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Multi-Tech Systems Mvp400, Mvp800 Voice/fax Over Ip Networks User Guide

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    							Chapter 4 - MultiVOIP 400/800 Software 
    						
    							52 MultiVOIP 400/800 User Guide
    Introduction
    This chapter describes various features of the MultiVOIP 400/800 software that enable you to change
    (update) the configuration of your MultiVOIP 400/800. The basic configuration parameters were
    established during the loading of the software (Chapter 3). The MultiVOIP 400/800 software and
    configuration utilities described in this chapter enable you to change that initial configuration as
    necessary.
    The primary interface to the MultiVOIP 400/800 software is the Main menu (MultiVOIP 800 Setup is
    on the title bar) which includes individual buttons that enable you to quickly and easily select a
    desired function. These features are discussed in detail in the MultiVOIP 400/800 Configuration
    section later in this chapter.
    The MultiVOIP 800 Configuration (Main menu) utility along with eight other configuration utilities
    provide full software functionality for your MultiVOIP 400/800. The Configuration Port Setup utility
    enables you to change the method by which you access the MultiVOIP 400/800 (i.e., through a direct
    connection of a PC to the Command Port on the MultiVOIP 400/800, or via your Internet or LAN
    connection to the LAN port on the MultiVOIP 400/800).  The Date and Time Setup utility enables you
    to easily set the date and time used for data logging in the MultiVOIP 400/800. The Download
    Factory Defaults utility enables you to return the configuration to the original factory settings. The
    Download Firmware utility enables you to download new versions of firmware as enhancements
    become available. The Download User Defaults utility enables you to repeat the download user
    defaults process (part of software installation) and update the MultiVOIP 400/800 configuration with
    any necessary changes. The Download Voice Coders utility enables you to download voice coders
    to the MultiVOIP 400/800 after repair or upgrade. The Download H.323 Stack utility enables you to
    download the H.323 protocols to the MultiVOIP 400/800 after repair or upgrade. The Uninstall
    MultiVOIP 800 Configuration utility removes most of the MultiVOIP 400/800 software from your PC.
    The MultiVOIP 400/800 software includes a context-sensitive Help system. Clicking a Help [ 
    ? ]
    button anywhere in the graphical user interface (GUI) will display definitions and recommended
    values for the buttons, options, and fields on that dialog box or menu. Clicking the green underlined
    text in the Helps displays a popup box of related supplementary information for that topic. Clicking the
    Search button (just below the Help menu bar) displays an Index tab with a list of entries. Click an
    entry, then click the Display button to display the text associated with that topic.
    Before You Begin
    The MultiVOIP 400/800 software operates in a Microsoft Windows environment. The MultiVOIP 400/
    800 program group contains icons for all the utilities described above. In Windows 2000/NT/98/95,
    you can access the individual utility programs either by clicking Start | Programs | MultiVOIP 800 |
    (utility), or by double-clicking the utility icon in the MultiVOIP 400/800 program group shown here:   
    						
    							53 Chapter 4 - MultiVOIP Software
    MultiVOIP 400/800 Configuration
    The MultiVOIP 800 Setup menu consists of 10 buttons, an Events window in the middle of the menu,
    and a status bar at the bottom of the menu.  The 10 buttons allow you to display and change the
    voice channels and IP protocol parameters, display and manage the Phone Book listing, view
    statistics and call progress, and change features such as SNMP Agent, Telnet Server, WEB Server,
    and assign a MultiVOIP 400/800 password.
    The Events window in the lower third of the Setup menu provides information about the boot process.
    The status bar at the bottom of the Setup menu displays the current status of the unit and shows, for
    example, if it is Running, the most recent date the unit was configured, the type of connection you
    have to the unit, and your rights. It shows if your PC is connected directly to the command port of the
    MultiVOIP 400/800  or is communicating with the Ethernet port.  The last field on the status bar is the
    Rights field which displays either Read/Write or Read only rights. The first user to communicate with
    the MultiVOIP 400/800 has Read/Write rights that enable the user to view and/or change the
    configuration of the MultiVOIP 400/800.  Any additional users have Read Only rights and can only
    display the configuration of the MultiVOIP 400/800 but are prohibited from changing the configuration.   
    						
    							54 MultiVOIP 400/800 User Guide
    Changing Channel Parameters
    The channel parameters include the interface type and its options, voice and fax settings, billing and
    security, and voice communications for the country and region in which the MultiVOIP 400/800 is
    operating. The Channel Setup dialog box, accessed by clicking the Voice Channels button on the
    Setup menu, has four tabs that display the following categories of channel information -- Interface,
    Voice/Fax, Billing/Misc, and Regional.
    Interface
    The Interface tab defines the parameters related to the physical interface of the voice/fax channel.
    Depending on the interface type selected (FXS, FXO, or E&M), other options on the interface tab will
    be grayed out (become inactive) indicating that they do not apply to the selected interface. The Inter
    Digit Time feature applies to all interface types.
    The Inter Digit Time (in seconds) option in the Dialing Options group defines the amount of time the
    MultiVOIP 400/800 waits between digits as they are entered by the user.  If this timer expires, the
    MultiVOIP 400/800 will immediately attempt to match the digits entered to an entry in the Phone
    Directory Database. The range for this option is 2 to 100 with a default of 2.
    If the interface type is FXO, the Regeneration group in the Dialing Options group defines how the
    MultiVOIP 400/800 recreates telephone numbers that were detected at the remote end. You can
    select Pulse (for rotary dial telephones) or DTMF Tone dialing (touch-tone), depending upon the
    dialing type that is supported by the PBX or exchange. When FXO is the Interface, the Flash Hook
    Timer field is enabled (activated)  Enter the time, in milliseconds, for the duration of the flash hook
    signals output on the FXO interface. The default is 600 milliseconds.
    FXS Interface
    The FXS Interface is used to connect telephones, fax machines, key telephone systems, etc., to the
    MultiVOIP 400/800.  In addition, you need to select either Loop Start or Ground Start.  Most of the
    equipment mentioned will use Loop Start which is the default.
    FXO Interface
    The FXO Interface is used to connect PBX extensions or central office telephone lines.  You also,
    need to select DTMF or Pulse dialing in the Regeneration field of the Dialing Options group.  If you
    are unsure of the correct selection, contact the personnel in charge of your PBX or your local
    telephone company to determine whether pulse or DTMF should be used.   
    						
    							55 Chapter 4 - MultiVOIP Software
    E&M Interface
    The E&M Interface is used to connect PBX E&M trunks.  You will need to select between Dial Tone or
    Wink signaling and also between 2-wire and 4-wire mode.  If wink signaling is selected, the wink
    timer field becomes active with a range from 100 to 350 milliseconds.  Contact the personnel in
    charge of your PBX to determine the proper configuration of these settings.
    FXO Disconnect On
    The FXO Disconnect On option applies when two MultiVOIP 400/800s are used in an FXO-to-FXO
    configuration.  When you have an FXO-to-FXO configuration, you need to determine the method of
    terminating the call.  Three methods of terminating the call are provided:  Current Loss, Tone
    Detection, or Silence Detection.  Current Loss is the preferred method.
    Current Loss has to be supported by your PBX or local telephone company.  Current Loss
    terminates the call when the PBX or local telephone company switch detects a person hanging up the
    phone and opens the local circuit for a minimum of 600 milliseconds.
    Tone Detection terminates the call when the party who wishes to disconnect enters a one or two
    digit sequence on the telephone keypad.  Valid digits are zero to nine, *, #, and A thru D.
    Note: A through D are extended DTMF tones supported by some PBX or central office equipment
    and are not the same as letters a - d on the standard telephone key pad.
    Silence Detection can be silence in one direction or silence in both directions for a specified amount
    of time.  The amount of time is defined by the entry in the Silence Timer.  The range of the Timer is
    from one to 65535 seconds (roughly 18 hours).  The default is 15 seconds.
    Ring Count
    This field enables you to enter the maximum number of rings output on the FXS interface (default is
    8)  before hanging up and releasing the line to another call or the number of rings (default is 2) that
    must be received before the FXO port answers an incoming call.
    A setting of 0 (zero) on the FXS interface disables the generation of rings. The caller will receive a
    “Busy” tone.
    A setting of 0 (zero) on the FXO interface causes the FXO port to ignore rings from the attached PBX
    or exchange, disabling access to the MultiVOIP 400/800. 
    						
    							56 MultiVOIP 400/800 User Guide
    Voice/Fax
    The Voice/Fax tab controls voice and DTMF gain, voice coder, fax settings, and advanced options.
    The Voice Gain group enables you to select the Input and Output voice gain. Gain is the increased
    signaling power that occurs as the signal is boosted by the MultiVOIP 400/800. The Input Gain drop-
    down list defines the input gain for this voice channel. Before your MultiVOIP 400/800 digitizes voice,
    the volume can be increased or decreased. Input gain modifies the level of the audio coming in to the
    voice channel before it is sent over the Internet to the remote MultiVOIP 400/800. The valid range for
    this option is +31dB to –31dB. The recommended and default value is 0. The Output Gain drop-down
    list defines the voice output gain for this voice channel. Before your MultiVOIP 400/800 converts
    digital voice back to analog, the volume can be increased or decreased. The output gain modifies the
    level of the audio being output to the device attached to the voice channel. The valid range for this
    option is +31dB to –31dB. The recommended and default value is 0.
    The DTMF Gain (Dual Tone Multi-Frequency) group controls the volume level of the digital tones sent
    out for Touchtone dialing.  The Gain High and Gain Low fields control the gain in dB (decibels) of the
    High and Low tones in the tone pairs; the default gain values are -4 dB and -7 dB, respectively.
    DTMF Gain should not be changed except under supervision of MultiTech’s Technical Support.
    The MultiVOIP 400/800 supports many state-of-the art ITU (International Telecommunications Union)
    voice coders. The Voice Coder drop-down menu enables you to 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 has to have the same voice coder selected; otherwise, you will always get a Busy signal.
    The Fax group enables a fax machine to transmit and receive faxes through the MultiVOIP 400/800. If
    a fax machine is connected to one of the voice/fax channels, the Max Baud Rate should be set to
    match the baud rate of the fax machine (refer to user documentation). The Fax Volume setting
    controls the output level of the fax tones, and this setting should be changed only under the direction
    of Multi-Tech’s Technical Support personnel (see Chapter 6 - Warranty, Service and Tech Support).
    The Jitter Value setting defines the inter-arrival packet deviation (in milliseconds) for the fax
    transmission.  A higher value will increase the amount of delay (allowing for a higher percentage of
    packets to be reassembled) and a lower value would decrease the amount of delay (a lower
    percentage of packets would be reassembled).
    The Advanced Features group allows you to enable Silence Compression so that a MultiVOIP 400/
    800 will not transmit voice packets when silence is detected, thereby reducing the amount of network
    bandwidth that is being used by the voice channel; Echo Cancellation for a particular  voice channel
    will remove echo and improve the quality of sound; and, Forward Error Correction allowing 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.   
    						
    							57 Chapter 4 - MultiVOIP Software
    Billing/Misc
    This tab controls the parameters for auto call, automatic disconnection, billing options, and dynamic
    jitter buffer.
    The Auto Call option allows the local MultiVOIP 400/800 to call a remote MultiVOIP 400/800 without
    the user having to dial a Phone Directory Database number.  As soon as you access the local
    MultiVOIP 400/800 voice/fax channel, the MultiVOIP 400/800 immediately connects to the remote
    MultiVOIP 400/800 that you identified in the Remote MultiVOIP 400/800 Phone Number field of this
    option.
    The Automatic Disconnection group provides three 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 20 milliseconds. A higher value means
    voice transmission will be more accepting of jitter. A lower value is less tolerant of jitter.
    Consecutive Packets Lost defines the number of consecutive packets that are lost after which the
    call is automatically disconnected. The default is 30 packets.
    Call Duration defines the maximum length of time (in seconds) that a call remains connected before
    the call is automatically disconnected. The default is 180 seconds.  A call limit of three minutes may
    be too short for most configurations. Therefore, you may want to increase this default value.
    Billing Options can be used to track the cost of Inbound and/or Outbound calls on any of the three
    interfaces (FXO, FXS, or E&M).  The amount to be charged in cents is entered in the Charge (   )
    Cents field together with the associated time duration in the Per (   ) Seconds field.  While a given
    call is active, the accumulated charges can then be viewed on the Call Progress dialog box. When
    the call ends, the charges are transferred to a Log File that can be viewed by highlighting the call
    event in the Log Entries dialog box and selecting Details.
    Dynamic Jitter Buffer defines a minimum and a maximum jitter value for voice communications.
    When receiving voice packets from a remote MultiVOIP 400/800, it is possible to experience varying
    delays between packets due to traffic conditions on the network. This is called Jitter. To compensate
    for this effect, the MultiVOIP 400/800 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the
    MultiVOIP 400/800 to wait for delayed voice packets by automatically adjusting the length of the Jitter
    Buffer between configurable minimum and maximum values. An Optimization Factor adjustment
    controls how quickly the length of the Jitter Buffer is increased when jitter increases on the network.
    The length of the jitter buffer directly effects the voice delay between MultiVOIP 400/800 gateways.
    The default minimum dynamic jitter buffer of 150 milliseconds is the minimum delay that would be
    acceptable over a low jitter network. The default maximum dynamic jitter buffer of 300 milliseconds is
    the maximum delay tolerable over a high jitter network.   
    						
    							58 MultiVOIP 400/800 User Guide
    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.
    The Optimization Factor can be configured in the range of 0 to 12 with a default setting of 7.
    Regional
    The Regional tab controls the voice communications for the country or region in which the MultiVOIP
    400/800 is being used.
    From the Country/Region drop-down list you can select the country or region for which you are
    configuring the MultiVOIP 400/800. The Tone Pairs group always displays the tones used in the
    country or region currently selected. In addition to Australia, Central America, Chile, Europe, France,
    Japan, UK, and USA, there is a Custom selection (with defaults identical to USA) that will make the
    Custom button active. Clicking the Custom button enables you to edit the Tone Pairs and establish
    custom sets of tone pairs for Dial Tone, Ring, and Busy on a Custom Tone Pair Settings dialog box.
    The Pulse Generation Ratio group contains two ratios: the 60/40 is for the USA, and the 67/33 ratio
    is for international applications.   
    						
    							59 Chapter 4 - MultiVOIP Software
    Changing the Phone Directory Database
    The Phone Directory Database dialog box displays all the phone numbers in your MultiVOIP 400/
    800 network.  The database displays the phone numbers in numerical order with destination details
    (i.e., IP Address (or H323 ID), channel assignment, and a brief description of the entry (e.g., New
    York Office 1). The method for changing the phone directory database is dependent on whether the
    Gatekeeper option or the Proprietary Phonebook option is enabled.
    Access this database by clicking the Phone Book button on the Main MultiVOIP 400/800 menu.
    Proprietary Phonebook Enabled
    With the Proprietary Phonebook option enabled, you can add, delete, or edit any entry in the
    MultiVOIP 400/800 phone directory database. You can change the master - slave relationship of the
    entry, view/print entries, and view the details of Hunt Groups.
    To add an entry to the Phone Directory database, perform the steps as described in Chapter 3. To
    edit an entry in the Phone Directory Database, click on the relevant entry in the Phone Direcory
    database dialog box and then click the Edit button. The Add/Edit Phone Entry dialog box will be
    displayed with relevant information.
     In the Station Information group, enter/change the unique phone number for the device in the
    Phone Number field (e.g., 101); enter/change the Channel number (1 - 4 or 1 - 8) in the Voice
    Channel field; and provide/change the optional Description. In the Station Identification group,
    select the Hunt group from the Hunt Group drop-down list and select the appropriate Hunt Group
    you want the phone number to be associated with; enter the appropriate IP Address of the entry in
    the IP Address field (e.g., 204.22.122.118); and accept the Port default.     
    						
    							60 MultiVOIP 400/800 User Guide
    In the Station Information group, the phone number does not have to be a conventional phone
    number; for example, it does not have to be 717-5565. It can be a single digit or several digits. For
    example, you could enter a phone number of 101 with a description of New York Office 1.
    The Station Identification group enables you to assign the entry to a Hunt Group, provide the IP
    Address, and accept the H.323 industry standard Port number.  A Hunt Group is a series of
    telephone lines organized in such a way that if the first line is busy the next line is hunted and so on
    until a free line is found. It is a set of links which provides a common resource and which is assigned
    a single hunt group designation. A user requesting that designation may then be connected to any
    member of the Hunt Group.
    To establish the phone directory database for a remote MultiVOIP 400/800, you do not have to enter
    phone numbers, but you have to check the Slave option in the Database Type group. When you
    click the Slave option, the Update Database From group becomes active. You need to enter the IP
    address of the MultiVOIP 400/800 that you established as the Master.
    Clicking the Slave Status button on the Phone Directory Database dialog box allows you to view
    the status of all the slave units in your VOIP network (Send Status Report to Master must be
    enabled on the Slave).  The Phone Number of each Slave is displayed with its IP Address, current
    line status, and the description of the phone number.
    You can view the details of the current Hunt Group (e.g., HUNT GROUP #1) configuration by clicking
    on the Phone Directory Database’s Hunt button. The current Phone numbers for HUNT GROUP #1
    are displayed.
    Highlight the Hunt Group you wish to view. The Phone no’s window displays the telephone numbers
    associated with that Hunt Group and the No. of Entries field displays the running total of entries.
    Note: You can change the name of the Hunt Group by clicking on the entry that you want to change,
    editing the change in the Hunt Group name window, and then clicking the Set button.     
    						
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