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MultiVOIP
®
Voice/Fax over IP Gateways
MVP130
MVP130-FXS
User Guide
User Guide
S000386D
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 Multi-Tech Systems, Inc. All rights reserved.
Copyright © 2011, by Multi-Tech Systems, Inc.
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
B
C
D
09/26/05
04/25/07
02/08/08
04/30/09
Doc re-organization. Follows S000249K. Describes 1.08 software release.
Update tech support contact list & revise warranty.
Format revision and software version 1.11 update.
Add link to Multi-Tech website for warranty update.
E
F
Patents
10/25/13
12/17/13
Removed references to product CD. Updated RoHS, safety, and other regulatory information.
Added UL translations.
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
Europe, Middle East, Africa:
U.S., Canada, all others:
By Email [email protected]
By Phone
(44) 118 959 7774
(800) 972-2439 or (763) 717-5863
Warranty
To read the warranty statement for your product, please visit: http://www.multitech.com
.
Multi-Tech Systems, Inc. 2
Chapter 1 – Product Overview ........................................................................................................................6
Introduction.................................................................................................................................................................. 6
Interface ....................................................................................................................................................................... 6
Front Panel LEDs ........................................................................................................................................................... 7
Computer Requirements............................................................................................................................................... 7
Specifications ................................................................................................................................................................ 8
Chapter 2 – Installing and Cabling the MultiVOIP ............................................................................................9
Introduction.................................................................................................................................................................. 9
Safety Warnings ............................................................................................................................................................ 9
Lithium Battery Caution ................................................................................................................................................ 9
Safety Warnings Telecom .............................................................................................................................................. 9
Package Contents ....................................................................................................................................................... 10
Cabling Procedure for MVP130 ................................................................................................................................... 10
Chapter 3 – Software Installation ................................................................................................................. 12
Introduction................................................................................................................................................................ 12
Installing the Software ................................................................................................................................................ 12
Setup Overview .......................................................................................................................................................... 12
Ethernet/IP .................................................................................................................................................................. 13
Voice/Fax ..................................................................................................................................................................... 14
Interface ...................................................................................................................................................................... 15
Call Signaling ................................................................................................................................................................ 18
Regional ....................................................................................................................................................................... 20
Phone Book ................................................................................................................................................................. 21
Save & Reboot ............................................................................................................................................................. 22
Chapter 4 – Configuring Your MultiVOIP ....................................................................................................... 23
Navigating the Software ............................................................................................................................................. 23
Web Browser Interface ............................................................................................................................................... 23
Configuration Information Checklist ........................................................................................................................... 24
Ethernet/IP ................................................................................................................................................................. 25
Voice/Fax .................................................................................................................................................................... 27
Configurable Payload Type .......................................................................................................................................... 31
Interface ..................................................................................................................................................................... 31
FXS Loop Start Parameters .......................................................................................................................................... 32
FXO Parameters ........................................................................................................................................................... 35
FXO Supervision ........................................................................................................................................................... 36
DID Parameters .......................................................................................................................................................... 39
Call Signaling............................................................................................................................................................... 40
H.323 ........................................................................................................................................................................... 40
SIP ................................................................................................................................................................................ 41
SPP ............................................................................................................................................................................... 43
SNMP ........................................................................................................................................................................... 44
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Regional ...................................................................................................................................................................... 45
SMTP .......................................................................................................................................................................... 48
RADIUS ....................................................................................................................................................................... 50
Logs/Traces ................................................................................................................................................................. 52
NAT Traversal ............................................................................................................................................................. 53
Supplementary Services .............................................................................................................................................. 54
Save Settings .............................................................................................................................................................. 57
Connection ................................................................................................................................................................. 57
Troubleshooting Software Issues ................................................................................................................................ 58
Chapter 5 – Phone Book Configuration ......................................................................................................... 59
Introduction................................................................................................................................................................ 59
Identify Remote VOIP Site to Call................................................................................................................................ 59
Identify VOIP Protocol to be Used .............................................................................................................................. 59
Phonebook Starter Configuration ............................................................................................................................... 60
Outbound Phonebook ................................................................................................................................................. 60
Inbound Phonebook .................................................................................................................................................... 61
Phone Book Descriptions ............................................................................................................................................ 63
Outbound Phone Book/List Entries ............................................................................................................................. 63
Inbound Phone Book/List Entries ................................................................................................................................ 67
Phone Book Save and Reboot ..................................................................................................................................... 70
Phonebook Examples.................................................................................................................................................. 70
North America ............................................................................................................................................................. 70
Europe ......................................................................................................................................................................... 74
Variations of Caller ID ................................................................................................................................................. 80
Chapter 6 – Using the Software .................................................................................................................... 83
Software Categories Covered in This Chapter ............................................................................................................. 83
System Information screen ......................................................................................................................................... 84
Statistics Section ......................................................................................................................................................... 85
Call Progress ................................................................................................................................................................ 85
Logs .............................................................................................................................................................................. 87
IP Statistics................................................................................................................................................................... 89
Link Management ........................................................................................................................................................ 91
Registered Gateway Details......................................................................................................................................... 92
Servers ......................................................................................................................................................................... 93
Advanced .................................................................................................................................................................... 94
MultiVOIP Program Menu Items ................................................................................................................................. 96
Updating Firmware ...................................................................................................................................................... 96
Implementing a Software Upgrade ............................................................................................................................. 97
Identifying Current Firmware Version ......................................................................................................................... 98
Downloading Firmware ............................................................................................................................................... 98
Downloading Factory Defaults .................................................................................................................................... 99
Downloading IFM Firmware ...................................................................................................................................... 100
Setting and Downloading User Defaults .................................................................................................................... 102
Setting a Password .................................................................................................................................................... 103
Upgrading Software ................................................................................................................................................... 105
FTP Server File Downloads ........................................................................................................................................ 105
Web Browser Interface ............................................................................................................................................. 109
SysLog Server Functions ............................................................................................................................................ 110
Appendix A – Cable Pin-outs ....................................................................................................................... 112
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Command Cable........................................................................................................................................................ 112
Ethernet Connector .................................................................................................................................................. 112
Voice/Fax Channel Connectors ................................................................................................................................. 112
Appendix B – TCP/UDP Port Assignments ................................................................................................... 113
Well Known Port Numbers ....................................................................................................................................... 113
Port Number Assignment List ................................................................................................................................... 113
Appendix C – Regulatory Information ......................................................................................................... 114
EMC, Safety, and R&TTE Directive Compliance ......................................................................................................... 114
FCC Part 15 Declaration ............................................................................................................................................ 114
FCC Part 68 Telecom ................................................................................................................................................. 115
Industry Canada ........................................................................................................................................................ 116
Canadian Limitations Notice ..................................................................................................................................... 116
Waste Electrical and Electronic Equipment Statement.............................................................................................. 117
WEEE Directive .......................................................................................................................................................... 117
Instructions for Disposal of WEEE by Users in the European Union ......................................................................... 117
Restriction of the Use of Hazardous Substances (RoHS) ............................................................................................ 118
Information on HS/TS Substances According to Chinese Standards .......................................................................... 119
Information on HS/TS Substances According to Chinese Standards (in Chinese) ....................................................... 120
依照中国标准的有毒有害物质信息 ....................................................................................................................... 120
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Introduction
The MultiVOIP gateway provides toll-free voice and fax communications over the Internet or Intranet. The
MVP130 and MVP130FXS models are a single-channel units. The MVP130FXS supports the FXS telephony interface only. Both units have a 10/100Mbps Ethernet interface and a command port for configuration.
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 minimized or eliminated altogether. Once the call destination is determined, the phonebook settings determine whether the destination
VOIP unit must strip off or add dialing digits to make the call appear at its destination to be a local call.
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.
LED
Power
Boot
Ethernet
TX
RX
XS
RS
Description
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 (that is, 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 network.
Transmit Signal. This indicator lights when the FXS-configured channel is off-hook or the FXOconfigured 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 FXOconfigured channel (MVP130 only) has taken the line off-hook.
Computer Requirements
The computer used to configure the MultiVOIP:
● Must have Windows operating system
● Must have an available COM port
This computer is only required for local configuring and monitoring. After initial setup, most configuring and monitoring can be done remotely via the IP network.
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Chapter 2 — Quick Start
Specifications
Operating Voltage/Current
Mains Frequencies
Power Consumption
Mechanical Dimensions
Weight
Ambient temperature range
Warranty
MVP130 & MVP130-FXS
100-240VAC / 1.0 A
50/60 Hz
4.5 watts (9.7 watts with phone off hook)
1.0” H x 4.3” W x 5.6” D (2.5 cm H x 10.9 cm W x 14.2 cm D)
8 oz. (23 g)
Maximum: 14060 degrees Celsius (140 degrees Fahrenheit) @ 20-
90% non-condensing relative humidity.
Minimum: 0 degrees Celsius (32 degrees Fahrenheit).
2 years
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Introduction
MVP130 MultiVOIP models must be installed by qualified service personnel in a restricted-access area, in accordance with Articles 110-16, 10-17, and 110-18 of the National Electrical Code, ANSI/NFPA 70.
Safety Warnings
Lithium Battery Caution
● A lithium battery located within the product provides backup power for the timekeeping. This battery has an estimated life expectancy of ten years.
● When this battery starts to weaken, the date and time may be incorrect. If the battery fails, the board must be sent back to Multi-Tech Systems for battery replacement.
● Lithium cells and batteries are subject to the Provisions for International Transportation. Multi-Tech
Systems, Inc. confirms that the Lithium batteries used in the Multi-Tech product(s) referenced in this manual comply with Special Provision 188 of the UN Model Regulations, Special Provision A45 of the ICAO-TI/IATA-
DGR (Air), Special Provision 310 of the IMDG Code, and Special Provision 188 of the ADR and RID (Road and
Rail Europe).
CAUTION: Risk of explosion if this battery is replaced by an incorrect type. Dispose of batteries according to instructions.
ATTENTION: Risque d'explosion si cette batterie est remplacée par un type incorrect. Jetez les batteries conformément aux instructions.
Safety Warnings Telecom
Before servicing, disconnect this product from its power source and telephone network. Also:
● Never install telephone wiring during a lightning storm.
● Never install a telephone jack in wet locations unless the jack is specifically designed for wet locations.
● Use this product with UL and cUL listed computers only.
● Never touch uninsulated telephone wires or terminals unless the telephone line has been disconnected at the network interface.
● Use caution when installing or modifying telephone lines.
● Avoid using a telephone during an electrical storm. There may be a remote risk of electrical shock from lightning.
● Do not use a telephone in the vicinity of a gas leak.
CAUTION: To reduce the risk of fire, use only 26 AWG or larger UL Listed or CSA Certified telecommunication
Line cord.
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Chapter 2— Installing and Cabling the MultiVOIP
Avertissements de sécurité télécom analogique
Avant de l'entretien, débrancher ce produit de son réseau d'alimentation et de téléphone. également:
● Ne jamais installer du câblage téléphonique pendant un orage électrique.
● Ne jamais installer de prises téléphoniques à des endroits mouillés à moins que la prise ne soit conçue pour de tels emplacements.
● Utilisez ce produit avec UL et cUL ordinateurs répertoriés seulement.
● Ne jamais toucher fils ou des bornes téléphoniques non isolés à moins que la ligne téléphonique n'ait été déconnectée au niveau de l'interface réseau.
● Faire preuve de prudence au moment d'installer ou de modifier des lignes téléphoniques.
● Éviter d'utiliser le téléphone pendant un orage électrique. Il peut y avoir un risque de choc électrique causé par la foudre.
● N'utilisez pas un téléphone à proximité d'une fuite de gaz.
ATTENTION: Pour réduire les risques d’incendie, utiliser uniquement des conducteurs de télécommunications 26 AWG au de section supérleure.
●
●
●
Package Contents
● MVP130 or MVP130-FXS
● DB9 to RJ45 cable
● Power transformer
Power cord
RJ-11 phone cord
Printed Cabling Guide
Cabling Procedure for MVP130
To connect the MultiVOIP to your LAN and telephone equipment:
1. Connect the power cord to the power connector on the back of the MultiVOIP and to a live AC outlet.
2.
Back connections for MVP130
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.
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Chapter 2— Installing and Cabling the MultiVOIP
3.
Connect a network cable to the ETHERNET connector on the back of the MultiVOIP. Connect the other end of the cable to your network.
4. a. For an FXS or FXO connection (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. (MVP130)
(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).
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.
Cable connections
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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).
Installing the Software
To download and install the software:
1.
Go to multitech.com/setup/product.go and select your model.
2.
Click the Software tab and download the MultiVOIP Manager file.
3.
Double-click mvm108d8.exe to extract the files.
4.
Double-click on c:\mvm108d8\Disk1\Setup.exe.
5. Click Yes to allow the program to make changes and launch the Installation Wizard.
6.
Click Next and follow the wizard instructions to install your MultiVOIP software.
7.
Click Finish to exit the wizard. Click Yes to launch the software.
Setup Overview
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.
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Chapter 3—Software Installation
Ethernet/IP
For basic operation, you need a unique LAN IP address the MultiVOIP, a subnet mask, and Gateway IP. 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.
IP settings
Actions
1.
If used, select Packet Prioritization and set 802.1 Priority Parameters as needed.
●
●
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)
2. Set the Frame Type to match the network. Options are TYPE II or SNAP.
3. Enter Gateway Name and if used, check to enable DHCP.
4. Enter IP Address.
5.
Enter Subnet IP Mask.
6.
Enter Gateway IP.
7.
Enable DNS if desired. If enabled, enter the DNS Server IP Address.
8.
Enable SRV support if needed.
9.
Diff Serv Parameters are for routers that are Diff Serv compatible.
Note: Setting both values to 0 effectively disables Diff Serv
10.
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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Voice & Fax settings
14
Chapter 3—Software Installation
Actions:
1.
Select Channel a. Choose channel 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.
2. b. The Copy Channel button is available for easily transferring these settings to the other channels
Repeat for all channels to be used
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 option available for all interface types is the inter digit timer option. This option defines the maximum amount of time that the unit will wait before mapping the dialed digits to an entry in the phone book database. If too much time elapses between digits, and the wrong numbers are mapped, you will hear rapid busy signal. If this happens, hang up and dial again.
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Chapter 3—Software Installation
Interface Parameters
Actions:
1.
Select Channel a. Select Interface Type: FXS, FXO, or DID (FXS only for the MVP130-FXS) b. Regeneration
● Choose how signal is regenerated; as Pulse or DTMF c. Inter Digit Timer
● Time the MultiVOIP waits between digits d. Message Waiting Indication is available if desired e. Inter Digit Regeneration Timer
2.
● Length of time between sent DTMF digits
Flash Hook Options a. Generation (used in conjunction with FXO) b. Detection Range (used in conjunction with FXS)
3. Caller ID
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Chapter 3—Software Installation a. Bellcore is the only option available b. CallerID Manipulation is available if needed
4.
Pass Through (opens an audio path through the MultiVOIP)
5. FXS Options a. Set Ring Count (the number of rings allowed before call abandoned; default is 8) b. Use Current Loss (MultiVOIP interrupts current to disconnect)
6. c. Generate Current Reversal (activates Answer/Disconnect Supervision to FXO)
FXO Options (not available for the MVP130-FXS) a. Ring Count (set number of rings before MultiVOIP answers) b. No Response Timer (set time to attempt call before abandoning)
c. 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)
7.
DID Options (not available for the MVP130-FXS) a. Start Modes (Immediate, Wink or Delay Dial) b. Wink Timer (in milliseconds)
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Chapter 3—Software Installation
Call Signaling
There are three choices for Call Signaling: H.323, SIP and SPP. 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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Signaling Protocols
18
Chapter 3—Software Installation
Actions:
1.
Configure your chosen Call Signal type a. H.323
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) b. SIP
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) c. SPP
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.
Regional Parameters
Actions:
1.
Select the choice that matches the location of the MultiVOIP from the Country/Region field d. If there is not a selection to fit your needs, you may select Custom and set the tones manually e. 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.
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Phone Book screens
21
Chapter 3—Software Installation
Actions
1.
Select Outbound Phone Book a. Select Add Entry b. Accept Any Number may be selected to allow unmatched destinations an alternative c. Enter the number necessary to get out from the PBX system followed by the calling code of the destination in the Destination Pattern field d. Enter the PBX access digit (same number as needed to get out of the PBX system) in the Remove Prefix field e. Any digits that need to be added should be put in the Add Prefix field f. Enter the IP address of the call destination (add a Description if you like) g. 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
2. h. The Advanced Button will allow an Alternate IP Address to be entered for outbound traffic
Select Inbound Phone Book a. Select Add Entry b. Accept Any Number for inbound traffic does not work when external routing devices are used c. Enter any access digits followed by the local calling code in the Remove Prefix field d. Enter any digits needed to access an outside line in the Add Prefix field e. Select Hunting in the Channel Number field to have the VOIP use the next available channel f. Add a description if you like g. Call Forward may be set up (details available in Chapter 5) h. Select Registration Option
3.
Repeat the Phone Book steps for any additional entries needed
Save & Reboot
After you change settings on the VOIP unit, choose the Save & Reboot option; otherwise all changes made will be lost when the MultiVOIP is reset or shutdown.
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Access your MultiVOIP through either a web browser or Windows software interface. There are eight parameters required for MultiVOIP to operate properly. To configure the device, you need the IP address, IP mask, Gateway IP, DNS, and the telephone interface type. Initially, the MultiVOIP must be configured locally. After initial configuration, make changes locally or remotely. Local configuration requires a connection between the MultiVOIP Command port and the computer’s
COM port. Use the MultiVOIP configuration software to do this.
Alternatively, MultiVOIP Manager is a Simple Network Management Protocol (SNMP) agent program that extends the capabilities of the MultiVOIP configuration software. MultiVOIP Manager allows the user to manage any number of VOIPs on a network, whereas the MultiVOIP configuration software manages only one. The MultiVOIP Manager can configure multiple VOIPs simultaneously. MultiVOIP Manager may reside on the same PC as the MultiVOIP configuration software.
Navigating 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.
There are several ways to arrive at the parameter that you want to use: through the left-hand panel, from the drop-down menu, clicking a taskbar icon (if available) or a keyboard shortcut (if available). Once the initial settings are entered, you may choose to configure the MultiVOIP through a Web browser instead.
Web Browser Interface
The MultiVOIP web browser interface gives access to the same commands and configuration parameters as are available in the MultiVOIP Windows interface except for logging functions. When using the web browser interface, logging can be done by email (the SMTP option).
Set up the Web Browser interface (Optional). After establishing an IP address for the MultiVOIP you can use the
MultiVOIP web browser to configure the unit. To configure using the web browser interface, you must set it up:
1.
Set IP address of MultiVOIP unit using the MultiVOIP Configuration program (the Windows interface).
2.
Save Setup in Windows interface.
3.
Close Windows interface.
4.
If your computer does not have Java, you can download it from multitech.com/setup/product.go. Select your model and click Drivers.
5. Open a web browser.
6.
Browse to IP address of MultiVOIP unit.
7.
If username and password have been established, enter them when prompted.
8.
Set browser to allow pop-ups. The MultiVOIP Web interface makes use of pop-up windows.
9.
The configuration screens in the web browser will have the same content as their counterparts in the software; only the presentation differs.
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Chapter 4—Configuring your VOIP
Configuration Information Checklist
To assist with the organization of the information needed, below is a chart summarizing what is necessary.
Info Info
Type of Configuration Info Gathered: Configuration screen where info is entered: Obtained?
Entered?
IP info for VOIP unit
IP address
Gateway
DNS IP (if used)
802.1p Prioritization (if used)
Interface Type
FXS/FXO*
DID-DPO
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 ).
DID info (only if DID used)
Wink
Immediate
Delay Dial
Country code
Email address for VOIP (optional)
Interface parameters
Regional parameters
SMTP parameters
Reminder : Be sure to Save Setup after entering configuration values.
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Chapter 4—Configuring your VOIP
Ethernet/IP
This section covers the Ethernet settings needed for the MultiVOIP unit. In each field, enter the values that fit the network to which the MultiVOIP will be connected to. For many of the settings, the default values will work best – try these settings first unless you know you definitely need to change a parameter.
Network parameters
The Ethernet/IP Parameters fields are described in the tables and text passages below. Note that both Diff Serv parameters (Call Control PHB and VOIP Media PHB) must be set to zero if you enable Packet Prioritization
(802.1p). Nonzero Diff Serv values negate the prioritization scheme.
Field Name
Ethernet Parameters
Packet Prioritization
(802.1p)
Frame Type
802.1p
Values
Y/N
Description
Select to activate prioritization under 802.1p protocol (described below).
Type II, SNAP Must be set to match network’s frame type. Default is Type II.
A draft standard of the IEEE about data traffic prioritization on Ethernet networks. The 802.1p draft is an extension of the 802.1D bridging standard. 802.1D determines how prioritization will operate within a
MAC-layer bridge for any kind of media. The 802.1Q draft for virtual local-area-networks (VLANs) addresses the issue of prioritization for Ethernet networks in particular.
802.1p enacts this Quality-of-Service feature using 3 bits. This 3-bit code allows data switches to reorder packets based on priority level. The descriptors for the 8 priority levels are given below.
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Chapter 4—Configuring your VOIP
Field Name Values Description
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
Sets the priority for signaling packets. Call Control Priority 0-7, where 0 is lowest priority
VOIP Media Priority 0-7, where 0 is lowest priority
Sets the priority for media packets.
Others (Priorities)
VLAN ID
0-7, where 0 is lowest priority
1 - 4094
Sets the priority for SMTP, DNS, DHCP, and other packet types.
The 802.1Q IEEE standard allows virtual LANs to be defined within a network. This field identifies each virtual LAN by number.
IP Parameter fields
Gateway Name
Enable DHCP
IP Address
IP Mask
Gateway
Diff Serv
Parameter fields alphanumeric
Y/N disabled by default
Descriptor of current VOIP unit to distinguish it from other units in system.
Dynamic Host Configuration Protocol is a method for assigning IP address and other
IP parameters to computers on the IP network in a single message with great flexibility. IP addresses can be static or temporary depending on the needs of the computer. n.n.n.n
n.n.n.n
The unique LAN IP address assigned to the MultiVOIP.
Subnetwork address that allows for sharing of IP addresses within a LAN.
n.n.n.n The IP address of the device that connects your MultiVOIP to the Internet.
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.
Call Control PHB
To disable Diff Serv, configure both fields to 0 decimal.
0 – 63 default = 34
Value is used to prioritize call setup IP packets.
Setting this parameter to 0, in conjunction with VOIP Media PHB disables Diff Serv.
VOIP Media PHB 0 – 63 default = 46
FTP Parameter fields
FTP Server
Enable
Y/N
Default = disabled
See “FTP Server
File Transfers” in
Chapter 6
DNS Parameter fields
Enable DNS Y/N
Default = disabled
Value is used to prioritize the RTP/RTCP audio IP packets.
Setting this parameter to 0, in conjunction with Call Control PHB disables 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.
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Chapter 4—Configuring your VOIP
Field Name
Enable SRV
Values
Y/N
Description
Enables ‘service record’ function. Service record is a category of data in the Internet
Domain Name System specifying information on available servers for a specific protocol and domain, as defined in RFC 2782.
Newer internet protocols like SIP, STUN, H.323, POP3, and XMPP may require SRV support from clients. Client implementations of older protocols, like LDAP and SMTP, may have been enhanced in some settings to support SRV.
IP address of specific DNS server to be used to resolve Internet computer names. DNS Server IP
Address
n.n.n.n
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.
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Voice/Fax parameters
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Chapter 4—Configuring your VOIP
Field Name
Default
Select Channel
Values
--
1-2 (210)
1-4 (410)
1-8 (810)
--
Description
When this button is clicked, all Voice/FAX parameters are set to their default values.
Channel to be configured is selected here.
Copy Channel
Voice Gain
Input Gain
Output Gain
DTMF Gain
--
+31dB to
–31dB
+31dB to
–31dB
--
Copies the Voice/FAX attributes of one channel to another channel. Attributes can be copied to multiple channels or all channels at once.
Signal amplification (or attenuation) in dB.
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.
DTMF Gain,
High Tones
DTMF Gain, Low
Tones
DTMF Parameters
Duration (DTMF)
+3dB to
-31dB & “mute”
+3dB to
-31dB & “mute”
DTMF
In/Out of Band
Out of Band Mode
60 – 3000 ms
Out of Band, or
Inband
RFC 2833,
SIP Info
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.
FAX Parameters
Fax Enable
Max Baud Rate
(Fax)
Fax Volume
Y/N
2400, 4800, 7200,
9600, 12000,
14400 bps
-18.5 dB to –3.5 dB
Jitter Value (Fax) Default =
400 ms
Mode (Fax) FRF 11; T.38
Enables or disables fax capability for a particular channel.
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 Multi-
Tech’s Technical Support. Default = -9.5 dB
Defines the inter-arrival packet deviation (in milliseconds) for the fax transmission. A higher value will increase the delay, allowing a higher percentage of packets to be reassembled. A lower value will decrease the delay allowing fewer packets to be reassembled.
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.
Coder Parameters
Coder Manual or
Automatic
Selected Coder G.711 a/u law 64 kbps;
G.726, @
16/24/32/40 kbps;
G.727, @ nine
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.
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Field Name
Selected Coder:
“Coder Priority”
Values 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
Chapter 4—Configuring your VOIP
Description
To make selections from the Selected Coder drop-down list, the Manual option must be enabled.
Max bandwidth
(coder)
11 – 128 kbps
Advanced Features
Silence
Compression
Y/N
Echo Cancellation Y/N
Forward Error
Correction
AutoCall/Offhook Alert Parameters
Auto Call / Offhook
Alert
AutoCall,
Offhook Alert
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Y/N
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, that is, 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 drop-down list is enabled only if the Coder is set to
Automatic.
If coder is to be selected automatically (“Auto” setting), then enter a value for maximum bandwidth.
Determines whether silence compression is enabled (checked) for this voice channel.
With Silence Compression enabled, the MultiVOIP will not transmit voice packets when silence is detected, thereby reducing the amount of network bandwidth that is being used by the voice channel. Default = on.
Determines whether echo cancellation is enabled for this voice channel.
Echo Cancellation removes echo and improves sound quality. Default = on.
Determines whether forward error correction is enabled (checked) for this voice channel.
Forward Error Correction enables some of the voice packets that were corrupted or lost to be recovered. FEC adds an additional 50% overhead to the total network bandwidth consumed by the voice channel. Default = off.
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
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Chapter 4—Configuring your VOIP
Field Name
Generate Local Dial
Tone
Y/N
Offhook Alert
Timer
Phone Number
Minimum Jitter
Value
Maximum Jitter
Value
Optimization
Factor
0 – 3000 seconds
--
Dynamic Jitter
Buffer
Dynamic Jitter
Automatic
Disconnection
Jitter Value
Auto Disconnect
--
1-65535
Call Duration
Consecutive Packets
Lost
Values
60 to 400 ms
60 to 400 ms
0 to 12
1-65535
1-65535
Network Disconnection 1 to 65535;
Default =
30 sec.
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Description phonebook of the local MultiVOIP and a matched setting in the Inbound Phonebook of the remote VOIP.
Used for AutoCall only. If selected, dial tone will be generated locally while the call is being established between gateways. The capability to generate dial tone locally would be particularly useful when there is a lengthy network delay.
The length of time that must elapse before the off hook alert is triggered and a call is automatically made to the phone number listed in the Phone Number field.
Phone number used for Auto Call function or Offhook Alert Timer function. This phone number must correspond to an entry in the Outbound Phonebook of the local
MultiVOIP and in the Inbound Phonebook of the remote MultiVOIP (unless a gatekeeper unit is used in the VOIP system).
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.
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Chapter 4—Configuring your VOIP
Configurable Payload Type
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 (that is, 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.
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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Telephony parameters
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Chapter 4—Configuring your VOIP
FXS Loop Start Parameters
The parameters applicable to FXS Loop Start are shown in the figure below and described in the table that follows.
Field Name Values
Dialing Options fields
FXS (Loop Start) Y/N
Inter Digit Timer 1 - 10 seconds
Message Waiting
Indication
Inter Digit
Regeneration Time
-- in milliseconds
FXS Options fields
FXS Ring Count, FXS 1-99
Current Loss
Generate Current
Reversal
Y/N
Y/N
FXS Loop Start parameters
Description
Enables FXS Loop Start interface type.
This is the length of time that the MultiVOIP will wait between digits. When the time expires, the MultiVOIP will look in the outbound phonebook for the number entered and place the call accordingly.
Default = 2.
See details below.
The length of time between the outputting of DTMF digits.
Default = 100 ms.
Maximum number of rings that the MultiVOIP will issue before giving up the attempted call.
When enabled, the MultiVOIP will interrupt loop current in the FXS circuit to initiate a 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.
When selected, this option implements Answer Supervision and Disconnect
Supervision to the FXO interface using current reversal to indicate events.
Applicable only when FXS and FXO interfaces are connected back to back.
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Chapter 4—Configuring your VOIP
Field Name Values
Flash Hook Options fields
Generation --
Detection Range for Min. and Max.,
50 - 1500 milliseconds
Pass Through Enable Y/N
Description
Not applicable to FXS interface
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
When enabled, this parameter creates an open audio path through the
MultiVOIP.
If the Pass-Through feature is enabled, the AutoCall feature must be enabled for this VOIP channel in the Voice/Fax Parameters screen
Type
Enable
CID Mode
Caller ID fields
CID Manipulation
Bellcore
Y/N
Enabled by default with Caller ID enable above
Disable
Transparent,
User CID,
Prefix,
Suffix
The MultiVOIP currently supports only one implementation of Caller ID. That implementation is Bellcore type 1 with Caller ID placed between the first and second rings of the call.
Caller ID information is a description of the remote calling party received by the called party. The description has three parts: name of caller, phone number of caller, and time of call. The ‘time-of-call’ portion is always generated by the receiving MultiVOIP unit (on FXS channel) based on its date and time setup.
The forms of the ‘Caller Name’ and ‘Caller Phone Number’ differ depending on the IP transmission protocol used (H.323, SIP, or SPP) and upon entries in the phonebook screens of the remote (CID generating) VOIP unit. The CID
Name and Number appearing on the phone at the terminating FXS end will come either from a central office switch (showing a PSTN phone number), or the phonebook of the remote (CID sending) VOIP unit.
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: the CID received from PSTN will be sent out as such, without any manipulation.
User CID: the CID received from PSTN will be replaced by this User CID value.
Prefix: the CID received from PSTN will be prefixed with this value.
Suffix: the CID received from PSTN will be suffixed with this value.
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Chapter 4—Configuring your VOIP
Message Waiting
Message Waiting Indication is a feature that displays an audible or visible indication that a message available. A type of message waiting is sounding a special dial tone (called stutter dial tone), lighting a light, or indicator on the phone.
When a user enables a subscription for message waiting indication, a subscription is made with the Voice Mail
Server (VMS) for that particular event. Whenever the Voice Mail Server finds a change in the state of a corresponding mailbox or some event happens (for example, 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.
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.
To send Subscriptions for Inbound Phone Book entries, all of the following conditions are met:
●
●
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
(that is, if the MultiVOIP gets a 401/407 response from a subscribe request. Then it will take the configured password, calculate the response, and resend the “SUBSCRIBE” request.
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Chapter 4—Configuring your VOIP
FXO Parameters
The parameters applicable to the FXO telephony interface type are shown in the figure below and described in the table that follows.
Field Name
Interface Type
Values
FXO
Dialing Options
Regeneration
Inter Digit Timer
Pulse, DTMF
1 to 10 seconds
Message Waiting
Indication
Inter Digit
Regeneration Time
--
50 to 20,000 milliseconds
FXO Options
FXO Ring Count 1-99
Multi-Tech Systems, Inc.
FXO parameters
Description
Enables FXO functionality
Determines whether digits generated and sent out will be pulse tones or DTMF.
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.
Not applicable to FXO interface
The length of time between the outputting of DTMF digits.
Default = 100 ms.
Number of rings required before the MultiVOIP answers the incoming call.
35
Field Name Values
No Response Timer 1 – 65535
(in seconds)
Flash Hook Options fields
Generation 50 - 1500 milliseconds
Detection Range --
Caller ID fields
Caller ID Type Bellcore
Caller ID enable
CID Manipulation
CID Mode
Y/N
Enabled by default with
Caller ID enable above
Disable
Transparent,
User CID,
Prefix,
Suffix
Chapter 4—Configuring your VOIP
Description
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.
The MultiVOIP currently supports only one implementation of Caller ID. That implementation is Bellcore type 1 with caller ID placed between the first and second rings of the call.
Caller ID information is a description of the remote calling party received by the called party. The description has three parts: name of caller, phone number of caller, and time of call. The ‘time-of-call’ portion is always generated by the receiving MultiVOIP unit (on FXS channel) based on its date and time setup. The forms of the ‘Caller Name’ and ‘Caller Phone Number’ differ depending on the IP transmission protocol used (H.323, SIP, or SPP) and upon entries in the phonebook screens of the remote (CID generating) VOIP unit. The CID Name and
Number appearing on the phone at the terminating FXS end will come either from a central office switch (showing a PSTN phone number), or the phonebook of the remote (CID sending) VOIP unit.
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: the CID received from PSTN will be sent out as such, without any manipulation.
User CID: the CID received from PSTN will be replaced by this User CID value.
Prefix: the CID received from PSTN will be prefixed with this value.
Suffix: the CID received from PSTN will be suffixed with this value.
FXO Supervision
When the selected Interface type is FXO, the Supervision button is active. Click on this button to access call answering supervision parameters and call disconnection parameters that relate to the FXO interface type.
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Field Name Values
Answer Supervision fields
Current Reversal Y/N
Answer Delay Y/N
FXO Supervision
Description
When this option is selected, the FXO interface sends notice to make connection upon detecting current reversal from the PBX (which occurs when the called extension goes off hook).
When this option is selected, the FXO interface sends the connection notice to the calling party only when the Answer Delay Timer expires. The connection notice is sent regardless of whether or not the called extension has gone off hook.
When Answer Delay is enabled, this value determines when the FXO interface sends the connection notice.
When selected, call disconnection will be triggered by a tone sequence
List from which tones can be chosen to signal call answer.
Answer Delay
Timer
Tone Detection
1 – 65535
(in seconds)
Y/N
Available Tones dial tone, ring tone, busy tone, unobtainable tone
(fast busy), survivability tone, re-order tone
Answer Tones any tone from
Available Tones list
Disconnect Supervision fields
Current Reversal Y/N
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.
Disconnection to be triggered by reversal of current from the PBX.
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Field Name
Current Loss
Values
Y/N
Current Loss Timer 200 to 2000
(in milliseconds)
Y/N Silence Detection
Enable
Silence Detection
Type
One-Way or
Two-Way
Silence Timer in seconds integer value
Disconnect Supervision fields
DTMF Tone
Chapter 4—Configuring your VOIP
Description
Disconnection to be triggered by loss of current. That is, when Current Loss is enabled (“Y”), the MultiVOIP will hang up the call at a specified interval after it detects a loss of current initiated by the attached device.
Determines the interval after detection of current loss at which the call will be disconnected.
Enables/disables silence-detection method of supervising call disconnection.
Disconnection to be triggered by silence in one direction only or in both directions simultaneously
Duration of silence required to trigger disconnection.
Enables supervision of call disconnection using DTMF tones.
DTMF Tone Pairs
High Tones
1
4
7
*
1209Hz
2
5
8
0
1336Hz
3
6
9
#
1447Hz
A
B
C
D
1633Hz
Low Tones
697Hz
770Hz
852Hz
941Hz
Disconnect Tone
Sequence
Tone Detection
1 st
tone pair
+
2 nd
tone pair
Y/N
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.
Enables supervision of call disconnection by detecting cessation of a prespecified tone from the PBX.
List from which tones can be chosen to signal call disconnection. Available Tones dial tone, ring tone, busy tone, unobtainable tone (fast busy), survivability tone, re-order tone
Disconnect
Tones any tone from Available
Tones list
Currently chosen disconnection supervision tone.
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DID Parameters
The parameters applicable to the Direct Inward Dial (DID) telephony interface type are shown in the figure below and described in the table that follows. The DID interface allows one phone line to direct incoming calls to any one of several extensions without a switchboard operator. Of course, one DID line can handle only one call at a time. The parameters described here pertain to the customer-premises side of the DID connection (DID-
DPO, dial-pulse originating); the network side of the DID connection (DID-DPT, dial-pulse terminating) is not supported.
Note: The FXS model does not support DID.
Field Name
Interface
Values
DID-DPO
DID Options
Start Modes
Wink Timer
(in ms)
Dialing Options
Inter Digit Timer
Message Waiting
Indication
Inter-Digit
Regeneration Timer
--
Integer values, in milliseconds
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Immediate Start,
Wink Start,
Delay Dial
Integer values, in milliseconds
Integer values, in seconds
DID parameters
Description
Enables the customer-premises side of DID functionality
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.
Not applicable to DID-DPO interface.
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.
H.323 call signaling
Field Name
Use Fast Start
Values
Y/N
Signaling Port
Register with
Gatekeeper port
Y/N
Allow Incoming
Calls Through
Gatekeeper Only
Y/N
GateKeeper RAS Parameters
Primary GK
Alternate GK
1 and 2
IP Address
--
-- n.n.n.n
RAS Port 1719
Gatekeeper Name alphanumeric
RAS TTL Value
Gatekeeper
Discovery Polling
Interval
Description
Enables the H.323 Fast Start procedure. May need to be enabled / disabled for compatibility with third-party VOIP gateways.
Default: 1720 (H.323)
Check this field to have traffic on current VOIP gateway controlled by a gatekeeper.
When selected, incoming calls are accepted only if those calls come through the gatekeeper.
This is the preferred gatekeeper for controlling the traffic of the current VOIP.
A first and a second alternate gatekeeper can be specified for use by the current VOIP for situations where the Primary GK is busy or otherwise unavailable.
IP address of the GateKeeper.
Well-known port number for GateKeepers. Must match port number (1719).
Optional. The name of the GateKeeper with which this MultiVOIP is trying to register. A primary gatekeeper and two alternate units are listed. seconds H.323 Gatekeeper Time to Live value. When the MultiVOIP gateway registers it starts a gatekeeper a countdown timer. 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 to maintain the connection. If the MultiVOIP does not register before the
TTL interval expires, the MultiVOIP gateway’s registration with the gatekeeper expires and the gatekeeper no longer permits call traffic to or from that gateway. Calls in progress continue to function even if the gateway becomes de-registered integer
60 - 300
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.
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Field Name
Use Online
Alternate
Gatekeeper List
Values Description
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.
H.323 Version 4 Options
H.323 Multiplexing Y/N
H.245 Tunneling
(Tun)
Parallel H.245
(FS + Tun)
Y/N
Y/N
Signaling for multiple phone calls can be carried on a single port rather than opening a separate signaling port for each. This conserves bandwidth resources.
H.245 messages are encapsulated within the Q.931 call-signaling channel. Among other things, the H.245 messages let the two endpoints tell each other what their technical capabilities are and determine who, during the call, will be the client and who the server.
Tunneling is the process of transmitting these H.245 messages through the Q.931 channel. The same TCP/IP socket (or logical port) already being used for the Call Signaling
Channel is then also used by the H.245 Control Channel. This encapsulation reduces the number of logical ports (sockets) needed and reduces call setup time.
FS (Fast Start) is a Q.931 feature of H.323v2 to hasten call setup as well as ‘pre-opening’ the media channel before the CONNECT message is sent. This pre-opening is a requirement for certain billing activities. Under Parallel H.245 FS + Tun, this Fast Connect feature can operate simultaneously with H.245 Tunneling.
Annex –E (AE) Y/N Multiplexed UDP call signaling transport. Annex E is helpful for high-volume VOIP system endpoints. Gateways with lesser volume can afford to use TCP to establish calls.
However, for larger volume endpoints, the call setup times and system resource usage under TCP can become problematic. Annex E allows endpoints to perform call-signaling functions under the UDP protocol, which involves substantially streamlined overhead
(this feature should not be used on the public Internet due to potential problems with security and bandwidth usage).
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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Field Name
SIP Proxy Parameters
Signaling Port
Proxy Polling
Interval
TTL Value
Values port
Use SIP Proxy
Allow Incoming
Calls Through SIP
Proxy Only
Primary Proxy
Y/N
Y/N
--
-- 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 n.n.n.n
Y/N port
Description
Port number on which the MultiVOIP UserAgent software module will be waiting for any incoming SIP requests. Default = 5060
Allows the MultiVOIP to work in conjunction with a proxy server.
When selected, incoming calls are accepted only if those calls come through the proxy.
This is the preferred SIP proxy server for controlling the traffic of the current VOIP.
A first and a second alternate SIP proxy server can be specified for use by the VOIP for situations where the Primary proxy server is otherwise unavailable.
Network address of the proxy server that the VOIP is using.
When checked, the domain name of the SIP Proxy serving the MultiVOIP gateway will be included as part of the User ID for that gateway. If unchecked, the SIP Proxy’s IP address will be included as part of the User ID instead of the SIP Proxy’s domain name.
Logical port number for proxy communications. Default = 5060
This is used as the default end point register with a Proxy. name If the Username is not populated in the Phone Book, this is the Username that will be used. This works the same for the password as well. password Password for proxy server function. See “Default Username” description above.
10–65535 seconds
60 - 300
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.
The interval between the VOIP gateway’s successive attempts to connect to and be governed by a higher level SIP proxy server. The Primary Proxy is the highest level gatekeeper. Alternate Proxy 1 is second; Alternate Proxy 2 is the lowest order SIP proxy server.
SIP proxy
“Time to
Live” value.
(in
seconds)
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 deregistered.
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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.
SPP call signaling
Field Name
Mode
Values
Direct,
Client, or
Registrar
Description
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.
Port
General Options
Re-transmission port The UDP port on which data transmission will occur. Each client VOIP has its own port. If two client VOIPs are both behind the same firewall, then they must have different ports assigned to them.
If there are two clients and each is behind a different firewall, then the clients could have different port numbers or the same port number. Default port number = 10000.
50 - 5000ms If packets are lost (as indicated by absence of an acknowledgment) then the endpoint will retransmit the lost packets after this designated time duration has elapsed. Default =
2000 milliseconds.
Max Retransmission
0 - 20
Client Options
Number of times the VOIP will re-transmit a lost packet if no acknowledgment has been received. Default = 3.
Client Option fields are active only in registrar/client mode and only for client VOIP units.
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Field Name
Primary Registrar
Alternate Registrar
1 and 2
Registrar IP
Address
Registrar Port
Polling Interval
Values
--
--
10000 or other integer
60 - 300
Registrar Options
Keep Alive n.n.n.n
30 – 300
(seconds)
Description
This is the preferred SPP registrar gateway for controlling the traffic of the current VOIP.
A first and a second alternate SPP Registrar gateway can be specified for use by the current VOIP for situations where the Primary Registrar gateway is busy or otherwise unavailable.
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 Parameters
Behind Proxy/NAT device
Y/N
Proxy/NAT Device
Parameters –
Public IP Address n.n.n.n
Enables MultiVOIP (running in SPP Registrar mode) to operate ‘behind’ a proxy/NAT device (NAT = Network Address Translation).
The public IP address of the proxy/NAT device which the MultiVOIP is behind.
SNMP
If you intend to manage your MultiVOIP remotely using the MultiVOIP Manager software, you will need to set the Simple Network Management Protocol parameters. To make the MultiVOIP controllable by a remote PC running the MultiVOIP Manager software, check the “Enable SNMP Agent” box on the SNMP Parameters screen.
The MVP130 MultiVOIPs only have limited SNMP functionality available. Contact Multi-Tech Support for help if you want to use this.
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Chapter 4—Configuring your VOIP
Field Name
Enable SNMP
Agent
Address
Community Name
Port Number
Values
Y/N
Trap Manager Parameters
n.n.n.n
--
162
Community Name
1
Permissions
Community Name
2
Permissions
Length = 19 characters (max.)
Case sensitive.
Read-Only,
Read/Write
Length = 19 characters (max.)
Case sensitive.
Read-Only,
Read/Write
Description
Enables the SNMP code in the firmware of the MultiVOIP. This must be enabled for the MultiVOIP to communicate with and be controllable by the MultiVOIP Manager
. Default: disabled
MultiVOIP Manager computer IP address.
A community is a group of VOIP endpoints that can communicate with each other.
Public is used to designate a grouping where all end users have access to entire
VOIP network. Calling permissions can be configured to restrict access as needed.
The default port number of the SNMP manager receiving the traps is the standard port 162.
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.
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.
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Chapter 4—Configuring your VOIP
Field Name
Country/Region
Advisory screen
Values
USA, Japan, UK,
Custom
Standard Tones fields
Type column
Frequency 1 dial tone, ring tone, busy tone, unobtainable tone
(fast busy), survivability tone, re-order tone freq. in Hertz
Frequency 2
Gain 1 freq. in Hertz gain in dB
+3dB to –31dB and “mute” setting
Gain 2
Cadence
(ms) On/Off gain in dB
+3dB to –31dB and “mute” setting n/n/n/n four integer time values in milliseconds; zero value for dial-tone indicates continuous tone
Custom (button) --
Description
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 tonepairing scheme worldwide can be accommodated.
Note 1: Intercept tone is applicable only when the FXS telephony interface has been chosen in the Interface screen and when the AutoCall / OffHook Alert field is set to OffHook Alert in the Voice/Fax Parameters screen. The time allowed for dialing before the automatic calling process begins is set in the OffHook Alert Timer field of the Voice/Fax Parameters screen.
Note 2: “Survivability” tone indicates a special type of call-routing redundancy & applies to MultiVantage VOIP units only
This message screen appears whenever the Country field is changed. It informs the operator that, upon change of the Country field value, all User Defined
Tones will be deleted.
Type of telephony tone-pair for which frequency, gain, and cadence are being presented.
Lower frequency of pair.
Higher frequency of pair.
Amplification factor of lower frequency of pair.
This applies to the dial, ring, busy and ‘unobtainable’ tones that the MultiVOIP outputs as audio to the FXS 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 Custom to bring up Custom Tone Pair Settings. 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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Field Name
Country Selection for Built-In Modem
(not applicable to
MVP130 or 130-
FXS)
Values country name
Description
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-Selectionfor-Built-In-Modem value.
Name of supervisory tone pair. Cannot be same as name of any standard tone pair.
User Defined Tones fields
Type column alphanumeric name
Frequency 1
Frequency 2
Gain 1
Gain 2
Cadence
(ms) On/Off
Freq. in Hertz
Freq. in Hertz
+3dB to –31dB and “mute” setting
+3dB to –31dB and “mute” setting n/n/n/n four integer time values in milliseconds; (zero value indicates continuous tone)
Lower frequency of pair.
Higher frequency of pair.
Amplification factor of lower frequency of pair.
This applies to any supervisory tones that the MultiVOIP outputs as audio to the
FXS 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 allows the user to specify tone-pair attributes that are not found in any of the standard national/regional telephony toning schemes. To access this customization feature, click on the Custom button on the Regional Parameters screen. The “Custom” button is active only when “Custom” is selected in the
Country/Region field.
Field Name Values Description
Tone Pair dial tone, busy tone ring tone, ‘unobtainable’ tone, survivability tone, re-order tone
Tone Pair Values
Frequency 1 Frequency in Hertz
Identifies the type of telephony signaling tone for which frequencies are being specified.
Frequency 2 Frequency in Hertz
Gain 1
Gain 2
Cadence 1
+3dB to –31dB and “mute” setting
+3dB to –31dB and “mute” setting integer time value in milliseconds; zero value
About Defaults: US telephony values are used as defaults on this screen.
Frequency of lower tone of pair.
This outbound tone pair enters the MultiVOIP at the input port.
Frequency of higher tone of pair.
This outbound tone pair enters the MultiVOIP at the input port.
Amplification factor of lower frequency of pair. This figure describes amplification that the MultiVOIP applies to outbound tones entering the MultiVOIP at the input port. Default: -16dB
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
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Field Name
Cadence 2
Cadence 3
Cadence 4
Values for dial-tone indicates continuous tone duration in milliseconds duration in milliseconds duration in milliseconds
Chapter 4—Configuring your VOIP
Description duration of first period of tone being “on” in the cadence of the telephony signal.
Cadence 2 is duration of first “off” period in signaling cadence.
Cadence 3 is duration of second “on” period in signaling cadence.
Cadence 4 is duration of second “off” period in the signaling cadence.
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 has its own email account (with Login Name and
Password) on some mail server connected to the IP network. Using this account, the MultiVOIP sends email messages containing log report information. The “Recipient” of the log report email is ordinarily the VOIP administrator. Because the MultiVOIP cannot receive email, you must set up a “Reply-To” address. Ordinarily, the “Reply-To” address belongs to 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 Reply-To address receives error or failure messages regarding the emailed reports.
The SMTP Parameters screen is shown below:
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Field Name
Enable SMTP
Requires
Authentication
Login Name
Password
Mail Server IP
Address
Port Number
Mail Type
Subject
Reply-To Address
Recipient Address
Values
Y/N
Y/N alpha-numeric alpha-numeric
n.n.n.n
25 text or html text email address email address
Mail Criteria
Description
In order to send log reports by email, this box must be checked. However, to enable SMTP functionality, you must also select “SMTP” in the Logs screen.
If this checkbox is checked, the MultiVOIP will send Authentication information to the SMTP server. The authentication information indicates whether or not the email sender has permission to use the SMTP server.
This is the User Name for the MultiVOIP unit’s email account.
Login password for MultiVOIP unit’s email account.
This is the mail server’s IP address. This mail server must be accessible on the IP network to which the MultiVOIP is connected.
25 is a standard port number for SMTP.
Mail type in which log reports will be sent.
User specified. Subject line that will appear for all emailed log reports for this
MultiVOIP unit.
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 receives log reports.
Criteria for sending log summary by email. The log summary email is sent when the user-specified number of log messages has accumulated, or once every day or multiple days, whichever comes first.
Number of Records
Number of Days integer integer
The number of log records that must accumulate to trigger the sending of a logsummary email.
The number of days that must pass before triggering the sending of a log-summary email.
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 Field Definitions
Field
Select All
Channel Number
Duration
Description
Log report to include all fields shown.
Data channel carrying call.
Length of call.
Packets Sent
Bytes Sent
Packets Lost
Outbound Digits
Total packets sent in call.
Total bytes sent in call.
Packets lost in call.
The DTMF dialing digits received by this
Field
Start Date,
Time
Call Mode
Description
Date and time the phone call began.
Voice or fax.
Packets
Received
Total packets received in call.
Bytes Received Total bytes received in call.
Coder Voice Coder /Compression Rate used for call will be listed in log.
Prefix Matched When selected, the phonebook prefix matched in processing the call will be listed in log.
Call Type Indicates the Call Signaling protocol used
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Received
Call Status
Call Direction
Server Details gateway from the remote gateway presuming that DTMF is set to "Out of
Band."
Successful or unsuccessful.
Indicates call’s originating party.
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.
DTMF
Capability 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-ofband 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.”
Disconnect Reason Indicates whether the call was disconnected simply because the desired conversation was done or some other irregular cause occasioned disconnection
(for example, a technical error or failure).
Values are "Normal" and "Local" disconnection.
From Details
Gateway Number
IP Address
Descript
Options
Originating gateway
IP address where call originated.
Identifier of site where call originated.
When selected, log will not Silence
Compression and Forward Error Correction by call originator.
Outbound
Digits Sent
To Details
Gateway Name Completing or answering gateway
IP Address IP address where call was completed or answered.
Descript
Options
Identifier of site where call was completed or answered.
When selected, log will not use Silence
Compression and Forward Error
Correction by party answering call.
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.
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RADIUS settings
The fields of the RADIUS screen are described in the table below.
Field Name Values Description
Enable Accounting
Server Address
Accounting Port
Retransmission
Interval
Number of
Retransmissions
Shared Secret
Select Attributes
(button)
Y/N
n.n.n.n
1 - 65535
0 - 255 alpha-numeric
--
When checked, the MultiVOIP will access the accounting functionality of the RADIUS server.
IP address of the RADIUS server that handles accounting (billing) for the current
MultiVOIP unit.
TDM time slot at which RADIUS accounting information will be transmitted 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.
Client encryption key for the current VOIP unit.
Gives access to RADIUS Attributes screen. On Attributes screen, one can specify the parameters to be tallied by the RADIUS server for accounting (usually billing) purposes.
The RADIUS dialog box has a secondary dialog box, RADIUS Attributes, 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 server.
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Field
Select All
Description
Log report to include all fields shown.
Data channel carrying call.
Field
Start Date, Time
Description
Date and time the phone call began.
Channel
Number
Duration
Packets Sent
Bytes Sent
Length of call.
Total packets sent in call.
Total bytes sent in call.
Call Mode
Packets Received
Bytes Received
Coder
Voice or fax.
Total packets received in call.
Total bytes received in call.
Voice Coder /Compression Rate used for call will be listed in log.
Packets Lost Packets lost in call. Prefix Matched When selected, the phonebook prefix matched in processing the call will be listed in log.
Successful or unsuccessful. Outbound
Digits Sent
DTMF digits received by this gateway from remote gateway (if that DTMF set to "Out of Band").
Call Status
Server Details The IP address of the traffic control server being used will be displayed here if the call is handled through that server. The Options field refers to non-mandatory server features that might be activated. For example, with
H.323, various H.323 Version 4 options might be listed.
From Details
Originating gateway Gateway
Number
IP Address
Descript
Options
IP address where call originated.
Identifier of where call originated.
When selected, log will not use Silence
Compression and Forward Error
Correction by call originator.
Gateway
Name
To Details
Completing or answering gateway
IP Address IP address where call was completed/answered.
Descript Identifier of where call was completed/answered.
Options When selected, log will not use Silence Compression and Forward Error Correction by party answering call.
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),
● through email (SMTP)
● at the MultiVOIP Manager remote VOIP system management program (SNMP).
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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).
Field Name
Enable Console
Messages
Filters (button)
Turn Off Logs
Logs Buttons
GUI
SNMP
SMTP
SysLog Server
Enable
IP Address
Port
Online Statistics
Updation Interval
Values Description
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.
Click to access secondary screen on where console messages can be included/excluded by category and on a per-channel basis.
Y/N Check to disable log-reporting function.
Only one of these three log reporting methods, GUI, SMTP, or SNMP, may be chosen.
User must view logs at the MultiVOIP configuration program.
Log messages will be delivered to the MultiVOIP Manager 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 IP address of computer, in VOIP network, on which SysLog Server program is running.
514 Logical port for SysLog Server. 514 is commonly used. integer Set the interval (in seconds) at which logging information will be updated.
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.
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Field Name
Enable (STUN)
Name/IP (Server)
Port (Server;
NAT/STUN)
Keep Alive (Timers;
NAT/STUN)
Values
Y/N
n.n.n.n
port; default=
3478
60 – 3600
(seconds)
Chapter 4—Configuring your VOIP
Description
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.
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.
Supplementary Services
Supplementary Services features derive from the H.450 standard, which brings to the VOIP telephony functionality once only available with PSTN or PBX telephony. Even though the H.450 standard refers only to
H.323, Supplementary Services are still applicable to the SIP and SPP VOIP protocols.
Of the features implemented under Supplementary Services, three are very closely related: Call Transfer, Call
Hold, and Call Waiting. Call Name Identification is similar but not identical to the premium PSTN feature commonly known as Caller ID.
Call Transfer. Call Transfer allows one party to re-connect the party with whom they have been speaking to a third party. The first party is disconnected when the third party becomes connected. Feature is used by a programmable phone keypad sequence (for example, #7).
Call Hold. Call Hold allows one party to maintain an idle (non-talking) connection with another party while receiving another call (Call Waiting), while initiating another call (Call Transfer), or while performing some other call management function. Feature is used by a programmable phone keypad sequence (for example, #7).
Call Waiting. Call Waiting notifies an engaged caller of an incoming call and allows them to receive a call from a third party while the party with whom they have been speaking is put on hold. Feature is used by a programmable phone keypad sequence (for example, #7).
Call Name Identification. When enabled for a given VOIP unit (the ‘home’ VOIP), this feature gives notice to remote VOIPs involved in calls. Notification goes to the remote VOIP administrator, not to individual phone stations. When the home VOIP is the caller, a plain English descriptor will be sent to the remote VOIP identifying the channel over which the call is being originated (for example, “Calling Party - Omaha Sales Office Line 2”). If that VOIP channel is dedicated to a certain individual, the descriptor could say that, as well (for example “Calling
Party - Harold Smith in Omaha”). When the home VOIP receives a call from any remote VOIP, the home VOIP sends a status message back to that caller. This message confirms that the home VOIP’s phone channel is either busy or ringing or that a connection has been made (for example, “Busy Party - Omaha Sales Office Line 2”).
These messages appear in the Statistics – Call Progress screen of the remote VOIP.
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Field Name
Select Channel
Call Transfer Enable
Transfer Sequence
Call Hold Enable
Hold Sequence
Call Waiting Enable
Retrieve Sequence
Call Name
Identification Enable
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Values
Channel 1
Y/N
Description
The channel to be configured is selected here.
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.
Any phone keypad character
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.
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*#.
Select to enable Call Hold function in VOIP unit. Y/N phone keypad characters
Y/N
Phone keypad characters, two characters in length
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 #).
Select to enable Call Waiting function in VOIP unit.
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.
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
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Chapter 4—Configuring your VOIP
Field Name
Calling Party, Allowed
Name Type (CNI)
Alerting Party,
Allowed Name Type
(CNI)
Busy Party, Allowed
Name Type (CNI)
Connected Party,
Allowed Name Type
(CNI)
Values Description
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.)
If the ‘home’ VOIP unit is originating the call and Calling Party is selected, then the identifier (from the
Caller Id field) will be sent to the remote VOIP unit being called. The Caller Id field gives the remote
VOIP administrator a plain-language identifier of the party that is originating the call occurring on a specific channel.
This field is applicable only when the ‘home’ VOIP unit is originating the call.
Example. Suppose a VOIP system has offices in both Denver and Omaha. In the Omaha VOIP unit (the
‘home’ VOIP in this example), Call Name Identification has been enabled, Calling Party has been enabled as an Allowed Name Type, and “Omaha Sales Office Voipchannel 2” has been entered in the
Caller Id field.
When channel 2 of the Omaha VOIP is used to make a call to any other VOIP phone station (for example, the Denver office), the message “Calling Party - Omaha Sales Office Voipchannel 2” will appear in the “Caller Id” field of the Statistics - Call Progress screen of the Denver VOIP.
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.
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Field Name
Caller ID
Default
Copy Channel
Chapter 4—Configuring your VOIP
Values Description
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.
This is the identifier of a specific channel of the ‘home’ VOIP unit. The Caller Id field typically describes a person, office, or location, for example, “Harry Smith,” or “Bursar’s Office,” or “Barnesville Factory.”
When this button is clicked, all Supplementary Service parameters are set to their default values.
Copies the Supplementary Service attributes of one channel to another channel. Attributes can be copied to multiple channels or all channels at once.
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.
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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 error message will appear.
Error pop-up
To change the COM port setting, use the COM Port Setup dialog box, by going to the Connection pull-down menu and choosing “Settings” or use the left side control panel. In the “Select Port” field, select a COM port that is available on the PC (if no COM ports are currently available, re-allocate COM port resources in the computer’s
MS Windows operating system to make one available).
Fixing a Cabling Problem
If the MultiVOIP cannot be located by the computer, three error messages will appear (saying “Multi-VOIP Not
Found”, “Phone Database Not Read” and “Password Phone Database Not Read).
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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Introduction
When a VOIP serves a PBX system, it’s important that the operation of the VOIP be transparent to the telephone end user. That is, the VOIP should not entail the dialing of extra digits to reach users elsewhere on the network that the VOIP serves. On the contrary, VOIP service more commonly reduces dialed digits by allowing users
(served by PBXs in facilities in distant cities) to dial their co-workers with 3-, 4-, or 5-digit extensions as if they were in the same facility.
Furthermore, the setup of the VOIP generally should allow users to make calls on a non-toll basis to any numbers accessible without toll by users at all other locations on the VOIP system. Consider, for example, a company with VOIP-equipped offices in New York, Miami, and Los Angeles, each served by its own PBX. When the VOIP phone books are set correctly, personnel in the Miami office should be able to make calls without toll not only to the company’s offices in New York and Los Angeles, but also to any number that’s local in those two cities.
To achieve transparency of the VOIP telephony system and to give full access to all types of non-toll calls made possible by the VOIP system, the VOIP administrator must properly configure the “Outbound” and “Inbound” phone-books of each VOIP in the system.
The “Outbound” phonebook for a particular VOIP unit describes the dialing sequences required for a call to originate locally (typically in a PBX in a particular facility) and reach any of its possible destinations at remote
VOIP sites, including non-toll calls completed in the PSTN at the remote site.
The “Inbound” phonebook for a particular VOIP unit describes the dialing sequences required for a call to originate remotely from any other VOIP sites in the system, and to terminate on that particular VOIP.
Briefly stated, the MultiVOIP’s Outbound phonebook lists the phone stations it can call; its Inbound phonebook
describes the dialing sequences that can be used to call that MultiVOIP and how those calls will be directed. The phone numbers are not literally “listed” individually, but are, instead, described by rule.
Identify Remote VOIP Site to Call
When you’re done installing the MultiVOIP, you’ll want to confirm that it is configured and operating properly.
To do so, it’s good to have another VOIP that you can call for testing purposes. You’ll want to confirm end-toend 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 non-standard protocol developed by Multi-Tech. SPP is not compatible with the “Proprietary” protocol used in Multi-Tech’s earlier generation of VOIP gateways.
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Chapter 5 — Phonebook Configuration
Phonebook Starter Configuration
This section will walk you through the phone book setup with examples that will aid in entering the correct numbers needed to have the MultiVOIP working correctly. To do this part of the setup, you need access to another VOIP that you can call to conduct a test. It should be at a remote location, typically somewhere outside of your building. You must know the phone number and IP address for that site. We are assuming here that the
MultiVOIP will operate in conjunction with a PBX.
You must configure both the Outbound Phonebook and the Inbound Phonebook. A starter configuration only means that two VOIP locations will be set up to begin the system and establish VOIP communication. Once this is accomplished, you can easily add other VOIP sites to the network.
Outbound Phonebook
1.
Open the MultiVOIP program. (Start | MultiVOIP 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, and so on) that you’ll be calling.
Follow the example that best fits your situation:
Euro, National Call Example North America,
Long-Distance Example
Technician in Seattle (area 206) must set up one VOIP there, another in Chicago (area 312, downtown).
Technician in central London
(area 0207) to set up VOIP there, another in Birmingham (area
0121).
Answer: write down 312. Answer: write down 0121.
Euro, International Call Example
Technician in Rotterdam
(country 31; city 010) to set up one VOIP there, another in
Bordeaux (country 33; area 05).
Answer: write down 3305.
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 (that is, 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 number.
Answer: write down 90.
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.
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4.
In the “Destination Pattern” field of the Add/Edit Outbound Phonebook screen, enter the digits from the previous examples.
Euro, National Call Example Euro, International Call Example North America,
Long-Distance Example
Seattle/Chicago system.
Answer: enter 81312 as
Destination Pat-tern in
Outbound Phone-book of Seattle VOIP.
London/Birmingham system.
Leading zero of Birmingham area code is dropped when combined with national-dialing access code. (Such practices vary by country.)
Answer: enter 90121 as
Destination Pattern in Outbound
Phonebook of London VOIP.
Not 900121.
Rotterdam/Bordeaux system.
Answer: enter 903305 as
Destination Pattern in
Outbound Phonebook of
Rotterdam VOIP.
5. In the “Remove Prefix” field, enter the initial PBX access digit (“8” or “9”).
6.
North America,
Long-Distance Example
Euro, National Call Example Euro, International Call Example
Seattle/Chicago system.
Answer: enter 8 in “Remove
Prefix” field of Seattle Outbound
Phonebook.
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.
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.
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.
7.
Click OK to exit from the Add/Edit Outbound Phonebook screen.
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, and so on) 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).
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Chapter 5 — Phonebook Configuration
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
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.
Euro, International Call Example
Rotterdam/Bordeaux system.
Rotterdam is country code 31, city code 010. Bordeaux employees must dial 903110 before dialing any Rotterdam number on the VOIP system.
Answer: 03110 is prefix to be removed by local
(Rotterdam) VOIP.
4.
In the “Add Prefix” field, enter any digits that must be dialed from your local VOIP to gain access to the
PSTN.
Euro, National Call Example Euro, International Call Example 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.
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 (that is, 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.
Euro, National Call Example Euro, International Call Example North America,
Long-Distance Example
Seattle/Chicago system.
Possible Description:
Free Seattle access, all employees
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.
Outbound Phone Book
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Add/Edit Outbound Phone Book
Accept Any
Number
Add/Edit screen
Enter Outbound Phone Book data for your MultiVOIP unit. Note that the Advanced button gives access to the
Alternate IP Routing feature, if needed. Alternate IP Routing can be implemented in a secondary screen (as described after the primary screen field definitions below).
Field Name Values Description
Y/N
Destination
Pattern prefixes, area codes, exchanges, line numbers, extensions
When checked, “Any Number” appears as the value in the Destination Pattern field.
How Any Number works depends on whether or not an external routing device is used.
When no external routing device is used. If Any Number is selected, calls to phone numbers that don’t match a listed Destination Pattern are 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 that don’t match a listed Destination Pattern are 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.
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.
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Field Name Values Description
Total Digits
Remove Prefix
Add Prefix
IP Address as needed dialed digits dialed digits
n.n.n.n
Description
Protocol Type alpha-numeric
SIP or H.323 or SPP
H.323 fields
Use Gatekeeper Y/N
Number of digits the phone user must dial to reach specified destination. This field not
used in North America
Portion of dialed number to be removed before completing call to destination.
Digits to be added before completing call to destination.
The IP address to which the call will be directed if it begins with the destination pattern given.
Describes the facility or geographical location at which the call will be completed.
Indicates protocol to be used in outbound transmission. Single Port Protocol (SPP) is a non-standard protocol designed by Multi-Tech.
Gateway H.323
ID alpha-numeric
Gateway Prefix numeric
H.323 Port
Number
1720
Indicates whether or not gatekeeper is used.
The H.323 ID assigned to the destination MultiVOIP. Only valid if “Use Gatekeeper” is enabled for this entry.
This number becomes registered with the GateKeeper. Call requests sent to the gatekeeper and preceded by this prefix will be routed to the VOIP gateway.
This parameter pertains to Q.931, which is the H.323 call signaling protocol for setup and termination of calls (aka ITU-T Recommendation I.451). H.323 employs only one
“well-known” port (1720) for Q.931 signaling. If Q.931 message-oriented signaling protocol is used, 1720 must be chosen as the H.323 Port Number.
SIP Fields
Use Proxy
Transport
Protocol
Y/N
TCP or
UDP
SIP Port Number 5060 or other
*See RFC 3087
(“Control of
Service Context using SIP Request-
URI,” by the
Network Working
Group).
SIP URL sip.userphone@ho
stserver, where
“userphone” is the telephone number and “hostserver” is the domain name or an address on the network
SPP Fields
Select if proxy server is used.
VOIP administrator must choose between UDP and TCP transmission protocols. UDP is a high-speed, low-overhead connectionless protocol where data is transmitted without acknowledgment, guaranteed delivery, or guaranteed packet sequence integrity. TCP is slower connection-oriented protocol with greater overhead, but having acknowledgment and guarantees delivery and packet sequence integrity.
The SIP Port Number is a UDP logical port number. The VOIP will “listen” for SIP messages at this logical port. If SIP is used, 5060 is the default, standard or “well known” port number to be used. If 5060 is not used, then the port number used is that specified in the SIP Request URI (Universal Resource Identifier).
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.
Use Registrar Y/N
Select this checkbox to use registrar when VOIP system is operating in the
“Registrar/Client” SPP mode. In this mode, one VOIP (the registrar, as set in Phonebook
Configuration screen) has a static IP address and all other VOIPs (clients) point to the registrar’s IP address as functionally their own. However, if your VOIP system overall is operating in “Registrar/Client” mode but you want to make an exception and use Direct mode for the destination pattern of this particular Add/Edit Phonebook entry, leave this checkbox unselected. Also do not select this if your overall VOIP system is operating in the Direct SPP mode – in this mode all VOIPs are peers with unique static IP addresses.
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Field Name
Port Number
Values numeric
Description
When operating in “Registrar/Client” mode, this is the port by which the gateway receives all SPP data and control messages from the registrar gateway. (This ability to receive all data and messages via one port allows the VOIP to operate behind a firewall with only one port open.)
When operating in “Direct” mode, this is the Port by which peer VOIPs receive data and messages.
Phone number associated with alternate IP routing. Alternate Phone
Number numeric
Remote Device is [legacy VOIP]
Y/N When checked, this MultiVOIP can operate with ‘first-generation’ MultiVOIP units in the same IP network. These include MVP-110/120/200/400/800.
Advanced button
Gives access to secondary screen where an Alternate IP Route can be specified for backup or redundancy of signal paths. For SIP & H.323 operation only.
Click on Advanced to bring up the Alternate Routing screen. This 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.
Advanced button
Alternate Routing Field Definitions
Field Name
Alternate IP
Address
Round Trip
Delay
Values n.n.n.n
Default is 300 milliseconds
Description
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 here, the data stream will be diverted to the alternate destination specified as the Alternate IP Address.
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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.
FXO
PSTN Line
4. Call completed via PSTN.
VOIP
IP
NETWORK
VOIP
PBX
FXS
1. Call originates.
2. IP network fails.
PSTN failover
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.
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Chapter 5 — Phonebook Configuration
Add/Edit Inbound Phone Book
Field Name
Accept Any
Number
Values
Y/N
Remove Prefix dialed digits
Add Prefix dialed digits
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Add/Edit Inbound Phone Book
Description
When checked, “Any Number” appears as the value in the Remove Prefix field.
The Any Number feature of the Inbound Phone Book does not work when an external routing device is used (Gatekeeper for H.323 protocol, Proxy for SIP protocol, Registrar for SPP protocol).
When no external routing device is used. If Any Number is selected, calls received from phone numbers not matching a listed Prefix (shown in the Remove Prefix column of the
Inbound Phone Book) will be admitted into the VOIP on the channel listed in the Channel
Number field. “Any Number” can be used in addition to one or more Prefixes. portion of dialed number to be removed before completing call to destination
(often a local PBX) digits to be added before completing call to destination
(often a local PBX)
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Chapter 5 — Phonebook Configuration
Field Name
Channel
Number
Description --
Call Forward Parameters
Enable Y/N
Forward
Condition
Forward
Destination
Ring Count
Registration
Option
Parameters
Values channel, or
“Hunting”
Unconditional,
Busy,
No Response
Description
Channel number to which the call will be assigned as it enters the local telephony equipment (often a local PBX). “Hunting” directs the call to any available channel.
Describes the facility or geographical location at which the call originated.
Click the check-box to enable the call-forwarding feature.
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
IP address, phone number, port number, etc
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,
(c) IP address: port number,
(d) phone number: IP address: port number,
(e) SIP URL, or
(f) phone #: IP address.
For SPP calls, the Forward Destination can be one of the following:
(a) phone number,
(b) IP address: port, or
(c) phone number: IP address: port. integer When “No Response” is condition for forwarding calls, this determines how many unanswered rings are needed to trigger the forwarding.
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.
Authorized User Name and Password for SIP
To enable the Registration Options on the Add/Edit Inbound Phone Book, 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.
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Phone Book Save and Reboot
When 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).
Phonebook Examples
North America
The following example demonstrates how Outbound and Inbound Phonebook entries work in a situation of multiple area codes. Consider a company with offices in Minneapolis and Baltimore.
Notice first the area code situation in those two cities: Minneapolis’s local calling area consists of multiple adjacent area codes; Baltimore’s local calling area consists of a base area code plus an overlay area code.
Company
VOIP/PBX
SIte
NW
Suburbs
763 Mpls
612
St. Paul
& Suburbs
651
...
SW Suburbs
952
Baltimore/
Outstate MD
Overlay
443
Company
VOIP/PBX
SIte
Baltimore
410
North America example
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An outline of the equipment setup in both offices is shown below.
Equipment setup example
The screen below shows Outbound Phonebook entries for the VOIP located in the company’s Baltimore facility.
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Baltimore example
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The entries in the Minneapolis VOIP’s Inbound Phonebook match the Outbound Phonebook entries of the
Baltimore VOIP, as shown below.
Minneapolis example
To call the Minneapolis/St. Paul area, a Baltimore employee must dial eleven digits. (In this case, we are assuming that the Baltimore PBX does not require an “8” or “9” to seize an outside phone line.)
If a Baltimore employee dials any phone number in the 612 area code, the call will automatically be handled by the company’s VOIP system. Upon receiving such a call, the Minneapolis VOIP will remove the digits “1612”. But before the suburban-Minneapolis VOIP can complete the call to the PSTN of the Minneapolis local calling area, it must dial “9” (to get an outside line from the PBX) and then a comma (which denotes a pause to get a PSTN dial tone) and then the 10-digit phone number which includes the area code (612 for the city of Minneapolis; which is different than the area code of the suburb where the PBX is actually located -- 763).
A similar sequence of events occurs when the Baltimore employee calls number in the 651 and 952 area codes because number in both of these area codes are local calls in the Minneapolis/St. Paul area.
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 generally matches the Outbound Phone
Book of the Minneapolis VOIP.
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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”.
Outbound Minneapolis example
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Chapter 5 — Phonebook Configuration
Europe
The most direct use of the VOIP system is making calls between the offices where the VOIPs are located.
Consider, for example, the Wren Clothing Company. This company has VOIP-equipped offices in London, Paris, and Amsterdam, each served by its own PBX. VOIP calls between the three offices completely avoid international long-distance charges. These calls are free. The phonebooks can be set up to allow all Wren
Clothing employees to contact each other using 3-, 4-, or 5-digit numbers, as though they were all in the same building.
United Kingdom
Wren Clothing Co.
VOIP/PBX Site
London
Wren Clothing Co.
VOIP/PBX Site
Amsterdam
The
Netherlands
Wren Clothing Co.
VOIP/PBX Site
Paris
Free VOIP Calls
France
Free VOIP calls
In another use of the VOIP system, the local calling area of each VOIP location becomes accessible to all of the
VOIP system’s users. As a result, international calls can be made at local calling rates. For example, suppose that
Wren Clothing buys its zippers from The Bluebird Zipper Company in the western part of metropolitan London.
In that case, Wren Clothing personnel in both Paris and Amsterdam could call the Bluebird Zipper Company without paying international long-distance rates. Only London local phone rates would be charged. This applies to calls completed anywhere in London’s local calling area. Generally, local calling rates apply only within a single area code, and, for all calls outside that area code, national rates apply. There are, however, some
European cases where local calling rates extend beyond a single area code. Local rates between Inner and Outer
London are one example of this. It is also possible, in some locations, that calls within an area code may be national calls - but this is rare.
United Kingdom
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
France
Local calling area
Calls at London local rates
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 greater Paris will be accessible to all company offices as local calls.
Vendors in Guildford, Lyon, and Rotterdam can be contacted as national calls by all company offices.
France Country Code: 33
Paris: Area 01
Lille
Reims
Rouen
Nantes Strasbourg
Bordeaux
Ly on
Toulouse
Marseille
UK & France codes
The Netherlands
Country Code: 31
Texel 0222
058
Leeuwarden
Den Helder 0223
050
Groningen
Beverwijk 0251
Haarlem 023
Aalsmeer0297
070
The Hague
038 Zwolle
0299 Purmerend
020 Amsterdam
0294 Weesp
010
Rotterdam
026
Arnhem
053
Enschede
0118
Middelburg 040
Eindhoven
043
Maastricht
Netherlands codes
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An outline of the equipment setup in these three offices is shown below.
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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.
London example outbound
The Inbound Phone Book for the London VOIP is shown below.
London example inbound
NOTE: Commas are allowed in the Inbound Phonebook, but not in the Outbound Phonebook. Commas denote a brief pause for a dial tone, allowing time for the PBX to get an outside line.
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The screen below shows Outbound Phone Book entries for the VOIP located in the company’s Paris facility.
Paris example outbound
The Inbound Phone Book for the Paris VOIP is shown below.
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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.
Amsterdam example outbound
The Inbound Phone Book for the Amsterdam VOIP is shown below.
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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:
VOIP Caller ID Case #1 – Call, through telco central office with standard CID, enters VOIP system.
CID Flow
Call is received here.
FXS
CID
Terminating
VoIP xxxyyyzzzz
J.Q. Public
Display shows:
Clock:
5-31,
1:42pm
CID Number: 763-555-8794
CID Name: Melvin Jones
Time Stamp: Date: 05/31
Time:1:42pm
IP
Network
H.323 or SPP
Protocol *
CID
Generating
VoIP FXO
Central Office with standard telephony
Caller ID service
Call originates here at 1:42pm, May 31.
phone of: xxxyyyzzzz
J.Q. Public
Melvin Jones
763-555-8794
* In x.06 release, when SIP protocol is used,
CID Name field will duplicate value in
CID Number field.
Caller ID example 1
VOIP Caller ID Case #2 – Call, through telco central office without standard CID, enters H.323 VOIP system.
Call is received here.
FXS
CID
Terminating
VoIP xxxyyyzzzz
J.Q. Public
Clock:
7/10, 4:19pm
Display shows:
CID Flow
IP
Network
H.323 Protocol
*
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 field.
CID
Generating
VoIP
Ch2
Ch1
FXO
Ch3
Ch4
Central Office without standard telephony
Caller ID service
Phone Book Configuration
Gateway Name: Anoka-Whse-VP3
Call originates here at 4:19pm, July 10.
xxxyyyzzzz
J.Q. Public phone of:
Wilda Jameson
763-555-4071
Q.931 Parameters Inbound Phone Book {Channel 2}
Remove Prefix Add Prefix Forward/Addr
Gatekeeper RAS Parameters
423
748
Caller ID example 2
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VOIP Caller ID Case #3 – Call, through telco central office without standard CID, enters SPP VOIP system.
CID Flow
Call is received here.
FXS
Terminating
VoIP x xxy yy zz zz
J.Q. Pu bl ic
Clock:
15:26, 5-31
Display shows:
IP
Network
SPP Protocol
Ch1
Generating
VoIP
Ch2
Ch3
Ch4
FXO
Central Office without standard telephony
Caller ID service
Call originates here at 5:47pm, Sept 27.
phone of: xx xyy yz zz z
J.Q. Pu bl ic
Henry Brampton
763-555-4077
CID Number: 423
CID Name: Shipping Dept
Time Stamp: Date: 0927
Time: 1747
... if “Description” field in Add/Edit
Inbound Phone Book is used
OR
Inbound Phone Book {Channel 2}
Remove Prefix Add Prefix Forward/Addr
423
748
Phone Book Configuration
Gateway Name: Anoka-Whse-VP3
CID Number: 423
CID Name: Anoka-Whse-VP3
Time Stamp: Date: 0927
Time: 1747
... if “Description” in Add/Edit
Inbound Phone Book is blank
Use as default entry
Remove Prefix: Gatekeeper RAS Parameters
Add Prefix:
Channel Number: Channel 2
Description:
Shipping Dept
Caller ID example 3
VOIP Caller ID Case #4 – Remote FXS call on H.323 VOIP system.
CID Flow
Call is received here.
FXS
CID
Terminating
VoIP xxxyyyzzzz
J.Q. Public
Clock:
10/03, 4:51pm
Display shows:
IP
Network
CID
Generating
VoIP
Ch2
Ch1
401
402
FXS
Ch3
403
Ch4
404
H.323 Protocol
*
*
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 value in
CID Number field.
xxxyyyzzzz
J.Q. Public
Call originates here at 4:51pm, Oct 3.
phone of: Nigel Thurston
763-555-9401
Phone Book Configuration
Gateway Name: Anoka-Whse-VP3
Q.931 Parameters Inbound Phone Book {Channel 2}
Remove Prefix Add Prefix Forward/Addr
Gatekeeper RAS Parameters
423
748
Caller ID example 4
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VOIP Caller ID Case #5 – Call through telco central office without standard CID enters DID channel in H.323 VOIP system.
Call is received here.
FXS
CID
Terminating
VoIP xxxyyyzzzz
J.Q. Public
Clock:
11/15, 6:17pm
Display shows:
CID Flow
IP
Network
H.323 Protocol
*
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
Generating
VoIP
Ch2
Ch1
DID
Ch3
Ch4
Central Office without standard telephony
Caller ID service
Phone Book Configuration
Gateway Name: Anoka-Whse-VP3
Call originates here at 6:17pm, Nov 15.
xxxyyyzzzz
J.Q. Public phone of:
Edwin Smith
763-743-5873
Q.931 Parameters Inbound Phone Book {Channel 2}
Remove Prefix Add Prefix Forward/Addr
Gatekeeper RAS Parameters
423
748
Caller ID example 5
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This chapter describes 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
System Information
Call Progress
Logs
IP Statistics
Link Management
Registered Gateway Details
Servers o H.323 GateKeepers o SIP Proxies o SPP Registrars
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.
Field Name
Boot Version
Firmware Version
Configuration Version
Phone Book Version
IFM Version
Mac Address
Up Time
Hardware ID
Values
nn.nn alphanumeric
nn alphanumeric numeric days: hours: mm:ss alphanumeric
nn.nn alphanumeric nn.nn.nn alphanumeric nn.nn.
nn.nn alphanumeric
System information
Description
Indicates the version of the code that is used at the startup (booting) of the VOIP. The boot code version is independent of the software version.
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.
Denotes the number assigned as the VOIP unit’s unique Ethernet address.
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).
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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Field Name
Channel
Values
1-n
Description
Number of data channel or time slot on which the call is carried. This is the channel for which call-progress details are being viewed.
Call Details
Duration
Mode
Voice Coder
IP Call Type
IP Call Direction
Packet Details
Packets Sent
Packets Rcvd
Bytes Sent
Bytes Rcvd
Packets Lost
Gateway Name (to)
H/M/S
Voice or FAX
G.723, G.729,
G.711, and so on
H.323, SIP, or SPP incoming, outgoing integer value integer value integer value integer value integer value
From – To Details
Gateway Name (from) alphanumeric string
IP Address (from)
Options n.n.n.n
SC, FEC
The length of the call in hours, minutes, and seconds (hh:mm:ss).
Indicates whether the call being described was a voice call or a FAX call.
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.
The number of data packets sent over the IP network during the call.
The number of data packets received over the IP network during the call.
The number of bytes of data sent over the IP network during the call.
The number of bytes of data received over the IP network during the call.
The number of voice packets from this call that were lost after being 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.
Displays VOIP transmission options in use on the current call. These may include
Forward Error Correction or Silence Compression.
Identifier for the VOIP gateway that handled the completion of this call.
IP Address (to)
Options alphanumeric string
n.n.n.n
SC, FEC
IP address to which the call was sent.
Displays VOIP transmission options in use on the current call. These may include
Forward Error Correction or Silence Compression.
DTMF/Other Details
Prefix Matched
Outbound Digits Sent
Outbound Digits
Received
Server Details
DTMF Capability specified dialing digits
0-9, #, *
0-9, #, *
n.n.n.n and/or other related descriptions inband, out of band
Expressions differ slightly for different Call
Signaling protocols (H.323,
SIP, or SPP).
Displays the dialed digits that were matched to a phonebook entry.
The digits transmitted by the MultiVOIP to the PBX/telco for this call.
Of the digits transmitted by the MultiVOIP to the PBX/telco for this call, these are the digits that were confirmed as being received.
The IP address (and so on) 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 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-ofband condition or "Inband" to indicate the in-band condition. For SPP it can display "Out of Band RFC2833" or "Inband".
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Field Name Values
Supplementary Services Status
Call on Hold alphanumeric
Call Waiting
Caller ID
Description alphanumeric
“Calling Party +
identifier”;
“Alerting Party +
identifier”;
“Busy Party
+ identifier”;
“Connected
Party +
identifier”
Describes held call by its IP address source, location/gateway identifier, and hold duration. Location/gateway identifiers come from Gateway Name field in Phone
Book Configuration screen of remote VOIP.
Describes waiting call by its IP address source, location/gateway identifier, and hold duration. Location/gateway identifiers come from Gateway Name field in
Phone Book Configuration screen of remote VOIP.
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.
Call Status fields
Call Status
Call Control Status hangup, active
Tun, FS + Tun,
AE, Mux
Silence Compression
Forward Error
Correction
SC
FEC
Shows condition of current call.
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
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Field Name
Log # column
Start Date, Time column
Values
1 or higher dd:mm:yyyy hh:mm:ss
Duration column
Type
Status column
IP Direction
Mode column
From column
To column
Special Buttons
Previous
Next
First
Last hh:mm:ss
H.323, SIP, SPP success or failure incoming, outgoing voice or FAX gateway name gateway name
--
--
--
--
-- Delete File
Call Details
Voice coder
Disconnect Reason
DTMF Capability
Coder protocol
"Normal" or
"Local" disconnection. inband, out of band
Expressions differ slightly for different Call
Signaling protocols.
Outbound Digits
Received
0-9, #, *
Outbound Digits Sent 0-9, #, *
Server Details n.n.n.n
Description
All calls are assigned an event number in chronological order, with the most recent call having the highest event number.
The starting time of the call. The date is presented as a day and a month of one or two digits, and a four-digit year. This is followed by a time-of-day in a two-digit hour, a two-digit minute, and a two-digit seconds value.
This describes how long the call lasted in hours, minutes, and seconds.
Indicates the Call Signaling protocol used for the call (H.323, SIP, or SPP).
Displays the status of the call (whether the call was completed or not).
Indicates whether the call is "incoming" or "outgoing" with respect to the gateway.
Indicates whether the event being described was a voice call or a FAX call.
Displays the name of the voice gateway that originates the call.
Displays the name of the voice gateway that completes the call.
Displays log entry before currently selected one.
Displays log entry after currently selected one.
Displays first log entry
Displays last log entry.
Deletes selected log file.
The voice coder being used on this call.
Indicates whether the call was disconnected simply because the desired conversation was done or some other irregular cause occasioned disconnection
(for example, 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-ofband condition or "Inband" to indicate the in-band condition. For SPP it can display "Out of Band RFC2833" or "Inband".
Packets sent
Packets received
Packets lost
Bytes sent
Bytes received integer value integer value integer value integer value integer value
The digits, sent by MultiVOIP to PBX/telco, that were acknowledged as having been received by the remote VOIP gateway.
The digits transmitted by the MultiVOIP to the PBX/telco for this call.
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.
Number of data packets sent over the IP network in the course of this call.
Number of data packets received over the IP network in the course of this call.
Number of voice packets from this call that were lost after being received from the IP network.
Number of bytes of data sent over the IP network in the course of this call.
Number of bytes of data received over the IP network in the course of this call.
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FROM Details
Gateway Name
IP Address
Options alphanumeric
n.n.n.n
FEC, SC
TO Details
Gateway Name
IP Address alphanumeric
n.n.n.n
Options
Supplementary Services Info
Call Transferred To
Call Forwarded To phone number phone number
IP Statistics
Identifier for the VOIP gateway that originated this call.
IP address of the VOIP gateway from which the call was received.
Displays VOIP transmission options used by the VOIP gateway originating the call.
These may include Forward Error Correction or Silence Compression.
Identifier for the VOIP gateway that completed (terminated) this call.
IP address of the VOIP gateway at which the call was completed.
Displays transmission options used by VOIP gateway terminating the call.
Number of party called in transfer.
Number of party called in forwarding.
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, connectionoriented transmission of data.
Both TCP and UDP split data into packets called “datagrams.” However, TCP includes extra headers in the datagram to enable retransmission of lost packets and reassembly of packets into their correct order if they arrive out of order. UDP does not provide this. Lost UDP packets are irretrievable; that is, out-of-order UDP packets cannot be reconstituted in their proper order.
Despite these obvious disadvantages, UDP packets can be transmitted much faster than TCP packets -- as much as three times faster. In certain applications, like audio and video data transmission, the need for high speed outweighs the need for verified data integrity. Sound or pictures often remain intelligible despite a certain amount of lost or disordered data packets (which comes through as static).
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“Clear” button --
Total Packets
Transmitted
Received integer value integer value
Received with
Errors
UDP Packets
Transmitted integer value
Received integer value integer value
Received with
Errors integer value
TCP Packets
Transmitted
Received integer value integer value
Received with
Errors integer value
RTP Packets
Transmitted
Received
Received with
Errors integer value integer value integer value
Received with
Errors
RTCP Packets
Transmitted
Received integer value integer value integer value
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Field Name
IP Address
Values
n.n.n.n
Description
IP address of the MultiVOIP. For an IP address to be displayed here, the MultiVOIP must have
DHCP enabled. Its IP address, in such a case, is assigned by the DHCP server.
Clears packet tallies from memory.
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.
Field Name Values
Monitor Link fields
IP Address to Ping n.n.n.n
Pings per Test 1-999
Response Timeout 500 – 5000 milliseconds
Ping Size in Bytes
Timer Interval between
Pings
32 – 128 bytes
0 or 30 – 6000 minutes
-- Start Now command button
Clear command button --
Link Status Parameters
IP Address column
No. of Pings Sent
No. of Pings Received
Round Trip Delay
(Min/Max/Avg)
Last Error n.n.n.n as listed as listed as listed, in milliseconds as listed
Link management
Description
This is the IP address of the target endpoint to be pinged.
This field determines how many pings will be generated by the Start Now command.
The duration after which a ping will be considered to have failed.
This field determines how long or large the ping will be.
This field determines how long of a wait there is between one ping and the next.
Initiates pinging.
Erases ping parameters in Monitor Link field group and restores default values.
These fields summarize the results of pinging.
Target of ping.
Number of pings sent to target endpoint.
Number of pings received by target endpoint.
Displays how long it took from time ping was sent to time ping response was received.
Indicates when last data error occurred.
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Registered Gateway Details
The Registered Gateway Details screen presents a real-time display of the special operating parameters of the
Single Port Protocol (SPP). These are configured in the Call Signaling screen and in the Add/Edit Outbound
Phone Book screen.
Field Name
Column Headings
Values
Description alphanumeric
IP Address
Port
Register
Duration
Status
No. of Entries
Details
Count of
Registered
Numbers
List of
Registered
Numbers
n.n.n.n n
Registered/ unregistered
Registered endpoints
Description
This is a descriptor for a particular VOIP gateway unit. This descriptor should generally identify the physical location of the unit (for example, city, building, and so on) and perhaps even its location in an equipment rack.
The RAS address for the gateway.
Port by which the gateway exchanges H.225 RAS messages with the gatekeeper.
The time remaining in seconds before the TimeToLive timer expires. If the gateway fails to reregister within this time, the endpoint is unregistered.
The current status of the gateway either registered or unregistered.
The number of gateways currently registered to the Registrar. This includes all SPP clients registered and the Registrar itself.
If a registered gateway is selected (by clicking on it in the screen), The "Count of Registered
Numbers" will indicate the number of registered phone numbers for the selected gateway.
When a client registers, all of its inbound phonebook's phone numbers become registered.
Lists all of the registered phone numbers for the selected gateway.
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Servers
H.323 GateKeepers
Field Name
IP Address
Port
GK Name
Type
Priority
Status n
Values
Column Headings
n.n.n.n
Description
The IP address of the gatekeeper.
TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it.
Identifier for gatekeeper alpha-numeric string
Primary,
Predefined n registered, not registered
This field describes the type of gateway as which the MultiVOIP is defined with respect to the gatekeeper
Priority level given.
The current status of the gateway either registered or unregistered.
SIP Proxies
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Field Name Values
Column Headings
IP Address n.n.n.n
Port port
Type
Status
Primary,
Alternate registered, not registered
Description
The IP address of the SIP proxy by which the MultiVOIP is governed.
TDMA time slot used for communication between MultiVOIP unit and the SIP Proxy that governs it.
This field describes the type of gateway as which the MultiVOIP is defined with respect to the gatekeeper.
The current status of the MultiVOIP gateway with respect to the SIP proxy either registered or unregistered.
SPP Registrars
Field Name
Column Headings
Values
IP Address n.n.n.n
Port
Type
Status port
Primary,
Predefined registered, not registered
Description
The IP address of the gatekeeper.
TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it.
This field describes the type of gateway as which the MultiVOIP is defined with respect to the gatekeeper.
The current status of the gateway either registered or unregistered.
Advanced
Packetization Time
You can use the Packetization Time screen to specify definite packetization rates for coders selected in the
Voice/FAX Parameters screen (in the “Coder Options” group of fields). The Packetization Time screen is accessible under the “Advanced” options entry in the sidebar list of the main VOIP software screen. In dealing with RTP parameters, the Packetization Time screen is closely related to both Voice/FAX Parameters and to IP
Statistics. It is located in the “Advanced” group for ease of use.
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Packetization time
Packetization rates can be set separately for each channel.
The table below presents the ranges and increments for packetization rates. The final column represents recommended settings (based on the most common found) when operating with third party devices.
Coder Types
Packetization Ranges and Increments
Range (in Kbps); {default} Increments (in Kbps)
Recommendations
Setting (in ms)
G711, G726, G727
G723
G729
NetCoder
5-120 {5}
30-120 {30}
10-120 {10}
20-120 {20}
5
30
10
20
20
30
20
20
After you set the packetization rate for one channel, you can copy it into other channels by using the Copy
Channel button on the Packetization Time screen. To do so, select the channels you want to copy the settings for.
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MultiVOIP Program Menu Items
After the MultiVOIP program is installed on the PC, it can be launched from the Programs group of the Windows
Start menu (Start | Programs | MultiVOIP n.nn | …). This section describes the software functions available on this menu.
Menu Selection Description
Configuration
Configuration Port Setup
Date and Time Setup
Select this to enter the Configuration program where values for IP, telephony, and other parameters are set.
Select this to access the COM Port Setup screen of the MultiVOIP Configuration program.
Select this for access to set calendar/clock used for data logging.
Download Factory Defaults
Download Firmware
Download IFM Firmware
Download User Defaults
Set Password
Uninstall
Upgrade Software
Select this to return the configuration parameters to the original factory values.
Select this to download new versions of firmware as enhancements become available.
Select this to download new versions of IFM firmware as enhancements become available. The Interface Module (IFM) is the telephony interface for analog MultiVOIP units. There is one IFM for each channel of the MultiVOIP unit. For each channel, the IFM handles the analog signals to and from the attached telephone, PBX or CO line.
To be used after a full set of parameter values, values specified by the user, have been saved (using Save Setup). This command loads the saved user defaults into the
MultiVOIP.
Select this to create a password for access to the MultiVOIP software programs (Program group commands, Windows interface, web browser interface, & FTP server). Only the
FTP Server function requires a password for access. The FTP Server function also requires that a username be set along with the password.
Select this to uninstall the MultiVOIP software (most, but not all components are removed from computer when this command is used).
Loads firmware (including H.323 stack) and settings from the controller PC to the
MultiVOIP unit. User can choose whether to load Factory Default Settings or Current
Configuration settings.
“Downloading” refers to transferring program files from the PC to the nonvolatile “flash” memory of the
MultiVOIP. Such transfers are made using the PC’s serial port. This is a “download” from the perspective of the
MultiVOIP unit.
When new versions of the MultiVOIP software become available, they are posted on Multi-Tech’s website.
Although transferring updated program files from the Multi-Tech website to the user’s PC can generally be considered a download (from the perspective of the PC), this type of download cannot be initiated from the
MultiVOIP software’s Program menu command set.
Generally, you must download updated firmware from the Multi-Tech website to the PC before it can be loaded from the PC to the MultiVOIP.
Updating Firmware
Generally, you must download updated firmware from the Multi-Tech website to the PC before it can be loaded from the PC to the MultiVOIP.
Note that the structure of the Multi-Tech website may change without notice. However, you can find most firmware updates by using standard web techniques. For example, you can access updated firmware by doing a search or by clicking on Support.
If you choose Support, you can select “MultiVOIP” in the Product Support menu and then click on Firmware to find MultiVOIP resources.
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Web locations
After you locate the updated firmware, you can download it from the website.
The firmware file is a self-extracting compressed file (with .zip extension), which you must expand
(decompressed, or “unzipped”) on the user’s PC in a user-specified directory. It is usually best to click the
Browse button and select a folder that is easy to get to and remember.
Extract files
Implementing a Software Upgrade
You can upgrade MultiVOIP software locally using a single command at the MultiVOIP Windows interface, namely Upgrade
Software. This command downloads firmware (including the H.323 stack), and factory default settings from the controller
PC to the MultiVOIP unit.
When using the MultiVOIP Windows interface, you can transfer firmware and factory default settings from controller PC to
MultiVOIP piecemeal using separate commands.
When using the MultiVOIP web browser interface to control/configure the VOIP remotely, upgrading of software must be done on a piecemeal basis using the FTP Server function of the MultiVOIP unit.
When performing a software upgrade (whether from the Windows interface or web browser interface), follow these steps in order:
1.
Identify Current Firmware Version
2.
Download Firmware
3. Download Factory Defaults
When upgrading firmware, implement the software commands “Download Firmware,” and “Download Factory Defaults” in order. If not done in order, the upgrade is incomplete.
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Identifying Current Firmware Version
Before implementing a MultiVOIP firmware upgrade, be sure to verify the firmware version currently loaded on it. The firmware version appears in the MultiVOIP Program menu. Go to Start | Programs | MultiVOIP n.nn. The final expression, n.nn, is the firmware version number.
When a new firmware version is installed, the MultiVOIP software can be upgraded in one step using the
Upgrade Software command, or piecemeal using the Download Firmware command and the Download Factory
Defaults command.
Download Firmware transfers the firmware (including the H.323 protocol stack) in the PC’s MultiVOIP directory into the nonvolatile flash memory of the MultiVOIP.
Download Factory Defaults sets all configuration parameters to the standard default values that are loaded at the Multi-Tech factory.
Upgrade Software implements both the Download Firmware command and the Download Factory Defaults command.
Downloading Firmware
1.
The MultiVOIP Configuration program must be off when invoking the Download Firmware command. If it is on, the command will not work.
2. To use the Download Factory Defaults command, go to Start | Programs | MultiVOIP n.nn | Download
Firmware.
3.
If a password has been established, the Password Verification screen will appear.
Type in the password and click OK.
4. The MultiVOIP n.nn Firmware screen appears saying
“MultiVOIP [model number] is up. Reboot to Download Firmware?”
Click OK to download the firmware.
The “Boot” LED on the MultiVOIP will light up and remain lit during the file transfer process.
5.
The program will locate the firmware “.bin” file in the MultiVOIP directory. Highlight the correct (newest)
“.bin” file and click Open.
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6.
Progress bars will appear at the bottom of the screen during the file transfer.
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.
7. 7. The Download Firmware procedure is complete.
Downloading Factory Defaults
1.
The MultiVOIP Configuration program must be off when invoking the Download Factory Defaults command.
If it is on, the command will not work.
2. To use the Download Factory Defaults command, go to Start | Programs | MultiVOIP n.nn. | Download
Factory Defaults.
3.
If a password has been established, the Password Verification screen will appear.
4.
Type in the password and click OK.
The MVP n.nn - Firmware screen appears saying “MultiVOIP [model number] is up. Reboot to Download
Firmware?”
Click OK to download the factory defaults.
The “Boot” LED on the MultiVOIP will light up and remain lit during the file transfer process.
5.
After the PC gets a response from the MultiVOIP, the Dialog – IP Parameters screen will appear.
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The user should verify that the correct IP parameter values are listed on the screen and revise them if necessary. Then click OK.
6. Progress bars will appear at the bottom of the screen during the data transfer.
7.
The MultiVOIP’s “Boot” LED will turn off at the end of the transfer.
The Download Factory Defaults procedure is complete.
Downloading IFM Firmware
The Interface Module (IFM) is the telephony interface for analog MultiVOIP units. There is one IFM for each channel of the
MultiVOIP unit. For each channel, the IFM handles the analog signals to and from the attached telephone, PBX or CO line.
The IFM communicates with the main processor indicating the status of the telephone line. For example, it might indicate that a phone is off hook (FXS) or that an incoming ring is present (FXO). The IFM receives operating instructions from the
VOIP’s main processor. For example, the IFM might be instructed to ring the phone (FXS) or seize the line (FXO). The IFM contains a codec (coder/decoder) to convert the incoming audio to a PCM stream (pulse code modulation) which it sends to the DSP (digital signal processor). The IFM’s codec also converts outgoing PCM to audio.
The firmware of the IFMs will change from time to time and you may need to upgrade the firmware on your MultiVOIP unit.
To do so, follow these instructions.
1.
In the System Information screen of the MultiVOIP Configuration software, check the version number of the
IFM firmware already installed on the MultiVOIP unit. Write down the version number.
2.
Exit the Configuration software program. The MultiVOIP Configuration program must be off when invoking the Download IFM Firmware command. If it is on, the command will not work.
3. To use the Download IFM Firmware command, go to Start | Programs | MultiVOIP n.nn | Download IFM
Firmware.
4. A warning window will appear: “Downloading IFM Firmware will reboot the MultiVOIP. Do you want to continue?” Click OK.
5.
The “Boot” LED on the front panel of the MultiVOIP will come on.
6.
The software will search for an IFM firmware file to use to upgrade the system; if the file found represents firmware newer than that already installed on the MultiVOIP (or if you want to overwrite the same version of firmware) click Open.
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7.
The IFM Firmware Download screen will appear. Select “Copy to All IFMs” and click OK. (Only in very special circumstances would different IFMs in the same VOIP be loaded with different IFM firmware.)
The main MultiVOIP Configuration screen appears. Progress bars can be seen at the bottom of the screen while files are being copied. Then a completion screen entitled IFM Test will appear.
8.
Click OK to close. The MultiVOIP reboots. When the reboot is complete, the MultiVOIP Configuration screen closes. The IFM firmware downloading process is complete.
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Setting and Downloading User Defaults
The Download User Defaults command allows you to maintain a known working configuration that is specific to your VOIP system. You can then experiment with alterations or improvements to the configurations confident that a working configuration can be restored if necessary.
1. Before you can use the Download User Defaults command, you must first save a set of configuration parameters by using the Save Setup command in the sidebar menu of the MultiVOIP software.
2.
Before the setup configuration is saved, you will be prompted to save the setup as the User Default
Configuration. Select the checkbox and click OK.
3.
A user default file will be created. The MultiVOIP unit will reboot itself.
To download the user defaults, go to Start | Programs | MultiVOIP n.nn | Download User Defaults.
4.
A confirmation screen will appear indicating that this action will entail rebooting the MultiVOIP. Click OK.
5.
When the file transfer process is complete, the Dialog / IP Parameters screen will appear.
6.
Set the IP values per your particular VOIP system. Click OK. Progress bars will appear as the MultiVOIP reboots itself.
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Setting a Password
Windows Interface
After a user name has been designated and a password has been set, that password is required to gain access to any functionality of the MultiVOIP software. Only one user name and password can be assigned to a VOIP unit.
The user name will be required when communicating with the MultiVOIP via the web browser interface.
Note: Record your user name and password in a safe place. If the password is lost, forgotten, or irretrievable, the user must contact Multi-Tech Tech Support in order to resume use of the MultiVOIP unit.
1. The MultiVOIP configuration program must be off when invoking the Set Password command. If it is on, the command will not work.
2.
To use the Set Password command, go to Start | Programs | MultiVOIP n.nn | Set Password.
3.
You will be prompted to confirm that you want to establish a password, which will entail rebooting the
MultiVOIP (which is done automatically).
Click OK to proceed with establishing a password.
4. The Password screen appears. If you intend to use the FTP Server function that is built into the MultiVOIP, enter a user name. (A User Name is not needed to access the local Windows interface, the web browser interface, or the commands in the Program group.) Type your password in the Password field of the
Password screen. Type this same password again in the Confirm Password field to verify the password you have chosen.
Click OK.
A message will appear indicating that a password has been set successfully.
After the password has been set successfully, the MultiVOIP will re-boot itself and, in so doing, its BOOT LED will light up.
5.
After the password has been set, the user will be required to enter the password to gain access to the web browser interface and any part of the MultiVOIP software listed in the Program group menu. User Name and Password are both needed for access to the FTP Server residing in the MultiVOIP.
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When MultiVOIP program asks for password, the program shuts down if you select CANCEL .
An error message appears if you enter invalid password is entered.
Web Browser Interface
A passwords is optional for the MultiVOIP web browser interface. Only one password can be assigned and it works for all MultiVOIP software function. If a password has been set, it is required to access the MultiVOIP web browser interface.
NOTE: If you lose the password, the contact Multi-Tech Tech Support to resume use of the MultiVOIP web browser interface.
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Upgrading Software
As noted earlier the Upgrade Software command transfers, from the controller PC to the MultiVOIP unit, firmware (including the H.323 stack) and settings. The settings can be either Factory Default Settings or Current
Configuration Settings.
NOTE: To upgrade a MultiVOIP from software version x.04 or earlier, an ftp primer file must first be sent to the VOIP. Contact Multi-Tech Technical Support if you need this the FTP_Primer.bin file.
CAUTION: You cannot go back to x.04 or earlier versions using FTP. You must use ‘upgrade software’ via the serial port.
NOTE: These ftp upgrade instructions do not apply to software release x.05 and above.
FTP Server File Downloads
Multi-Tech has built an FTP server into the MultiVOIP unit. Therefore, you can use an FTP client program or even a browser to transfer files from the controller PC to the VOIP unit.
The terminology of “downloads” and “uploads” gets a bit confusing in this context. File transfers from a client to a server are typically considered “uploads.” File transfers from a large repository of data to machines with less data capacity are considered “downloads.” In this case, these metaphors are contradictory: the FTP server is actually housed in the MultiVOIP unit, and the controller PC, which is actually the repository of the info to be transferred, uses an FTP client program. In this situation, we have chosen to call the transfer of files from the PC to the VOIP “downloads.” (Be aware that some FTP client programs may use the opposite terminology, that is, they may refer to the file transfer as an “upload “)
You can download firmware, CAS telephony protocols, default configuration parameters, and phonebook data for the MultiVOIP unit with this FTP functionality. These downloads are done over a network, not by a local serial port connection. Consequently, VOIPs at distant locations can be updated from a central control point.
The phonebook downloading feature greatly reduces the data-entry required to establish inbound and outbound phonebooks for the VOIP units within a system. Although each MultiVOIP unit will require some unique phonebook entries, most will be common to the entire VOIP system. After the phonebooks for the first few VOIP units have been compiled, phonebooks for additional VOIPs become much simpler: you copy the common material by downloading and then do data entry for the few phonebook items that are unique to that particular VOIP unit or VOIP site.
To transfer files using the FTP server function in the MultiVOIP:
1. Establish Network Connection and IP Addresses. Both the controller PC and the MultiVOIP unit(s) must be connected to the same IP network. An IP address must be assigned for each.
2. Establish User Name and Password. Establish a user name and (optionally) a password for contacting the
VOIP over the IP network. (When connection is made through a local serial connection between the PC and the VOIP unit, no user name is needed.)
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3.
As shown above, the user name and password can be set in the web interface as well as in the Windows interface.
Install FTP Client Program or Use Substitute. You should install an FTP client program on the controller PC.
FTP file transfers can be done using a web browser (for example, Netscape or Internet Explorer) in conjunction with a local Windows browser a (for example, Windows Explorer), but this approach is somewhat clumsy (it requires use of two application programs rather than one) and it limits downloading to only one VOIP unit at a time. With an FTP client program, multiple VOIPs can receive FTP file transmissions in response to a single command (the transfers may occur serially however).
Although Multi-Tech does not provide an FTP client program with the MultiVOIP software or endorse any particular FTP client program, we remind our readers that adequate FTP programs are readily available under retail, shareware and freeware licenses. (Read and observe any End-User License Agreement carefully.) Two examples of this are the “WSFTP” client and the “SmartFTP” client, with the former having an essentially text-based interface and the latter having a more graphically oriented interface, as of this writing.
User preferences will vary.
4. Enable FTP Functionality. Go to the IP Parameters screen and click on the “FTP Server: Enable” box.
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5.
Identify Files to be Updated. Determine which files you want to update. Six types of files can be updated using the FTP feature. In some cases, the file to be transferred has “Ftp” as the part of its filename just before the suffix (or extension). So, for example, the file “mvpt1Ftp.bin” can be transferred to update the bin file (firmware) residing in the MultiVOIP. Similarly, the file “fxo_loopFtp.cas” could be transferred to enable use of the FXO Loop Start telephony interface in one of the analog VOIP units and the file
“r2_brazilFtp.cas” could be transferred to enable a particular telephony protocol used in Brazil. Note, however, that before any CAS file can be used as an update, it must be renamed to CASFILE.CAS so that it overwrites and replaces the default CAS file.
File Type firmware “bin” file factory defaults
CAS file inbound phonebook outbound phonebook
File Names mvpt1Ftp.bin fdefFtp.cnf fxo_loopFtp.cas, em_winkFtp.cas, r2_brazilFtp.cas r2_chinaFtp.cas
InPhBk.tmr
OutPhBk.tmr
Description
This is the MultiVOIP firmware file. Only one file of this type will be in the directory.
This file contains factory default settings for user-changeable configuration parameters. Only one file of this type will be in the directory.
These telephony files are for Channel Associated Signaling. The directory contains many CAS files, some labeled for specific functionality, others for countries or regions where certain attributes are standard. Any CAS file used must first be renamed to
“CASFILE.CAS.”
This file updates the inbound phonebook in the MultiVOIP unit.
This file updates the outbound phonebook in the MultiVOIP unit.
6. Contact MultiVOIP FTP Server. You must make contact with the FTP Server in the VOIP using either a web browser or FTP client program. Enter the IP address of the MultiVOIP’s FTP Server. If you are using a browser, the address must be preceded by “ftp://” (otherwise you’ll reach the web interface within the
MultiVOIP unit).
7.
Log In. Use the User Name and password. The login screens differ, depending on whether the FTP file transfer is to be done with a web browser (shown below) or with an FTP client program (varies).
8. Use Download. Downloading can be done with a web browser or with an FTP client program.
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Download with Web Browser: a. In the local Windows browser, locate the directory holding the MultiVOIP program files. The default location will be C:\Program Files \Multi-Tech Systems \MultiVOIP xxxx yyyy (where x and y represent
MultiVOIP model numbers and software version numbers). b. Drag-and-drop files from the local Windows browser (for example, Windows Explorer) to the web browser. c. You may be asked to confirm the overwriting of files on the MultiVOIP. Do so. d. File transfer between PC and VOIP will look like transfer within VOIP directories.
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Download with FTP Client Program: a. In the local directory browser of the FTP client program, locate the directory holding the MultiVOIP program files. The default location will be C:\Program Files \Multi-Tech Systems \MultiVOIP xxxx yyyy
(where x and y represent MultiVOIP model numbers and software version numbers). b. In the FTP client program window, drag-and-drop files from the local browser pane to the pane for the
MultiVOIP FTP server. FTP client interface operations vary. In some cases, you can choose between immediate and queued transfer. In some cases, there may be automated capabilities to transfer to multiple destinations with a single command.
9.
Verify Transfer. The files transferred appear in the directory of the MultiVOIP.
10.
Log Out of FTP Session. Whether the file transfer was done with a web browser or with an FTP client program, you must log out of the FTP session before opening the MultiVOIP Windows interface.
Web Browser Interface
You can control the MultiVOIP unit with a graphic user interface (interface) based on the common web browser platform. Qualifying browsers are Internet Explorer 6+, Netscape 6+, and Mozilla Firefox 1.0+.
Function Remote configuration and control of MultiVOIP units.
Configuration Prerequisite
Browser Version Requirement
Local Windows interface must be used to assign IP address to MultiVOIP.
Internet Explorer 6.0 or higher; or
Netscape 6.0 or higher; or
Mozilla Firefox 1.0 or higher.
Java Requirement Java Runtime Environment version 1.4.0_01 or higher
(this application program is included with MultiVOIP)
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The initial configuration step of assigning the VOIP unit an IP address must still be done locally using the
Windows interface. However, all additional configurations can be done via the web interface.
The content and organization of the web interface is directly parallel to the Windows interface. For each screen in the Windows interface, there is a corresponding screen in the web interface. The fields on each screen are the same, as well.
The Windows interface gives access to commands via icons and pull-down menus whereas the web interface does not. The web interface, however, cannot perform logging in the same direct mode done in the Windows interface. However, when the web interface is used, logging can be done by email (SMTP).
The graphic layout of the web interface is also somewhat larger-scale than that of the Windows interface. For that reason, it’s helpful to use as large of a video monitor as possible.
The primary advantage of the web interface is remote access for control and configuration. The controller PC and the MultiVOIP unit itself must both be connected to the same IP network and their IP addresses must be known.
To use the web interface, install a Java application program on the controller PC. Java supports drop-down menus and multiple windows in the web interface.
When installation is complete, the Java program runs automatically in the background as a plug-in supporting the MultiVOIP web interface. No user actions are required.
After the Java program has been installed, you can access the MultiVOIP using the web browser interface. Close the MultiVOIP Windows interface. Start the web browser. Enter the IP address of the MultiVOIP unit. Enter a password when prompted. (A password is needed here only if password has been set for the local Windows interface or for the MultiVOIP’s FTP Server function. See “Setting a Password -- Web Browser interface” earlier in this chapter.) The web browser interface offers essentially the same control over the VOIP as can be achieved using the Windows interface. As noted earlier, logging functions cannot be handled via the web interface. And, because network communications will be slower than direct communications over a serial PC cable, command execution will be somewhat slower over the web browser interface than with the Windows interface.
SysLog Server Functions
Multi-Tech has built SysLog server functionality into the software of the MultiVOIP units. SysLog is a de facto standard for logging events in network communication systems.
The SysLog Server resides in the MultiVOIP unit itself. To implement this functionality, you will need a SysLog client program (sometimes referred to as a “daemon”). SysLog client programs, both paid and freeware can be obtained from Kiwi Enterprises (search the Internet for kiwi syslog daemon), among other firms. Read the End-
User License Agreement carefully and observe license requirements. SysLog client programs essentially give you a means of structuring console messages for convenience and ease of use.
Multi-Tech Systems does not endorse any particular SysLog client program. A SysLog client programs from qualified providers most likely work with MultiVOIP units.
Before using a SysLog client program, enable the SysLog function within the MultiVOIP in the Logs menu under
Configuration.
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The IP Address used is that of the MultiVOIP itself.
In the Port field, entered by default, is the standard (‘well-known’) logical port, 514.
Configuring the SysLog Client Program. Configure the SysLog client program for your own needs. In various
SysLog client programs, you can define where log messages are saved and archived, opt for interaction with an
SNMP system (like MultiVOIP Manager), set the content and format of log messages, determine disk space allocation limits for log messages, and establish a hierarchy for the seriousness of messages (normal, alert, critical, emergency, and so on.).
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Command Cable
RJ-45 Connector End-to-End Pin Info
RJ-45 connector plugs into Command Port of MultiVOIP.
DB-9 connector plugs into serial port of command PC (which runs MultiVOIP configuration software).
Ethernet Connector
The functions of the individual conductors of the MultiVOIP’s Ethernet port are shown on a pin-by-pin basis below.
RJ-45 Ethernet Connector Pin Circuit Signal Name
3
6
1
2
TD+ Data Transmit Positive
TD- Data Transmit Negative
RD+ Data Receive Positive
RD- Data Receive Negative
Voice/Fax Channel Connectors
Multi-Tech Systems, Inc.
FXS Pin Description
2 N/C
3
4
5
Ring
Tip
N/C
FXO Pin Description
2 N/C
3
4
5
Tip
Ring
N/C
112
Well Known Port Numbers
The following description of port number assignments for Internet Protocol (IP) communication is taken from the Internet Assigned Numbers Authority (IANA) web site (www.iana.org).
“The Well Known Ports are assigned by the IANA and on most systems can only be used by system (or root) processes or by programs executed by privileged users. Ports are used in the TCP
[RFC793] to name the ends of logical connections which carry long term conversations. For the purpose of providing services to unknown callers, a service contact port is defined. This list specifies the port used by the server process as its contact port. The contact port is sometimes called the "well-known port". To the extent possible, these same port assignments are used with the UDP [RFC768]. The range for assigned ports managed by the IANA is 0-1023.”
Well-known port numbers especially pertinent to MultiVOIP operation are listed below.
Port Number Assignment List
Function telnet tftp
Port Number
23
69 snmp snmp tray gatekeeper registration
H.323
SIP
SysLog
161
162
1719
1720
5060
514
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EMC, Safety, and R&TTE Directive Compliance
The CE mark is affixed to this product to confirm compliance with the following European Community Directives:
Council Directive 2004/108/EC of 15 December 2004 on the approximation of the laws of Member States relating to electromagnetic compatibility; and
Council Directive 2006/95/EC of 12 December 2006 on the harmonization of the laws of Member States relating to electrical equipment designed for use within certain voltage limits; and
Council Directive 1999/5/EC of 9 March 1999 on radio equipment and telecommunications terminal equipment and the mutual recognition of their conformity.
FCC Part 15 Declaration
This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to part
15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference in which case the user will be required to correct the interference at his own expense. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
● Reorient or relocate the receiving antenna.
● Increase the separation between the equipment and receiver.
●
●
Plug the equipment into an outlet on a circuit different from that to which the receiver is connected.
Consult the dealer or an experienced radio/TV technician for help.
This device complies with Part 15 of the 47 CFR rules. Operation of this device is subject to the following conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference that may cause undesired operation.
Warning: Changes or modifications to this unit not expressly approved by the party responsible for compliance could void the user’s authority to operate the equipment.
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Appendix C — Regulatory Information
FCC Part 68 Telecom
1. This equipment complies with Part 68 of the 47 CFR rules and the requirements adopted by the ACTA.
Located on this equipment is a label that contains, among other information, the registration number and
Ringer Equivalence Number (REN) for this equipment or a product identifier in the format:
2.
For current products: US:AAAEQ##Txxxx.
For legacy products: AU7USA-xxxxx-xx-x
If requested, this number must be provided to the telephone company.
A plug and jack used to connect this equipment to the premises wiring and telephone network must comply with the applicable 47 CFR Part 68 rules and requirements adopted by the ACTA. It’s designed to be connected to a compatible modular jack that is also compliant.
3.
The Ringer Equivalence Number (REN) is used to determine the number of devices that may be connected to a telephone line. Excessive RENs on a telephone line may result in the devices not ringing in response to an incoming call. In most but not all areas, the sum of RENs should not exceed five (5.0). To be certain of the number of devices that may be connected to a line, as determined by the total RENs, contact the local telephone company. For products approved after July 23, 2001, the REN for this product is part of the product identifier that has the format US:AAAEQ##Txxxx. The digits represented by ## are the REN without a decimal point (e.g., 03 is a REN of 0.3). For earlier products, the REN is separately shown on the label.
4.
If this equipment causes harm to the telephone network, the telephone company will notify you in advance that temporary discontinuance of service may be required. But if advance notice isn't practical, the telephone company will notify the customer as soon as possible. Also, you will be advised of your right to file a complaint with the FCC if you believe it is necessary.
5. The telephone company may make changes in its facilities, equipment, operations or procedures that could affect the operation of the equipment. If this happens, the telephone company will provide advance notice in order for you to make necessary modifications to maintain uninterrupted service.
6.
If trouble is experienced with this equipment, please contact Multi-Tech Systems, Inc. at the address shown below for details of how to have the repairs made. If the equipment is causing harm to the telephone network, the telephone company may request that you disconnect the equipment until the problem is resolved.
7.
Connection to party line service is subject to state tariffs. Contact the state public utility commission, public service commission or corporation commission for information.
8.
No repairs are to be made by you. Repairs are to be made only by Multi-Tech Systems or its licensees.
Unauthorized repairs void registration and warranty.
9.
If your home has specially wired alarm equipment connected to the telephone line, ensure the installation of this equipment does not disable your alarm equipment. If you have questions about what will disable alarm equipment, consult your telephone company or a qualified installer.
10.
Connection to party line service is subject to state tariffs. Contact the state public utility commission, public service commission or corporation commission for information.
11.
This equipment is hearing aid compatible.
12.
Manufacturing Information on telecommunications device (modem):
Manufacturer:
Trade Name:
Model Number:
Multi-Tech Systems
MultiVOIP
MTIFM
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Registration Number:
Ringer Equivalence:
Modular Jack (USOC):
Service Center in USA:
US:AU7CN06BMTIFM
0.6B
RJ-48C
Multi-Tech Systems, Inc.
2205 Woodale Drive
Mounds View, MN 55112 USA
(763)785-3500
(763) 785-9874 (Fax)
Industry Canada
This Class A digital apparatus meets all requirements of the Canadian Interference-Causing Equipment Regulations.
Cet appareil numérique de la classe A respecte toutes les exigences du Reglement Canadien sur le matériel brouilleur.
Canadian Limitations Notice
Notice: This equipment meets the applicable Industry Canada Terminal Equipment Technical Specifications. This is confirmed by the registration number. The abbreviation, IC, before the registration number signifies that registration was performed based on a Declaration of Conformity indicating that Industry Canada technical specifications were met. It does not imply that Industry Canada approved the equipment.
Notice: The REN assigned to each terminal equipment provides an indication of the maximum number of terminals allowed to be connected to a telephone interface. The termination on an interface may consist of any combination of devices subject only to the requirement that the sum of the Ringer Equivalence Numbers of all the devices does not exceed five.
Restrictions concernant le raccordement de matériel
Avis: Le présent matériel est conforme aux spécifications techniques d’Industrie Canada applicables au matériel terminal. Cette conformité est confirmée par le numéro d'enregistrement. Le sigle IC, placé devant le numéro d'enregistrement, signifie que l’enregistrement s’est effectué conformément à une déclaration de conformité et indique que les spécifications techniques d'Industrie Canada ont été respectées. Il n’implique pas qu’Industrie
Canada a approuvé le matériel.
Avis: L'IES assigné à chaque dispositif terminal indique le nombre maximal de terminaux qui peuvent être raccordés à une interface téléphonique. La terminaison d'une interface peut consister en une combinaison quelconque de dispositifs, à la seule condition que la somme d'indices d'équivalence de la sonnerie de tous les dispositifs n'excède pas 5.
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Appendix C — Regulatory Information
Waste Electrical and Electronic Equipment Statement
WEEE Directive
The WEEE Directive places an obligation on EU-based manufacturers, distributors, retailers, and importers to take-back electronics products at the end of their useful life. A sister directive, ROHS (Restriction of Hazardous
Substances) complements the WEEE Directive by banning the presence of specific hazardous substances in the products at the design phase. The WEEE Directive covers all Multi-Tech products imported into the EU as of
August 13, 2005. EU-based manufacturers, distributors, retailers and importers are obliged to finance the costs of recovery from municipal collection points, reuse, and recycling of specified percentages per the WEEE requirements.
Instructions for Disposal of WEEE by Users in the European Union
The symbol shown below is on the product or on its packaging, which indicates that this product must not be disposed of with other waste. Instead, it is the user's responsibility to dispose of their waste equipment by handing it over to a designated collection point for the recycling of waste electrical and electronic equipment.
The separate collection and recycling of your waste equipment at the time of disposal will help to conserve natural resources and ensure that it is recycled in a manner that protects human health and the environment.
For more information about where you can drop off your waste equipment for recycling, please contact your local city office, your household waste disposal service or where you purchased the product.
July, 2005
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Appendix C — Regulatory Information
Restriction of the Use of Hazardous Substances (RoHS)
Multi-Tech Systems, Inc.
Certificate of Compliance
2011/65/EU
Multi-Tech Systems confirms that its embedded products comply with the chemical concentration limitations set forth in the directive 2011/65/EU of the European Parliament (Restriction of the use of certain Hazardous
Substances in electrical and electronic equipment - RoHS)
These Multi-Tech products do not contain the following banned chemicals 1 :
●
●
●
●
●
●
Lead, [Pb] < 1000 PPM
Mercury, [Hg] < 1000 PPM
Hexavalent Chromium, [Cr+6] < 1000 PPM
Cadmium, [Cd] < 100 PPM
Polybrominated Biphenyl, [PBB] < 1000 PPM
Polybrominated Diphenyl Ether, [PBDE] < 1000 PPM
Environmental considerations:
●
●
Moisture Sensitivity Level (MSL) =1
Maximum Soldering temperature = 260C (in SMT reflow oven)
1 Lead usage in some components is exempted by the following RoHS annex, therefore higher lead concentration would be found in some modules (>1000 PPM);
–Resistors containing lead in a glass or ceramic matrix compound.
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Appendix C — Regulatory Information
Information on HS/TS Substances According to Chinese
Standards
In accordance with China’s Administrative Measures on the Control of Pollution Caused by Electronic
Information Products (EIP) # 39, also known as China RoHS, the following information is provided regarding the names and concentration levels of Toxic Substances (TS) or Hazardous Substances (HS) which may be contained in Multi-Tech Systems Inc. products relative to the EIP standards set by China’s Ministry of Information Industry
(MII).
Hazardous/Toxic Substance/Elements
Component Name Lead Mercury Cadmium Hexavalent Polybrominated Polybrominated
(PB)
(Hg) (CD) Chromium Biphenyl Diphenyl Ether
(CR6+) (PBB) (PBDE)
Printed Circuit Boards O O O O O O
Resistors X O O O O O
Capacitors X O O O O O
Ferrite Beads O O O O O O
Relays/Opticals O O O O O O
ICs O O O O O O
Diodes/ Transistors O O O O O O
Oscillators and Crystals X O O O O O
Regulator O O O O O O
Voltage Sensor O O O O O O
Transformer O O O O O O
Speaker O O O O O O
Connectors O O O O O O
LEDs O O O O O O
Screws, Nuts, and other
Hardware
AC-DC Power Supplies
X O O O O O
O O O O O O
Software /
Documentation CDs
Booklets and
Paperwork
Chassis
X
O
O
O
O
O
O
O
O
O
O
O
O
O O O O O O
Represents that the concentration of such hazardous/toxic substance in all the units of homogeneous material of such component is higher than the SJ/Txxx-2006 Requirements for
Concentration Limits.
O Represents that no such substances are used or that the concentration is within the aforementioned limits.
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Appendix C — Regulatory Information
Information on HS/TS Substances According to Chinese
Standards (in Chinese)
依照中国标准的有毒有害物质信息
根据中华人民共和国信息产业部 (MII) 制定的电子信息产品 (EIP)
标准-中华人民共和国《电子信息产品污染控制管理办法》(第 39 号),也称作中国
RoHS,下表列出了 Multi-Tech Systems, Inc. 产品中可能含有的有毒物质 (TS) 或有害物质 (HS)
的名称及含量水平方面的信息。
有害/有毒物质/元素
成分名称 铅 汞 镉 六价铬 多溴联苯 多溴二苯醚
(PB) (Hg) (CD) (CR6+) (PBB) (PBDE)
印刷电路板
电阻器
电容器
铁氧体磁环
继电器/光学部件
O
X
X
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
IC
O
O
O
O
O
O
O
O
O
O
O
O
二极管/晶体管
振荡器和晶振
调节器
O
X
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
电压传感器
变压器
扬声器
连接器
LED
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
螺丝、螺母以及其它五金件
交流-直流电源
软件/文档 CD
手册和纸页
底盘
X
X
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
O
表示所有使用类似材料的设备中有害/有毒物质的含量水平高于 SJ/Txxx-2006 限量要求。
O 表示不含该物质或者该物质的含量水平在上述限量要求之内。
Multi-Tech Systems, Inc. 120
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Table of contents
- 2 Record of Revisions
- 6 Chapter 1 – Product Overview
- 6 Introduction
- 6 Interface
- 7 Front Panel LEDs
- 7 Computer Requirements
- 8 Specifications
- 9 Chapter 2 – Installing and Cabling the MultiVOIP
- 9 Introduction
- 9 Safety Warnings
- 9 Lithium Battery Caution
- 9 Safety Warnings Telecom
- 10 Avertissements de sécurité télécom analogique
- 10 Package Contents
- 10 Cabling Procedure for MVP130
- 12 Chapter 3 – Software Installation
- 12 Introduction
- 12 Installing the Software
- 12 Setup Overview
- 13 Ethernet/IP
- 13 Actions
- 14 Voice/Fax
- 15 Actions:
- 15 Interface
- 16 Actions:
- 18 Call Signaling
- 19 Actions:
- 20 Regional
- 20 Actions:
- 21 Phone Book
- 22 Actions
- 22 Save & Reboot
- 23 Chapter 4 – Configuring Your MultiVOIP
- 23 Navigating the Software
- 23 Web Browser Interface
- 24 Configuration Information Checklist
- 25 Ethernet/IP
- 27 Voice/Fax
- 31 Configurable Payload Type
- 31 Interface
- 32 FXS Loop Start Parameters
- 34 Message Waiting
- 35 FXO Parameters
- 36 FXO Supervision
- 39 DID Parameters
- 40 Call Signaling
- 40 H.323
- 41 SIP
- 43 SPP
- 44 SNMP
- 45 Regional
- 47 Setting Custom Tones and Cadences (optional)
- 48 SMTP
- 50 RADIUS
- 52 Logs/Traces
- 53 NAT Traversal
- 54 Supplementary Services
- 57 Save Settings
- 57 Save & Reboot
- 57 Connection
- 57 Settings
- 58 Troubleshooting Software Issues
- 58 Fixing a COM Port Problem
- 58 Fixing a Cabling Problem
- 59 Chapter 5 – Phone Book Configuration
- 59 Introduction
- 59 Identify Remote VOIP Site to Call
- 59 Identify VOIP Protocol to be Used
- 60 Phonebook Starter Configuration
- 60 Outbound Phonebook
- 61 Inbound Phonebook
- 63 Phone Book Descriptions
- 63 Outbound Phone Book/List Entries
- 64 Add/Edit Outbound Phone Book
- 66 Alternate Routing Field Definitions
- 67 Inbound Phone Book/List Entries
- 68 Add/Edit Inbound Phone Book
- 69 Authorized User Name and Password for SIP
- 70 Phone Book Save and Reboot
- 70 Phonebook Examples
- 70 North America
- 74 Europe
- 80 Variations of Caller ID
- 83 Chapter 6 – Using the Software
- 83 Software Categories Covered in This Chapter
- 84 System Information screen
- 85 Statistics Section
- 85 Call Progress
- 87 Logs
- 89 IP Statistics
- 91 Link Management
- 92 Registered Gateway Details
- 93 Servers
- 93 H.323 GateKeepers
- 93 SIP Proxies
- 94 SPP Registrars
- 94 Advanced
- 94 Packetization Time
- 96 MultiVOIP Program Menu Items
- 96 Updating Firmware
- 97 Implementing a Software Upgrade
- 98 Identifying Current Firmware Version
- 98 Downloading Firmware
- 99 Downloading Factory Defaults
- 100 Downloading IFM Firmware
- 102 Setting and Downloading User Defaults
- 103 Setting a Password
- 103 Windows Interface
- 104 Web Browser Interface
- 105 Upgrading Software
- 105 FTP Server File Downloads
- 109 Web Browser Interface
- 110 SysLog Server Functions
- 112 Appendix A – Cable Pin-outs
- 112 Command Cable
- 112 Ethernet Connector
- 112 Voice/Fax Channel Connectors
- 113 Appendix B – TCP/UDP Port Assignments
- 113 Well Known Port Numbers
- 113 Port Number Assignment List
- 114 Appendix C – Regulatory Information
- 114 EMC, Safety, and R&TTE Directive Compliance
- 114 FCC Part 15 Declaration
- 115 FCC Part 68 Telecom
- 116 Industry Canada
- 116 Canadian Limitations Notice
- 116 Restrictions concernant le raccordement de matériel
- 117 Waste Electrical and Electronic Equipment Statement
- 117 WEEE Directive
- 117 Instructions for Disposal of WEEE by Users in the European Union
- 118 Restriction of the Use of Hazardous Substances (RoHS)
- 119 Information on HS/TS Substances According to Chinese Standards
- 120 Information on HS/TS Substances According to Chinese Standards (in Chinese)
- 120 依照中国标准的有毒有害物质信息