email messages.

The Reply-To address provides a destination for returned messages

indicating the status of messages sent by the MultiVOIP (esp. to indicate

when log report email was undeliverable or when an error has occurred).

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.

sending of a log-summary email.

log-summary email.

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The SMTP Parameters dialog box has a secondary dialog box, accessed by the Select Fields button, which 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. Channel Number Duration Length of call. Packets

Packets Sent T otal packets sent in call. Bytes Bytes Sent Total bytes sent in call. Coder Voice Coder /Compression Rate used Packets Lost Packets lost in call. Prefix

Outbound Digits Received

Call Status Successful or unsuccessful. Call Direction Indicates call’s originating party. Server Details

Disconnect Reason

Gateway Number IP Address IP address where call originated. IP Address IP address where call was completed

Descript Identifier of site where call orig inated. Descript Identifier of site where call was Options When selected, log will not Silence

Data channel carrying call. Call Mode Voice or fax.

The DTMF dialing digits received by this

gateway from the remote gateway

presuming that DTMF is set to "Out of

Band."

The IP address of the traffic control

server (if any) being used (whether an

H.323 gatekeeper, a SIP proxy, or an

SPP registrar gateway) will be displayed

here if the call is handled through that

server.

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.

From Details To Details

Originating gateway Gateway

Compression and Forward Error

Correction by call originator.

Start Date,

Time

Received Received

Matched

Call Type Indicates the Call Signaling protocol

DTMF Capability

Outbound Digits Sent

Name

Options When selected, log will not use Silence

Date and time the phone call began.

Total packets received in call. Total bytes received in call.

for call will be listed in log. When selected, the phonebook prefix matched in processing the call will be listed in log.

used for the call (H.323, SIP, or SPP).

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 dialing digits sent by this gateway to the remote gateway presuming that DTMF is set to "Out of Band.”

Completing or answering gateway

or answered. completed or answered. Compression and Forward Error

Correction by party answering call.

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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 4-15: 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 1 - 65535 TDM time slot at which RADIUS accounting information will be transmitted

Retransmission Interval

Number of Retransmissions

Shared Secret alpha-numeric Client encryption key for the current VOIP unit. Select Attributes

(button)

n.n.n.n

0 - 255

-- Gives access to RADIUS Attributes screen. On Attributes screen, one can

IP address of the RADIUS server that handles accounting (billing) for the current MultiVOIP unit.

and received. 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.

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, which 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 se rver.

“RADIUS Attributes” Definitions

Field Description Field Description

Select All Log report to include all fields

shown. Channel Number Duration Length of call. Packets Received Total packets received in call. Packets Sent Total packets sent in call. Bytes Received Total bytes received in call. Bytes Sent Total bytes sent in call. Coder Voice Coder /Compression Rate used for

Packets Lost Packets lost in call. Prefix Matched When selected, the phonebook prefix

Outbound Digits Sent

Server Details The IP address of the traffic control server being used will be displayed here if the call is handled

Gateway Number IP Address IP address where call originated. IP Address IP address where call was completed/answered. Descript Identifier of where call originated. Descript Identifier of where call was completed/answered.

Options When selected, log will not use

Data channel carrying call. Call Mode Voice or fax.

DTMF digits received by this

gateway from remote gateway

(if that DTMF set to "Out of

Band").

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

Originating gateway Gateway

Silence Compression and Forward Error Correction by call originator.

Start Date, Time Date and time the phone call began.

call will be listed in log. matched in processing the call will be listed

in log.

Call Status Successful or unsuccessful.

Completing or answering gateway

Name

Options When selected, log will not use Silence

Compression and Forward Error Correction by party answering call.

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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 4-16: 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

Filters (button) Click to access secondary screen on where console messages can be

Turn Off Logs Y/N Check to disable log-reporting function. Logs Buttons Only one of these three log reporting methods, GUI, SMTP, or SNMP, may be chosen.

GUI

SNMP

SMTP

SysLog Server Enable

IP Address Port 514 Logical port for SysLog Server. 514 is commonly used. Online Statistics

Updation Interval

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.

included/excluded by category and on a per-c hannel basis.

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.

•

Y/N This box must be checked if logging is to be done in conjunction with a SysLog Server

program.

n.n.n.n

integer Set the interva l (in seconds) at which logging information will be updated.

IP address of computer, in VOIP network, on which SysLog Server program is running.

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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) is a protocol for assisting devices behind a NAT firewall or router with their packet routing.

Figure 4-17: 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) Port (Server;

NAT/STUN)

Keep Alive (Timers; NAT/STUN)

n.n.n.n port;

default= 3478

60 – 3600 (seconds)

IP address of the STUN server.

The data port (TDM time slot) at which STUN info will be transmitted and

received.

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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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 allo ws 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.

The Supplementary Services fields are described in the tables below.

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Figure 4-18: Supplementary Services

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Supplementary Services Parameter Definitions

Field Name Values Description

Select Channel Channel 1 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

Call Hold Enable Y/N Select to enable Call Hold function in VOIP unit.

Hold Sequence phone keypad

characters

Call Waiting Enable Y/N Select to enable Call Waiting function in VOIP unit. Retrieve Sequence Phone keypad

characters, two characters in length

Call Name Identification Enable

Table is continued on next page…

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.)

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 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.

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 #).

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.

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Supplementary Services Definitions (continued)

Field Name Description

Calling Party, Allowed Name Type (CNI)

Alerting Party, Allowed Name Type (CNI)

Busy Party, Allowed Name Type (CNI)

Connected Party, Allowed Name Type (CNI)

Caller ID This is the iden tifier of a specific channel of the ‘home’ VOIP unit. The Caller Id field typically

Default When this button is clicked, all Supplementary Service parameters are set to their default valu es. Copy Channel Copies the Supplementary Service attributes of one channel to another channel. Attributes can be

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 ca ll. 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.

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.

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.

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.

describes a person, office, or location, for example, “Harry Smith,” or “Bursar’s Office,” or “Barnesville Factory.”

copied to multiple channels or all channels at once.

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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.

Chapter 4: Configuring your VOIP

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Figure 4-19: COM port setup

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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 erro r message will appear.

Figure 4-20: Error pop-up

To change the COM port setting, use the COM Port Setup dialog box, by going to the Connection pulldown 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 4-21: 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 basi s 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, it is highly desirable to use the same VOIP protocol for all VOIP units in the system. SPP is a nonstandard protocol developed by Multi-Tech. SPP is not compatible with the “Proprietary” protocol used in MultiTech’s earlier generation of VOIP gateways.

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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 n.nn | 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

Technician in Seattle (area 206) must set up one VOIP there, another in Chicago (area 312, downtown).

Answer: Write down 312.

Euro, National Call Example Euro, International Call

Example

Technician in central London (area 0207) to set up VOIP there, another in Birmingham (area

0121).

Answer: write down 0121.

Technician in Rotterdam (country 31; city 010) to set up one VOIP there, another in Bordeaux (country 33; area 05).

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

Seattle/Chicago system. Seattle VOIP works with PBX

that uses “8” for all VOIP calls. “1” must immediately precede area code of dialed number.

Answer: write down 81.

Euro, National Call Example Euro, International Call

Example

London/Birmingham system. London VOIP works with PBX

that uses “9” for all out-ofbuilding calls whether by VOIP or by PSTN. “0” must immediately precede area code of dialed

Rotterdam/Bordeaux system. Rotterdam VOIP works with PBX where “9” is used for all out-ofbuilding calls. “0” must precede all international calls.

Answer: write down 90.

number.

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Answer: write down 90.

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5. In the “Destination Pattern” field of the Add/Edit Outbound Phonebook screen, enter the digits from step 4 followed by the digits from step 3.

North America, Long-Distance Example

Seattle/Chicago system.

Answer: enter 81312 as

Destination Pat-tern in Outbound Phone-book of Seattle VOIP.

Euro, National Call Example Euro, International Call

Example

London/Birmingham system. Leading zero of Birmingham area

code is dropped when combined with national-dialing access code. (Such practices vary by

Rotterdam/Bordeaux system.

Answer: enter 903305 as

Destination Pattern in Outbound Phonebook of Rotterdam VOIP.

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.

Answer: enter 8 in “Remove Prefix”

field of Seattle Outbound Phonebook.

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.

Euro, National Call Example Euro, International Call

Example

London/Birmingham system.

Answer: enter 9 in “Remove Prefix”

field of London Outbound Phonebook.

Rotterdam/Bordeaux system.

Answer: enter 9 in “Remove Prefix”

field of Outbound Phonebook for Rotterdam VOIP.

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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Inbound Phonebook

1. Open the MultiVOIP program. (Start | MultiVOIP n.nn | 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

Seattle/Chicago system. Seattle is area 206. Chicago

employees must dial 81 before dialing any Seattle number on the VOIP system.

Answer: 1206 is prefix to be

removed by local (Seattle) VOIP.

Euro, National Call Example Euro, International Call

Example

London/Birmingham system. Inner London is 0207 area.

Birmingham employees must dial 9 before dialing any London number on the VOIP system.

Answer: 0207 is prefix to be

removed by local (London) VOIP.

Rotterdam/Bordeaux system. Rotterdam is country code 31, city code 010. Bordeaux employees must dial 903110 before dialing any Rotterdam number on the VOIP system.

Answer: 03110 is prefix to be

removed by local (Rotterdam) VOIP.

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

Seattle/Chicago system. On Seattle PBX, “9” is used to get an

outside line. Answer: 9 is prefix to be added by

local (Seattle) VOIP.

Euro, National Call Example Euro, International Call

Example

London/Birmingham system. On London PBX, “9” is used to get

an outside line. Answer: 9 is prefix to be added by

local (London) VOIP.

Rotterdam/Bordeaux system. On Rotterdam PBX, “9” is used to get an outside line.

Answer: 9 is prefix to be added by

local (Rotterdam) VOIP.

5. In the “Channel Number” field, enter “Hunting.” A “hunting” value means the VOIP unit will assign the call to the first available channel. If desired, specific channels can be assigned to specific incoming calls (i.e., to any set of calls received with a particular incoming dialing pattern).

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

Seattle/Chicago system. Possible Description:

Free Seattle access, all employees

Euro, National Call Example Euro, International Call

Example

London/Birmingham system. Possible Description:

Local-rate London access, all employees

Rotterdam/Bordeaux system. Possible Description:

Local-rate Rotterdam access, all employees

7. Repeat steps 2-6 for each inbound phonebook entry. When all entries are complete, go to step 8.

8. Click OK to exit the inbound phonebook screen.

9. Click on Save Setup. Highlight Save and Reboot. Click OK.

Your starter inbound phonebook configuration is complete.

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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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Add/Edit Outbound Phone Book

Chapter 5: Phonebook Configuration

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 fields of the Add/Edit Outbound Phone Book screen are described in the table below.

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Figure 5-2: Add/Edit screen

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Chapter 5: Phonebook Configuration

Add/Edit Outbound Phone Book: Field Definitions

Field Name Values Description

Accept Any Number

Destination Pattern

Total Digits as needed Number of digits the phone user must dial to reach specified destination. This

Remove Prefix dialed digits Portion of dialed number to be removed before completing call to destination. Add Prefix dialed digits Digits to be added before completing call to destination. IP Address

Description alp ha-numeric Describes the facility or geographical l ocation at which the call will be

Protocol Type SIP or H.323

H.323 fields Use Gatekeeper Y/N Indicates whether or not gatekeeper is used. Gateway H.323

ID Gateway Prefix numeric This number becomes registered with the GateKeeper. Call requests sent to

H.323 Port Number

Table is continued on next page…

Y/N W hen 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.

prefixes, area codes, exchanges, line numbers, extensions

n.n.n.n

or SPP

alpha-numeric The H.323 ID assigned to the destination MultiVOIP. Only valid if “Use

1720 This parameter pertains to Q.931, which is the H.323 call signaling protocol

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.

field not used in North America

The IP address to which the call will be directed if it begins with the destination pattern given.

completed. Indicates protocol to be used in outbound transmission. Single Port Protocol

(SPP) is a non-standard protocol designed by Multi-Tech.

Gatekeeper” is enabled for this entry.

the gatekeeper and preceded by this prefix will be routed to the VOIP gateway.

for setup and termination of calls (aka ITU-T Recommendation I.451). H.323 employs only one “well-known” port (1720) for Q.931 signaling. If Q.931 message-oriented signaling protocol is used, 1720 must be chosen as the H.323 Port Number.

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Chapter 5: Phonebook Configuration

Add/Edit Outbound Phone Book: Field Definitions (continued)

Field Name Values Description

SIP Fields

Use Proxy Y/N Select if proxy server is used. Transport

Protocol

SIP Port Number

SIP URL sip.userphone@hostserver,

Use Registrar

Port Number numeric When operating in “Registrar/Client” mode, this is the port by which the

Alternate Phone Number Remote Device is [legacy VOIP] Advanced button

TCP or UDP

5060 or other *See RFC 3087 (“Control of Service Context using SIP Request-URI,” by the Network Working Group).

where “userphone” is the telephone number and “hostserver” is the domain name or an address on the network

SPP Fields

Y/N

numeric Phone number associated with alternate IP routing.

Y/N When checked, this MultiVOIP can operate with ‘first-generation’

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.

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 connectionoriented protocol with greater overhead, but having acknowledgment and guarantees delivery and packet sequence integrity.

The SIP Port Number is a UDP logical port number. The VOIP will “listen” for SIP messages at this logical port. If SIP is used, 5060 is the default, standard or “well known” port number to be used. If 5060 is not used, then the port number used is that specified in the SIP Request URI (Universal Resource Identifier).

Looking similar to an email address, a SIP URL identifies a user's address.

In SIP communications, each caller or recipient 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.

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.

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.

MultiVOIP units in the same IP network. These include MVP110/120/200/400/800.

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Chapter 5: Phonebook Configuration

Clicking on the Advanced button brings up the Alternate Routing secondary screen. This fea ture 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

Field Name Values Description

Alternate IP Address

Round Trip Delay

n.n.n.n

Default is 300 milliseconds

Alternate destination for outbound data traffic in case of excessive delay in data transmission.

The Round Trip Delay is the criterion for judging when a data pathway is considered blocked. When the delay exceeds the threshold specified her e, 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.

3. Call diverts to Alt IP address in voip accessing PSTN line.

FXS

1. Call originates.

FXO

VOIP

PSTN Line

NETWORK

2. IP network fails.

VOIP

4. Call completed via PSTN.

PBX

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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Add/Edit Inbound Phone Book

Chapter 5: Phonebook Configuration

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Figure 5-6: Add/Edit Inbound Phone Book

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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

Remove Prefix dialed digits portion of dialed number to be removed before completing call to destination Add Prefix dialed digits digits to be added before completing call to destination Channel

Number Description -- Describes the facility or geographical location at which the call originated.

Call Forward Parameters

Enable Y/N Click the check-box to enable the call-forwarding feature. Forward Condition

Forward Destination

Ring Count integer When “No Response” is condition for forwarding calls, this determin es how many

Registration Option Parameters

Y/N When checked, “Any Number” appears as the value in the Remove Prefix field.

The Any Number feature of the Inboun d 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.

(often a local PBX)

(often a local PBX) channel, or “Hunting”

Unconditional, Busy, No Response

IP address, phone number, port number, etc

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.

In a SIP VOIP system, gateways can register with the SIP Proxy. In an SPP VOIP system, gateways can register with the SPP Registrar VOIP unit.

Channel number to which the call will be assigned as it enters the local telephon y equipment (often a local PBX). “Hunting” directs the call to any available channel.

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.

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,

(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

unanswered rings are needed to trigger the forwarding.

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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.

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.

Chapter 5: Phonebook Configuration

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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Phonebook Examples

North America

The following example demonstrates how Outbound and Inbound Phoneboo k 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

Suburbs

763

SIte

Mpls

612

SW Suburbs

952

Outstate MD

St. Paul

& Suburbs

651

...

Company

VOIP/PBX

SIte

Baltimore

410

Figure 5-7: North America example

Baltimore/

Overlay

443

An outline of the equipment setup in both offices is shown below.

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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.

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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Figure 5-10: Minneapolis example

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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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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.

VOIP/PBX Site

London

Wren Clothing Co.

VOIP/PBX Site

Amsterdam

The

Netherlands

Wren Clothing Co.

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 Lon don. 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 applie s 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

Bluebird Zipper Co.

London

Wren Clothing Co.

VOIP/PBX Site

London

Wren Clothing Co.

VOIP/PBX Site

Amsterdam

The

Netherlands

Wren Clothing Co.

VOIP/PBX Site

Paris

Calls at London local rates Local Calling Area

France

Figure 5-14: Local calling area

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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 g reater 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.

Chapter 5: Phonebook Configuration

France Country Code: 33

Lille

Paris: Area 01

Rouen

Nantes

Reims

Strasbourg

Figure 5-15: UK & France codes

The Netherlands

Country Code: 31

Texe l 02 22

Den Helder 0223

Beverwijk 0251

Haarlem 023

Aalsmeer0297

070

The Hague

Rotterdam

0299 Purmerend

020 Amsterdam

0294 Weesp

010

058

Leeuwarden

038 Zwolle

Arnhem

026

050

Groningen

053

Enschede

Bordeaux

Toulouse

Lyon

Marseille

0118

Middelburg

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040

Eindhoven

043

Maastricht

Figure 5-16: Netherlands codes

Page 84

An outline of the equipment setup in these three offices is shown below.

Chapter 5: Phonebook Configuration

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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.

NOTE: Commas are allowed in the Inbound Phonebook, but not in the Outbound Phonebook. Comma s denote a

brief pause for a dial tone, allowing time for the PBX to get an outside line.

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Figure 5-19: London example inbound

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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.

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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.

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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:

Call is received

here.

xxxyyyzzzz J.Q. Public

Display shows:

FXS

CID

Terminating

VoIP

Clock:

5-31,

1:42pm

CID Number: 763-555-8794 CID Name: Melvin Jones

Tim e 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.

CID Flow

CID

Generating

VoIP

FXO

Network

H.323 or SPP

Protocol

Figure 5-24: Caller ID example 1

Central Office

with

standard telephony

Caller ID ser v ic e

Call originates here

at 1:42pm, May 31.

xxxyyyzzzz J.Q. Public

phone of:

Melvin Jones 763-555-8794

Figure 5-25: VOIP Caller ID Case #1 – Call, through telco central office with standard CID, enters VOIP system.

Call is received

here.

xxxyyyzzzz J.Q. Public

FXS

CID

Terminating

VoIP

Clock:

7/10, 4:19pm

Display shows:

CID Number: 423 CID Name: Anoka-Whse-VP3

Time Stamp: Date: 7/10 Time: 4:19pm

In x.06 release, when SIP protocol is used,

CID Name field will duplicate value in CID Number fi eld.

CID Flow

Network

H.323 Protocol

CID

VoIP

Ch2 Ch3

Ch4

Ch1

FXO

Generating

Phone Book Configuration

Gateway Name:

Q.931 Parameters

Gatekeeper RAS Parameters

Anoka-Whse-VP3

Call originates here

Central Office

without

standard telephony

Caller ID service

at 4:19pm, July 10.

xxxyyyzzzz

J.Q. Public

phone of:

Wilda Jameson 763-555-4071

Inbound Phone Book

Remove Prefix Add Prefix Forward/Addr

423 748

{Channel 2}

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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Call is received

here.

x xxyyy zzzz J.Q. Publ ic

Display shows:

FXS

Term inating

VoIP

Clo ck:

1 5:26, 5-31

CID Flow

Network

SPP Protocol

Ch1

Generating

VoIP

Ch2 Ch3 Ch4

FXO

st andard tele phony

Central Office

without

Cal l er ID ser vi c e

Call originates here

at 5:47p m , Sept 27.

xx xyyyz zzz J.Q. Publ ic

phone of:

Henry Brampton 763-555-4077

CI D Numbe r: 423 CID Name: Shipping Dept Time St amp: Date: 0927

Time: 1747

... if “De s c rip tion” f ield in Add/Edit Inbound Phon e Book is used

CI D Numbe r: 423 CI D Name: Anoka -Wh s e -VP3 Time St amp: Date: 0927

Time: 1747

... if “Description” in Add/Edit Inbou nd P hon e Book is b lank

Inbound Phone Book

Re m ove Prefix Add P refix For ward/Addr

423

748

Phone Boo k Configurati on

Gate w ay Name :

Add/Edit Inbound Phone Book

Use as default entry

Remove Prefix:

Add Prefix:

Channel Number:

Description:

Q.931 Parameters

Gatekeeper RAS Parameters

Channel 2

Shi pp i ng Dept

Anoka-Whse-VP3

{Channel 2}

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.

Call is received

here.

xxxyyyzzzz J.Q. Public

FXS

CID

Terminating

VoIP

Clock:

10/03, 4:51pm

Display shows:

CID Number: 423 CID Name: Anoka-Whse-VP3

Time Stamp: Date: 10/03 Time: 4:51pm

In x.06 release, when SIP protocol is used,

CID Name field will duplicate val ue in CID Number field.

CID Flow

Network

H.323 Protocol

CID

Generating

VoIP

Ch2 Ch3

Ch4

Ch1 401

402 403

404

FXS

Phone Book Configuration

Gateway Name:

Q.931 Parameters

Gatekeeper RAS Parameters

Anoka-Whse-VP3

xxxyyyzzzz J.Q. Public

phone of:

Inbound Phone Book

Remove Prefix Add Prefix Forward/Addr

Figure 5-27: Caller ID example 4

Figure 5-28: VOIP Caller ID Case #4 – Remote FXS call on H.323 VOIP system.

Call originates here

at 4:51pm, Oct 3.

Nigel Thurston 763-555-9401

{Channel 2}

423 748

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Call is received

here.

xxxyyyzzzz J.Q. Public

FXS

CID

Terminating

VoIP

Clock:

11/15, 6:17pm

Display shows:

CID Number: 423 CID Name: Anoka-Whse-VP3 Time Stamp: Date: 11/15

Time: 6:17pm

In x.06 release, when SIP protocol is used,

CID Name field will duplicate value in CID Number field.

CID Flow

Network

H.323 Protoco l

CID

VoIP

Ch2 Ch3

Ch4

Ch1

DID

Generating

Phone Book Configuration

Gateway Name:

Q.931 Parameters

Gatekeeper RAS Parameters

Anoka-Whse-VP3

Call originates here

Central Office

without

standard telephony

Caller ID service

at 6:17pm, Nov 15.

xxxyyyzzzz J.Q. Public

phone of:

Edwin Smith 763-743-5873

Inbound Phone Book

Remove Prefix Add Prefix Forward/Addr

423

748

{Channel 2}

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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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 Values Description

Boot Version

Firmware Version

Configuration Version

Phone Book Version

IFM Version

Mac Address numeric Denotes the number assigned as the VOIP unit’s unique Ethernet address. Up Time days:

Hardware ID alpha-

nn.nn alphanumeric

nn.nn.nn

alphanumeric

nn.nn. nn.nn

alphanumeric

nn.nn alphanumeric

nn alphanumeric

hours: mm:ss

numeric

Indicates the version of the code that is used at the startup (booting) of the VOIP. The boot code version is independent of the software version.

Indicates the version of the MultiVOIP firmware.

Indicates the version of the MultiVOIP configuration software.

Indicates the version of the MultiVOIP phone book being used.

Indicates version of the IFM module, the device that performs the transformation between telephony signals and IP signals.

Indicates how long the VOIP has been running since its last booting.

Indicates version of the MultiVOIP circuit board assembly being used.

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The frequency with which the System Information screen is updated is determined by a setting in the Logs/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

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Figure 6-3: Call progress screen

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Call Progress Details: Field Definitions

Field Name Values Description

Channel 1-n Number of data channel or time slot on which the call is carried. This is the

channel for which call-progress details are being viewed.

Call Details Duration H/M/S The length of the call in hours, minutes, and seconds (hh:mm:ss). Mode Voice or FAX Indicates whether the call being described was a voice call or a FAX call.

Voice Coder G.723, G.729,

G.711, etc.

IP Call Type H.323, SIP, or

SPP

IP Call Direction incoming,

outgoing

Packet Details

Packets Sent integer value The number of data packets sent over the IP net work in the course of this

Packets Rcvd integer value The number of data packets received over the IP network in the course of

Bytes Sent integer value The number of bytes of data sent over the IP network in the course of this

Bytes Rcvd integer value The number of bytes of data received over the IP network in the course of

Packets Lost integer value The number of voice packets from this call that were lost after being

From – To Details Description Gateway Name (from) IP Address (from) Options SC, FEC Displays VOIP transmission options in use on the current call. These may

Gateway Name (to) alphanumeric IP Address (to)

Options SC, FEC Displays VOIP transmission options in use on the current call. These may

DTMF/Other Details

Prefix Matched specified Outbound Digits Sent 0-9, #, * The digits transmitted by the MultiVOIP to the PBX/telco for this call.

Outbound Digits Received

Server Details

DTMF Capability inb and,

Table is continued on next page…

alphanumeric string

n.n.n.n

string n.n.n.n

dialing digits

0-9, #, * Of the digits transmitted by the MultiVOIP to the PBX/telco for this call,

n.n.n.n and/or other related descriptions

out of band Expressions differ slightly for different Call Signaling protocols (H.323, SIP, or SPP).

The voice coder being used on this call.

Indicates the Call Signaling protocol used for the call (H.323, SIP, or SPP).

Indicates whether the call in question is an incoming call or an outgoing call.

call.

this call.

call.

this call.

received from the IP network.

Identifier for the VOIP gateway that handled the origination of this call. IP address from which the call was received. include Forward Error Correction or Silence Compression.

Identifier for the VOIP gateway that handled the completion of this call. IP address to which the call was sent. include Forward Error Correction or Silence Compression. Displays the dialed digits that were matched to a phonebook entry.

these are the digits that were confirmed as being received. 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.

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Call Progress Details: Field Definitions (continued)

Field Name Values Description

Supplementary Services Status

Call on Hold alphanumeric Describes held call by its IP address source, location/gateway identifier,

Call Waiting alphanumeric Describes waiting call by its IP address source, location/gateway identifier,

Caller ID “Calling Party

+ identifier”; “Alerting Party + identifier”; “Busy Party + identifier”; “Connected Party + identifier”

Call Status fields Call Status hangup, active Shows condition of current call. Call Control Status Tun, FS + Tun,

AE, Mux

Silence Compression SC

Forward Error Correction

FEC

and hold duration. Location/gateway identifiers come from Gateway Name field in Phone Book Configuration screen of remote VOIP.

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.

Displays the H.323 version 4 features in use for the selected call. These include tunneling (Tun), Fast Start with tunneling (FS + Tun), Annex E multiplexed UDP call signaling transport (AE), and Q.931 Multiplexing (Mux).

“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. “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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Logs Screen Details: Field Definitions

Field Name Values Description

Log # column 1 or higher All calls are assigned an event number in chronological order, with the

most recent call having the highest event number. Start Date, Time column

Duration column hh:mm:ss This describes how long the call lasted in hours, minutes, and seconds. Type H.323, SIP, SPP Indicates the Call Signaling protocol used for the call (H.323, SIP, or SPP). Status column success or failure Displays the status of the call (whether the call was completed or not). IP Direction incoming,

Mode column voice or FAX Indicates whether the event b eing described was a voice call or a FAX call. From column gateway name Displays the name of the voice gateway that originates the call. To column gateway name Displays the name of the voice gateway that completes the call.

Special Buttons

Previous -- Displays log entry before currently selected one. Next -- Displays log entry after currently selected one. First -- Displays first log entry Last -- Displays last log entry. Delete File -- Deletes selected log file.

Call Details Voice coder Coder protocol The voice coder being used on this call. Disconnect Reason "Normal" or

DTMF Capability inband,

Outbound Digits Received Outbound Digits Sent Server Details

Packets sent integer value Number of data packets sent over the IP network in the cou r se of this call. Packets received integer value Number of data packets received over the IP network in the course of this

Packets lost integer value Number of voice packets from this call that were lost after being received Bytes sent integer value

Bytes received integer value Number of bytes of data received over the IP network in the course of this

FROM Details

Gateway Name alphan umeric Identifier for the VOIP gateway that originated this call. IP Address Options FEC, SC Displays VOIP transmission options used by the VOIP gatewa y originating

TO Details

Gateway Name alphanumeric Identifier for the VOIP gateway that completed (terminated) this call. IP Address Options Displays transmission options used b y VOIP gateway terminating the call.

Supplementary Services Info

Call Transferred To phone number Number of party called in transfer. Call Forwarded To phone number Number of party called in forwarding.

dd:mm:yyyy hh:mm:ss

outgoing

"Local" disconnection.

out of band Expressions differ slightly for different Call Signaling protocols. 0-9, #, * The digits, sent by MultiVOIP to PBX/telco, that were acknowledged as

0-9, #, * The digits transmitted by the MultiVOIP to the PBX/telco for this call.

n.n.n.n

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.

Indicates whether the call is "incoming" or "outgoing" with respect to the gateway.

Indicates whether the call was disconnected simply because the desired conversation was done or some other irregular cause occasioned disconnection (e.g., a technical error or failure). Indicates whether the DTMF dialing digits are carried "Inband" or "Out of Band." The corresponding field values differ for the 3 different VOIP protocols. 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".

having been received by the remote VOIP gateway.

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.

call. from the IP network.

Number of bytes of data sent over the IP network in the course of this call. call.

IP address of the VOIP gateway from which the call was received. the call. These may include Forward Error Correction or Silence

Compression.

IP address of the VOIP gateway at which the call was completed.

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IP Statistics

Chapter 6: Using the Software

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, conne ctionoriented transmission of data.

Both TCP and UDP split data into packets called “datagrams.” However, TCP includes extra headers in th e 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 -- a s 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 “Clear” button -- Clears packet tallies from memory.

Total Packets

Transmitted integer Received integer Received with

Errors

UDP Packets

Transmitted integer

Received integer Received with

Errors

TCP Packets

Transmitted integer

Received integer Received with

Errors

RTP Packets

Transmitted integer

Received integer Received with

Errors

RTCP Packets

Transmitted integer

Received integer Received with

Errors

n.n.n.n

value value

integer value

value

value integer value

value

value integer value

value

value integer value

value

value integer value

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.

Sum of data packets of all types. Total number of packets transmitted by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software. Total number of packets received by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software. Total number of error-laden packets received by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software.

User Datagram Protocol packets. Number of UDP packets transmitted by this VOIP gateway since the last “clearing” or

resetting of the counter within the MultiVOIP software. Number of UDP packets received by this VOIP gateway since the last “clearing” or

resetting of the counter within the MultiVOIP software. Number of error-laden UDP packets received by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software.

Transmission Control Protocol packets. Number of TCP packets transmitted by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software.

Number of TCP packets received by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software. 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. Number of RTP packets transmitted by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software.

Number of RTP packets received by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software. 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.

Number of RTCP packets transmitted by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software.

Number of RTCP packets received by this VOIP gateway since the last “clearing” or resetting of the counter within the MultiVOIP software. 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 1-999 This field determines how many pings will be generated b y the Start Now

Response Timeout 500 – 5000

Ping Size in Bytes 32 – 128 bytes This field determines how long or large the ping will be. Timer Interval

between Pings Start Now command

button Clear command

button

Link Status Parameters IP Address column No. of Pings Sent as listed Number of pings sent to target endpoint. No. of Pings

Received Round Trip Delay

(Min/Max/Avg) Last Error as listed Indicates when last data error occurred.

n.n.n.n

milliseconds

0 or 30 – 6000 minutes

-- Initiates pinging.

-- Erases ping parameters in Monitor Link field group and restores default

n.n.n.n

as listed Number of pings received by target endpoint.

as listed, in milliseconds

This is the IP address of the target endpoint to be pinged.

command. The duration after which a ping will be considered to have failed.

This field determines how long of a wait there is between one ping and the next.

values. These fields summarize the results of pinging. Target of ping.

Displays how long it took from time ping was sent to time ping response was received.

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Registered Gateway Details

The Registered Gateway Details screen presents a real-time display of the special op erating 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 Values Description

Column Headings Description alphanumeric This is a descriptor for a particular VOIP gateway unit. This descriptor should

generally identify the physical location of the unit (e.g., city, building, etc.) and

perhaps even its location in an equipment rack. IP Address Port Register Duration Status Registered/

No. of Entries The number of gateways currently registered to the Registrar. This includes all SPP

Count of Registered Numbers

List of Registered Numbers

n.n.n.n n

The time remain ing in seconds be fore the Time ToLive ti mer expire s. If the gateway

unregistered

Details

If a registered gateway is selected (by clicking on it in the screen), The "Count of

Lists all of the registered phone numbers for the selected gateway.

The RAS address for the gateway.

Port by which the gateway exchanges H.225 RAS messages with the gatekeeper.

fails to reregister within this time, the endpoint is unregistered.

The current status of the gateway either registered or unregistered.

clients registered and the Registrar itself.

Registered Numbers" will indicate th e number o f reg istere d phone numbers fo r the

selected gateway. When a cli ent registers, all of its inbound phonebook's phone

numbers become registered.

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