Identify MVP-SS Unit’s Role in SIP VOIP System ...29 Placement ...30 Command/Control Computer Setup (Specs & Settings) ...30 Quick Hookup for MVP410-SS & MVP810-SS...31 Quick Hookup for MVP210-SS ...32 Load MultiVOIP Control Software onto PC...33 Phone/IP Starter Configuration...34 Phonebook Starter Configuration (with remote voip)...40 Outbound Phonebook ...40...
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CHAPTER 4: SOFTWARE INSTALLATION ...77 ...78 NTRODUCTION VOIP S OADING ULTI OFTWARE ONTO THE VOIP C NSTALLING THE ULTI CHAPTER 5: TECHNICAL CONFIGURATION...88 VOIP ...89 ONFIGURING THE ULTI ...92 OCAL ONFIGURATION Pre-Requisites...92 IP Parameters...92 Telephony Interface Parameters ...93 SMTP Parameters (for email call log reporting)...94 Config Info CheckList ...95 Local Configuration Procedure (Summary) ...96 Local Configuration Procedure (Detailed)...97...
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About IP Statistics... 306 About Link Management... 311 About Registered Gateway Details ... 314 About Alternate Server Statistics ... 317 About Packetization Time ... 321 ULTI ROGRAM TEMS Configuration Option... 326 Configuration Port Setup... 326 Date and Time Setup... 327 Obtaining Updated Firmware...
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APPENDIX B: TCP/UDP PORT ASSIGNMENTS... 385 ... 386 NOWN UMBERS ... 386 UMBER SSIGNMENT INDEX ... 388...
Overview MultiVOIP User Guide About This Manual This manual is about Voice-over-IP products made by Multi-Tech Systems, Inc. It describes three analog MultiVOIP units with SIP- survivability features, models MVP810SS, MVP410SS, and MVP210SS These MultiVOIP units can inter-operate with other contemporary analog MultiVOIP units (MVP130, MVP130FXS, MVP210, MVP410, and MVP810), with contemporary BRI MultiVOIP units (MVP410ST &...
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Digital MultiVOIP Products MVP- Description 2410 24-48 Model Function digital digital VOIP VOIP unit add-on card Capacity channels added channels Chassis/ 19” 1U circuit Mounting rack card mount only 3010 30-60 digital digital VOIP VOIP unit add-on card added channels channels 19”...
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Analog MultiVOIP Products Description Model analog add-on analog Function voip card voip Capacity 4 added channels channels channels Chassis/ 19” 1U circuit 19” 1U Mounting rack card rack mount only mount Description Model 810 SS 410SS Function analog voip; acts analog voip;...
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ISDN/BRI MultiVOIP Products Description MVP810ST MVP410ST Model ISDN-BRI voip ISDN-BRI voip Function 4 ISDN lines 2 ISDN lines Capacity (8 B-channels) (4 B-channels) Chassis/ 19” 1U rack mount 19” 1U rack mount Mounting 1. “BRI” means Basic Rate Interface.
Introduction to Analog MultiVOIPs with SIP Survivability Features (MVP-210SS/410SS/810SS) VOIP: The Free Ride. We proudly present Multi-Tech's MVP- 210SS/410SS/810SS MultiVOIP Voice-over-IP Gateways. These three models allow voice/fax communication to be transmitted at no additional expense over your existing IP network, which has ordinarily been data only.
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Capacity. MultiVOIP model MVP810SS is an eight-channel unit, the model MVP410SS is a four-channel unit, and the MVP210SS is a two- channel unit. All three of these MultiVOIP units have a 10/100Mbps Ethernet interface and a command port for configuration. SIP Survivability.
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Overview MultiVOIP User Guide for the MVP810SS) constitutes a practical limitation on their capacity to support PSTN access for other gateways. Systems must be scaled to match required capacity by including additional MultiVOIP-SS units. Mounting. Mechanically, the MVP410SS and MVP810SS MultiVOIPs are designed for a one-high industry-standard EIA 19-inch rack enclosure.
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MultiVOIP User Guide Overview MultiVOIP web browser GUI. All of these control software packages are included on the Product CD. While the web GUI’s appearance differs slightly, its content and organization are essentially the same as that of the Windows GUI (except for logging).
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Overview MultiVOIP User Guide The primary advantage of the web GUI 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.
MultiVOIP Front Panel LEDs LED Types. The MultiVOIPs have two types of LEDs on their front panels: (1) general operation LED indicators (for power, booting, and ethernet functions), and (2) channel operation LED indicators that describe the data traffic and performance in each VOIP data channel. Active LEDs.
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LED Descriptions for MultiVOIP-SS Units Front Panel LED Definitions DESCRIPTION LED NAME General Operation LEDs (one set on each MultiVOIP model) Power Indicates presence of power. Boot 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.
Computer Requirements The computer on which the MultiVOIP’s configuration program is installed must meet these requirements: • must be IBM-compatible PC with MS Windows operating system; • must have an available COM port for connection to the MultiVOIP. However, this PC does not need to be connected to the MultiVOIP permanently.
Specifications Parameter MVP410SS /Model Operating 100-240 VAC 1.2 - 0.6 A Voltage/ Current Mains 50/60 Hz Frequencies 29 watts Power Consumption 1.75” H x Mechanical 17.4” W x Dimensions 8.5” D 4.5cm H x 44.2 cm W x 21.6 cm D 7.1 lbs.
Viewing and printing a user guide from the Web also requires that you have the Acrobat Reader loaded on your system. To select the MultiVOIP User Guide from the Multi-Tech Systems home page, click Documents and then click MultiVOIP Family in the product list drop-down window. All documents for this MultiVOIP Product Family will be displayed.
Introduction This chapter contains streamlined instructions to get the MultiVOIP up and running quickly. These start-up instructions include assistance on setting up the MultiVOIP’s Inbound and Outbound Phonebooks. These sections of the Quick Start Instructions may be particularly useful for phonebook configuration: Phonebook Starter Configuration Phonebook Tips...
MultiVOIP Startup Tasks Task Collecting Phone/IP Details ( vital! ) Placement Command/Control Computer Setup: Specs & Settings Hookup Software Installation Phone/IP Starter Configuration Phonebook Starter Configuration Connectivity Test Troubleshooting Summary The MultiVOIP must be configured to interface with your particular phone system and IP network.
Phone/IP Details *Absolutely Needed* Before Starting the Installation The MultiVOIP will interface with both the IP network and the phone system. You must gather information about the IP network and about the phone system so that the MultiVOIP can be configured to operate with them properly.
Phone/IP Details *Absolutely Needed* Gather Telephone Information Telephony Parameters Ask phone company or telecom manager. Analog Telephony Interface Parameters: • Which interface type is used? E&M _____ FXS/FXO • If FXS, determine whether the line will be used for a phone, fax, or KTS (key telephone system) •...
MultiVOIP User Guide QS: Gathering Phone/IP Details Phone/IP Details Often Needed/Wanted Obtain Email Address for VOIP (for email call log reporting) required if log reports of VOIP call traffic Optional are to be sent by email SMTP Parameters Preparation Task: Ask Mail Server To: I.T.
Config Info CheckList Type of Config Info Gathered IP info for voip unit ● IP address ● Gateway ● DNS IP (if used) ● 802.1p Prioritization (if used) Interface Type (Choices: E&M, FXS/FXO*, DIP, DPO) E&M info (only if E&M is used) ●...
Identify MVP-SS Unit’s Role in SIP VOIP System The MVP210-SS/410-SS/810-SS unit always uses the SIP protocol. However, the MVP-SS units are equipped to play an additional role in the voip system -- the role of a SIP server. And as a SIP server, the MVP-SS unit can operate in either “stand-alone”...
Placement Mount your MultiVOIP in a safe and convenient location where cables for your network and phone system are accessible. Rack-mounting instructions are in Chapter 3: Mechanical Installation & Cabling of the User Guide. Command/Control Computer Setup (Specs & Settings) The computer used for command and control of the MultiVOIP (a) must be an IBM-compatible PC, (b) must use a Microsoft operating system,...
Load MultiVOIP Control Software onto PC For more details, see Chapter 4: Software Installation in User Guide. 1. MultiVOIP must be properly cabled. Power must be turned on. 2. Insert MultiVOIP CD into drive. Allow 10-20 seconds for Autorun to start.
Phone/IP Starter Configuration This is a summary. For full details, see “Technical Configuration” chapter of User Guide. 1. Open MultiVOIP program: Start | MultiVOIP xxx | Configuration. 2. Go to Configuration | Ethernet/IP. Enter the IP parameters for your voip site. Activate Packet Prioritization (802.1p) if desired. If you use a Domain Name Server (DNS), specify its IP address.
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4. Web Browser GUI Setup (Optional). To do configuration and operation procedures using the web browser GUI, you must first set it up. To do so, follow these steps. (The browser used must be Internet Explorer 6.0 or above; or Netscape 6.0 or above; or FireFox 1.0 or above.) A.
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Phone/IP Starter Configuration (continued) 5. Go to Configuration | Voice/Fax. Select Coder | “Automatic.” At the right-hand side of the dialog box, click OK. If you know any specific parameter values that will apply to your system, enter them. Click Copy Channel. Select Copy to All. Click Copy. At main Voice/Fax Parameters screen, click OK to exit from the dialog box.
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Whenever email log messages are sent out, they must have a standard Subject line. Something like “Phone Logs for Voip N” is useful. If you have more than one MultiVoip unit in the building, you’ll need a unique identifier for each one (select a useful name or number for “N”).
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Phone/IP Starter Configuration (continued) 12. Enable premium (H.450) telephony features. Go to Supplementary Services For Call Hold, Call Transfer, & Call Waiting, specify the key sequence that the phone user will press to invoke the feature. For Call Name Identification, specify the allowed name types to be used and a caller- id descriptor.
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MultiVOIP User Guide QS: Phone/IP Starter Config. 17. Go to Save Setup | Save and Reboot. Click OK. This will save the parameter values that you have just entered. The MultiVOIP’s “BOOT” LED will light up while the configuration file is being saved and loaded into the MultiVOIP. Don’t do anything to the MultiVOIP until the “BOOT “LED is off (a loss of power at this point could cause the MultiVOIP unit to lose the configuration settings you have made).
Phonebook Starter Configuration (with remote voip) If the topic of voip phone books is new to you, it may be helpful to read the PhoneBook Tips section (page 47) before starting this procedure. To do this part of the quick setup, you need to know of another voip that you can call to conduct a test.
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4. Suppose you want to call a phone number outside of your building using a phone station that is an extension from your PBX system (if present). What digits must you dial? Often a “9” or “8” must be dialed to “get an outside line” through the PBX (i.e., to connect to the PSTN).
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5. In the “Destination Pattern” field of the Add/Edit Outbound Phonebook screen, enter the digits from step 4 followed by the digits from step 3. North America, Long-Distance Example Seattle-Chicago system. : enter 81312 as Answer Destination Pat- tern in Outbound Phone-book of Seattle voip.
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6. In the “Remove Prefix” field, enter the initial PBX access digit (“8” or “9”). North America, Long-Distance Example Seattle-Chicago system. : enter 8 in “Remove Answer Prefix” field of Seattle Outbound Phonebook. Euro, International Call Example Rotterdam/Bordeaux system. : enter 9 in “Remove Prefix” field of Outbound Answer Phonebook for Rotterdam voip.
Inbound Phonebook 1. Open the MultiVOIP program. ( Start | MultiVOIP xxx | Configuration ) 2. Go to Phone Book | Inbound Phonebook | Add Entry. 3. In the “Remove Prefix” field, enter your local calling code (area code, country code, city code, etc.) preceded by any other “access digits” that are required to reach your local site from the remote voip location (think of it as though the call were being made through the PSTN –...
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4. In the “Add Prefix” field, enter any digits that must be dialed from your local voip to gain access to the PSTN. North America, Long-Distance Example Seattle-Chicago system. On Seattle PBX, “9” is used to get an outside line. Answer: 9 is prefix to be added by local (Seattle) voip.
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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.”...
Phonebook Tips Preparing the phonebook for your voip system is a complex task that, at first, seems quite daunting. These tips may make the task easier. Use Dialing Patterns, Not Complete Phone Numbers generally enter complete phone numbers in the voip phonebook. Instead, you’ll enter “destination patterns”...
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QS: Phonebook Tips MultiVOIP User Guide . There are digits (PSTN access codes) that must be access codes dialed to gain access to an operator, to access the publicly switched ‘long-distance’ calling system(North America), to access the publicly switched ‘national’ calling system (Europe and elsewhere), or to access the publicly switched ‘international’...
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Knowing When to Drop Digits When calling area codes and access codes are used in combination, a leading “1” or “0” must sometimes be dropped. Phonebook Entry Example Area code for Inner London is listed as “0207.” However, in international calls the leading “0”...
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Using a Comma Commas are used in telephone dialing strings to indicate a pause to allow a dial tone to appear (common on PBX and key systems). Commas may be used only in the “Add Prefix” field of the Inbound Phonebook. .
Phonebook Example Boise Office Area: 208 PBX System. Main Number: 333-2700 PSTN 90 extensions 204.16.49.73 24-Channel Digital VoIP (MVP2410) Inbound Phonebook Each contains Inbound Phonebook tw o entries. The first entry (4 digits) specifies how incoming calls from the other voip sites w ill be handled if they go out onto the local PSTN .
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Voip Sites with Phonebooks Boise Office Boise Voip Inbound Phonebook Area: 208 Prefix to Prefix PBX System. to Add Remove Main Number: 91208 333-2700 PSTN 90 extensions 204.16.49.73 24-Channel Digital VoIP (MVP2410) Network Santa Fe Voip Santa Fe Voip Inbound Phonebook Outbound Phonebook Prefix to Prefix...
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Sample Phonebooks Enlarged Boise Voip Inbound Phonebook Prefix to Prefix Description Destin. Total to Add Incoming Calls Digits Remove Pattern 91208 Incoming calls 91505 to PSTN, Boise Area i ncoming calls to extensions of company’s PBX system in Boise 91520 Santa Fe Voip Inbound Phonebook Prefix...
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Phonebook Worksheet Voip Location/ID:____________________________ Inbound Phonebook Outbound Phonebook Prefix Description Total Prefix to Destin. to Add Incoming Calls Digits Remove Pattern Other Details: Voip Location/ID:____________________________ Inbound Phonebook Prefix to Prefix Description Destin. to Add Incoming Calls Remove Pattern Other Details: Voip Location/ID:____________________________ Inbound Phonebook Outbound Phonebook...
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MultiVOIP User Guide QS: Phonebook Example Enlarged Phonebook Worksheet...
Connectivity Test The procedures “Phone/IP Starter Configuration” and “Phonebook Starter Configuration” must be completed before you can do this procedure. 1. These connections must be made: MultiVOIP to local phone station –OR-- MultiVOIP to extension of key phone system MultiVOIP to command PC MultiVOIP to Internet 2.
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4. You now need to free up the COM port connection (currently being used by the MultiVOIP program) so that the HyperTerminal program can use it. To do this, you can either (a) click on Connection in the sidebar and select “Disconnect” from the drop-down box, or (b) close down the MultiVOIP program altogether.
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7. Make VOIP call. Make call on a local phone line accessing PSTN directly or through key system. 8. Read console messages recorded on HyperTerminal. Console Messages from Originating VOIP. The voip unit that originates the call will send back messages like that shown below. [00026975] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[1] TimeStamp : 26975 [00027190] CAS[0] : TX : ABCD = 1, 1, 1, 1...
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Console Messages from Terminating VOIP. The voip unit connected to the phone where the call is answered will send back messages like that shown below. [00170860] H323[0]: New incoming call [00170860] PSTNIF : Placing call on channel 0 Outbound digit 7175662 [00170885] CAS[0] : TX : ABCD = 1, 1, 1, 1 [00171095] H323IF [0]: MasterSlaveStatus=Master [00171105] CAS[0] : RX : ABCD = 1, 1, 1, 1,Pstn State[7]...
Troubleshooting If you cannot establish connectivity between two voips in the system, follow the steps below to determine the problem. 1. Ping both MultiVOIP units to confirm connectivity to the network. 2. Verify the telephone connections. Check cabling. Are connections well seated? To correct receptacle? Are telephone Interface Parameter settings correct? 3.
When the battery starts to weaken, the date and time may be incorrect. If the battery fails, the board must be sent back to Multi-Tech Systems for battery replacement. Warning: There is danger of explosion if the battery is incorrectly replaced.
MultiVOIP User Guide Mechanical Installation & Cabling Unpacking Your MultiVOIP When unpacking your MultiVOIP, check to see that all of the items shown are included in the box. For the various MultiVOIP models, the contents of the box will be different. Study the particular illustration below that is appropriate to the model you have purchased.
MultiVOIP User Guide Mechanical Installation & Cabling Unpacking the MVP210-SS MultiVOIP Cabling Guide Figure 3-2: Unpacking the MVP210-SS...
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Mechanical Installation & Cabling MultiVOIP User Guide Rack Mounting Instructions for MVP410-SS & MVP810-SS The MultiVOIPs can be mounted in an industry-standard EIA 19-inch rack enclosure, as shown in Figure 3-3. Figure 3-3: Rack-Mounting (MVP410SS or MVP810SS)
MultiVOIP User Guide Mechanical Installation & Cabling Safety Recommendations for Rack Installations Ensure proper installation of the unit in a closed or multi-unit enclosure by following the recommended installation as defined by the enclosure manufacturer. Do not place the unit directly on top of other equipment or place other equipment directly on top of the unit.
19-Inch Rack Enclosure Mounting Procedure Attaching the MultiVOIP to a rack-rail of an EIA 19-inch rack enclosure will certainly require two persons. Essentially, the technicians must attach the brackets to the MultiVOIP chassis with the screws provided, as shown in Figure 3-4, and then secure unit to rack rails by the brackets, as shown in Figure 3-5.
Cabling Procedure for MVP-410SS/810SS Cabling involves connecting the MultiVOIP to your LAN and telephone equipment. . If all channels of your MultiVOIP will be For DID channels only using either FXS, FXO, or E&M telephony interfaces, skip to step 2. For any channel on which you are using the DID interface type, you must change the jumper on the MultiVOIP circuit card.
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d. Identify the channels on which the DID interface will be used. Upper Circuit Card MVP810 only Ch 6 Ch 7 Ch 8 Ch 5 Jumpers 5-8 Main Circuit Card MVP-410/810 Ch 1 Ch 2 Ch 3 Ch 4 Jumpers 1-4 Figure 3-7.
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2. Connect the power cord supplied with your MultiVOIP to a live AC outlet and to the power connector on the back of the MultiVOIP as shown at top right in Figure 3-8. Command Modem connector for remote configuration CO MMAN D E &M FX S /FXO E&...
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Connect the other end to the trunk line. Verify that the E&M Type in the E&M Options group of the Interface dialog box is the same as the E&M trunk type supported by the telephone switch. See Appendix B for an E&M cabling pinout. For a DID connection.
Cabling Procedure for MVP210-SS Cabling involves connecting the MultiVOIP to your LAN and telephone equipment. . If both channels of your MVP210-SS For DID channels only MultiVOIP will be using either FXS, FXO, or E&M telephony interfaces, skip to step 2.
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R 56 R5 5 MVP210SS Circuit Board Figure 3-10. MVP210-SS Channel Jumper Settings e. Position the jumper for each DID channel so that it does not connect the two jumper posts. For DID operation of a voip channel, the MultiVOIP will work properly if you simply remove the jumper...
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MultiVOIP and to a live AC outlet as shown in Figure 3-11. Figure 3-11: Cabling for MVP210-SS 3. Connect the MultiVOIP to a PC by using a RJ-45 (male) to DB-9 (female) cable. Plug the RJ-45 end of the cable into the Command port of the MultiVOIP and the other end into the PC serial port.
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For an E&M connection. (E&M Example: trunk line from telephone switch.) Connect one end of an RJ-45 phone cord to the Channel 1 E&M connector on the back of the MultiVOIP. Connect the other end to the trunk line. Verify that the E&M Type in the E&M Options group of the Interface dialog box is the same as the E&M trunk type supported by the telephone switch.
Introduction Configuring software for your MultiVOIP entails three tasks: (1) loading the software onto the PC (this is “Software Installation and is discussed in this chapter), (2) setting values for telephony and IP parameters that will fit your system (this is “Technical Configuration” and it is discussed in Chapter 5), and (3) establishing “phonebooks”...
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2. Insert the MultiVOIP CD into your CD-ROM drive. The CD should start automatically. It may take 10 to 20 seconds for the Multi-Tech CD installation window to display. If the Multi-Tech Installation CD window does not display automatically, click My Computer, then right click the CD ROM drive icon, click Open, and then click the Autorun icon.
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Software Installation MultiVOIP User Guide 4. A ‘welcome’ screen appears. Press Enter or click Next to continue.
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MultiVOIP User Guide Software Installation 5. Follow the on-screen instructions to install your MultiVOIP software. The first screen asks you to choose the folder location of the files of the MultiVOIP software. Choose a location and click Next.
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Software Installation MultiVOIP User Guide 6. At the next screen, you must select a program folder location for the MultiVOIP software program icon. Click Next. Transient progress screens will appear while files are being copied.
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7. On the next screen you can select the COM port that the command PC will use when communicating with the MultiVoip unit. After software installation, the COM port can be re-set in the MultiVOIP Software (from the sidebar menu, select Connection | Settings to access the COM Port Setup screen or use the keyboard shortcut Ctrl + G).
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8. Transient screens will flash by as files are being copied. Then a completion screen will appear. Click Finish. 9. When setup of the MultiVOIP software is complete, you will be prompted to run the MultiVOIP software to configure the VOIP. Software installation is complete at this point.
Un-Installing the MultiVOIP Configuration Software 1. To un-install the MultiVOIP configuration software, go to Start | Programs and locate the entry for the MultiVOIP program. Select Uninstall. 2. Two confirmation screens will appear. Click Yes and OK when you are certain you want to continue with the uninstallation process.
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Software Installation MultiVOIP User Guide 3. A special warning message similar to that shown below may appear concerning the MultiVOIP software’s “.bin” file. Click Yes.
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MultiVOIP User Guide Software Installation 4. A completion screen will appear. Click Finish.
MultiVOIP User Guide Technical Configuration Configuring the MultiVOIP There are two ways in which the MultiVOIP must be configured before operation: technical configuration and phonebook configuration. Technical Configuration. First, the MultiVOIP must be configured to operate with technical parameter settings that will match the equipment with which it interfaces.
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Technical Configuration MultiVOIP User Guide Local configuration is done through a connection between the “Command” port of the MultiVOIP and the COM port of the computer; the MultiVOIP configuration program is used. Remote configuration is done through a connection between the MultiVOIP’s Ethernet (network) port and a computer connected to the same network.
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Functional Equivalence of Interfaces. The MultiVOIP configuration program is required to do the initial configuration (that is, setting an IP address for the MultiVOIP unit) so that the VOIP unit can communicate with the MultiVoipManager program or with the web browser GUI. Management of the VOIP after that point can be done from any of these three programs since they all offer essentially the same functionality.
Local Configuration This manual primarily describes local configuration with the Windows GUI. After IP addresses have been set locally using the Windows GUI, most aspects of configuration (logging functions are an exception) can be handled through the web browser GUI, as well (see the Operation and Maintenance chapter of this manual).
Write down the values for these IP parameters. You will need to enter these values in the “IP Parameters” screen in the Configuration section of the MultiVOIP software. You must have this IP information about every VOIP in the system. Telephony Interface Parameters The following parameters must be known about the PBX or telco central office equipment to which the analog MultiVOIP will connect:...
Technical Configuration MultiVOIP User Guide SMTP Parameters (for email call log reporting) required if log reports of VOIP call traffic Optional are to be sent by email SMTP Parameters Preparation Task: Ask Mail Server To: I.T. Department re: email account for VOIP administrator to set up email account (with password) for the...
Config Info CheckList Type of Configuration MultiVOIP Info Gathered Configuration screen on which to enter the Info IP Info for voip unit Ethernet/IP Parameters ● IP address ● Gateway ● DNS IP (if used) ●802.1p Prioritization (if used) Interface Type Interface (Choices: E&M, Parameters...
Local Configuration Procedure (Summary) After the MultiVOIP configuration software has been installed in the ‘Command’ PC (which is connected to the MultiVOIP unit), several steps must be taken to configure the MultiVOIP to function in its specific setting. Although the summary below includes all of these steps, some are optional.
20. Set SIP Server Configuration parameters. 21. Set SIP Server PreDefined Endpoint parameters. 22. View System Info screen and set updating interval (optional). 23. Save the MultiVOIP configuration. 24. Create a User Default Configuration (optional). When technical configuration is complete, you will need to configure the MultiVOIP’s inbound and outbound phonebooks.
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Technical Configuration MultiVOIP User Guide 3. Confirm Connection. If the MultiVOIP is set for an available COM port and is correctly cabled to the PC, the MultiVOIP main screen will appear. (If the main screen appears grayed out and seems inaccessible, go to step 4.)
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MultiVOIP User Guide Technical Configuration 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.
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4. Solving Common Connection Problems. A. 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.
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4B. Fixing a Cabling Problem. If the MultiVOIP cannot be located by the computer, four error messages will appear (saying “MultiVOIP- SS Not Found,” “Phone Database Not Read,” “SIP Endpoint Database Not Read,” and “Password Phone Database Not Read”). 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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Technical Configuration MultiVOIP User Guide 6. Set Ethernet/IP Parameters. This dialog box can be reached by pulldown menu, toolbar icon, keyboard shortcut, or sidebar. Accessing “Ethernet/IP Parameters” Pulldown Icon Shortcut Sidebar Ctrl + Alt + I...
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MultiVOIP User Guide Technical Configuration In each field, enter the values that fit your particular network.
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The Ethernet/IP Parameters fields are described in the tables and text passages below. Note that both DiffServ parameters (Call Control PHB and VoIP Media PHB) must be set to zero if you enable Packet Prioritization (802.1p). Nonzero DiffServ values negate the prioritization scheme.
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Ethernet/IP Parameter Definitions (cont’d) Field Name Values Ethernet Parameters 802.1p 4 – Controlled Load : applications subject to some form of (continued) “Admission Control”, such as preplanning of Network requirement, characterized by bandwidth reservation per flow. Traffic characterized by 5 – Video : delay <...
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Ethernet/IP Parameter Definitions (cont’d) Field Name Values IP Parameter fields Gateway alphanumeric Name Enable DHCP disabled by default IP Address 4-places, 0-255 IP Mask 4-places, 0-255 Gateway 4-places, 0-255. Description Descriptor of current voip unit to distinguish it from other units in system. Dynamic Host Configuration Protocol is a method for assigning IP...
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Ethernet/IP Parameter Definitions (cont’d) Field Name Values DiffServ PHB (Per Hop Behavior) values DiffServ pertain to a differential prioritizing Parameter system for IP packets as handled by fields DiffServ-compatible routers. 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;...
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The IP Datagram with Header, Its Type-of-Service field, & DiffServ bits => VERS HLEN TYPE OF SERVICE IDENTIFICATION TIME TO LIVE PROTOCOL SOURCE IP ADDRESS DESTINATION IP ADDRESS IP OPTIONS (if any) The TOS field consists of eight bits, of which only the first six are used. These six bits are called the “Differentiated Service Codepoint”...
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Ethernet/IP Parameter Definitions (cont’d) Field Name Values FTP Parameter fields FTP Server Enable Default = disabled See “FTP Server File Transfers” in Operation & Maintenance chapter. DNS Parameter fields Enable DNS Default = disabled Enable SRV DNS Server IP 4-places, 0-255. Address Description MultiVOIP unit has an...
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About Service Records An SRV record holds the following information: • Service: the symbolic name of the desired service. • Protocol: this is usually either TCP or UDP. • Domain name: the domain for which this record is valid. • TTL: standard DNS time to live field.
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7. Set up the Web Browser GUI (Optional). After an IP address for the MultiVOIP unit has been established, you can choose to do any further configuration of the unit (a) by using the MultiVOIP web browser GUI, or (b) by continuing to use the MultiVOIP Windows GUI. If you want to do configuration work using the web browser GUI, you must first set it up.
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8. Set Voice/FAX Parameters. This dialog box can be reached by pulldown menu, toolbar icon, keyboard shortcut, or sidebar. Accessing “Voice/FAX Parameters” Pulldown Shortcut Ctrl + H Icon Sidebar...
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MultiVOIP User Guide Technical Configuration In each field, enter the values that fit your particular network.
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Technical Configuration MultiVOIP User Guide Note that Voice/FAX parameters are applied on a channel-by-channel basis. However, once you have established a set of Voice/FAX parameters for a particular channel, you can apply this entire set of Voice/FAX parameters to another channel by using the Copy Channel button and its dialog box.
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The Voice/FAX Parameters fields are described in the tables below. Voice/Fax Parameter Definitions Field Name Values Description Default When this button is clicked, all Voice/FAX parameters are set to their default values. Select Channel to be configured is selected 1-2 (210) Channel here.
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Voice/Fax Parameter Definitions (cont’d) Field Name Values Description DTMF Parameters Duration 60 – 3000 When DTMF: Out of Band is selected, (DTMF) this setting determines how long each DTMF digit ‘sounds’ or is held. Default = 100 ms. Not supported in 5.02c BRI software.
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Voice/Fax Parameter Definitions (cont’d) Field Name Valuee Description FAX Parameters (cont’d) Fax Volume -18.5 dB Controls output level of fax tones. To Default = to –3.5 dB be changed only under the direction of -9.5 dB Multi-Tech’s Technical Support. Jitter Value Default = Defines the inter-arrival packet (Fax)
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Voice/Fax Parameter Definitions (cont’d) Coder Parameters Coder Manual or Determines whether selection of Auto- coder is manual or automatic. matic 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.
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Voice/Fax Parameter Definitions (cont’d) Field Name Values Description Advanced Features Silence Determines whether silence Compression 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.
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Voice/Fax Parameter Definitions (cont’d) Field Name Values Description AutoCall/Offhook Alert Parameters Auto Call / AutoCall, The AutoCall option enables the local Offhook Offhook MultiVOIP to call a remote MultiVOIP Alert Alert 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...
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Voice/Fax Parameter Definitions (cont’d) Field Name Values Description AutoCall/Offhook Alert Parameters Auto Call / AutoCall, (continued from previous page) Offhook Offhook Both functions apply on a channel-by- Alert Alert 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.
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Voice/Fax Parameter Definitions (cont’d) Field Name Values Description AutoCall/Offhook Alert Parameters Offhook 0 – 3000 The length of time that must elapse Alert Timer seconds before the offhook alert is triggered and a call is automatically made to the phone number listed in the Phone Number field.
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Voice/Fax Parameter Definitions (cont’d) ) Field Name Values Description Dynamic Jitter Dynamic Dynamic Jitter defines a minimum Jitter Buffer and a maximum jitter value for voice communications. When receiving voice packets from a remote between packets may occur due to network traffic problems.
Voice/Fax Parameter Definitions (cont’d) Field Name Values Dynamic Jitter Maximum 60 to 400 Jitter Value Optimizat- 0 to 12 ion Factor Modem Relay To place modem traffic onto the voip network (an application called “modem relay”), use Coder G.711 mu-law at 64kbps. Description The maximum dynamic jitter buffer of 400 milliseconds is the maximum...
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Voice/Fax Parameter Definitions (cont’d) ) Field Name Values Description Auto Disconnect Automatic The Automatic Disconnection Disconnect- group provides four options which can be used singly or in any combination. Jitter Value 1-65535 The Jitter Value defines the average milli- inter-arrival packet deviation (in seconds milliseconds) before the call is automatically disconnected.
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Technical Configuration MultiVOIP User Guide 9. Set Telephony Interface Parameters. This dialog box can be reached by pulldown menu, toolbar icon, keyboard shortcut, or sidebar. Accessing Telephony Interface Parameters Pulldown Icon Shortcut Sidebar Ctrl + Alt + N...
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MultiVOIP User Guide Technical Configuration In each field, enter the values that fit your particular network. The kinds of parameters for which values must be chosen depend on the type of telephony supervisory signaling or interface used (FXO, E&M, etc.). We present here the various parameters grouped and organized by interface type.
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Technical Configuration MultiVOIP User Guide Note that Interface parameters are applied on a channel-by-channel basis. However, once you have established a set of Interface parameters for a particular channel, you can apply this entire set of Voice/FAX parameters to another channel by using the Copy Channel button and its dialog box.
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FXS Loop Start Parameters. The parameters applicable to FXS Loop Start are shown in the figure below and described in the table that follows. FXS Loop Start Interface: Parameter Definitions Field Name Values FXS (Loop Start) Description Enables FXS Loop Start interface type.
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FXS Loop Start Interface: Parameter Definitions (cont’d) Field Name Values Dialing Options fields Inter Digit 1 - 10 seconds Timer Message Waiting Indication Inter Digit in milliseconds Regeneration Time FXS Options fields FXS Ring 1-99 Count , FXS Current Loss Generate Current Reversal...
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FXS Loop Start Interface: Parameter Definitions (cont’d) Field Name Values Flash Hook Options fields Generation Detection . and Range 50 - 1500 milliseconds Pass Through Enable Caller ID fields Type Bellcore Enable 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...
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FXS Loop Start Interface: Parameter Definitions (cont’d) Field Name Values Caller ID fields Enable (cont’d) The Caller ID feature has dependencies on both the telco central office and the MultiVOIP phone book. See the diagram series after the FXO Parameters section below. Description The forms of the ‘Caller Name’...
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MultiVOIP User Guide Technical Configuration FXO Parameters. The parameters applicable to the FXO telephony interface type are shown in the figure below and described in the table that follows.
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FXO Interface: Parameter Definitions* Field Name Values Interface Type Dialing Options Regeneration Pulse, DTMF Inter Digit 1 to 10 seconds Timer Message Waiting Indication Inter Digit 50 to 20,000 Regeneration milliseconds Time FXO Options FXO Ring 1-99 Count No Response 1 –...
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FXO Interface: Parameter Definitions (cont’d) Field Name Values Flash Hook Options fields Generation 50 - 1500 milliseconds Detection Range Caller ID fields Caller ID Type Bellcore Caller ID enable Description Length of flash hook that will be generated and sent out when the remote end initiates a flash hook and it is regenerated locally.
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The Caller ID feature has dependencies on both the telco central office and the MultiVOIP phone book. See the diagram series below. CID Flow Call is received here. Terminating VoIP Network xxxyyyzzzz J.Q. Public Clock: 5-31, 1:42pm Display shows: H.323 or SPP Protocol CID Number: 763-555-8794 CID Name:...
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CID Flow Call is received here. Terminating Generating VoIP VoIP Network x xxy yy zz zz J.Q. Pu b l i c Clock: 15:26, 5-31 Display shows: SPP Protocol CID Number: 423 CID Name: Shipping Dept Time Stamp: Date: 0927 Time: 1747 ...
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CID Flow Call is received here. Generating Terminating VoIP xxxyyyzzzz Network J.Q. Public Clock: 11/15, 6:17pm Display shows: H.323 Protocol CID Number: 423 Phone Book Configuration CID Name: Anoka-Whse-VP3 Gateway Name: Anoka-Whse-VP3 Time Stamp: Date: 11/15 Time: 6:17pm Q.931 Parameters In x.06 release, when SIP protocol is used, CID Name field will duplicate value in Gatekeeper RAS Parameters...
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MultiVOIP User Guide Technical Configuration 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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FXO Supervision Parameter Definitions Field Name Values Answer Supervision fields Current Reversal Answer Delay Answer Delay integer values Timer (in seconds) Range = 1 - 65535 Tone Detection Available dial tone, Tones ring tone, busy tone, unobtainable tone (fast busy), survivability tone, re-order tone...
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FXO Supervision Parameter Definitions Field Name Values Disconnect Supervision fields Current Reversal Current Loss Current Loss 200 to 2000 Timer (in milliseconds) Silence Detection Enable Silence One-Way or Detection Type Two-Way Silence Timer integer value in seconds Description There are four possible criteria for disconnection under FXO: current reversal, current loss, tone detection, and silence...
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FXO Supervision Parameter Definitions Field Name Values Disconnect Supervision fields DTMF Tone DTMF Tone Pairs High Tones 1209Hz 1336Hz 1447Hz 1633Hz Disconnect tone pair Tone Sequence tone pair Tone Detection Available dial tone, Tones ring tone, busy tone, unobtainable tone (fast busy), survivability tone, re-order tone...
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MultiVOIP User Guide Technical Configuration E&M Parameters. The parameters applicable to the E&M telephony interface type are shown in the figure below and described in the table that follows.
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E&M Interface Parameter Definitions Field Name Values Interface E&M Type Types 1-5. Mode 2-wire or 4-wire Signal Dial Tone or Wink Wink Timer integer values, (in ms) in milliseconds No Response integer values (in Timer seconds) Disconnect on Call Progress Tone Pass Through Enable...
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E&M Interface Parameter Definitions (cont’d) Field Name Values Dialing Options Inter Digit integer values, Timer in seconds Message Light or None Waiting Indication Inter Digit milliseconds Regeneration Timer Description This is the length of time that the MultiVOIP will wait between digits.
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E&M Interface Parameter Definitions (cont’d) Field Name Values Dialing Options (cont’d) Flash Hook Options fields Generation integer values, in milliseconds Detection . and Range 50 1500 milliseconds Description Length of flash hook that will be generated and sent out when the remote end initiates a flash hook and it is regenerated locally.
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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.
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DID Interface Parameter Definitions ( cont’d ) Field Name Values DID Options (cont’d) Start Modes Immediate Start, Wink Start, Delay Dial Wink Timer integer values, (in ms) in milliseconds Dialing Options Inter Digit integer values, Timer in seconds Message Waiting Indication Inter-Digit integer values,...
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MultiVOIP User Guide Technical Configuration 10. Set Call Signaling Parameters. This dialog box addresses SIP Call Signaling parameters. It can be reached by pulldown menu, keyboard shortcut, or a sidebar menu. Accessing “Call Signaling Parameters” Pulldown Shortcut Sidebar Ctrl + Alt + Shft + P...
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The tables below describes all fields in the general SIP Call Signaling screen. SIP Call Signaling Parameter Definitions Field Name Values SIP Proxy Parameters Signaling Port Use SIP Proxy Description Port number on which the MultiVOIP UserAgent software module will be waiting for any incoming SIP requests.
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SIP Call Signaling Parameter Definitions (cont’d) Field Name Values Description SIP Proxy Parameters Allow When selected, incoming calls Incoming Calls are accepted only if those calls Through SIP come through the gatekeeper. Proxy Only Primary Proxy This is the preferred SIP proxy server for controlling the traffic of the current voip.
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SIP Call Signaling Parameter Definitions (cont’d) Field Name Values & Description SIP Proxy Parameters Password Values : alphanumeric Description : Password for proxy server function. See “User Name” description above. Values : numeric (in seconds) Registration Description : This is the timeout interval for Time registration of the MultiVOIP with a SIP proxy server.
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MultiVOIP User Guide Technical Configuration 12. Set Regional Parameters (Phone Signaling Tones & Cadences). This dialog box can be reached by pulldown menu, keyboard shortcut, or sidebar. Accessing “Regional Parameters” Pulldown Icon Shortcut Sidebar Ctrl + R...
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Technical Configuration MultiVOIP User Guide The Regional Parameters screen will appear. 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. Each MVP410 and MVP810 Remote Configuration/Command Modem.
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The Regional Parameters fields are described in the table below. “Regional Parameter” Definitions Field Name Values Country/ USA, Japan, UK, Region Custom Description Name of a country or region that uses a certain set of tone pairs for dial tone, ring tone, busy tone (fast busy tone), unobtainable tone (tone heard...
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“Regional Parameter” Definitions Field Name Values Country/ USA, Japan, UK, Region Custom Note “Survivability” tone indicates a special type of call-routing redundancy & applies to MultiVantage voip units only. Advisory screen This message screen appears whenever the Country field is changed. It informs the operator that, upon change of the Country field value, all User Defined Tones will be deleted.
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“Regional Parameter” Definitions (cont’d) Field Name Values Standard Tones fields (cont’d) Gain 1 gain in dB +3dB to –31dB and “mute” setting Gain 2 gain in dB +3dB to –31dB and “mute” setting Cadence n/n/n/n (msec) On/Off four integer time values in milli-seconds;...
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“Regional Parameter” Definitions (cont’d) Field Name Values Country country name Selection for Built-In Modem (not applicable to MVP- 130/130FXS MVP210, MVP410ST, or MVP810ST) User Defined Tones fields Type column alphanumeric name specified by user Frequency 1 freq. in Hertz Frequency 2 freq.
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MultiVOIP User Guide Technical Configuration 13. Set 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 (fast busy signal) or “re-order” tones (telling the user that she must hang up an off-hook phone) or “survivability”...
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The Custom Tone-Pair Settings fields are described in the table below. Custom Tone-Pair Settings Definitions Field Name Values Tone Pair dial tone, busy tone, ring tone, ‘unobtainable’ tone, survivability tone, re-order tone TONE PAIR VALUES Frequency 1 frequency in Hertz Frequency 2 frequency in Hertz...
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Custom Tone-Pair Settings Definitions Field Name Values Cadence 1 integer time value in milli-seconds; zero value for dial-tone indicates continuous tone Cadence 2 duration in milliseconds Cadence 3 duration in milliseconds Cadence 4 duration in milliseconds Description On/off pattern of tone durations used to denote phone ringing, phone busy, dial tone (“0”...
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14. Set 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.).
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The SMTP Parameters screen is shown below “SMTP Parameters” Definitions Field Name Values Enable SMTP Requires Authentication Login Name alpha- numeric, per email domain Description In order to send log reports by email, this box must be checked. However, to enable SMTP functionality, you must also select “SMTP”...
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“SMTP Parameters” Definitions (cont’d) Field Name Values Password alpha- numeric Mail Server IP n.n.n.n Address for n= 0 to Port Number Mail Type text or html Subject text Reply-To email address Address Recipient email address Address Mail Criteria Number of integer Records Number of...
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The SMTP Parameters dialog box has a secondary dialog box, Custom Fields, that allows you to customize email log messages for the MultiVOIP. 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.
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“Custom Fields” Definitions (cont’d) Field Description Field Total bytes sent in Bytes Bytes Sent Received call. Packets lost in Packets Coder Lost call. The DTMF dialing Outbound Prefix Digits Matched digits received by Received this gateway from the remote gateway presuming that DTMF is set to "Out of Band."...
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“Custom Fields” Definitions (cont’d) Field Description Field The IP address of Server Outbound Digits Sent Details the traffic control server (if any) being used (whether an H.323 gatekeeper, a SIP proxy, or an SPP registrar gateway) will be displayed here if the call is handled through that server.
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Technical Configuration MultiVOIP User Guide...
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15. Set Log Reporting Method. The Logs 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: A.
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Technical Configuration MultiVOIP User Guide “Filters” button and using the Console Messages Filter Settings screen (see subsequent page). 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.
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“Logs” Screen Definitions Field Name Values Description Enable Allows MultiVOIP debugging messages to be Console read via a basic terminal program like Messages HyperTerminal ™ or equivalent. Normally, this should be disabled because it uses MultiVOIP processing resources. Console messages are meant for tech support personnel.
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Technical Configuration MultiVOIP User Guide To customize console messages by category and/or by channel, click on “Filters” and use the Console Messages Filters Settings screen.
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16. Set Supplementary Services Parameters. This dialog box can be reached by pulldown menu, keyboard shortcut, or sidebar. Accessing “Supplementary Services” Parameters Pulldown Shortcut Ctrl + Alt +H Supplementary Services features derive from the H.450 standard, which brings to voip telephony functionality once only available with PSTN or PBX telephony.
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In each field, enter the values that fit your particular network. 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.
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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.
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The Supplementary Services fields are described in the tables below. Supplementary Services Parameter Definitions Field Name Values Description Select The channel to be configured is 1-4 (410SS) Channel selected here. 1-8 (810SS) Call Select to enable the Call Transfer Transfer function in the voip unit.
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Supplementary Services Definitions (cont’d) Field Name Values Description Call Hold Select to enable Call Hold function in Enable voip unit. Call Hold allows one party to maintain an idle (non-talking) connection with another party while receiving another call (Call Waiting), while initiating another call (Call Transfer), or while performing some other call management function.
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Supplementary Services Definitions (cont’d) Field Name Values Description Call Name Enables CNI function. Call Name Identification is not the same as Caller Identification Enable ID. When enabled on a given voip unit currently being controlled by the MultiVOIP GUI (the ‘home voip’), Call Name Identification sends an identifier and status information to the administrator of the remote voip...
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Supplementary Services Definitions (cont’d) Field Name Values Description Calling If the ‘home’ voip unit is originating Party, the call and Calling Party is selected, Allowed then the identifier (from the Caller Id Name Type field) will be sent to the remote voip (CNI) unit being called.
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Supplementary Services Definitions (cont’d) Field Name Values Description Alerting If the ‘home’ voip unit is receiving the Party, call and Alerting Party is selected, Allowed then the identifier (from the Caller Id Name Type field) will tell the originating remote (CNI) voip unit that the call is ringing.
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Supplementary Services Definitions (cont’d) Field Name Values Description Busy Party, If the ‘home’ voip unit is receiving a Allowed call directed toward an already Name Type engaged channel or phone station and (CNI) 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.
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Supplementary Services Definitions (cont’d) Field Name Values Description Connected If the ‘home’ voip unit is receiving a Party, call and Connected Party is selected, Allowed then the identifier (from the Caller Id Name Type field) will tell the originating remote (CNI) voip unit that the attempted call has been completed and the connection is...
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Supplementary Services Definitions (cont’d) Field Name Values Description Caller ID This is the identifier of a specific channel of the ‘home’ voip unit. The Caller Id field typically describes a person, office, or location, for example, “Harry Smith,” or “Bursar’s Office,”...
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Technical Configuration MultiVOIP User Guide 17. Set NAT Traversal parameters. NAT (Network Address Translation) parameters are applicable only when the MultiVOIP is operating in SIP mode. The use of STUN (Simple Traversal of UDP NATs) servers to aid networks with NAT devices is described in RFC 3489.
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MultiVOIP User Guide Technical Configuration Accessing “NAT Traversal” Parameters Pulldown Icon Shortcut Sidebar Ctrl + Alt + Sft + VH Descriptions for NAT Traversal screen fields are presented in the table below.
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NAT Traversal Definitions (cont’d) Field Name Values Description Enable Enables STUN client functionality in (STUN) 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.
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18. Set RADIUS parameters. In general, RADIUS is concerned with authentication, authorization, and accounting. The MultiVOIP-SS supports the authentication functions. In the Attributes secondary screen (accessed by clicking on Select Attributes), the voip administrator can select the parameters to be tallied by the RADIUS server.
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Technical Configuration MultiVOIP User Guide The fields of the RADIUS screen are described in the table below.
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RADIUS Screen Field Definitions Field Name Values Description Server n.n.n.n IP address of the RADIUS server that Address 0 – 255 handles accounting for the current MultiVOIP unit. Accounting TDM time slot at which RADIUS numeric; Port accounting information will be 1 - 65535 transmitted and received.
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The RADIUS Parameters dialog box has a secondary dialog box, Custom Fields, that allows you to customize accounting information sent to the RADIUS server by the MultiVOIP. The MultiVOIP software logs data about many aspects of the call traffic going through the MultiVOIP.
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“Custom Fields” Definitions (cont’d) Field Description Field Bytes Bytes Total bytes sent in Received Sent call. Packets Packets lost in Coder Lost call. The DTMF dialing Prefix Outbound Digits digits received by Matched Sent this gateway from the remote gateway presuming that DTMF is set to "Out of Band."...
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“Custom Fields” Definitions (cont’d) Field Description Field From Details Gateway Originating Gatew N. Number gateway IP Addr IP address where IP Addr call originated. Descript Identifier of site Descript where call originated. Options When , log Options selected will not use Silence Compression and Forward Error...
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MultiVOIP User Guide Technical Configuration 19. 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.
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Technical Configuration MultiVOIP User Guide 20. Set SIP Server Configuration parameters. Accessing SIP Server Configuration Parameters Sidebar...
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SIP Server Configuration Parameter Definitions Field Name Values Description Operating In “Survivability” mode, the MVP-SS surviv., Mode unit can function as a SIP server for stnd-alone other gateways in its network in case that network loses contact with the network’s main SIP server (typically a PBX).
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SIP Server Configuration Parameter Definitions Field Name Values Description Registrar Options If undefined registrations are allowed Allow (value= Y), then gateways other than Undefined those listed in the PreDefined Registrations Endpoints list can register with the MVP-SS voip unit as it functions in its SIP server mode.
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SIP Server Configuration Parameter Definitions Field Name Values Description Registrar Options integer The time after which the MultiVOIP Registration values; in UserAgent Client is supposed to register Time seconds; with the proxy server. default is Expiration of the registration interval 3600 means that the gateway has lost contact with the main SIP server and that the...
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The main screen for Predefined Endpoints is a list. If you click on function buttons to entries in this list of endpoints, a secondary screen will Edit appear and allow you to add new endpoints or edit existing endpoint entries. When your work with the list is complete, click SIP Server Predefined Endpoints Parameter Definitions Field Name...
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neither make nor receive calls. integer The time after which the MultiVOIP Re-Registration Interval values; in UserAgent Client is supposed to register seconds; with the proxy server. default is Expiration of the registration interval 3600 means that the gateway has lost contact with the main SIP server and that the MVP-SS unit will enter its ‘survivability’...
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22. View System Information screen and set updating interval (optional). This dialog box can be reached by pulldown menu, keyboard shortcut, or sidebar. Accessing “System Information” Screen Pulldown Shortcut Ctrl + Alt +Y Icon Sidebar...
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This screen presents vital system information at a glance. Its primary use is in troubleshooting. System Information Parameter Definitions Field Name Values Description Boot Indicates the version of the code that nn.nn Version is used at the startup (booting) of the voip.
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System Information Parameter Definitions (cont’d) Field Name Values Description Configur- Indicates version of MultiVOIP nn.nn.nn. ation Configuration software (which Version includes screens for IP Parameters, alpha- SMTP Parameters, Regional numeric Parameters, etc.). Phone Book Indicates the version of the inbound numeric Version and outbound phonebook portion of...
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The frequency with which the System Information screen is updated is determined by a setting in the Logs screen 23. 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.
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Technical Configuration MultiVOIP User Guide 24. 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.
T1 Versus E1 Telephony Environments Phonebooks for Series II analog MultiVOIP units (MVP130, MVP130FXS, MVP210, MVP410, and MVP810) can be operated in either a North American telephony standards environment (potentially operating with T1 digital MultiVOIPs) or in a European telephony standards environment (potentially operating with E1 digital MultiVOIPs).
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MultiVOIP User Guide T1 PhoneBook Configuration 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.
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Phonebook configuration screens can be accessed using icons or the sidebar menu. Phonebook Icons Description Phonebook Configuration Inbound Phonebook Entries List Add Inbound Phonebook Entry Edit selected Inbound Phonebook Entry Outbound Phonebook Entries List Add Outbound Phonebook Entry Edit selected Outbound Phonebook Entry...
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Phonebook Pulldown Menu Inbound Phonebook Shortcut Alt + I Phonebook Sidebar Menu Outbound Phonebook Shortcut Alt + O...
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T1 Phonebook Configuration MultiVOIP User Guide 1. Select Outbound Phone Book/List Entries. Fields in the “Details” section describe various SIP parameters. Click Add.
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MultiVOIP User Guide T1 PhoneBook Configuration 2. The Add/Edit Outbound PhoneBook screen appears. Enter Outbound PhoneBook 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).
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The fields of the Add/Edit Outbound Phone Book screen are described in the table below. Add/Edit Outbound Phone Book: Field Definitions Field Name Values Description Accept Any When checked, “Any Number Number” appears as the value in the Destination Pattern field. The Any Number feature works differently depending on whether or not an external...
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Add/Edit Outbound Phone Book: Field Definitions (cont’d) Field Name Values Description Destination prefixes, Defines the beginning of Pattern area codes, dialing sequences for calls exchanges, that will be connected to line another VOIP in the system. numbers, Numbers beginning with extensions these sequences are diverted from the PTSN and carried...
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Add/Edit Outbound Phone Book: Field Definitions (cont’d) Field Name Values Description SIP Fields Use Proxy Select if proxy server is used. Transport TCP or Voip administrator must choose Protocol between UDP and TCP transmission protocols. UDP is a high-speed, low-overhead connectionless protocol where data is transmitted without acknowledgment, guaranteed...
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MultiVOIP User Guide T1 PhoneBook Configuration Clicking on the Advanced button brings up the Alternate Routing secondary screen. This feature provides an alternate path for calls if the primary IP network cannot carry the traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN. However, this feature could also be used to divert traffic to a redundant (backup) unit in case one voip unit fails.
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Alternate Routing Field Definitions Field Values Name Alternate n.n.n.n where Address n= 0-255 Round milliseconds Trip Delay 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.
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MultiVOIP User Guide T1 PhoneBook Configuration Inbound PhoneBook | List Entries 3. Select...
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T1 Phonebook Configuration MultiVOIP User Guide 4. The Add/Edit Inbound PhoneBook screen appears.
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Enter Inbound PhoneBook data for your MultiVOIP. The fields of the Add/Edit Inbound PhoneBook screen are described in the table below. Add/Edit Inbound Phone Book: Field Definitions Field Name Values Description Accept Any : Y/N Values Number When checked, “Any Number” Description: appears as the value in the Remove Prefix field.
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Add/Edit Inbound Phone Book: Field Definitions (cont’d) Field Name Values Description Call Forward Parameters Forward Uncondit.; Unconditional all calls received will be Condition Busy forwarded. No Resp. Busy will be forwarded when station is busy. No Response calls will be forwarded if called party does not answer after a specified number of rings, as specified in Ring...
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Add/Edit Inbound Phone Book: Field Definitions (cont’d) Field Name Values and Description Ring Count When “No Response” is 0, 1, 2, 3, etc. condition for forwarding calls, this determines how many unanswered rings are needed to trigger the forwarding. Registration In a SIP voip system, gateways can register Option with the SIP Proxy.
T1 Phonebook Examples 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. The system depicted is H.323. However, the phonebook entries presented are still applicable for SIP systems.
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An outline of the equipment setup in both offices is shown below. Local-Call Area Codes: 612, 651, Company HQ. Minneapolis North Sub. area 763 Digital VoIP -5174 200.2.10.3 -5173 -5172 -5171 717-5170 Network Overlay Area Code: Baltimore Sales Ofc. area 410 Digital VoIP -7003...
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T1 Phonebook Configuration MultiVOIP User Guide The screen below shows Outbound PhoneBook entries for the VOIP located in the company’s Baltimore facility. The entries in the Minneapolis VOIP’s Inbound PhoneBook match the Outbound PhoneBook entries of the Baltimore VOIP, as shown below.
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MultiVOIP User Guide T1 PhoneBook Configuration 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.
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T1 Phonebook Configuration MultiVOIP User Guide Similarly, the Inbound PhoneBook for the Baltimore VOIP (shown first below) generally matches the Outbound PhoneBook of the Minneapolis VOIP (shown second below). 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.
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MultiVOIP User Guide T1 PhoneBook Configuration The Outbound PhoneBook 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” .
Configuring Mixed Digital/Analog VOIP Systems Analog MultiVOIP units, like the MVP-210/410/810/410SS/810SS are compatible with digital MultiVOIP units like the MVP2410. In many cases, digital and analog VOIP units will appear in the same telephony/IP system. In addition to MVP-210/410xx/810xx MultiVOIP units (Series II units), legacy analog VOIP units (Series I units made by MultiTech) may be included in the system, as well.
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MultiVOIP User Guide T1 PhoneBook Configuration The Series I analog VOIP phone book resides in the “Host” VOIP unit at Site B. It applies to both of the Series I analog VOIP units. Each of the Series II analog MultiVOIPs (the MVP210 and the MVP410) requires its own inbound and outbound phonebooks.
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These seven phone books are shown below. Phone Book for Series I Analog VOIP Host Unit (Site B) VOIP Dir # IP Address Channel -OR- Destination Pattern 200.2.9.8 200.2.9.8 200.2.9.6 200.2.9.7 1615 200.2.9.9 (Note 2.) xxxx 3xxx 200.2.9.9 (Note 1.) 1402 200.2.9.5 140226374...
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Note 1. The “x” is a wildcard character. Note 2. By specifying “Channel 0,” we instruct the MVP2400/2410 to choose any available data channel to carry the call. Note 3. Note that Site F key system has only 30 extensions (x7400-7429).
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Outbound Phone Book for MVP2410 Digital VOIP (Site D) Destin. Remove Pattern Prefix Prefix Address 200.2.9.7 1507 1507 101# 200.2.9.8 Note 3. 200.2.9.8 200.2.9.6 1402 200.2.9.5 1402 200.2.9.5 1402 200.2.9.5 1402 200.2.9.5 Note 3. The pound sign (“#”) is a delimiter separating the VOIP number from the standard telephony phone number.
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Inbound Phonebook for MVP2410 Digital VOIP (Site D) Remove Channel Comment Prefix Prefix Number 1615 Allows phone users at remote Note 4. voip sites to call non-toll Note 5. numbers within the Site D area code (615; Pierre, SD) over the VOIP network.
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Outbound Phone Book for MVP410 Analog VOIP (Site F) Destin. Remove Pattern Prefix Address Prefix 200.2.9.7 1507 1507 101# 200.2.9.8 Note 3. 200.2.9.8 200.2.9.6 1615 200.2.9.9 1615 200.2.9.9 Note 3. The pound sign (“#”) is a delimiter separating the VOIP number from the standard telephony phone number. Comment To originate calls to Site A...
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Inbound Phonebook for MVP410 Analog VOIP (Site F) Remove Channel Comment Prefix Prefix Number 1402 Access to Lincoln local PSTN by users at remote VOIP locations via FXO port at Site F. 1402 Gives remote voip users access 263740 to extension of key phone 1402 system at Site F (Lincoln).
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Outbound Phone Book for MVP210 Analog VOIP (Site E) Destin. Remove Pattern Prefix Prefix Address 200.2.9.7 1507 1507 101# 200.2.9.8 Note 3. 200.2.9.8 1402 200.2.9.5 1402 200.2.9.5 1615 200.2.9.9 1615 200.2.9.9 Note 3. The pound sign (“#”) is a delimiter separating the VOIP number from the standard telephony phone number.
Inbound Phonebook for MVP210 Analog VOIP (Site E) Remove Channel Prefix Prefix Number Call Completion Summaries Site A calling Site C, Method 1 1. Dial 101. 2. Hear dial tone from Site B. 3. Dial 7175662. 4. Await completion. Talk. Site A calling Site C, Method 2 1.
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Site D calling Site C 1. Dial 9,15077175662. 2. “9” gets outside line. On some PBXs, an “8” may be used to direct calls to the VOIP, while “9” directs calls to the PSTN. However, some PBX units can be programmed to identify the destination patterns of all calls to be directed to the VOIP.
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Site D calling Site F A voip call from Pierre PBX to extension 7424 on the key telephone system in Lincoln, Nebraska. A. The required entry in the Pierre Outbound Phonebook to facilitate origination of the call, would be 1402263742. The call would be directed to the Lincoln voip’s IP address, 200.2.9.5.
Site F calling Site D A voip call from a Lincoln key extension to extension 3117 on the PBX in Pierre, South Dakota. A. The required entry in the Lincoln Outbound Phonebook to facilitate origination of the call, would be “31”. The string “1615492” would have to be added as a prefix.
E1 Versus T1 Telephony Environments Phonebooks for Series II analog MultiVOIP units (MVP130, MVP130FXS, MVP210, MVP410, MVP810, MVP210SS, MVP410SS, and MVP810SS) can be operated in either an environment of either North American telephony standards (potentially operating with T1 digital MultiVOIPs) or of European telephony standards (potentially operating with E1 digital MultiVOIPs).
MultiVOIP User Guide E1 PhoneBook Configuration Free Calls: One VOIP Site to Another 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.
Local Rate Calls: Within Local Calling Area of Remote VOIP In the second 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.
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Similarly, the VOIP system allows Wren Clothing employees in London and Amsterdam to call anywhere in Paris at local rates; it allows Wren Clothing employees in Paris and London to call anywhere in Amsterdam at local rates. United Kingdom Wren Clothing Co. Wren Clothing Co.
National Rate Calls: Within Nation of Remote VOIP Site In the third use of the VOIP system, the national 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 national calling rates. Again, significant savings are possible.
Similarly, the VOIP system allows Wren Clothing employees in London and Amsterdam to call anywhere in France at French national rates; it allows Wren Clothing employees in Paris and Amsterdam to call anywhere in the United Kingdom at its national rates. United Kingdom Wren Clothing Co.
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E1 Phonebook Configuration MultiVOIP User Guide In order for any VOIP phone call to be made, there must be both an Inbound Phonebook entry and an Outbound Phonebook entry that describe the end-to-end connection. The phone station originating the call must be connected to the VOIP system. The Outbound Phonebook for that VOIP unit must have a destination pattern entry that includes the ‘called’...
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Phonebook configuration screens can be accessed using icons or the sidebar menu. Phonebook Icons Description Phonebook Configuration Inbound Phonebook Entries List Add Inbound Phonebook Entry Edit selected Inbound Phonebook Entry Outbound Phonebook Entries List Add Outbound Phonebook Entry Edit selected Outbound Phonebook Entry...
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Phonebook Pulldown Menu Inbound Phonebook Shortcut Alt + I Phonebook Sidebar Menu Outbound Phonebook Shortcut Alt + O...
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E1 Phonebook Configuration MultiVOIP User Guide 2. The Add/Edit Outbound PhoneBook screen appears. Enter Outbound PhoneBook 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).
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The fields of the Add/Edit Outbound Phone Book screen are described in the table below. Add/Edit Outbound Phone Book: Field Definitions Field Name Values Description Accept Any When checked, “Any Number Number” appears as the value in the Destination Pattern field. The Any Number feature works differently depending on whether or not an external...
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Add/Edit Outbound Phone Book: Field Definitions Field Name Values Description Destination prefixes, Defines the beginning of Pattern area codes, dialing sequences for calls exchanges, that will be connected to line another VOIP in the system. numbers, Numbers beginning with extensions these sequences are diverted from the PTSN and carried on Internet or other IP...
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Add/Edit Outbound Phone Book: Field Definitions (cont’d) Field Name Values Description SIP Fields Use Proxy Select if proxy server is used. Transport TCP or Voip administrator must choose Protocol between UDP and TCP transmission protocols. UDP is a high-speed, low-overhead connectionless protocol where data is transmitted without acknowledgment, guaranteed...
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E1 Phonebook Configuration MultiVOIP User Guide Clicking on the Advanced button brings up the Alternate Routing secondary screen. This feature provides an alternate path for calls if the primary IP network cannot carry the traffic. Often in cases of failure, call traffic is temporarily diverted into the PSTN. However, this feature could also be used to divert traffic to a redundant (backup) unit in case one voip unit fails.
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Alternate Routing Field Definitions Field Values Description Name Alternate n.n.n.n Alternate destination for outbound data traffic where in case of excessive delay in data transmission. Address n= 0-255 Round milliseconds The Round Trip Delay is the criterion for Trip judging when a data pathway is considered Delay blocked.
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E1 Phonebook Configuration MultiVOIP User Guide 4. The Add/Edit Inbound PhoneBook screen appears. Enter Inbound PhoneBook data for your MultiVOIP unit. The fields of the Add/Edit Inbound PhoneBook screen are described in the table below.
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Add/Edit Inbound Phone Book: Field Definitions Field Values Description Name Accept When checked, “Any Number” appears as the value in the Remove Prefix field. Number The Any Number feature of the Inbound Phone Book does not work when an external routing device is used (Gatekeeper for H323 protocol, Proxy for SIP protocol, Registrar for SPP protocol).
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Add/Edit Inbound Phone Book: Field Definitions (cont’d) Field Name Values Description E1 channel number to which Channel 1-30, or the call will be assigned as it Number “Hunting” enters the local telephony equipment (often a local PBX). “Hunting” directs the call to any available channel.
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Add/Edit Inbound Phone Book: Field Definitions (cont’d) Field Name Values Description Forward Phone number or IP address to which calls Destination will be directed. IP address, For H.323 calls, the Forward Destination can phone number, be either a Phone Number of an IP Address. port number, For SIP calls, the Forward Destination can be etc.
Remember that the initial MultiVOIP setup must be done locally or via the built-in Remote Configuration/Command Modem using the MultiVOIP program. However, 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 GUI software program or the MultiVOIP program (in conjunction with the built-in modem).
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MultiVOIP User Guide E1 PhoneBook Configuration France Country Code: 33 Lille Paris: Area 01 Reims Rouen Nantes Strasbourg Lyon Bordeaux Toulouse Marseille...
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E1 Phonebook Configuration MultiVOIP User Guide The Netherlands Country Code: 31 Groningen Leeuwarden Texel 0222 Den Helder 0223 038 Zwolle 0299 Purmerend Beverwijk 0251 Haarlem 023 020 Amsterdam Aalsmeer0297 0294 Weesp Enschede The Hague Arnhem Rotterdam 0118 Middelburg Eindhoven Maastricht...
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An outline of the equipment setup in these three offices is shown below. Wren Clothing Co. London Office Country Code: +44 Area Code: 0208 Digital VoIP -5174 200.2.10.3 -5173 -5172 -5171 979-5170 Wren Clothing Co. Paris Office Country Code: +33 Area Code: 01 Digital VoIP...
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The screen below shows Outbound PhoneBook entries for the VOIP located in the company’s London facility The Inbound PhoneBook for the London VOIP is shown below. 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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MultiVOIP User Guide E1 PhoneBook Configuration The screen below shows Outbound PhoneBook entries for the VOIP located in the company’s Paris facility. The Inbound PhoneBook for the Paris VOIP is shown below.
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E1 Phonebook Configuration MultiVOIP User Guide The screen below shows Outbound PhoneBook entries for the VOIP in the company’s Amsterdam facility. The Inbound PhoneBook for the Amsterdam VOIP is shown below.
Configuring Digital & Analog VOIPs in Same System Analog MultiVOIP units, like the MVP-210/410/810 are compatible with digital MultiVOIP units like the MVP3010. In many cases, digital and analog VOIP units will appear in the same telephony/IP system. In addition to MVP-210/410/810 MultiVOIP units (Series II units), legacy analog VOIP units (Series I units made by MultiTech) may be included in the system, as well.
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The Series I analog VOIP phone book resides in the “Host” VOIP unit at Site B. It applies to both of the Series I analog VOIP units. Each of the Series II analog MultiVOIPs (the MVP210 and the MVP410) requires its own inbound and outbound phonebooks. The MVP3010 digital MultiVOIP requires its own inbound and outbound phonebooks, as well.
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Phone Book for Analog VOIP Host Unit (Site B) (continued) VOIP Dir # IP Address Channel -OR- Destination Pattern 0207 200.2.9.9 (Note 2.) xxxx 0208 200.2.9.9 (Note 2.) xxxx Note 1. The “x” is a wildcard character. Note 2. By specifying “Channel 0,” we instruct the MVP3010 to choose any available data channel to carry the call.
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The Outbound PhoneBook of the MVP3010 is shown below. Outbound Phone Book for MVP3010 Digital VOIP (Site D) Destin. Remov Pattern Prefix Address Prefix 200.2.9.7 901189 901189 101# 200.2.9.8 Note 3. 200.2.9.6 90182 90182 200.2.9.5 90182 200.2.9.5 90182 200.2.9.5 200.2.9.8 Note 3.
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The Inbound PhoneBook of the MVP3010 is shown below. Inbound Phone Book for MVP3010 Digital VOIP (Site D) Remove Channel Prefix Prefix Number 0207 Note 4. Note 5. 0208 Note 4. Note 5. 0207 39883 Note 4. “9” gives PBX station users access to outside line. Note 5.
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Outbound Phone Book for MVP410 Analog VOIP (Site F) Destin. Remove Pattern Prefix Prefix Address 200.2.9.7 01189 0118 101# 200.2.9.8 Note 3. 200.2.9.8 200.2.9.6 0207 200.2.9.9 0208 200.2.9.9 0207 200.2.9.9 Note 3. The pound sign (“#”) is a delimiter separating the VOIP number from the standard telephony phone number.
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Inbound Phonebook for MVP410 Analog VOIP (Site F) Remove Channel Comment Prefix Prefix Number 01822 Calls to Tavistock local PSTN through FXO port (Port #4) at Site F. 0182 Gives remote voip users, access to extensions of key phone system atTavistock office. 0182 Because call is completed at key system, abbreviated dialing (3-...
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Outbound Phone Book for MVP210 Analog VOIP (Site E) Destin. Remove Pattern Prefix Prefix Address 200.2.9.7 01189 0118 101# 200.2.9.8 Note 3. 200.2.9.8 01822 01822 200.2.9.5 0182 200.2.9.5 26374 0207 0207 200.2.9.9 0207 200.2.9.9 Note 3. The pound sign (“#”) is a delimiter separating the VOIP number from the standard telephony phone number.
Inbound Phonebook for MVP210 Analog VOIP (Site E) Remove Channel Prefix Prefix Number Call Completion Summaries Site A calling Site C, Method 1 Dial 101. Hear dial tone from Site B. Dial 9435632. Await completion. Talk. Site A calling Site C, Method 2 Dial 101#9435632 Await completion.
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Site D calling Site C 1. Dial 901189435632. 2. “9” gets outside line. On some PBXs, an “8” may be used to direct calls to the VOIP, while “9” directs calls to the PSTN. However, some PBX units can be programmed to identify the destination patterns of all calls to be directed to the VOIP.
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Site D calling Site F A voip call from Inner London PBX to extension 7424 on the key telephone system in Tavistock, UK. A. The required entry in the London Outbound Phonebook to facilitate origination of the call, would be 90182263742. The call would be directed to the Tavistock voip’s IP address, 200.2.9.5.
Site F calling Site D A voip call from a Tavistock key extension to extension 3117 on the PBX in Inner London. A. The required entry in the Tavistock Outbound Phonebook to facilitate origination of the call, would be “3”. The string 02073988 is added, preceding the “3”.
International Telephony Numbering Plan Resources Due to the expansion of telephone number capacity to accommodate pagers, fax machines, wireless telephony, and other new phone technologies, numbering plans have been changing worldwide. Many new area codes have been established; new service categories have been established (for example, to accommodate GSM, personal numbering, corporate numbering, etc.).
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http://kropla.com/phones.htm http://www.numberplan.org/ http://www.eto.dk/ http://www.eto.dk/ETNS.htm http://www.regtp.de/en/reg_tele/start /fs_05.html Description Guide to international use of modems. National and international numbering plans based on direct input from regulators worldwide. Includes lists of telecom carriers per country. European Telecommunications Office. Primarily concerned with mobile/wireless radiotelephony, GSM, etc.
Operation and Maintenance Although most Operation and Maintenance functions of the software are in the Statistics group of screens, an important summary appears in the System Information of the Configuration screen group. Also, the SIP Server | Endpoint Statistics screen presents statistical information unique to the MVP SS MultiVOIP units.
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SIP Server Endpoint Statistics Parameter Definitions Field Name Values Description Endpoint Identifier for gateway within SIP voip alpha- Name system. Max. length is 33 characters. numeric Status Indicates the SIP server that is server controlling traffic for this endpoint. identifier Max.
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SIP Server Endpoint Statistics Parameter Definitions Field Name Values Description Initiated Indicates how many calls were numeric Call Count initiated by phones connected to this endpoint. Received Indicates how many calls were numeric Call Count received by phones connected to this endpoint.
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MultiVOIP User Guide Operation & Maintenance The illustration below shows the SIP Server Endpoint Statistics screen for an active SIP phone system in web GUI format.
Operation & Maintenance MultiVOIP User Guide System Information screen This screen presents vital system information at a glance. Its primary use is in troubleshooting. This screen is accessible via the Configuration pulldown menu, the Configuration sidebar menu, or by the keyboard shortcut Ctrl + Alt + Y...
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System Information Parameter Definitions Field Name Values Description Boot Indicates the version of the code that nn.nn Version is used at the startup (booting) of the alpha- voip. The boot code version is numeric independent of the software version. Firmware Indicates the version of the nn.nn.nn Version...
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Operation & Maintenance MultiVOIP User Guide The frequency with which the System Information screen is updated is determined by a setting in the Logs screen...
Statistics Screens 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. About Call Progress Accessing Call-Progress Statistics Channel Icons (Main Screen Lower Left) Channel icons are green when data traffic is present, red when idle.
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Operation & Maintenance MultiVOIP User Guide The Call Progress Details Screen...
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Call Progress Details: Field Definitions Field Name Values Description Channel 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 Hours: The length of the call in hours, Minutes: minutes, and seconds (hh:mm:ss).
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Call Progress Details: Field Definitions Field Name Values Description Packet Details Packets Sent integer value The number of data packets sent over the IP network in the course of this call. Packets Rcvd integer value The number of data packets received over the IP network in the course of this call.
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Call Progress Details: Field Definitions (cont’d) From – To Details Gateway alphanumeric Identifier for the VOIP gateway Name (from) string that handled the origination of this call. IP Address x.x.x.x, IP address from which the call (from) where x has a was received.
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Call Progress Details: Field Definitions (cont’d) DTMF/Other Details Field Name Values Prefix specified Displays the dialed digits that Matched dialing digits were matched to a phonebook entry. Outbound 0-9, #, * The digits transmitted by the Digits Sent MultiVOIP to the PBX/telco for this call.
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Call Progress Details: Field Definitions (cont’d) Field Name Values Description Supplementary Services Status Call on Hold alphanumeric Describes held call by its IP address source, location/gateway identifier, and hold duration. Location/gateway identifiers comes from Gateway Name field in Phone Book Configuration screen of remote voip.
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Call Progress Details: Field Definitions (cont’d) Field Name Values Description Call Status fields Call Status hangup, active Shows condition of current call. Call Control Tun, FS + Tun, Displays the H.323 version 4 Status AE, Mux features in use for the selected call.
MultiVOIP User Guide Operation & Maintenance About Logs The Logs Accessing “Statistics: Logs” Pulldown Icon Shortcut Sidebar Ctrl + O...
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Operation & Maintenance MultiVOIP User Guide The Logs Screen...
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Logs Screen Details: Field Definitions Field Name Values Log # column 1 or higher Start Date,Time dd:mm:yyyy column hh:mm:ss Duration column hh:mm:ss Type H.323, SIP, or SPP Status column success or failure IP Direction incoming, outgoing Mode column voice or FAX From column gateway name To column...
Logs Screen Details: Field Definitions (cont’d) Field Name Values Call Details Voice coder G.723, G.729, G.711, etc. Disconnect Values are Reason "Normal" and "Local" disconnection. DTMF Capability inband, out of band Expressions differ slightly for different Call Signaling protocols (H.323, SIP, or SPP).
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Logs Screen Details: Field Definitions (cont’d) Field Name Values Call Details Server Details n.n.n.n for n= 0-255 Packets sent integer value Packets received integer value Packets loss integer value (lost) Bytes sent integer value Bytes received integer value Description When the MultiVOIP is operating in the non-direct mode (with Gatekeeper in H.323 mode;...
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Logs Screen Details: Field Definitions (cont’d) Field Name Values Call Details (cont’d) FROM Details Gateway Name alphanumeric string IP Address x.x.x.x, where x has a range of 0 to 255 Options FEC, SC TO Details Gateway Name alphanumeric string IP Address x.x.x.x, where x has a range of 0 to 255...
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Logs Screen Details: Field Definitions (cont’d) Supplementary Services Info Call Transferred phone number string Call Forwarded phone number string Number of party called in transfer. Number of party called in forwarding.
Operation & Maintenance MultiVOIP User Guide About IP Statistics Accessing IP Statistics Pulldown Icon Shortcut Sidebar Ctrl + P...
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IP Statistics Screen IP Statistics: Field Definitions Field Values Description Name UDP versus TCP Protocol versus Transmission Control Protocol). UDP provides unguaranteed, connectionless transmission of data across an IP network. By contrast, TCP provides reliable, connection-oriented transmission of data.. . (User Datagram...
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IP Statistics: Field Definitions Field Values Description Name UDP versus TCP 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.
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IP Statistics: Field Definitions (cont’d) Field Values Description Name Total Packets Sum of data packets of all types. (cont’d) Received integer Total number of error-laden packets with value received by this VOIP gateway since the Errors last “clearing” or resetting of the counter within the MultiVOIP software.
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IP Statistics: Field Definitions (cont’d) RTP Packets Voice signals are transmitted in Realtime Transport Protocol packets. RTP packets are a type or subset of UDP packets. Transmit integer Number of RTP packets transmitted by value this VOIP gateway since the last “clearing”...
MultiVOIP User Guide Operation & Maintenance About 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.
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Link Management screen Field Definitions Field Name Values Description Monitor Link fields IP Address to a.b.c.d This is the IP address of the target Ping 0-255 endpoint to be pinged. Pings per Test 1-999 This field determines how many pings will be generated by the Start Now command.
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Link Management screen Field Definitions (cont’d) Field Name Values Description Link Status Parameters These fields summarize the results of pinging. IP Address a.b.c.d Target of ping. column 0-255 No. of Pings as listed Number of pings sent to target Sent endpoint.
Operation & Maintenance MultiVOIP User Guide About 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 PhoneBook screen.
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MultiVOIP User Guide Operation & Maintenance...
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Registered Gateway Details: Field Definitions Field Values Description Name Column Headings Description alphanumeric This is a descriptor for a particular voip gateway unit. This descriptor should generally identify the physical location of the unit (e.g., city, building, etc.) and perhaps even its location in an equipment rack.
MultiVOIP User Guide Operation & Maintenance About Alternate Server Statistics Accessing Alternate Server Statistics Pulldown Shortcut Sidebar Ctrl + Alt + 4...
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H.323 Gatekeepers (Statistics, Servers): Field Definitions Field Values Description Name Column Headings IP Address n.n.n.n, The IP address of the gatekeeper. for n = 0-255 Port TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it. GK Name alpha-numeric Identifier for gatekeeper.
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SIP Proxies (Statistics, Servers): Field Definitions Field Values Description Name Column Headings IP Address n.n.n.n, The IP address of the SIP proxy by which the MultiVOIP is governed. for n = 0-255 Port TDMA time slot used for communication between MultiVOIP unit and the SIP Proxy that governs it.
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SPP Registrars (Statistics, Servers): Field Definitions Field Values Description Name Column Headings IP Address n.n.n.n, The IP address of the gatekeeper. for n = 0-255 Port TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it. This field describes the type of gateway as Type Primary,...
MultiVOIP User Guide Operation & Maintenance About 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.
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Packetization Time Screen Packetization rates can be set separately for each channel. The table below presents the ranges and increments for packetization rates. Packetization Ranges and Increments Coder Types Range (in Kbps); {default value} G711, G726, G727 5-120 G723 30-120 G729 10-120 Netcoder...
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MultiVOIP User Guide Operation & Maintenance Once the packetization rate has been set for one channel, it can be copied into other channels.
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 ____ | … ). In this section, we describe the software functions available on this menu.
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MultiVOIP Program Menu (cont’d) Menu Selection Description Download Factory Defaults Select this to return the configuration parameters to the original factory values. Download Firmware Select this to download new versions of firmware as enhancements become available. Download IFM Firmware Select this to download new versions of IFM firmware as enhancements become available.
“Downloading” here refers to transferring program files from the PC to the nonvolatile “flash” memory of the MultiVOIP. Such transfers are made via the PC’s serial port. This can be understood as a “download” from the perspective of the MultiVOIP unit. When new versions of the MultiVoip software become available, they will be posted on MultiTech’s web or FTP sites.
Date and Time Setup The dialog box below allows you to set the time and date indicators of the MultiVOIP system. Obtaining Updated Firmware Generally, updated firmware must be downloaded from the MultiTech web/FTP site to the user’s PC before it can be downloaded from that PC to the MultiVOIP.
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Operation & Maintenance MultiVOIP User Guide If you conduct a search, for example, on the word “MultiVoip,” you will be directed to a list of firmware that can be downloaded. 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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MultiVOIP User Guide Operation & Maintenance Once the updated firmware has been located, it can be downloaded from the web/ftp site using normal PC/Windows procedures. While the next 3 screens below pertain to the MVP3010, similar screens will appear for any MultiVOIP model described in this manual.
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Operation & Maintenance MultiVOIP User Guide Generally, the firmware file will be a self-extracting compressed file (with .zip extension), which must be expanded (decompressed, or “unzipped”) on the user’s PC in a user-specified directory.
Implementing a Software Upgrade MultiVOIP software can be upgraded locally using a single command at the MultiVOIP Windows GUI, 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 GUI, firmware and factory default settings can also be transferred from controller PC to MultiVOIP piecemeal using separate commands.
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 MultiTech factory. Upgrade Software implements both the Download Firmware command and the Download Factory Defaults command.
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3. If a password has been established, the Password Verification screen will appear. Type in the password and click OK. 4. The MultiVOIP ___- Firmware screen appears saying “MultiVOIP [model number] is up. Reboot to Download Firmware?” Click OK to download the firmware. The “Boot”...
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5. The program will locate the firmware “.bin” file in the MultiVOIP directory. Highlight the correct (newest) “.bin” file and click Open. 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.
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 invoke the Download Factory Defaults command, go to Start | Programs | MVP____ x.xx | Download Factory Defaults.
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3. If a password has been established, the Password Verification screen will appear. Type in the password and click OK. 4. The MVP____- Firmware screen appears saying “MultiVOIP [model number] is up. Reboot to Download Firmware?” Click OK to download the factory defaults. The “Boot”...
5. After the PC gets a response from the MultiVOIP, the Dialog – IP Parameters screen will appear. The user should verify that the correct IP parameter values are listed on the screen and revise them if necessary. Then click OK. 6.
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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).
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6. 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.) 7. The main MultiVOIP Configuration screen will appear. Progress bars can be seen at the bottom of the screen while files are being copied.
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8. Then a completion screen entitled IFM Test will appear. Click OK. 9. The MultiVOIP will reboot itself. When the reboot is complete, the MultiVOIP Configuration screen will close. 10. The IFM firmware downloading process is complete.
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.
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Operation & Maintenance MultiVOIP User Guide 3. To download the user defaults, go to Start | Programs | MultiVOIP xxx | Download User Defaults...
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4. A confirmation screen will appear indicating that this action will entail rebooting the MultiVOIP. Click OK. 5. Progress bars will appear during the file transfer process. 5. When the file transfer process is complete, the Dialog-- IP Parameters screen will appear. 6.
Setting a Password (Windows GUI) 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.
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4. The Password screen will appear. If you intend to use the FTP Server function that is built into the MultiVOIP, enter a user name. (A User Name is not needed to access the local Windows GUI, the web browser GUI, or the commands in the Program group.) Type your password in the Password field of the Password screen.
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Operation & Maintenance MultiVOIP User Guide 6. After the password has been set, the user will be required to enter the password to gain access to the web browser GUI 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.
Setting a Password (Web Browser GUI) Setting a password is optional when using the MultiVOIP web browser GUI. Only one password can be assigned and it works for all MultiVOIP software functions (Windows GUI, web browser GUI, FTP server, and all Program menu commands, e.g., Upgrade Software – only the FTP Server function requires a User Name in addition to the password).
Un-Installing the MultiVOIP Software 1. To un-install the MultiVOIP configuration software, go to Start | Programs and locate the MultiVOIP entry. Select Uninstall MVP____ vx.xx (versions may vary). 2. Two confirmation screens will appear. Click Yes and OK when you are certain you want to continue with the uninstallation process.
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MultiVOIP User Guide Operation & Maintenance 3. A special warning message similar to that shown below may appear for the MultiVOIP software’s “.bin” file. Click Yes. 4. A completion screen will appear. Click Finish.
Upgrading Software As noted earlier (see the section Implementing a Software Upgrade above), 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.
MultiVOIP User Guide Operation & Maintenance FTP Server File Transfers (“Downloads”) MultiTech has built an FTP server into the MultiVOIP unit. Therefore, file transfers from the controller PC to the voip unit can be done using an FTP client program or even using a browser (e.g., Internet Explorer, Netscape, or FireFox, used in conjunction with Windows Explorer).
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To transfer files using the FTP server functionality in the MultiVOIP, follow these directions. 1. Establish Network Connection and IP Addresses. Both the controller PC and the MultiVOIP unit(s) must be connected to the same IP network. An IP address must be assigned for each. IP Address of Control PC IP Address of voip unit #1 IP address of voip unit #n...
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MultiVOIP User Guide Operation & Maintenance 3. Install FTP Client Program or Use Substitute. You should install an FTP client program on the controller PC. FTP file transfers can be done using a web browser (e.g., Netscape or Internet Explorer) in conjunction with a local Windows browser a (e.g., Windows Explorer), but this approach is somewhat clumsy (it requires use of two application programs rather than one) and it limits downloading to only one VOIP...
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5. Identify Files to be Updated. Determine which files you want to update. Six types of files can be updated using the FTP feature. In some cases, the file to be transferred will have “Ftp” as the part of its filename just before the suffix (or extension).
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MultiVOIP User Guide Operation & Maintenance 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://”...
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Operation & Maintenance MultiVOIP User Guide 7. Log In. Use the User Name and password established in item #2 above. The login screens will differ depending on whether the FTP file transfer is to be done with a web browser (see first screen below) or with an FTP client program (see second screen below).
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8A. Download with Web Browser. 8A1. 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).
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Operation & Maintenance MultiVOIP User Guide You may be asked to confirm the overwriting of files on the MultiVOIP. Do so. File transfer between PC and voip will look like transfer within voip directories.
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8B. Download with FTP Client Program. 8B1. 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).
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Operation & Maintenance MultiVOIP User Guide Some FTP client programs are more graphically oriented (see previous screen), while others (like the “WS-FTP” client) are more text oriented. 9. Verify Transfer. The files transferred will appear in the directory of the MultiVOIP. 10.
Web Browser Interface You can control the MultiVOIP unit with a graphic user interface (GUI) based on the common web browser platform. Qualifying browsers are InternetExplorer6, Netscape6, and Mozilla FireFox 1.0. MultiVOIP Web Browser GUI Overview Function Configuration Prerequisite Browser Version Requirement Java Requirement Video Usability...
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Operation & Maintenance MultiVOIP User Guide The initial configuration step of assigning the voip unit an IP address must still be done locally using the Windows GUI. However, all additional configuration can be done via the web GUI. The content and organization of the web GUI is directly parallel to the Windows GUI.
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MultiVOIP User Guide Operation & Maintenance The web GUI, however, cannot perform logging in the same direct mode done in the Windows GUI. However, when the web GUI is used, logging can be done by email (SMTP). The graphic layout of the web GUI is also somewhat larger-scale than that of the Windows GUI.
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Operation & Maintenance MultiVOIP User Guide To install the Java program, go to the Java directory on the MultiVOIP product CD. Double-click on the EXE file to begin the installation. Follow the instructions on the Install Shield screens.
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MultiVOIP User Guide Operation & Maintenance During the installation, you must specify which browser you’ll use in the Select Browsers screen. When installation is complete, the Java program becomes accessible in your Start | Programs menu (Java resources are readily available via the web).
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Operation & Maintenance MultiVOIP User Guide After the Java program has been installed, you can access the MultiVOIP using the web browser GUI. Close the MultiVOIP Windows GUI. 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 GUI or for the MultiVOIP’s FTP Server function.
SysLog Server Functions MultiTech 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”).
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Operation & Maintenance MultiVOIP User Guide Before a SysLog client program is used, the SysLog functionality must be enabled within the MultiVOIP in the Logs menu under Configuration. The IP Address used will be that of the MultiVOIP itself. In the Port field, entered by default, is the standard (‘well-known’) logical port, 514.
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MultiVOIP User Guide Operation & Maintenance Configuring the SysLog Client Program. Configure the SysLog client program for your own needs. In various SysLog client programs, you can define where log messages will be saved/archived, opt for interaction with an SNMP system (like MultiVoipManager), set the content and format of log messages, determine disk space allocation limits for log messages, and establish a hierarchy for the seriousness of messages (normal, alert, critical, emergency, etc.).
Limited Warranty Multi-Tech Systems, Inc. (“MTS”) warrants that its products will be free from defects in material or workmanship for a period of two years from the date of purchase, or if proof of purchase is not provided, two years from date of shipment.
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Warranty, Service, & Tech Support MultiVOIP User Guide For out-of-warranty repair charges, go to www. multitech.com/documents/warranties Extended two-year overnight replacement service agreements are available for selected products. Please call MTS at (888) 288-5470, extension 5308, or visit our web site at www.multitech.com/programs/orc for details on rates and coverages.
Technical Support Multi-Tech Systems has an excellent staff of technical support personnel available to help you get the most out of your Multi-Tech product. If you have any questions about the operation of this unit, or experience difficulty during installation you can contact Tech Support via the...
EMC, Safety, and R&TTE Directive Compliance The CE mark is affixed to this product to confirm compliance with the following European Community Directives: Council Directive 89/336/EEC of 3 May 1989 on the approximation of the laws of Member States relating to electromagnetic compatibility, Council Directive 73/23/EEC of 19 February 1973 on the harmonization of the laws of Member States relating to electrical equipment designed for use within certain voltage limits,...
6. If trouble is experienced with this equipment (the model of which is indicated below), please contact Multi-Tech Systems, Inc. at the address shown below for details of how to have repairs made. If the equipment is causing harm to the network, the telephone company...
7. 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. 8. Manufacturer:...
Regulatory Information MultiVOIP User Guide WEEE Statement (Waste Electrical and Electronic Equipment) July, 2005 The WEEE directive places an obligation on EU-based manufacturers, distributors, retailers and importers to take-back electronics products at the end of their useful life. A sister Directive, ROHS (Restriction of Hazardous Substances) compliments the WEEE Directive by banning the presence of specific hazardous substances in the products at the design phase.
Appendix A: Cable Pinouts Command Cable RJ-45 Connector To Command Port Connector 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.
ISDN BRI RJ-45 Pinout Information The S/T interface uses an 8-conductor modular cable terminated with an 8-pin RJ-45 plug. An 8-pin RJ-45 jack located on the terminal is used to connect the terminal to the DSL (Digital Subscriber Loops) using this modular cable.
ISDN Interfaces: “ST” and “U” The MVP410ST and MVP810ST are ISDN-BRI voip units that use an S/T outlet interface. You will need an NT1 device to connect these units to any network equipment that has the “U” ISDN interface. In the UK, and in many European countries, the telco supplies an NT1 device for ISDN-BRI service.
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.
INDEX 802.1p Priority Levels ... 104, 105 abbreviated dialing, inter-office E1... 243 T1... 207 Accept Any Number (inbound) E1... 259 T1... 219 Accept Any Number (outbound) field E1... 253 T1... 212 Accept Registrations for domains field SIP Server Configuration parameters... 196 Accept Registrations for IP addresses field SIP Server Configuration...
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... 116 Fax Enable field... 116 fax machine connecting to analog voip (MVP210-SS) ... 76 connecting to MVP210-SS voip . 75 connecting to voip (MVP- 410SS/810SS) ... 71, 72 FAX Parameters... 116 fax tones, output level... 117 Fax Volume field ...
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Inter Digit Timer (dialing) ... 134 Message Waiting Indication ... 134 No Response Timer ... 134 Regeneration (dialing) ... 134 Tone Detection... 134 FXO interface(MVP210-SS) uses of ... 75 FXO interface(MVP-410SS/810SS) uses of ... 71 FXO Parameter fields Caller ID enable... 135 Caller ID Type ...
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Gateway Name (Ethernet/IP Parameters) field... 106 Gateway Number, From Details (RADIUS Attributes) field... 192 Gateway Number, From Details (SMTP logs) field ... 167 Gateway Number, To Details (RADIUS Attributes) field... 192 Gateway Number, To Details (SMTP logs) field... 167 Generate Current Reversal FXS Loop Start ...
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DID-DPO... 148 E&M ... 145 FXO ... 134 FXS Loop Start ... 130 Intercept Tone (Regional Params) and Offhook Alert (Voice/Fax Params) ... 155 Intercept Tone and required Interface & Voice/Fax settings ... 155 Interface field (DID-DPO)... 147 Interface field (E&M) ... 144 interface parameters, accessing...
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MVP-410SS/810SS ... 70 Keep Alive (Timers; NAT/STUN) 186 key system connecting to analog voip (MVP- 410SS/810SS) ... 71 connecting to MVP210-SS ... 75 Knowledge Base (online, for MultiVOIPs) ... 8 Last button Logs (Statistics) screen ... 301 Last Error (Link Management) field ...
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Options (caller) ... 304 Outbound digits ... 304 Outbound Digits Recvd ... 302 Outbound Digits Sent ... 302 Packets lost ... 303 Packets recvd ... 303 Packets sent... 303 Packets Sent... 301 Server Details... 303 Start Date, Time... 301 Status ... 301 Supplementary Services info ...
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Outbound Digits Received (call progress) field ... 296 Outbound Digits Received (statistics, logs) field... 302 Outbound Digits Received(SMTP logs) field... 166 Outbound Digits Sent (call progress) field... 296 Outbound Digits Sent (RADIUS Attributes) field... 191 Outbound Digits Sent (SMTP logs) field...
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... 380 T1/E1 connector... 381 Voice/FAX connector ... 381 placement of voip Quick Start Instructions ...30 polarity sensitivity DID lines and (MVP210-SS) ...76 DID lines and (MVP- 410SS/810SS) ...72 pop-ups allowing with Web GUI ... 111 Port (Contact Info) SIP Server Predefined Endpoint Parameters...
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