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

							41 Chapter 3 - Software Loading and Configuration
Configuring Your Slave MultiVOIP 400/800s
If the Proprietary Phonebook option on the Phone Directory Database dialog box was enabled,
then you will need to configure all remote H.323 endpoints as “Slave” units. For example, the
MultiVOIP 400/800 at the company’s subsidiary office in London would need to be configured as a
“Slave.”.
CAUTION: If you are installing a MultiVOIP 400/800 behind a Firewall, the Firewall must support
H.323. Refer to your Firewall user documentation to enable H.323 support.
1. Disconnect the PC from the Command port of the Master MultiVOIP 400/800 and connect it to
the Command port on the Slave MultiVOIP 400/800.
2.Win 3.1 users - from the Program Manager, click the MultiVOIP 800 Configuration icon in the
MultiVOIP 400/800 Program Group.  The Main menu is displayed.
Win2000/NT/98/95 users - from your desktop, click Programs I MultiVOIP 800 I MultiVOIP 800
Configuration. The Main menu is displayed.
3. Click IP to display the IP Setup dialog box.
Click (check) the Diffserv box to enable differentiated services on routers that support this
service. This feature provides priority to voice packets so that they are not delayed because of
large data files being downloaded.
The default Frame Type is TYPE_II. If this does not match your IP network, change the Frame
Type by clicking on the drop-down arrow. The Frame Type choices are TYPE_II and SNAP.     
						

							42 MultiVOIP 400/800 User Guide
4. In the Port Address group, enter the IP Address and IP Mask. In the Gateway Address group,
enter the gateway IP address for the slave unit.
The IP Address is the unique IP address that you assign to the MultiVOIP 400/800, and the
Gateway Address is the IP address of the device (e.g., network router) connected to the Internet/
Intranet.
Click OK when you are finished. The Main menu is displayed.
5. From the Main menu, click Voice Channels to display the Channel Setup dialog box. The
Channel Setup dialog box is displayed. The four tabs in this dialog box define the channel
interface, voice/fax parameters, Billing/Misc parameters, and regional telephone parameters for
each channel.
Configure each channel for the type of interface you are connecting to. The Interface tab defaults
to Channel 1 in the Select Channel field. To change the channel number, click the drop-down
arrow and the list of channels is displayed. Highlight the channel you want to configure.
Feature options are enabled or disabled (grayed out) according to the interface type that you
select. The one option available for all interface types is the Inter Digit Time 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 Directory Database. If too much time elapses between digits, and the wrong
numbers are mapped, you will hear a rapid busy signal. If this happens, it will be necessary to
hang up and dial again. The default is 2 seconds.
6. The Interface group defaults to FXS (Loop Start).  Select the interface option that corresponds
to the interface type being connected to the Voice/Fax Channel 1 jack on the back panel of the
MultiVOIP 400/800.
FXS (Loop Start): If a station device; e.g., an analog telephone, fax machine, or KTS (Key
Telephone System) is connected to the Voice/Fax connector on the back of the unit, FXS (Loop
Start) will likely be the correct Interface option.
FXS (Ground Start): If the station device uses ground start, then choose the FXS (Ground Start)
option. Refer to the device’s user documentation.
For both FXS Loop Start and FXS Ground Start , the Ring Count FXS window allows you to set
the maximum number of rings output on the FXS interface before hanging up and releasing the
line to another call. The default setting is 8 counts.
Note: Zero (0) means no rings - caller hears a busy tone.
FXO: If you are using an analog extension from your PBX, then choose the FXO option.  Check
with your in-house phone personnel to verify the connection type.   
						

							43 Chapter 3 - Software Loading and Configuration
If FXO is selected, the Dialing Options Regeneration, Flash Hook Timer, and Ring Count
groups are enabled.  Check with your local in-house phone personnel to verify whether your local
PBX dial signaling is Pulse or tone (DTMF). Then, set the Regeneration option accordingly. The
Flash Hook Timer allows you to eter the time, in milliseconds, for the duration of the flash hook
signals output on the FXO interface. The default setting is 600 milliseconds. The Ring Count
FXO window allows you to set the number of rings received on the FXO interface before the
MultiVOIP 400/800 answers the incoming call. The default setting is 2 counts.
Note: Zero (0) means that the MultiVOIP 400/800 never answers.
For FXO-to-FXO communications, you can enable a specific type of FXO Disconnect; Current
Loss, Tone Detection, or Silence Detection. (Check with your in-house phone personnel to
verify the preferred type of disconnect to use.) Enabling Tone Detection activates the
Disconnect Tone Sequence options. For Disconnect Tone Sequence, you can select from drop-
down lists either one or two tones that will cause the line to be disconnected; the person hanging
up a call must then hit the key(s) that will produce those tones. For Silence Detection, select
One Way or Two Way, then set the timer for the number of seconds of silence before disconnect.
Note that the default value of 15 seconds may be shorter than desired for your application.
E&M: If you are connecting to an analog E&M trunk on your PBX, then choose the E&M interface
option to enable the E&M Options group.  Check with your local in-house phone personnel to
determine if the signaling is Dial Tone or Wink and if the connection is 2-wire or 4-wire.  If Wink
signaling is used, then the Wink Timer is enabled with a default of 250 milliseconds.  The range
of the Wink Timer is from 100 to 350 milliseconds. Consult with your local in-house phone
personnel for this timer setting.
Note: After configuring a given channel, you can copy that channel’s configuration by clicking the
Copy button and everything on the Interface tab will be copied to the designated channel.
7. Repeat the above step to configure the interface type for voice/fax channels 2 - 4 or 2 - 8.
8. The Voice/Fax tab displays the parameters for the voice gain, DTMF (Dual Tone Multi-
Frequency) gain, voice coder, faxing, and advanced features such as Silence Compression,
Echo Cancellation, and Forward Error Correction.
9. You can set up the input and output voice gain so that 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; and, output gain modifies the level of the
audio being output to the device attached to the voice channel. Make your selections from the
Input and Output drop-down lists in the Voice Gain group. The valid range  is +31dB to –31dB
with a recommended/default value of 0.   
						

							44 MultiVOIP 400/800 User Guide
You can also set up the DTMF gain (or output level in decibels - dB) for the higher and lower
frequency groups of the DTMF tone pair. Make your selections in the drop-down lists in the
DTMF Gain group.
Note: Only change the DTMF gain under the direction of Multi-Tech Technical Support
supervision.
10. To change the voice coder, first select the channel by clicking the Select Channel down arrow
(highlighting the channel number) then click Manual in the Coder group. To select the appropriate
coder, click the Selected Coder down arrow and highlight your new voice coder entry.
If you changed the voice coder, ensure that the same voice coder is used on the voice/fax
channel you are calling; otherwise, you will always get a busy signal.
Note: If you allow the Coder to be selected automatically, then you need to select the Max
Bandwidth from the drop-down list. Check with your VOIP administrator to determine how much
bandwidth is available.
11. The Fax group enables you to send/receive faxes on the selected voice/fax channel. You can set
the maximum baud rate for faxes and the fax volume in the two drop-down lists and change the
jitter value in milliseconds.
When receiving fax packets from a remote MultiVOIP 400/800, it is possible for individual packets
to be delayed or received out of order due to traffic conditions on the network. To compensate for
this effect, the MultiVOIP 400/800 uses a Jitter Buffer. The Jitter Value field allows the MultiVOIP
400/800 to wait a user-definable period of time, in milliseconds, for delayed or out of order fax
packets. The range of allowable Jitter Values is 0 to 400 with a default of 50 milliseconds.
If you do not plan to send or receive faxes on a given voice/fax channel, you can disable faxes in
the Fax group.
12. You can enable the voice/fax advanced features by clicking (checking) the silence compression,
echo cancellation, or forward error correction options.
The Silence Compression option defines whether silence compression is enabled (checked) for
this voice channel. If silence compression is enabled, the 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.
The Echo Cancellation option defines whether echo cancellation is enabled (checked) for this
voice channel. If echo cancellation is enabled, the MultiVOIP 400/800 will remove echo-delay
which improves the quality of sound.
The Forward Error Correction (FEC) option defines whether forward error correction is enabled
(checked) for this voice channel. The FEC feature allows 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.
Note: After configuring a given channel, you can copy that channel’s configuration by clicking the
Copy button and everything on the Voice/Fax tab will be copied to the designated channel. 
						

							45 Chapter 3 - Software Loading and Configuration
The Billing/Misc tab displays the parameters for auto call, automatic disconnection, billing
options and dynamic jitter buffer.
13. If you want to dedicate a local voice/fax channel to a remote voice/fax channel (so you will not
have to dial the remote channel), click the Auto Call Enable option in the Auto Call group. Then
enter the phone number of the remote MultiVOIP 400/800 in the Phone Number field.
14. The Automatic Disconnection group provides three options to be used singly or in combination.
The Jitter Value defines the average inter-arrival packet deviation (in milliseconds) before the
call is automatically disconnected. Jitter is the inter-arrival packet deviation (phase shift of digital
pulses) over the transmission medium that causes voice breakup which can be particularly
disruptive to voice communications. The default setting is 20 milliseconds. A higher value means
that the voice transmission will be more accepting of jitter. A lower value will be 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 setting 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 setting 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.
15. You can set billing options for inbound and/or outbound calls by checking them in the Billing
Options group and then entering the charge in cents per number of seconds.
16. A minimum and maximum set of values can be set for Dynamic Jitter Buffer. 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 Minimum Jitter Value default setting is 150 milliseconds, the Maximum Jitter Value default
setting is 300 milliseconds, and the Optimization Factor default setting is 7.   
						

							46 MultiVOIP 400/800 User Guide
Note: After configuring a given channel, you can copy that channel’s configuration to the other
channel by clicking the Copy button. Everything on the Billing/Misc tab will be copied to the
designated channel.
If your country/region is not the default USA, click the Regional tab and proceed to step 17;
otherwise, proceed to step 18 to begin building your phone directory database.
17. To change the Tone Pairs on the Regional tab, click the Country/Region down arrow and
highlight your specific country or region.
Note: If your country or region is not listed, click the Custom button to define it.
The Tone Pairs group enables you to select/modify the parameters according to choice.  Click
OK when finished and proceed to step 6 to begin building your phone directory database.
18. From the Main menu, click Phone Book to display the Phone Directory Database dialog box.
Make certain the Proprietary Phonebook option is enabled and in the Database Type group,
click the Slave option. The Master IP Address field becomes active.
Note: After you have enabled the Slave option, the Slave Status button is replaced by the
Update button. Once your Phone Directory database has been established, you can click this
button to refresh the entries in the Phone Directory Database window.     
						

							47 Chapter 3 - Software Loading and Configuration
19. Enter the IP address (204.022.122.118) of the New York Office MultiVOIP 400/800 in the Master
IP Address field and enable the Send Status Report to Master so that status reports are sent
to the Master MultiVOIP 400/800.
Note:  In a Dial-On-Demand (DOD) network, you should leave Send Status Report to Master
disabled (not checked). This allows the router to disconnect whenever there is no voice activity.
Note that Slaves with Send Status Report to Master disabled will show up as “Unknown” when
viewing Slave status on the Master.
20. Click OK to return to the Main menu.
21. Click Download Setup to write the new configuration to the slave unit. The Save Setup dialog
box is displayed.
22. Select (check) the Save Current Setup as User Default Configuration and click OK. The Writing
Setup dialog box is displayed as the setup configuration is written to the MultiVOIP 400/800.
After the setup is written to the MultiVOIP 400/800, it reboots.
23. Check that the Boot LED on the MultiVOIP 400/800 is off after the download is complete. This
may take several minutes as the MultiVOIP 400/800 reboots.
24. You are returned to the Main menu.
Your MultiVOIP 400/800 is operational at this time.
Repeat the process for each of the slave units. When all slaves have been configured, go to
“Deploying the Network.”       
						

							48 MultiVOIP 400/800 User Guide
Deploying the VOIP Network
With the Proprietary Protocol option enabled on the Phone Directory Database dialog box, the
VOIP Administrator must develop the VOIP Dialing Directory and deploy the pre-configured slave
MultiVOIP 400/800s to their remote sites. The remote site administrators need only connect power to
the pre-configured MultiVOIP 400/800, connect the MultiVOIP 400/800 to their Ethernet LAN and
predefined telephone equipment, and then wait for the phone directory database to be downloaded.
With the Gatekeeper option enabled on the Phone Directory Database dialog box, all MultiVOIP
400/800s are configured as “Master” and cannot be downloaded. In this case, each MultiVOIP 400/
800 Phone Book will be programmed with phone numbers for its own channels. These phone
numbers are remotely registered or pre-registered with the H.323 Gatekeeper (See the “Registering
with a Gatekeeper Phone Directory” section discussed earlier.
VOIP Administrator
The VOIP administrator is responsible for the following two steps.
1. Establish your VOIP Dialing Directory based on your Phone Directory Database for the numbers
to connect the MultiVOIP 400/800s to your VOIP network and the telephone extension number
you need to connect to the Voice/Fax channels. A sample VOIP Dialing Directory is provided
below for your consideration and use.
Note: The # character in the dialing sequence in the table below acts as a delimiter between the
VOIP phone number and the PBX extension.
To call from
Headquarters to
Sales OfficeCall Process
Pick up telephone and dial a trunk extension
number (e.g., 1, 2, 3, or4).
Second dial tone is generated, dial the Sales
Office MultiVOIP (201).
Attendent connects you to Sales office
extension of called partyDialing
Sequence
1#
201#
5123
Headquarters to
Marketing OfficePick up telephone and dial an extension
number (e.g., 1,2,3, or4).
Second dial tone is generated, dial the
Marketing Office MultiVOIP (401).
Third dial tone is generated, dial the line access
code (9).
Then dial the  extension number of
your calling party (6128).2#
401#
9#
6128
Headquarters to
Regional OfficePick up telephone and dial an extension
number (e.g., 1,2,3, or 4).
Second dial tone is generated, dial the
Regional Office MultiVOIP (301).3#
301
Marketing Office
         to
HeadquartersPick up telephone and dial a trunk extension
number (e.g., 9 or 10).
Second dial tone is generated, dial the
Headquarters MultiVOIP (101)
Third dial tone is generated, dial extension
number of your calling party (4124).9#
101#
4124
Marketing Office
         to
Regional Office10#
301
VOIP Dialing Directory
Pick up telephone and dial a trunk extension
number (e.g., 9 or 10).
Second dial tone is generated, dial the
regional MultiVOIP (301) and telephone rings.
2. Send the slave MultiVOIP 400/800s to their remote sites. 
						

							49 Chapter 3 - Software Loading and Configuration
Remote Site Administrator
The following steps are for MultiVOIP 400/800 H.323 endpoints. For non-MultiVOIP 400/800 H.323
endpoints, refer to the appropriate installation documentation.
3. Unpack your MultiVOIP 400/800.
4. Connect one end of the power supply to a live AC outlet and connect the other end to the Power
connection on your MultiVOIP 400/800 (See Figure 3-1).
FXSE&M
FXO
PSTNEthernet Connection
Power Connection
10BASET
ETHERNET
POWER
Voice/Fax Channel
      Connections
E&MFXO
FXS
Figure 3-1.  Remote Site Cable Connection
5. Connect a network cable to the ETHERNET 10Base-T (RJ-45) connector on the back of your
MultiVOIP 400/800.
6. If you are connecting a station device (e.g., analog telephone, fax machine, or Key Telephone
System (KTS) to your MultiVOIP 400/800, connect the smaller end of a special adapter cable
(supplied) to the Voice/Fax Channel 1 FXS connector on the back of the MultiVOIP 400/800 and
the other end to the station device.
If you are connecting a PBX extension to your MultiVOIP 400/800, connect the smaller end of a
special adapter cable (supplied) to the Voice/Fax Channel 1 FXO connector on the back of the
MultiVOIP 400/800 and the other end to the PBX extension.
If you are connecting an E&M trunk from a telephone switch to your MultiVOIP 400/800, connect
one end of an RJ-45 phone cord to the Voice/Fax Channel 1 E&M connector on the back of the
unit and the other end to the trunk phone jack.
If you are connecting to an E&M trunk, you need to ensure that the E&M trunk jumper is in the
correct position for the E&M type trunk. The default E&M jumper position is E&M type 2. To
change the E&M jumper position, perform the E&M jumper block positioning procedure.
7. Repeat the above step to connect the remaining telephone equipment to each Voice/Fax
Channel on your MultiVOIP.
8. Turn on power to the MultiVOIP 400/800 by placing the ON/OFF switch on the back panel to the
ON position. Wait for the BOOT LED on the MultiVOIP 400/800 to go OFF before proceeding.
This may take a couple of minutes.
9. At this time your VOIP network should be fully operational. Dial one of the sites in your network
using the dialing directory supplied by your network administrator. 
						

							50 MultiVOIP 400/800 User Guide
If your H.323 endpoint is not a MultiVOIP 400/800 (e.g., a PC with Netmeeting software), the
Systems Administrator for your VOIP network should do the following.
1. Make certain that the H.323 endpoint (e.g., PC with Netmeeting, router with voice, etc.) have
been properly connected to the network. Refer to the appropriate end user documentation.
2. Acquire the relevant IP Addresses and/or H323 IDs (H323 IDs are required if Gatekeeper is
enabled) from the other H.323 endpoints that will participate in H.323 calls.
3. Enter these IP Addresses and H323 IDs (H323 IDs are required if Gatekeeper is enabled) in your
phonebook directory.