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Lincoln 1999 Continental Automobile User Manual

1. Tabs a PimayPoy sO AkenaePow OO AnematePow2 oO Registrer Options FT Append SIP Proxy Domada Name in Uses 1D Keep Alive in sec 50 Defaut Subscriber ARA Behind ProwNAT davies Detak Username Prom NAT Device Parameters Public IP Addiege 0 0 0 0 Password Posy Poli om A _ cal feo Re AegistialionT me Figure 3 10 Signaling Protocols Multi Tech Systems Inc 25 Chapter 3 Software Installation Actions e Configure your chosen Call Signal type o H 323 not supported by SS and FX series Use Fast Start may be needed for third party vendor compatibility Signaling Port default is 1720 Register with Gatekeeper needed if the VOIP is to be controlled by a gatekeeper Allow Incoming Calls Through Gatekeeper Only Gatekeeper RAS Parameters e Enter parameters for Primary and any Alternate Gatekeepers e RAS TTL Value Time To Live in seconds e Gatekeeper Discovery Polling Interval time between attempts connecting to gatekeepers e Use Online Alternate Gatekeeper List H 323 Version 4 Options detailed descriptions of these can be found in Chapter 4 o SIP Signaling Port default is 5060 Use SIP Proxy enable to work with a proxy server Allow Incoming Calls Through SIP Proxy Only SIP Proxy Parameters Enter information for Primary and any Alternate Proxy servers Append SIP Proxy Domain Name in User ID Enter User Name and Password Re Registration Ti
2. Table is continued on next page Multi Tech Systems Inc 41 Chapter 4 Configuring your VOIP FXS Loop Start Interface Parameter Definitions continued Field Name Values Description Flash Hook Options fields Generation Not applicable to FXS interface Detection Range for Min and Max 50 1500 milliseconds For a received flash hook to be regarded as such by the MultiVOIP its duration must fall between the minimum and maximum values given here Pass Through Enable Y N When enabled this parameter creates an open audio path through the MultiVOIP If the Pass Through feature is enabled the AutoCall feature must be enabled for this VOIP channel in the Voice Fax Parameters screen Caller ID fields Type Bellcore The MultiVOIP currently supports only one implementation of Caller ID That implementation is Bellcore type 1 with Caller ID placed between the first and second rings of the call Enable Y N Caller ID information is a description of the remote calling party received by the called party The description has three parts name of caller phone number of caller and time of call The time of call portion is always generated by the receiving MultiVOIP unit on FXS channel based on its date and time setup The forms of the Caller Name and Caller Phone Number differ depending on the IP transmission protocol used H 323
3. Multi Tech Systems Inc 42 Chapter 4 Configuring your VOIP Message Waiting Message Waiting Indication is a feature that displays an audible or visible indication that a message available A type of message waiting is sounding a special dial tone called stutter dial tone lighting a light or indicator on the phone When a user enables a subscription for message waiting indication a subscription is made with the Voice Mail Server VMS for that particular event Whenever the Voice Mail Server finds a change in the state of a corresponding mailbox or some event happens e g when a new voice message is recorded or a message is deleted then the VMS server sends a notification to the gateway lts indication to the user is a flashing LED or sounding a stutter dial tone The message waiting feature is active when the Use SIP Proxy option is selected on the Call Signaling SIP screen a Primary Proxy IP address is entered in the SIP Proxy Parameters Primary Proxy field the Voice Mail Server Domain Name or IP Address is entered in the SIP Voice Mail Server Parameters Group and the Interface Type is set to FXS Loop start Then the FXS Options Group becomes active The Message Waiting Indication options are None Light or Stutter Dial Tone Message Waiting Indication None v Password Figure 4 5 Message Waiting To receive messages from the VMS Voice Mail Server System the subscription needs to be enabled and the voic
4. File Type File Names Description firmware bin file mvpt1Ftp bin This is the MultiVOIP firmware file Only one file of this type will be in the directory factory defaults fdefFtp cnf This file contains factory default settings for user changeable configuration parameters Only one file of this type will be in the directory CAS file fxo_loopFtp cas These telephony files are for Channel Associated Signaling em_winkFtp cas The directory contains many CAS files some labeled for r2_brazilFtp cas specific functionality others for countries or regions where r2_chinaFtp cas certain attributes are standard Any CAS file used must first be renamed to CASFILE CAS inbound phonebook InPhBk tmr This file updates the inbound phonebook in the MultiVOIP unit outbound phonebook OutPhBk tmr This file updates the outbound phonebook in the MultiVOIP unit 6 Contact MultiVOIP FTP Server You must make contact with the FTP Server in the VOIP using either a web browser or FTP client program Enter the IP address of the MultiVOIP s FTP Server If you are using a browser the address must be preceded by ftp otherwise you ll reach the web interface within the MultiVOIP unit P ftp 192 168 2 200 Microsoft Internet Explorer gt File Edit View Go F a gt iT y Back Figure 6 36 FTP address 7 Log In Use the User Name and password established in item 2 above
5. Chapter 4 Configuring your VOIP SIP Server Predefined Endpoint Parameters In this screen you will specify the VOIP gateways that will depend on the MVP SS unit either as their primary SIP server if the MVP SS is used in Stand Alone mode as set in the SIP Server Configuration screen or as their backup SIP server if the MVP SS is used in Survivability mode as set in the SIP Server Configuration screen The main screen for Predefined Endpoints is a list If you click on function buttons to Add or Edit entries in this list of endpoints a secondary screen will appear and allow you to add new endpoints or edit existing endpoint entries When your work with the list is complete click Save SIP Server Endpcents warehouse Opoa Add Predetined Endporn Endpor Name z E OK Edi Predetined Endpori cone Ervdpoe Name myehcuse Regra Type C Stance Dynamis OK Passvand otra Corraci Intommance Ade Regra Type Stare C Dinamo ema Cornact Ironia Addes 23 45 981 Pot Resegitiahon Interval 3500 Por 50 Reregerabon Irtervat Figure 4 16 Endpoint parameters SIP Server Predefined Endpoints Parameter Definitions Field Name Values Description Endpoint Name name Identifier for gateway within SIP VOIP system Max length is 33 characters Password password This password is for authentication of gateway to SIP server Registration Type Static Static registrations are fixed and th
6. Country Selection for Built In Modem not applicable to MVP210 country name MultiVOIP units operating with the X 06 software release and above include a built in modem The administrator can dial into this modem to configure the MultiVOIP unit remotely The country name values in this field set telephony parameters that allow the modem to work in the listed country This value may be different than the Country Region value For example a user may need to choose Europe as the Country Region value but Denmark as the Country Selection for Built In Modem value User Defined Tones fields Type column alphanumeric name Name of supervisory tone pair Cannot be same as name of any standard tone pair Frequency 1 Freq in Hertz Lower frequency of pair Frequency 2 Freq in Hertz Higher frequency of pair Gain 1 3dB to 31dB Amplification factor of lower frequency of pair and mute This applies to any supervisory tones that the MultiVOIP outputs as setting audio to the FXS FXS or E amp M port Default Mute Gain 2 3dB to 31dB Amplification factor of higher frequency of pair and mute This applies to any supervisory tones that the MultiVOIP outputs as setting audio to the FXS FXO or E amp M port Default Mute Cadence n n n n On off pattern of tone durations used to denote supervisory tones ms On Off four integer time specified by user Supervisory
7. 5 Download IFM Firmware 2 Download User Defaults PL set Password 3 Uninstall 5 Upgrade Software Internet Explorer MSN Windows Media Play Windows Messenger 6 O lt P Tour Windows XP All Programs Log OF Turn Off Computer Figure 6 12 Program menu Several basic software functions are accessible from the MultiVOIP software menu as shown below MultiVOIP Program Menu Menu Selection Description Configuration Select this to enter the Configuration program where values for IP telephony and other parameters are set Configuration Port Setup Select this to access the COM Port Setup screen of the MultiVOIP Configuration program Date and Time Setup Select this for access to set calendar clock used for data logging 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 The Interface Module IFM is the telephony interface for analog MultiVOIP units There is one IFM for each channel of the MultiVOIP unit For each channel the IFM handles the analog signals to and from the attached telephone PBX or CO line Download User Defaults To be used after a full set of parameter values values specified by the user have been saved u
8. High 6 dB Low 8 gt aB Fax Volume 35 d Defaut Jitter Value 400 ms Duration i00 ms FRF 1 gt Help DTMF Que Df Band Fixed Duration y ila Out Of Band Mode Rfc2833 y r Coder Advanced Features Manual Dai Silence Compression Selected Coder 6 71 1 6 729 V Echo Cancellation Max id ij TF Forward Error Correction Auto Call OffHook Alert Auto Call OffHook Alert Auto Call y I Generate Local Dial Tone Phone Number Dynamic Jitter Buffer Minimum Jitter Value 80 ms Maximum Jitter alue 300 ms Optimization Factor 7 Automatic Disconnection IV Jitter Value 350 ms V Consecutive Packets Lost 30 Iv Call Duration fiso secs Network Disconnection 300 secs Configurable Payload Type DTMF RFC 2833 36 RTP Redundancy 104 FRF11 Fax 101 Modem Relay 105 Fax Bypass 102 Modem Bypass 103 Figure 3 8 Voice amp Fax settings Multi Tech Systems Inc 21 Chapter 3 Software Installation Actions e Select Channel o Choose channel parameters Set the Fax parameters to meet your needs e Set Max Baud Rate to match fax machine 2400 to 14400 bps e Fax Volume should not be changed as it may impair function e Jitter Value affects the time for packet reassembly e Mode Select T 38 or FRF 11 Modem Relay Enable allows modem traffic through the VOIP system Adjusting Voice Gain and DTMF should not be done as it may adversely affect quality Select a Coder or allow Automatic nego
9. VOIP this feature gives notice to remote VOIPs involved in calls Notification goes to the remote VOIP administrator not to individual phone stations When the home VOIP is the caller a plain English descriptor will be sent to the remote VOIP identifying the channel over which the call is being originated for example Calling Party Omaha Sales Office Line 2 If that VOIP channel is dedicated to a certain individual the descriptor could say that as well for example Calling Party Harold Smith in Omaha When the home VOIP receives a call from any remote VOIP the home VOIP sends a status message back to that caller This message confirms that the home VOIP s phone channel is either busy or ringing or that a connection has been made for example Busy Party Omaha Sales Office Line 2 These messages appear in the Statistics Call Progress screen of the remote VOIP Note that Supplementary Services parameters are applied on a channel by channel basis However once you have established a set of supplementary parameters for a particular channel you can apply this entire set of parameters to another channel by using the Copy Channel button and its dialog box to copy a set of Supplementary Services parameters to all channels select Copy to All and click Copy Supplementary Services Parameters Select Channel Channel 1 z Call Transfer Call Name Identification V Enable M Enable Boo Allowed Name Type e yl I
10. ua E MuliTech System InPhBk tmr into browser window 3 Multi Tech Systen a Lhsstrs A MultiWOIP 24 i LogsMaster log A MultiVOIP 24 Mfc42 dll Pi New file from PC SB NUM DIP 24 will overwrite old flle E MultivOIP 30 2 MukwoIP 30 on voip unit Y MultvolP 5 91 Y MultivOIP 6 0 A MultiVoip300C E Mtvoip 4 2 mupz400 id UD Te KULEA Figure 6 38 Drag and drop file e You may be asked to confirm the overwriting of files on the MultiVOIP Do so A This folder already contains a file called mwptl ftp bin gt Would you like to replace the existing file Ey 0 bytes 0 bytes Tuesday January 01 1980 12 00 PM with this one Ey 1 79 MB 1 881 364 bytes Monday September 09 2002 7 41 PM Yes to All No Cancel Figure 6 39 Overwrite confirmation e File transfer between PC and VOIP will look like transfer within VOIP directories Copying ES A Copying fdefftp cnf From C Program Files Multi Tech Systems Multi OIP 2410 4 03 to Figure 6 40 Copy screen Multi Tech Systems Inc 130 Chapter 6 Using the Software Download with FTP Client Program e 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 e I
11. 80 G727 24 Kbps 80 v NetCoder 8 Kbps 80 G727 32 16 Kbps 80 v NetCoder 8 8Kbps B0 G727 32 24 Kbps 80 NetCoder 9 6Kbps ep v G727 32 Kbps 80 v Figure 6 11 Packetization time Packetization rates can be set separately for each channel The table below presents the ranges and increments for packetization rates The final column represents recommended settings based on the most common found when operating with third party devices Packetization Ranges and Increments Recommendations Coder Types Range in Kbps default Increments in Kbps Setting in ms G711 G726 G727 5 120 5 5 20 G723 30 120 30 30 30 G729 10 120 10 10 20 NetCoder 20 120 20 20 20 Once the packetization rate has been set for one channel it can be copied into other channels by using the Copy Channel button on the Packetization Time screen Simply click the boxes next to the channels you wish to copy the settings for Multi Tech Systems Inc 115 Chapter 6 Using the Software 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 x xx In this section we describe the software functions available on this menu Configuration e Configuration A Configuration Port Setup A Date and Time Setup ES Download Factory Defaults 5 Download Firmware
12. Cancel i Figure 6 21 Download IFM firmware 5 The Boot LED on the front panel of the MultiVOIP will come on 6 The software will search for an IFM firmware file to use to upgrade the system if the file found represents firmware newer than that already installed on the MultiVOIP or if you want to overwrite the same version of firmware click Open A a Look in B Muvo eo y acem R File name ngav1 11 ifm Files of type IFM Files ifm y Cancel i Figure 6 22 IFM firmware file 7 The IFM Firmware Download screen will appear Select Copy to All IFMs and click OK Only in very special circumstances would different IFMs in the same VOIP be loaded with different IFM firmware Multi Tech Systems Inc 121 Chapter 6 Using the Software IFM Firmware Download M M M E eS Eee iFM Eg Figure 6 23 IFM firmware download 8 The main MultiVOIP Configuration screen will appear Progress bars can be seen at the bottom of the screen while files are being copied 9 Then a completion screen entitled IFM Test will appear Figure 6 24 IFM test screen Click OK 10 The MultiVOIP will reboot itself When the reboot is complete the MultiVOIP Configuration screen will close 11 The IFM firmware downloading process is complete Multi Tech Systems Inc 122 Chapter 6 Using the Software Setting and Downloading User Defaults The Download User Defaults command allows you
13. IP Parameters J Enable DHCP Ok IP Address 192 168 3 143 a IP Mask 255 25 255 0 i Help Gateway Figure 6 28 Dialog screen 7 Set the IP values per your particular VOIP system Click OK Progress bars will appear as the MultiVOIP reboots itself Multi Tech Systems Inc 123 Chapter 6 Using the Software Setting a Password Windows Interface After a user name has been designated and a password has been set that password is required to gain access to any functionality of the MultiVOIP software Only one user name and password can be assigned to a VOIP unit The user name will be required when communicating with the MultiVOIP via the web browser interface NOTE Record your user name and password in a safe place If the password is lost forgotten or irretrievable the user must contact Multi Tech Tech Support in order to resume use of the MultiVOIP unit 1 The MultiVOIP configuration program must be off when invoking the Set Password command If it is on the command will not work 2 To use the Set Password command go to Start Programs MultiVOIP x xx Set Password 3 You will be prompted to confirm that you want to establish a password which will entail rebooting the MultiVOIP which is done automatically Click OK to proceed with establishing a password 4 The Password screen will appear If you intend to use the FTP Server function that is built into the MultiVOIP enter a u
14. Note 1 Intercept tone is applicable only when the FXS telephony interface has been chosen in the Interface screen and when the AutoCall OffHook Alert field is set to OffHook Alert in the Voice Fax Parameters screen The time allowed for dialing before the automatic calling process begins is set in the OffHook Alert Timer field of the Voice Fax Parameters screen Note 2 Survivability tone indicates a special type of call routing redundancy amp applies to MultiVantage VOIP units only Custom Advisory screen Multi OIP Regional Setup Supervision Tones have been set to default values in Interface Page Standard Tones fields xx This message screen appears whenever the Country field is changed It informs the operator that upon change of the Country field value all User Defined Tones will be deleted Type column dial tone ring tone busy tone unobtainable tone fast busy survivability tone re order tone Type of telephony tone pair for which frequency gain and cadence are being presented Frequency 1 freq in Hertz Lower frequency of pair Frequency 2 freq in Hertz Higher frequency of pair Gain 1 gain in dB Amplification factor of lower frequency of pair 3dB to 31dB This applies to the dial ring busy and unobtainable tones that the and mute setting MultiVOIP outputs as audio to the FXS FXS or E amp M port Default 16dB Gain 2 gain in dB Amplification f
15. Seattle Chicago system London Birmingham system Possible Description Local rate Rotterdam access all employees Possible Description Local rate London access all employees Possible Description Free Seattle access all employees 7 Repeat steps 2 6 for each inbound phonebook entry When all entries are complete go to step 8 8 Click OK to exit the inbound phonebook screen 9 Click on Save Setup Highlight Save and Reboot Click OK Your starter inbound phonebook configuration is complete Multi Tech Systems Inc 80 Chapter 5 Phonebook Configuration Phone Book Descriptions Outbound Phone Book List Entries Fields in the Details section will differ depending on the protocol H 323 SIP or SPP of the selected list entry to which the details pertain F Outbound Phone Book IP Address Protocol__ Description Altemal 130 192 168 1 130 H 323 21 192 168 2 210 H 323 81 192 168 2 81 H 323 gt Number of Entries 3 Details Remove Prefix Add Add Prefix eu Gatekeeper not used Edit Gateway H 323 ID Gateway Prefi _Delete H 323 Port 1720 Close Transport Protocol E SIP URL Help Round Trip Delay 300 ms Alternate Phone Number Figure 5 1 Outbound Phone Book Multi Tech Systems Inc 81 Chapter 5 Phonebook Configuration Add Edit Outbound Phone Book Add Edit Dutbound Phone Book Phone Number Details Accept Any Number Destination Patte
16. portion is always generated by the receiving MultiVOIP unit on FXS channel based on its date and time setup The forms of the Caller Name and Caller Phone Number differ depending on the IP transmission protocol used H 323 SIP or SPP and upon entries in the phonebook screens of the remote CID generating VOIP unit The CID Name and Number appearing on the phone at the terminating FXS end will come either from a central office switch showing a PSTN phone number or the phonebook of the remote CID sending VOIP unit CID Manipulation Enabled by This is not implemented in the SS series VOIPs default with Caller ID Manipulation is used whenever the user wants to manipulate Caller ID enable the Caller ID before sending it to the remote end Caller ID above Manipulation is activated on the Interface Screen By enabling Caller Disable ID option you can set manipulation to Transparent User CID Prefix Suffix or Prefix and Suffix Caller ID Manipulation is a feature where the Caller ID detected from the PSTN line can be changed and then sent to the remote side over IP CID Mode Transparent The MultiVOIP is not allowed to modify the caller ID info and then User CID send it to the PSTN side It only allows it to detect the caller ID Prefix from the PSTN line modify it and then send them via IP to the Suffix remote end point Transparent the CID received from PSTN will be sent out as such without any
17. Chapter 2 Installing and Cabling the MultiVOIP Jumper Configurations fenlarged ___ ss Esda otad aici 1or5 Upper Circuit Card i z MVP810 only oyo For DID 2or6 Da Interface type ool Ch5 Ch6 Ch7 ChB n7 i307 Jumpers 5 8 cca dal T For non DID Y ies e Aor _ Interface type fe EELEE EF pF o Main Circuit Gard 9 T MVP 410 810 Ed For channels using the DID e interface the jumper must Ch1 eee Ch3 Ch4 e not straddle across the ey e cross hatched area between a y the jumper posts rely For channels using any non DID interface it is acceptable that the Jumpers 1 4 jumper straddles across the E dep 7 ef E A spree FX 7 cross hatched area between k the jumper posts Figure 2 8 MVP 410 810 Channel Jumper Settings 5 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 altogether but that is inadvisable because the jumper might be needed later if a different telephony interface is used for that VOIP channel 6 Slide the main circuit card back into the MultiVOIP chassis and replace the three screws Multi Tech Systems Inc 15 Chapter 3 Software Installation Introduction Configuring software for your MultiVOIP entails three tasks Loading the software
18. a PBX extensions or analog telco lines are FXO e DID DPO E amp M info only if E amp M used Interface parameters e Type 1 5 e 2 or 4 wires e Dial Tone or Wink Country code Regional parameters Email address for VOIP optional SMTP parameters Reminder Be sure to Save Setup after entering configuration values Multi Tech Systems Inc 31 Chapter 4 Configuring your VOIP Ethernet IP This section covers the Ethernet settings needed for the MultiVOIP unit In each field enter the values that fit the network to which the MultiVOIP will be connected to For many of the settings the default values will work best try these settings first unless you know you definitely need to change a parameter Ethernet IP Parameters Ethernet Parameters W Packet Prioritization 802 1p Frame Type TYPE 802 1p Parameters Priority Call Control 6 Voice v VolP Media 3 Excellent Effort v Lanca Others O Best Effort v VLAN ID 1 d Help IP Parameters Gateway Name Multi olP Enable DHCP Diff Serv Parameters i Call Control PHB IP Address 192 168 3 143 WolP Media PHB E li IP Mask 255 255 255 0 ETP Server V Enable Gateway Enable SRY DNS Server IP Address TDM Routing Option Use TDM Routing For Intra Gateway calls Figure 4 1 Network parameters The Ethernet IP Parameters fields are described in the tables and text p
19. the date and time may be incorrect If the battery fails the board must be sent back to Multi Tech Systems for replacement Warning There is danger of explosion if the battery is incorrectly replaced Safety Warnings Telecom 1 Never install telephone wiring during a lightning storm 2 Never install a telephone jack in wet locations unless the jack is specifically designed for wet locations 3 This product is to be used with UL and UL listed computers 4 Never touch un insulated telephone wires or terminals unless the telephone line has been disconnected at the network interface Use caution when installing or modifying telephone lines 6 Avoid using a telephone other than a cordless type during an electrical storm There may be a remote risk of electrical shock from lightning 7 Do not use a telephone in the vicinity of a gas leak 8 To reduce the risk of fire use only a UL listed 26 AWG or larger telecommunication line cord al Unpacking Your MultiVOIP When unpacking your MultiVOIP check to see that all of the items are included in the box For the various MultiVOIP models the contents of the box will be different If any box contents are missing contact Multi Tech Tech Support at 1 800 972 2439 MVP210 models content list MVP210 DB9 to RJ45 cable Power transformer Power cord Printed Cabling Guide e Product CD MVP410 810 models content list MVP410 or MVP810 DB9 to DB25 cable Mounting brackets
20. 1 400 Frequency 2 1000 amp K User Defined Tones Frequency2 Cadence secs On Off Disconnect 400 1000 0 400 2 000 0 000 0 000 16 16 Add Cadence 1 400 ad Here you can add the tones LL tor FXO User defined tones can be used to supervise Edit Supervision the answering and disconnection of calls Delete Country Selection For Builtln Modem United States US Figure 4 19 Regional parameters Multi Tech Systems Inc 62 Chapter 4 Configuring your VOIP Regional Parameter Definitions Field Name Values Description Country Region USA Name of a country or region that uses a certain set of tone pairs for dial tone ring Japan tone busy tone unobtainable tone fast busy tone survivability tone tone UK heard briefly 2 seconds after going off hook denoting survivable mode of VOIP unit re order tone a tone pattern indicating the need for the user to hang up the phone and intercept tone a tone that warns an a party that has gone off hook but has not begun dialing within a prescribed time that an automatic emergency or attendant number will be called the automatic call can be used to direct an attendant s attention to a disabled or distressed caller allowing an appropriate response to be made In some cases the tone pair scheme denoted by a country name may also be used outside of that country The Custom option button assures that any tone pairing scheme worldwide can be accommodated
21. Add Edit Inbound Phone Book Channel 2 Remove Prefix Add Prefix Forward Addr Phone Book Configuration al Inbound Phone Book is used Gateway Name Anoka Whse VP3 OR Add Edit Inbound Phone Book CID Number 423 O Use as default entry xc CID Name Anoka Whse VP3 ple meets Add Profa ESNY ee e Channel Number cram JF A Description ERRE DR f Description in Add Edit Inbound Phone Book is blank Figure 5 26 Caller ID example 3 Figure 5 27 VOIP Caller ID Case 3 Call through telco central office without standard CID enters SPP VOIP system lt CID Flow Call is received Call originates here here at 4 51pm Oct 3 phone of Nigel Thurston 763 555 9401 Display shows CID Number 423 CID Name Anoka Whse VP3 Time Stamp Date 10 03 Time 4 51pm r Phone Book Configuration Gateway Name Anoka Whse VP3 Eo Parameters Inbound Phone Book Channel 2 Remove Prefix Add Prefix Forward Addr 423 748 In x 06 release when SIP protocol is used CID Name field will duplicate value in CID Number field Gatekeeper RAS Para Figure 5 27 Caller ID example 4 Figure 5 28 VOIP Caller ID Case 4 Remote FXS call on H 323 VOIP system Multi Tech Systems Inc 100 Chapter 5 Phonebook Configuration CID Flow Call originates here at 6 17pm Nov 15 Call is received Generating Central Office MOP
22. 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 The user must specify the IP address of the alternate route for each destination pattern entry in the Outbound Phonebook Add Prefix IP Address Description SIF C H 323 e Alternate Routing Cancel OK Alternate IP Address OF E 0 A Round Trip Delay 1300 ms Help ainsport Protocol Figure 5 3 Advanced button Alternate Routing Field Definitions Field Name Values Description Alternate IP n n n n Alternate destination for outbound data traffic in case of excessive delay in data Address transmission Round Trip Default is The Round Trip Delay is the criterion for judging when a data pathway is Delay 300 considered blocked When the delay exceeds the threshold specified here the milliseconds data stream will be diverted to the alternate destination specified as the Alternate IP Address The Alternate Routing function facilitates PSTN Failover protection that is it allows you to re route VOIP calls automatically over the PSTN if the VOIP system fails The MultiVOIP can be programmed to respond to excessive delays in the transmission of v
23. Comections Channels 1 4 Bottom MVP410 810 Channels 5 8 Top MVP810 Only E amp M FXS FXO o Ethernet Connection A j i 1 F 1 AN IL gt le lt Command Port Connection Figure 2 5 Cabling for MVP 410 810 2 Connect the MultiVOIP to a PC by using a DB 25 male to DB 9 female cable Plug the DB 25 end of the cable into the Command port of the MultiVOIP and the other end into the PC serial port See Figure 2 5 3 Connect a network cable to the ETHERNET 10BASET connector on the back of the MultiVOIP Connect the other end of the cable to your network a Foran FXS or FXO connection SS and FX series FXS Examples analog phone fax machine FXO Examples PBX extension POTS line from central office Connect one end of an RJ 11 phone cord to the Channel 1 FXS FXO connector on the back of the MultiVOIP Connect the other end to the device or phone jack b For an E amp M connection E amp 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 amp M Type in the E amp M Options group of the Interface dialog box is the same as the E amp M trunk type supported by the telephone switch See Appendix B for an E amp M cabling pin out c Fora DID connection DID Examples DID fax system or DID voice phone lines Connect one end of an RJ 11 phone cord to the
24. Count set number of rings before MultiVOIP answers o No Response Timer set time to attempt call before abandoning o Supervision Button for call answering and disconnection settings Answer Fields Current Reversal use current reversal to answer Answer Delay Answer Delay Timer in seconds Tone Detection allow tone sequence to disconnect Available Tones Answer Tones shows current selection from Available Tones Disconnect Fields Current Reversal use current reversal to disconnect Current Loss loss of current will trigger disconnect Current Loss Timer time after current loss to disconnect in milliseconds Silence Detection Enable use silence detection to disconnect Silence Detection Type one way or two way Silence Timer time of silence needed to trigger disconnect in seconds DTMF Tone use tones to disconnect Disconnect Tone Sequence select tone pairs to use for disconnecting Tone Detection disconnect from termination of tone Available Tones Disconnect Tones shows current selection from Available Tones e E amp M Options not supported by SS and FX series o Type o Mode 2 wire or 4 wire o Signal Dial Tone or Wink o Wink Timer range is 100 to 350 milliseconds default is 250 o No Response Timer time in seconds after which an FXO call would be disconnected o Disconnect on Call Progress Tone allows disconnection when PBX issues call progress tone o Pass Through Enable creates an op
25. Direct mode has IP addresses assigned to the gateways The table below describes all fields in the general SPP Call Signaling screen The SS models do not support SPP SPP Parameters Mode Client v M General Options Signaling Port f 0000 Retransmission in ms io Max Retransmission 3 m Client Options IP Address Part Primary Registrar 0 0 0 0 110000 Altemate Registrar 1 0 0 0 D0 110000 Altemate Registrar 2 0 0 0 110000 Polling Interval f 80 secs gocoeseoesoesesecceoesocoesosoeosososesesooossecoesesosoesesesg ARA E AA Proxy NAT Device Parameters Public IP Address 0 0 0 0 OK Cancel Help Figure 4 17 SPP call signaling Multi Tech Systems Inc 59 Chapter 4 Configuring your VOIP SPP Call Signaling Parameter Definitions Field Name Values Description Mode Direct In direct mode all VOIP gateways have static IP addresses assigned to them Client or In registrar client mode one VOIP gateway serves as registrar and all other Registrar gateways being its clients point to that registrar The registrar assigns IP addresses dynamically General Options Port port The UDP port on which data transmission will occur Each client VOIP has its own port If two client VOIPs are both behind the same firewall then they must have different ports assigned to them If there are two clients and each is behind a different firewall th
26. IP Address Protocol Description 1612 200 2 10 3 H 323 1651 200 2 10 3 H 323 1763 200 2 10 3 H 323 1952 200 2 10 3 H 323 Number of Entries 4 etails Remove Prefix 1612 Add Prefix 9612 not used Minneapolis St Paul Minneapolis N Suburbs Minneapolis S Suburbs Add Edit Delete Close Help H 323 Port 1720 Round Trip Delay 30 ms Figure 5 9 Baltimore example The entries in the Minneapolis VOIP s Inbound Phonebook match the Outbound Phonebook entries of the Baltimore VOIP as shown below Inbound Phone Book 1612 9612 Not Used 1651 9651 Not Used 1763 3 Not Used 17637175 5 Not Used 1952 9952 Not Used Number of Entries 5 Details e e Channel No Hunting Description Local calls to Minneapolis Registration Options Subscription Options Figure 5 10 Minneapolis example To call the Minneapolis St Paul area a Baltimore employee must dial eleven digits In this case we are assuming that the Baltimore PBX does not require an 8 or 9 to seize an outside phone line If a Baltimore employee dials any phone number in the 612 area code the call will automatically be handled by the company s VOIP system Upon receiving such a call the Minneapolis VOIP will remove the digits 1612 But before the suburban Minneapolis VOIP can complete the call to the PSTN of the Minneapolis local calling area it must dial 9 to get an outside line from the PBX and then a comm
27. L Grounding Screw Telco POTS Line Figure 2 6 MVP410 810 connections for ground amp modem Ensure that the unit is properly connected to earth ground by verifying that it is reliably grounded when mounted within a rack This can be accomplished by connecting a grounding wire between the chassis grounding screw see Figure 2 6 and a metallic object that will provide an electrical ground Turn on power to the MultiVOIP by placing the ON OFF switch on the back panel to the ON position Wait for the Boot LED on the MultiVOIP to go off before proceeding This may take a few minutes 10 Proceed to Chapter 3 to load the MultiVOIP software For DID channels only For any channel on which you are using the DID interface type you must change the jumper on the MultiVOIP circuit card DID is not supported on the SS or FX models 1 2 4 Disconnect power Unplug the AC power cord from the wall outlet or from the receptacle on the MultiVOIP unit Using a 1 Phillips driver remove the three screws at back of unit that attach the main circuit card to the chassis of the MultiVOIP Remove these Figure 2 7 MVP 410 810 Rear Screw Locations Pull the main circuit card out about 5 inches the power connection to the board prevents it from being removed entirely from the chassis Identify the channels on which the DID interface will be used Multi Tech Systems Inc 14
28. 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 Link Management Monitor Link IP Address to Ping 0 Pings per Test CS Response Timeout 1000 32 Time Interval between Tests 0 min Ping Size in Bytes ms Start Now Clear Link Status IP Address lt gt Exit Figure 6 6 Link management Link Management screen Field Definitions Field Name Values Description Monitor Link fields IP Address to Ping n n n n This is the IP address of the target endpoint to be pinged Pings per Test 1 999 This field determines how many pings will be generated by the Start Now command Response Timeout 500 5000 The duration after which a ping will be considered to have failed milliseconds Ping Size in Bytes 32 128 bytes This field determines how long or large the ping will be Timer Interval between Pings 0 or 30 6000 minutes This field determines how long of a wait there is between one ping and the next Start Now command button Initiates pinging Clear command button Erases ping parameters in Monitor Link field group and restores default values Link Status Parameters These fields summarize the results of pinging IP Address column n n n n Target of ping No of Pings Sent as lis
29. Pulse DTMF Determines whether digits generated and sent out will be pulse tones or DTMF Inter Digit Timer 1 to 10 seconds This is the length of time that the MultiVOIP will wait between digits When the time expires the MultiVOIP will look in the phonebook for the number entered Default 2 Message Waiting Not applicable to FXO interface Indication Inter Digit 50 to 20 000 The length of time between the outputting of DTMF digits Regeneration Time milliseconds Default 100 ms FXO Options FXO Ring Count 1 99 Number of rings required before the MultiVOIP answers the incoming call No Response 1 65535 Length of time before call connection attempt is abandoned Timer in seconds Flash Hook Options fields Generation 50 1500 Length of flash hook that will be generated and sent out when the milliseconds remote end initiates a flash hook and it is regenerated locally Default 600 ms Detection Range Not applicable to FXO Caller ID fields Caller ID Type Bellcore The MultiVOIP currently supports only one implementation of Caller ID That implementation is Bellcore type 1 with caller ID placed between the first and second rings of the call Caller ID enable Y N Caller ID information is a description of the remote calling party received by the called party The description has three parts name of caller phone number of caller and time of call The time of call
30. SIP or SPP and upon entries in the phonebook screens of the remote CID generating VOIP unit The CID Name and Number appearing on the phone at the terminating FXS end will come either from a central office switch showing a PSTN phone number or the phonebook of the remote CID sending VOIP unit CID Manipulation Enabled by default with Caller ID enable above This is not implemented in the SS series VOIPs Caller ID Manipulation is used whenever the user wants to manipulate the Caller ID before sending it to the remote end Caller ID Manipulation is activated on the Interface Screen By enabling Caller Disable ID option you can set manipulation to Transparent User CID Prefix Suffix or Prefix and Suffix Caller ID Manipulation is a feature where the Caller ID detected from the PSTN line can be changed and then sent to the remote side over IP CID Mode Transparent The MultiVOIP is not allowed to modify the caller ID info and then User CID send it to the PSTN side It only allows it to detect the caller ID Prefix from the PSTN line modify it and then send them via IP to the Suffix remote end point Transparent the CID received from PSTN will be sent out as such without any manipulation User CID the CID received from PSTN will be replaced by this User CID value Prefix the CID received from PSTN will be prefixed with this value Suffix the CID received from PSTN will be suffixed with this value
31. Select Add Entry o Accept Any Number may be selected to allow unmatched destinations an alternative o Enter the number necessary to get out from the PBX system followed by the calling code of the destination in the Destination Pattern field o Enter the PBX access digit same number as needed to get out of the PBX system in the Remove Prefix field o Any digits that need to be added should be put in the Add Prefix field Enter the IP address of the call destination add a Description if you like o Select a Protocol type SS models use SIP only FX models do not support H 323 O For H 323 e Enter Gateway settings For SIP e Select Transport Protocol Proxy and URL if needed For SPP e Enter Registrar settings if needed o The Advanced Button will allow an Alternate IP Address to be entered for outbound traffic e Select Inbound Phone Book o Select Add Entry Accept Any Number for inbound traffic does not work when external routing devices are used Enter any access digits followed by the local calling code in the Remove Prefix field Enter any digits needed to access an outside line in the Add Prefix field Select Hunting in the Channel Number field to have the VOIP use the next available channel Add a description if you like Call Forward may be set up details available in Chapter 5 Select Registration Option e Repeat the Phone Book steps for any additional entries needed 000000 0 Save amp Reboot Any time that you change sett
32. Seven FUMGUOMS mesa A 134 Appendix A Cable Pin QU S siria 135 Appendix B TCP UDP Port ASSIQnNMents cccsseceeeesseeeeeseneeeenseseeeenssseeeeessseaeseseseaeeessseeeeesesseaesnsseaees 136 Appendix C Installation Instructions for MVP428 Upgrade Card oococcccnnnnnnoncccconecnnnnnnnancnnnnnnnnnnnnnnncnnes 137 Appendix D Regulatory INTOFMAtION lt 140 Appendix E Waste Electrical and Electronic Equipment WEEE Statement cccsssseeessseeeeees 142 Appendix F C ROHS HT TS Substance Concentration 0 0 ccccccccesseeceeceeeeeeeeeeeeeeee seen eeseeeseaeeeseeeseeeeeenenees 143 INDEX oras e Nai 144 Multi Tech Systems Inc 5 Chapter 1 Description and Specifications Introduction The MultiVOIP gateways MVP210 MVP410 and MVP810 provide toll free voice and fax communications over the Internet or an Intranet By integrating voice and fax into your existing data network you can realize substantial savings on inter office long distance toll charges MultiVOIP gateways connect directly to phones fax machines key systems PSTN lines or a PBX to provide real time toll quality voice connections to any office on your VOIP network The SS series models only support the SIP protocol through the FXS FXO interface with SIP survivability as well Figure 1 2 MVP 210 Chassis The MultiVOIP model MVP210 is a two channel unit the model MVP410 is a four channel and the MVP810 is an
33. Supplementary Services features derive from the H 450 standard which brings to the VOIP telephony functionality once only available with PSTN or PBX telephony Even though the H 450 standard refers only to H 323 Supplementary Services are still applicable to the SIP and SPP VOIP protocols Of the features implemented under Supplementary Services three are very closely related Call Transfer Call Hold and Call Waiting Call Name Identification is similar but not identical to the premium PSTN feature commonly known as Caller ID Call Transfer Call Transfer allows one party to re connect the party with whom they have been speaking to a third party The first party is disconnected when the third party becomes connected Feature is used by a programmable phone keypad sequence for example 7 Call Hold Call Hold allows one party to maintain an idle non talking connection with another party while receiving another call Call Waiting while initiating another call Call Transfer or while performing some other call management function Feature is used by a programmable phone keypad sequence for example 7 Call Waiting Call Waiting notifies an engaged caller of an incoming call and allows them to receive a call from a third party while the party with whom they have been speaking is put on hold Feature is used by a programmable phone keypad sequence for example 7 Call Name Identification When enabled for a given VOIP unit the home
34. The SMTP Parameters screen is shown below SMTP Parameters geseccneceeenssescsnssacenccneey Y Requires Authentication Login Name MultWolP Cancel Password Help Mail Server lP ddress 192 168 1 5 Port Number 25 Select Fields nikte Mail Type C Text e HTML Subject Reply To Address Recipient Address Multi olP multitech com Mail Criteria Number of Records i 00 V Number of Days l4 Figure 4 20 SMTP parameters Multi Tech Systems Inc 65 Chapter 4 Configuring your VOIP SMTP Parameters Definitions Field Name Values Description Enable SMTP Y N In order to send log reports by email this box must be checked However to enable SMTP functionality you must also select SMTP in the Logs screen Requires Y N If this checkbox is checked the MultiVOIP will send Authentication Authentication information to the SMTP server The authentication information indicates whether or not the email sender has permission to use the SMTP server Login Name alpha numeric This is the User Name for the MultiVOIP unit s email account Password alpha numeric Login password for MultiVOIP unit s email account Mail Server IP n n n n This is the mail server s IP address This mail server must be accessible on Address the IP network to which the MultiVOIP is connected Port Number 25 25 is a standard port number for SMTP Mail Type text or html Mail type in which log repor
35. The login screens will differ depending on whether the FTP file transfer is to be done with a web browser shown below or with an FTP client program varies gt Could not login to the FTP server anonymously Enter a user name and password and press Login to continue FTP Server 192 168 2 200 User Name m Password After you login you can return to this FTP server easily by adding it to your Favorites List J ean Anonymously I Save Password Cancel Figure 6 37 FTP log in 8 Use Download Downloading can be done with a web browser or with an FTP client program Multi Tech Systems Inc 129 Chapter 6 Using the Software Download with Web Browser e 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 e Drag and drop files from the local Windows browser e g Windows Explorer to the web browser loring Multi OIP 2410 4 03 Exploting MukiVOIP 2410 403 EA fip voip1 192 168 2 2007 Microsoft Intermet Explorer File Edit View Go Favorites Tools Help al 9 de Back Forward Up Cut Copy ddress 1 C Program Files Multi Tech Systems Multi OIF a a a 2 x Name casfile cas factdefenf H323 pdl mvptl ftp bin Java Web Start a fks_loopFtp cas J Microsoft Office h323 pdl ul
36. Tone not applicable for Wink signaling Dialing Options Inter Digit Timer 1 10 seconds This is the length of time that the MultiVOIP will wait between digits When the time expires the MultiVOIP will look in the phonebook for the number entered Default 2 Message Waiting Light or None Allows MultiVOIP to pass mode code sequences between Avaya Indication Magix PBXs to turn on and off the message waiting light on a PBX extension phone Mode codes 53 PBX extension 3 turns message light on 53 PBX extension 3 turns message light off Signals to turn message waiting lights on off are not sent to phones connected directly to the MultiVOIP on FXS channels not to other non Avaya Magix PBX phone stations on the VOIP network Inter Digit 50 20000 The length of time between the outputting of DTMF digits Regeneration milliseconds Default 100 ms Timer Flash Hook Options fields Generation 50 1500 Length of flash hook that will be generated and sent out when the milliseconds remote end initiates a flash hook and it is regenerated locally Default 600 ms Detection Range for Min and For a received flash hook to be regarded as such by the MultiVOIP its Max 50 1500 duration must fall between the minimum and maximum values given milliseconds here Multi Tech Systems Inc 50 E amp M Interface Types Chapter 4 Configuring your VOIP There are five different types of
37. and screws Power cord Printed Cabling Guide Product CD Multi Tech Systems Inc 9 Chapter 2 Installing and Cabling the MultiVOIP Rack Mounting Instructions for MVP410 amp MVP810 The MultiVOIPs can be mounted in an industry standard EIA 19 inch rack enclosure 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 If installing the unit in a closed or multi unit enclosure ensure adequate airflow within the rack so that the maximum recommended ambient temperature is not exceeded Ensure that the unit is properly connected to earth ground by verifying that it is reliably grounded when mounted within a rack If a power strip is used ensure that the power strip provides adequate grounding of the attached apparatus When mounting the equipment in the rack make sure mechanical loading is even to avoid a hazardous condition The rack used should safely support the combined weight of all the equipment it supports Ensure that the mains supply circuit is capable of handling the load of the equipment See the power label on the equipment for load requirements full specifications for MultiVOIP models are presented in chapter 1 of this manual This equipment should only be installed by
38. configurable minimum and maximum values An Optimization Factor adjustment controls how quickly the length of the Jitter Buffer is increased when jitter increases on the network The length of the jitter buffer directly affects the voice delay between MultiVOIP gateways Minimum Jitter Value 60 to 400 The minimum dynamic jitter buffer of 60 milliseconds is the minimum delay ms that would be acceptable over a low jitter network Default 150 ms Maximum Jitter Value 60 to 400 The maximum dynamic jitter buffer of 400 milliseconds is the maximum delay ms tolerable over a high jitter network Default 300 ms Optimization Factor 0 to 12 The Optimization Factor determines how quickly the length of the Dynamic Jitter Buffer is changed based on actual jitter encountered on the network Selecting the minimum value of 0 means low voice delay is desired but increases the possibility of jitter induced voice quality problems Selecting the maximum value of 12 means highest voice quality under jitter conditions is desired at the cost of increased voice delay Default 7 Auto Disconnect Automatic The Automatic Disconnection group provides four options which can be Disconnection used singly or in any combination Jitter Value 1 65535 The Jitter Value defines the average inter arrival packet deviation in milliseconds before the call is automatically disconnected The default is 300 milliseconds A higher value means voice transmiss
39. 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 Euro National Call Example London Birmingham system On London PBX 9 is used to get an outside line Answer 9 is prefix to be added by Euro International Call Example Rotterdam Bordeaux system On Rotterdam PBX 9 is used to get an outside line Answer 9 is prefix to be added by local Rotterdam VOIP local Seattle VOIP 5 In the Channel Number field enter Hunting A hunting value means the VOIP unit will assign the call to the first available channel If desired specific channels can be assigned to specific incoming calls i e to any set of calls received with a particular incoming dialing pattern local London VOIP 6 In the Description field it is useful to describe the ultimate destination of the calls For example in a New York City VOIP system incoming calls to Manhattan office might describe a phonebook entry as might the descriptor incoming calls to NYC local calling area The description should make the routing of calls easy to understand For this 40 characters are the maximum Euro International Call Example Rotterdam Bordeaux system North America Euro National Call Example Long Distance Example
40. not for H 323 Whenever you interoperate with older MultiVOIP products i e earlier than release x 11 for backward compatibility make sure to configure the payload type values to default ones which match the values of older MultiVOIP s Multi Tech Systems Inc 39 Interface Chapter 4 Configuring your VOIP The Telephony Interface parameters are set individually for each channel and include the line types as well as some specific situational settings for those that need them The kinds of parameters for which values must be chosen depend on the type of telephony supervisory signaling or interface used FXO E amp M etc Here you will find the various parameters grouped and organized by interface type Note that the SS and FX models only support FXS FXO In each field enter the values that fit your particular setup 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 To copy a set of Interface parameters to all channels select Copy to All and click Copy The screen below shows more options available than are actually used for clarity Your settings will determine what fields are available The SS series of MultiVOIPs do not support Caller ID Manipulation Interface Parameters Select Channel Channel 1 X FXS Options FXS Ring Count V Current Loss E Y
41. properly qualified service personnel Only connect like circuits connect SELV Secondary Extra Low Voltage circuits to SELV circuits and TN Telecommunications Network circuits to TN circuits 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 2 1 and then secure unit to rack rails by the brackets as shown in Figure 2 2 Because equipment racks vary screws for rack rail mounting are not provided Follow the instructions of the rack manufacturer and use screws that fit 1 Position the right rack mounting bracket on the MultiVOIP using the two vertical mounting screw holes Secure the bracket to the MultiVOIP using the two screws provided Position the left rack mounting bracket on the MultiVOIP using the two vertical mounting screw holes Secure the bracket to the MultiVOIP using the two screws provided Remove feet 4 from the MultiVOIP unit Mount the MultiVOIP in the rack enclosure per the rack manufacture s mounting procedure oun WN Rack Mounting Setup Figure 2 2 Attaching MultiVOIP to Rack Rail MVP410 amp MVP810 Multi Tech Systems Inc 10 Chapter 2 Installing and Cabling the MultiVOIP Cabling Procedure for MVP210 Cabling involves connecting the MultiVOIP to your LAN and tele
42. screen lets you pick which aspects will be included in the email log reports Custom Fields Definitions Field Description Field Description Select All Log report to Start Date Date and time the phone call began include all fields shown Time Channel Data channel carrying call Call Mode Voice or fax Number Duration Length of call Packets Total packets received in call Received Packets Sent Total packets sent in call Bytes Total bytes received in call Received Bytes Sent Total bytes sent in call Coder Voice Coder Compression Rate used for call will be listed in log Packets Lost Packets lost in call Prefix When selected the phonebook prefix Matched matched in processing the call will be listed in log Outbound The DTMF dialing digits received by this Call Type Indicates the Call Signaling protocol Digits gateway from the remote gateway used for the call H 323 SIP or SPP Received presuming that DTMF is set to Out of Band Call Status Successful or unsuccessful DTMF Indicates whether the DTMF dialing Call Direction Indicates call s originating party Capability digits are carried Inband or Out of Server The IP address of the traffic control Band The corresponding field values Details server if any being used whether an differ for the 3 different VOIP H 323 gatekeeper a SIP proxy or an protocols SPP registrar gateway will be displayed For H 323
43. server typically a PBX When in Survivability mode the unit is a backup SIP server In Stand Alone mode the MVP SS functions as a primary SIP server for other gateways In this mode the MVP SS operate to technical advantage with smart SIP phones Such smart SIP phones can choose the SIP server under which they operate and consequently can be controlled by either the SIP based PBX or by the MVP SS Survivability Register One of two status check packets is sent to the main SIP Proxy servers to which the Status Options MVP SS serves as a backup This packet determines whether the MVP SS needs to Check take over SIP server functions or stay in its normal backup mode Options and Register are two distinct SIP request methods The Options method solicits information but does not set up a connection The Register method conveys information about a user s location to the SIP server The Register method may entail more data overhead than the Options method If both of these methods are supported by your SIP server it is OK to use either one If only one is supported use the supported method Registrar Options Allow Y N If undefined registrations are allowed then gateways other than those listed in the Undefined Predefined Endpoints list can register with the MVP SS unit as it functions in its SIP Registrations server mode If undefined registrations are not allowed then incoming registrations will be allowe
44. split data into packets called datagrams However TCP includes extra headers in the datagram to enable retransmission of lost packets and reassembly of packets into their correct order if they arrive out of order UDP does not provide this Lost UDP packets are irretrievable that is out of order UDP packets cannot be reconstituted in their proper order Despite these obvious disadvantages UDP packets can be transmitted much faster than TCP packets as much as three times faster In certain applications like audio and video data transmission the need for high speed outweighs the need for verified data integrity Sound or pictures often remain intelligible despite a certain amount of lost or disordered data packets which comes through as static Multi Tech Systems Inc 108 Chapter 6 Using the Software IP Statistics Field Definitions Field Name Values Description IP Address n n n n IP address of the MultiVOIP For an IP address to be displayed here the MultiVOIP must have DHCP enabled Its IP address in such a case is assigned by the DHCP server Clear button Clears packet tallies from memory Total Packets Sum of data packets of all types Transmitted integer Total number of packets transmitted by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received integer Total number of packets received by this
45. that of the Windows interface except for logging These will be addressed in the following chapters Front Panel LEDs Active LEDs On both the MVP410 and MVP810 models there are eight sets of channel operation LEDs However on the MVP410 only the lower four sets of channel operation LEDs are functional On the MVP810 all eight sets are functional Figure 1 3 MVP410 810 LEDs Similarly the MVP210 models have the general operation indicator LEDs and two sets of channel operation LEDs Multi lec Mutti Syslens Velec Fax over IP Nebwer lt s Elere Voics Fex 1 Vo ve Fiax 2 o O 0000 0000 XMI 22V XSGRS7 XMI RSV KEG ESG Figure 1 4 MVP210 LEDs Front Panel LED Definitions LED Description 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 FDX LED indicates whether Ethernet connection is half duplex or full duplex FDX and in half duplex mode indicates occurrence of data collisions LED is on constantly for full duplex mode LED is off constantly for half duplex mode When operating in half duplex mode the LED will flash during Ethernet data collisions LNK Link Activity LED This LED is lit if Ethernet connection has been made It is off when the link is down i e when no Ethernet c
46. the MultiVOIP unit Using a 1 Phillips driver remove the screw at bottom of unit near the back cover end that attaches the main circuit card to the chassis of the MVP210 Pull the main circuit card out about half way Identify the channels on which the DID interface will be used as config ured for DID Interface i H I4 y Ch 1 Jumper Block as shipped for non DID interfaces Ch 2 Jumper as configured for DID Interface Figure 2 4 MVP210 Channel Jumper Settings 5 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 altogether but that is inadvisable because the jumper might be needed later if a different telephony interface is used for that VOIP channel Slide the main circuit card back into the MultiVOIP chassis and replace the screw at the bottom of the unit Multi Tech Systems Inc 12 Chapter 2 Installing and Cabling the MultiVOIP Cabling Procedure for MVP 410 810 Cabling involves connecting the MultiVOIP to your LAN and telephone equipment 1 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 the figure below The E amp M jacks are not present on the SS and FX models Command Modem connector for remote configuration Voice Fax Channel
47. the specified Unzip folder press the Unzip button X Run WinZip Unzip to folder C Acme Inc MVP3000 firm Browse pz M Overwrite files without prompting About Help PPE Figure 6 14 Extract files Multi Tech Systems Inc 117 Chapter 6 Using the Software Implementing a Software Upgrade MultiVOIP software can be upgraded locally using a single command at the MultiVOIP Windows interface namely Upgrade Software This command downloads firmware including the H 323 stack and factory default settings from the controller PC to the MultiVOIP unit When using the MultiVOIP Windows interface firmware and factory default settings can also be transferred from controller PC to MultiVOIP piecemeal using separate commands When using the MultiVOIP web browser interface to control configure the VOIP remotely upgrading of software must be done on a piecemeal basis using the FTP Server function of the MultiVOIP unit When performing a software upgrade whether from the Windows interface or web browser interface follow these steps in order 1 Identify Current Firmware Version 2 Download Firmware 3 Download Factory Defaults When upgrading firmware the software commands Download Firmware and Download Factory Defaults must be implemented in order else the upgrade is incomplete Identifying Current Firmware Version Before implementing a MultiVOIP firmware upgrade be sure to verify the firmware ver
48. this field can display Out here if the call is handled through that of Band or Inband For SIP it can server display either Out of Band RFC2833 or Out of Band SIP INFO to indicate the out of band condition or Inband to indicate the in band condition For SPP it can display Out of Band RFC2833 or Inband Disconnect Indicates whether the call was Outbound The dialing digits sent by this gateway Reason disconnected simply because the Digits Sent to the remote gateway presuming that desired conversation was done or some DTMF is set to Out of Band other irregular cause occasioned disconnection e g a technical error or failure Values are Normal and Local disconnection From Details To Details Gateway Originating gateway Gateway Completing or answering gateway Number Name IP Address IP address where call originated IP Address IP address where call was completed or answered Descript Identifier of site where call originated Descript Identifier of site where call was completed or answered Options When selected log will not Silence Options When selected log will not use Silence Compression and Forward Error Compression and Forward Error Correction by call originator Correction by party answering call Multi Tech Systems Inc 67 Chapter 4 Configuring your VOIP RADIUS In general RADIUS is concerned with authentication authorization and accounting The MultiVOIP supports the accounting and authentic
49. to configure the VOIP protocol in detail 8 Click OK to exit from the Add Edit Outbound Phonebook screen Multi Tech Systems Inc 79 Chapter 5 Phonebook Configuration 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 even though it will not be North America Long Distance Example Seattle Chicago system Seattle is area 206 Chicago employees must dial 81 before dialing any Seattle number on the VOIP system Answer 1206 is prefix to be removed by local Seattle VOIP Euro National Call Example London Birmingham system Inner London is 0207 area Birmingham employees must dial 9 before dialing any London number on the VOIP system Answer 0207 is prefix to be removed by local London VOIP Euro International Call Example Rotterdam Bordeaux system Rotterdam is country code 31 city code 010 Bordeaux employees must dial 903110 before dialing any Rotterdam number on the VOIP system Answer 03110 is prefix to be removed by local Rotterdam VOIP 4 In the Add Prefix field enter any digits that
50. to maintain a known working configuration that is specific to your VOIP system You can then experiment with alterations or improvements to the configurations confident that a working configuration can be restored if necessary 1 Before you can use the Download User Defaults command you must first save a set of configuration parameters by using the Save Setup command in the sidebar menu of the MultiVOIP software Figure 6 25 Save 8 Reboot 2 Before the setup configuration is saved you will be prompted to save the setup as the User Default Configuration Select the checkbox and click OK A user default file will be created The MultiVOIP unit will reboot itself 3 To download the user defaults go to Start Programs MultiVOIP x xx Download User Defaults 4 A confirmation screen will appear indicating that this action will entail rebooting the MultiVOIP Multi OIP 410 xj Downloading User Defaults will Reboot the MultivOIP 410 Do you want to continue cms Figure 6 26 Confirmation screen Click OK 5 Progress bars will appear during the file transfer process by A 0c o E Downloading Configuration Packets Sent 2 Acks received 2 Errors 0 BENE Figure 6 27 Progress bars 6 When the file transfer process is complete the Dialog IP Parameters screen will appear T IP Parameters Diff Serv Parameters 34 Call Control PHB Frame Type TYPE I y gt 46 VoIP Media PHB
51. to request the user to disconnect the equipment Users should ensure for their own protection that the electrical ground connections of the power utility telephone lines and internal metallic water pipe system if present are connected together This precaution may be particularly important in rural areas Caution Users should not attempt to make such connections themselves but should contact the appropriate electric inspection authority or electrician as appropriate Multi Tech Systems Inc 140 Appendix D Regulatory Information FCC Part 68 Telecom This equipment complies with part 68 of the Federal Communications Commission Rules On the outside surface of this equipment is a label that contains among other information the FCC registration number This information must be provided to the telephone company As indicated below the suitable jack Universal Service Order Code connecting arrangement for this equipment is shown If applicable the facility interface codes FIC and service order codes SOC are shown An FCC compliant telephone cord and modular plug is provided with this equipment This equipment is designed to be connected to the telephone network or premises wiring using a compatible modular jack that is Part 68 compliant See installation instructions for details If this equipment causes harm to the telephone network the telephone company will notify you in advance that temporary discontinuance of service may b
52. to the Direct Inward Dial DID telephony interface type are shown in the figure below and described in the table that follows The DID interface allows one phone line to direct incoming calls to any one of several extensions without a switchboard operator Of course one DID line can handle only one call at a time The parameters described here pertain to the customer premises side of the DID connection DID DPO dial pulse originating the network side of the DID connection DID DPT dial pulse terminating is not supported The SS and FX models do not support DID Interface Type DID DPO v Dialing Options Regeneration i Inter Digit Timer 2 secs a Inter Digit Regeneration Timer 100 ms DID Options Start Modes Wink Start Wink Timer 200 Figure 4 12 DID parameters DID Interface Parameter Definitions Field Name Values Description Interface DID DPO Enables the customer premises side of DID functionality DID Options MultiVOIP s use of DID applies only for incoming DID calls The Start Mode used by the MultiVOIP must match that used by the originating telephony equipment else DID calls cannot be completed Start Modes Immediate Start For Immediate Start the VOIP detects the off hook condition initiated Wink Start by the telco central office call and becomes ready to receive dial digits Delay Dial immediately For Wink Start the VOIP detects the off hook condition Then the VOIP reverses battery p
53. 0 secs Configurable Payload Type DTMF RFC 2833 96 RTP Redundancy 104 FRF11 Fax 101 Modem Relay 105 Fax Bypass 102 Modem Bypass 103 Figure 4 2 Voice Fax parameters The Voice FAX Parameters settings are described in the tables below Multi Tech Systems Inc 35 Chapter 4 Configuring your VOIP Voice Fax Parameter Definitions Field Name Values Description Default When this button is clicked all Voice FAX parameters are set to their default values Select Channel 1 2 210 Channel to be configured is selected here 1 4 410 1 8 810 Copy Channel Copies the Voice FAX attributes of one channel to another channel Attributes can be copied to multiple channels or all channels at once Voice Gain Signal amplification or attenuation in dB Input Gain 31dB to Modifies audio level entering voice channel before it is sent over the network to the 31dB remote VOIP The default amp recommended value is 0 dB Output Gain 31dB to Modifies audio level being output to the device attached to the voice channel The 31dB default and recommended value is 0 dB DTMF Gain The DTMF Gain Dual Tone Multi Frequency controls the volume level of the DTMF tones sent out for Touch Tone dialing DTMF Gain 3dB to Default value 4 dB Not to be changed except under supervision of Multi Tech High Tones 31dB 8 Technical Support mute DTMF Gain Low 3dB to Default value 7 dB Not to be changed except
54. 2833 or Out of Band SIP INFO to Signaling indicate the out of band condition or Inband to indicate the in band protocols condition For SPP it can display Out of Band RFC2833 or Inband Outbound Digits 0 9 The digits sent by MultiVOIP to PBX telco that were acknowledged as Received having been received by the remote VOIP gateway Outbound Digits 0 9 The digits transmitted by the MultiVOIP to the PBX telco for this call Sent Server Details n n n n When the MultiVOIP is operating in the non direct mode with Gatekeeper in H 323 mode with proxy in SIP mode or in the client server configuration of SPP mode this field shows the IP address of the server that is directing IP phone traffic Packets sent integer value Number of data packets sent over the IP network in the course of this call Packets received integer value Number of data packets received over the IP network in the course of this call Packets lost integer value Number of voice packets from this call that were lost after being received from the IP network Bytes sent integer value Number of bytes of data sent over the IP network in the course of this call Bytes received integer value Number of bytes of data received over the IP network in the course of this call FROM Details Gateway Name alphanumeric Identifier for the VOIP gateway that originated this call IP Address n n
55. 410 models MVP810 or MVP410 428 Operating Voltage Current External transformer 100 240 VAC 100 240 VAC dd Sone S 3A 5V 1 2 0 6 A 1 2 0 6 A Mains Frequencies 50 60 Hz 50 60 Hz 50 60 Hz Power Consumption 19 watts 29 watts 46 watts 1 4 H 1 75 Hx 1 75 Hx 6 2 W x 17 4 Wx 17 4 W x 9 Dx 8 5 D 8 5 D Mechanical Dimensions 3 6cm H 4 5cm H x 4 5cm H x 15 8cm W x 44 2 cm W x 44 2 cm Wx 22 9cm D x 21 6 cm D 21 6 cm D 1 8lbs 82kg 7 1 Ibs 7 7 Ibs Weight 2 6lbs 1 17kg 3 2 kg 3 5 kg with transformer Ambient temperature range Maximum 40 degrees Celsius 104 degrees Fahrenheit 20 90 non condensing relative humidity Minimum O degrees Celsius 32 degrees Fahrenheit Warranty 2 years Multi Tech Systems Inc Chapter 2 Installing and Cabling the MultiVOIP Introduction The MVP210 MultiVOIP models are tabletop units that can be handled easily by one person However the MVP410 and MVP810 MultiVOIPs are somewhat heavier units When these units are to be installed into a rack two able bodied persons should participate Please read the safety notices before beginning installation Safety Warnings Lithium Battery Caution A lithium battery on the voice fax channel board provides backup power for the timekeeping capability The battery has an estimated life expectancy of ten years When the battery starts to weaken
56. 47 FXS Loop Start parameters 41 H H 323 Call Signaling parameter definitions 54 Identifying current firmware version 118 IFM firmware 121 Multi Tech Systems Inc IP Statistics fields 109 LED descriptions 7 Link Management fields 110 Logs Statistics field definitions 107 N NAT Traversal screen fields 71 P Packet Prioritization 802 1p 33 Packetization rates 115 R RADIUS Screen field definitions 69 Regional parameter definitions 62 S Saving the MultiVOIP Configuration 75 Set Baud Rate 75 Set Log Reporting Method 70 Set SNMP parameters 61 Set Telephony Interface parameters 40 Setting Ethernet IP parameters 32 Setting password 124 Setting user defaults 123 SIP Call Signaling parameter definitions 56 SMTP parameters definitions 66 Specifications 8 SPP Call Signaling parameter definitions 59 STUN clients and servers 71 Supervisory signaling 40 Supplementary Services parameter definitions 72 Survivable SIP 57 SysLog Server function enabling 134 T T 38 36 U Updating firmware 117 V Voice FAX parameter definitions 35 144
57. 9 Miroda ainaani a solices 9 Safely WaAMINGS Eonian n N a e due vesagenuuetiaabenes 9 Unpacking Your MUV OI r N AA AA tu teeaea 9 Rack Mounting Instructions for MVP410 8 MVP8 10 ococccccinccccconcnonoccnoncncnnnnnnnn nn nono cnn nn rra nn rn cren 10 Cabling Procedure tor MMP 2 lWissaiosian tai A A A tid buns ita 11 Cabling Procedure for MVP 41 0 81 cta dt ii 13 Chapters Software Install A tan 16 INTOdUCIO Noria last 16 Loading MultiVOlP Software onto the PC isiicrirciicas rana err idea a L ENA a 16 SETUP OVER AAA A a 19 A A a O TA 20 BLOTE SI O a O are ret A rere eee Cnr ry eee 21 A O TO 23 Call SIQMENING 0 lala do 25 Regionalista la di 27 Phone BOOK cesaron ase ea e cn e e o o e e re eee 28 A a 29 Chapter 4 Configuring Your MUultIVOIP sns A aa aE NENAS 30 INTOUCIO a EE EAE TE E O E E AE A A AE ATE 30 Software Categories Covered in This Chapter coincide E 30 Howto Navigate Through the So MW iria coda 31 Wep Browser INtedTace in ideada 31 Gontiguration Information Checklist aiii a aaa eta 31 O 32 A A RO 35 contiqgurable Payload Perito eanaeunasetieas 39 LENEE EA said er ore rere 40 EAS boo Stat Pafametlors cti lips 41 Message Wales as 43 FARO Parameters usual si ias 44 E amp M Parameters usual ees 49 DID Paramotor a endo sees noe hs fens ein da doc a a Gage Sea 52 Gall SIGMA II Gs sur ia 53 PE o2 et ieee seetanseseie ct Seaeaeceaanes sans acceihetenia see cans ticassnents cost eacs ts sntcacasug ns teebarecectGasces Sue
58. BX i e to connect to the PSTN Generally 1 or 11 or 0 must be dialed as a prefix for calls outside of the calling code area long distance calls national calls or international calls On a sheet of paper write down the digits you must dial before you can dial a remote area code Euro International Call Example Rotterdam Bordeaux system Rotterdam VOIP works with PBX where 9 is used for all out of building calls 0 must precede all international calls North America Euro National Call Example Long Distance Example Seattle Chicago system Seattle VOIP works with PBX that uses 8 for all VOIP calls 1 must immediately precede area code of dialed number London Birmingham system London VOIP works with PBX that uses 9 for all out of building calls whether by VOIP or by PSTN 0 must immediately precede area code of dialed number 3 Answer write down 90 Answer write down 81 Answer write down 90 Multi Tech Systems Inc 78 Chapter 5 Phonebook Configuration 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 Euro National Call Example Euro International Call Long Distance Example Example Seattle Chicago system London Birmingham system Rotterdam Bordeaux system Answer enter 81312 as Leading zero of Birmingham a
59. Calling Party Busy Party Call Hold Alerting Party Connected Party V Enable Name Identification Hold Sequence 2 Call Waiting gt lena OK Default Help Retrieve Sequence 3 Cancel Copy Channel Figure 4 24 Supplementary Services The Supplementary Services fields are described in the tables below Multi Tech Systems Inc 72 Chapter 4 Configuring your VOIP Supplementary Services Parameter Definitions Field Name Values Description Select Channel 1 2 210 The channel to be configured is selected here 1 4 410 1 8 810 Call Transfer Enable Y N Select to enable the Call Transfer function in the VOIP unit This is a blind transfer and the sequence of events is as follows Callers A and B are having a conversation Caller A wants to put B into contact with C Caller A dials call transfer sequence Caller A hears dial tone and dials number for caller C Caller A gets disconnected while Caller B gets connected to caller C A brief musical jingle is played for the caller on hold Transfer Sequence Any phone keypad The numbers and or symbols that the caller must press on the phone character keypad to initiate a call transfer The call transfer sequence can be 1 to 4 characters in length using any combination of digits or characters or The sequences for call transfer call hold and call waiting can be from 1 to 4 digits in length consisting of any combination o
60. Channel V Enable iL Peele G C CID Manipulation F Disable CID Manipulation p CID Mode TransParent bd Pass Through Options MV Enable Figure 4 4 FXS Loop Start parameters FXS Loop Start Interface Parameter Definitions Field Name Values Description Dialing Options fields FXS Loop Start Y N Enables FXS Loop Start interface type Inter Digit Timer 1 10 seconds This is the length of time that the MultiVOIP will wait between digits When the time expires the MultiVOIP will look in the outbound phonebook for the number entered and place the call accordingly Default 2 Message Waiting Not applicable to SS series MultiVOIPs Indication Inter Digit in milliseconds The length of time between the outputting of DTMF digits Regeneration Time Default 100 ms FXS Options fields FXS Ring Count 1 10 Maximum number of rings that the MultiVOIP will issue before giving FXS up the attempted call Current Loss Y N When enabled the MultiVOIP will interrupt loop current in the FXS circuit to initiate a disconnection This tells the device connected to the FXS port to hang up The Multi VOIP cannot drop the call the FXS device must go on hook Generate Current Y N When selected this option implements Answer Supervision and Reversal Disconnect Supervision to the FXO interface using current reversal to indicate events Applicable only when FXS and FXO interfaces are connected back to back
61. Channel 1 FXS FXO connector on the back of the MultiVOIP Connect the other end to the DID jack NOTE DID lines are polarity sensitive If during testing the DID line rings busy consistently you will need to reverse the polarity of one end of the connector swap the connections of the wires to the two middle pins of one RJ 11 connector 4 Repeat step 3 to connect the remaining telephone equipment to each channel on your MultiVOIP Although a MultiVOIP s channels are often all configured identically each channel is individually configurable So for example some channels of a MultiVOIP might use the FXO interface and others the FXS some might use the DID interface and others E amp M etc Multi Tech Systems Inc 13 Chapter 2 Installing and Cabling the MultiVOIP If you intend to configure the MultiVOIP remotely using the MultiVOIP Windows interface connect an RJ 11 phone cable between the Command Modem connector not available on the SS or FX series and a receptacle served by a telco POTS line See Figure 2 6 below The Command Modem is built into the MVP410 and 810 units only To configure the MultiVOIP remotely using its Windows interface you must call into the MultiVOIP s Command Modem Once a connection is made the configuration process is identical to local configuration with the Windows interface Command Modem connector for remote configuration M VP 4 1 0 8 1 o Rear Panel FAT
62. Consider for example the Wren Clothing Company This company has VOIP equipped offices in London Paris and Amsterdam each served by its own PBX VOIP calls between the three offices completely avoid international long distance charges These calls are free The phonebooks can be set up to allow all Wren Clothing employees to contact each other using 3 4 or 5 digit numbers as though they were all in the same building United Kingdom __ Wren Clothing Co a VOIP PBX Site Wren Clothing Co London VOIP PBX Site Amsterdam The Netherlands Wren Clothing Co VOIP PBX Site Paris Free VOIP Calls Figure 5 13 Free VOIP calls In another use of the VOIP system the local calling area of each VOIP location becomes accessible to all of the VOIP system s users As a result international calls can be made at local calling rates For example suppose that Wren Clothing buys its zippers from The Bluebird Zipper Company in the western part of metropolitan London In that case Wren Clothing personnel in both Paris and Amsterdam could call the Bluebird Zipper Company without paying international long distance rates Only London local phone rates would be charged This applies to calls completed anywhere in London s local calling area Generally local calling rates apply only within a single area code and for all calls outside that area code national rates apply There are however some European cases w
63. Diff Serv VOIP Media 0 63 Value is used to prioritize the RTP RTCP audio IP packets PHB default 46 Setting this parameter to 0 in conjunction with Call Control PHB above will disable Diff Serv FTP Parameter fields FTP Server Enable Y N Default disabled See FTP Server File Transfers in Chapter 6 MultiVOIP unit has an FTP Server function so that firmware and other important operating software files can be transferred to the VOIP via the network DNS Parameter fields Address Enable DNS Y N Enables Domain Name Space System function where computer names are resolved Default using a worldwide distributed database disabled Enable SRV Y N Enables service record function Service record is a category of data in the Internet Domain Name System specifying information on available servers for a specific protocol and domain as defined in RFC 2782 Newer internet protocols like SIP STUN H 323 POP3 and XMPP may require SRV support from clients Client implementations of older protocols like LDAP and SMTP may have been enhanced in some settings to support SRV DNS Server IP n n n n IP address of specific DNS server to be used to resolve Internet computer names Multi Tech Systems Inc 34 Chapter 4 Configuring your VOIP Voice Fax Setting the Voice FAX Parameters The Voice Fax section needs to be set for each channel to be used However once you have established a
64. Edit Description Baltimore metro Delete Close Help Registration Options Subscription Options Figure 5 11 Inbound Baltimore example Notice the extended prefix to be removed 14103257 This entry allows Minneapolis users to contact Baltimore co workers as though they were in the Minneapolis facility using numbers in the range 7000 to 7999 Note also that a comma as in the entry 9 443 denotes a delay in dialing A one second delay is commonly used to allow a second dial tone to be generated for calls going outside of the facility s PBX system The Outbound Phone Book for the Minneapolis VOIP is shown below The third destination pattern 7 facilitates reception of co worker calls using local appearing extensions only In this case the Add Prefix field value for this phonebook entry would be 1410325 Outbound Phone Book Destination Pattern IP Address Protocol Description Alternate IP Address 1410 200 2 9 7 H 323 Baltimore 1443 200 2 9 7 H 323 Baltimore overlay 7 200 2 9 7 H 323 Baltimore Office Extensions Number of Entries 3 Details Remove Prefix 1410 Add Add Prefix 9 not used Edit Delete H 323 Port 1720 Close Help Round Trip Delay 300 ms Figure 5 12 Outbound Minneapolis example Multi Tech Systems Inc 92 Chapter 5 Phonebook Configuration Europe The most direct use of the VOIP system is making calls between the offices where the VOIPs are located
65. Generate Current Reversat FXO Options FXO Ring Count 2 No Response Timer 180 secs E amp M Options Wink Timer Type TYPE II y Mode e Zire C awie No Response Timer 60 secs IV Disconnect on Call Progress Tone DID Options Start Modes wink Start v Wink Timer 200 Figure 4 3 Telephony parameters Interface Type xs Loop Start Dialing Options Regeneration Inter Digit Timer 2 secs c Inter Digit Regeneration Timer 100 ms J Message Waiting Indication Light Flash Hook Options Generation e00 ms Detection Range Min f DO ms Max 11000 ms Caller ID Type BellCore v V Enable CID Manipulation M Disable CID Manipulation CID Mode TransParent v JE a Ml Pass Through Options V Enable Password Cancel Default Help Copy Channel dde Multi Tech Systems Inc 40 Chapter 4 Configuring your VOIP FXS Loop Start Parameters The parameters applicable to FXS Loop Start are shown in the figure below and described in the table that follows Interface Parameters Select Channel Channel 1 X Interface Type FXS Loop Start v 5 Dialing Options FXS Options FXS Ring Count B Regeneration Inter Digit Timer 2 secs VC Inter Digit Regeneration Timer 100 ms e Message Waiting Indication Light a Password Flash Hook Options ra Detection Range Cancel Min 100 ms Max 1000 ms Default Help Caller ID Type BellCore y 7 Copy
66. ID example 1 Figure 5 25 VOIP Caller ID Case 1 Call through telco central office with standard CID enters VOIP system CID Flow Call originates here at 4 19pm July 10 Call is received here CID Terminating Generating Central Office ver Volp without standard telephony Clock Ch3 Caller ID service 7 10 4 19pm S Display shows Z phone of Wilda Jameson 763 555 4071 CID Number 423 CID Name Anoka Whse VP3 Time Stamp Date 7 10 Time 4 19pm Phone Book Configuration Gateway Name Anoka Whse VP3 Q 931 Parameters Inbound Phone Book Channel 2 Remove Prefix Add Prefix Forward Addr In x 06 release when SIP protocol is used CID Name field will duplicate value in CID Number field Gatekeeper RAS Para Figure 5 25 Caller ID example 2 Figure 5 26 VOIP Caller ID Case 2 Call through telco central office without standard CID enters H 323 VOIP system Multi Tech Systems Inc 99 Chapter 5 Phonebook Configuration lt CID Flow Call is received Call originates here here 2 lt Ch1 at 5 47pm Sept 27 Terminating Generating Central Office A VolP VoIP cho with out e e standard telephony wd Clock A Ch3 Caller ID service 15 26 5 31 Ch4 SPP Protocol phone of Henry Brampton 763 555 4077 Display shows CID Number 423 CID Name Shipping Dept Time Stamp Date 0927 Time 1747 f Description field in
67. IP you ll want to confirm that it is configured and operating properly To do so it s good to have another VOIP that you can call for testing purposes You ll want to confirm end to end connectivity You ll need IP and telephone information about that remote site If this is the very first VOIP in the system you ll want to coordinate the installation of this MultiVOIP with an installation of another unit at a remote site Identify VOIP Protocol to be Used Will you use H 323 SIP or SPP Each has advantages and disadvantages Although it is possible to mix protocols in a single VOIP system it is highly desirable to use the same VOIP protocol for all VOIP units in the system SPP is a non standard protocol developed by Multi Tech SPP is not compatible with the Proprietary protocol used in Multi Tech s earlier generation of VOIP gateways The SS series of MultiVOIPs only support the SIP protocol The FX models do not support H 323 Multi Tech Systems Inc 77 Chapter 5 Phonebook Configuration Phonebook Starter Configuration This section will walk you through the phone book setup with examples that will aid in entering the correct numbers needed to have the MultiVOIP working correctly To do this part of the setup you need access to another VOIP that you can call to conduct a test It should be at a remote location typically somewhere outside of your building You must know the phone number and IP address for that site We are a
68. MVP x xx Firmware screen appears saying MultiVOIP model number is up Reboot to Download Firmware Click OK to download the factory defaults The Boot LED on the MultiVOIP will light up and remain lit during the file transfer process 5 After the PC gets a response from the MultiVOIP the Dialog IP Parameters screen will appear m IP Parameters r Diff Serv Parameters 1 434 TYPE I1 Call Control PHB Frame Type TYPE I y 46 YolP Media PHB MIP Parameters J Enable DHCP IP Address 192 168 IP Mask 255 255 255 0 Gateway Figure 6 19 Dialog screen The user should verify that the correct IP parameter values are listed on the screen and revise them if necessary Then click OK 6 Progress bars will appear at the bottom of the screen during the data transfer by o m 1S00VB6EOOO6OOEOIEEC OOOC IOOPSPETCNAA Downloading Configuration Packets Sent 2 Acks received 2 Errors 0 TT Figure 6 20 Progress bars The MultiVOIP s Boot LED will turn off at the end of the transfer 7 The Download Factory Defaults procedure is complete Multi Tech Systems Inc 120 Chapter 6 Using the Software Downloading IFM Firmware The Interface Module IFM is the telephony interface for analog MultiVOIP units There is one IFM for each channel of the MultiVOIP unit For each channel the IFM handles the analog signals to and from the atta
69. MultiVOIP Voice Fax over IP Gateways MVP210 410 810 MVP210 410 810 SS MVP210 410 810 FX User Guide Multi lech Systems User Guide S000383D Analog MultiVOIP Units Models MVP210 MVP410 MVP810 Models MVP210 SS MVP410 SS MVP810 SS Models MVP210 FX MVP410 FX MVP810 FX Upgrade Unit Model MVP428 This publication may not be reproduced in whole or in part without prior expressed written permission from Multi Tech Systems Inc All rights reserved Copyright O 2009 by Multi Tech Systems Inc Multi Tech Systems Inc makes no representations or warranty with respect to the contents hereof and specifically disclaims any implied warranties of merchantability or fitness for any particular purpose Furthermore Multi Tech Systems Inc reserves the right to revise this publication and to make changes from time to time in the content hereof without obligation of Multi Tech Systems Inc to notify any person or organization of such revisions or changes Check Multi Tech s Web site for current versions of our product documentation Record of Revisions Revision Date Description A 09 26 05 Doc re organization Follows S000249K Describes 6 08 software release B 04 25 07 Update tech support contact list amp revise warranty Cc 02 18 08 Format revision and software version x 11 update Add SS amp FX series D 04 21 09 Temperature change remove outdated sections Patents This Product is covered by one or more of
70. P address as functionally their own However if your VOIP system overall is operating in Registrar Client mode but you want to make an exception and use Direct mode for the destination pattern of this particular Add Edit Phonebook entry leave this checkbox unselected Also do not select this if your overall VOIP system is operating in the Direct SPP mode in this mode all VOIPs are peers with unique static IP addresses Port Number numeric When operating in Registrar Client mode this is the port by which the gateway receives all SPP data and control messages from the registrar gateway This ability to receive all data and messages via one port allows the VOIP to operate behind a firewall with only one port open When operating in Direct mode this is the Port by which peer VOIPs receive data and messages Alternate numeric Phone number associated with alternate IP routing Phone Number Remote Y N When checked this MultiVOIP can operate with first generation Device is MultiVOIP units in the same IP network These include MVP legacy 110 120 200 400 800 VOIP This is not available for the SS series of MultiVOIPs Advanced Gives access to secondary screen where an Alternate IP Route can be specified for backup or button redundancy of signal paths For SIP amp H 323 operation only Multi Tech Systems Inc 84 Chapter 5 Phonebook Configuration Clicking on the Advanced button brings up the
71. SE AU a AR MID ill EY E TAR A ETP AE PAB A RSE AUS CHL as Fee ll IE 395 thee E RoHS Fi 7 Multi Tech Systems Inc PARE RARA BE TS RA FWA HS EAS ASIRBMIITCA RDA 0 El AE ESE 32R EM PB CD CR6 PBB PBDE LEER CS ESTA E AM ee BR BS X RIE IMA BC PARADA E BO A S T xxx 2006 RE BER O ARMARIO A BATA ER A Multi Tech Systems Inc 143 INDEX Auto Disconnect 39 AutoCall Offhook 38 Cabling 210 11 410 810 13 Call Hold 72 Call Name Identification 72 Call Progress fields 105 Call Transfer 72 Call Waiting 72 Coder Parameters fields 37 Creating a User Default Configuration 75 Custom Tones and Cadences 64 D DID Interface Parameters 52 DID DPO Interface parameter definitions 52 Diff Serv PHB value 34 DTMF inband 36 DTMF out of band 36 Dynamic Jitter 39 E E amp M parameter definitions 50 E amp M Parameters 49 Email log reports 65 Error message Comm Port Unavailable 76 MultiVOIP Not Found 76 Phone Database not Read 76 Expansion card 4 to 8 channel installation 137 F FRF11 36 FTP Server function 127 FTP Server logging out 131 FXO Interface parameter definitions 45 FXO Parameters 44 FXO Supervision parameter definitions
72. VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received with integer Total number of error laden packets received by this VOIP gateway since the last Errors value clearing or resetting of the counter within the MultiVOIP software UDP Packets User Datagram Protocol packets Transmitted integer Number of UDP packets transmitted by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received integer Number of UDP packets received by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received with integer Number of error laden UDP packets received by this VOIP gateway since the last Errors value clearing or resetting of the counter within the MultiVOIP software TCP Packets Transmission Control Protocol packets Transmitted integer Number of TCP packets transmitted by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received integer Number of TCP packets received by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received with integer Number of error laden TCP packets received by this VOIP gateway since the last Errors value clearing or resetting of the counter within the MultiVOIP software RTP Packets Voice signals are transmi
73. VOIPs Subscriber This is used as the default end point register with a Proxy Default name If the Username is not populated in the Phone Book this is the Username that will Username be used This works the same for the password as well Password password Password for proxy server function See Default Username description above Re Registration 10 65535 This is the timeout interval for registration of the MultiVOIP with a SIP proxy server Time seconds The time interval begins the moment the MultiVOIP gateway registers with the SIP proxy server and ends at the time specified by the user in the Re Registration Time field this field When if registration lapses call traffic routed to from the MultiVOIP through the SIP proxy server will cease However calls in progress will continue to function until they end Proxy Polling 60 300 The interval between the VOIP gateway s successive attempts to connect to and Interval be governed by a higher level SIP proxy server The Primary Proxy is the highest level gatekeeper Alternate Proxy 1 is second Alternate Proxy 2 is the lowest order SIP proxy server TTL Value SIP proxy As soon as a MultiVOIP gateway registers with a SIP proxy server allowing the Time to proxy server to control its call traffic a countdown timer begins The TTL Value is Live the interval of the countdown timer Before the TTL countdown expires the value MultiVOIP gateway needs to register with the gatekeeper in ord
74. a which denotes a pause to get a PSTN dial tone and then the 10 digit phone number which includes the area code 612 for the city of Minneapolis which is different than the area code of the suburb where the PBX is actually located 763 A similar sequence of events occurs when the Baltimore employee calls number in the 651 and 952 area codes because number in both of these area codes are local calls in the Minneapolis St Paul area Multi Tech Systems Inc 91 Chapter 5 Phonebook Configuration The simplest case is a call from Baltimore to a phone within the Minneapolis St Paul area code where the company s VOIP and PBX are located namely 763 In that case that local VOIP removes 1763 and dials 9 to direct the call to its local 7 digit PSTN Finally consider the longest entry in the Minneapolis Inbound Phonebook 17637175 Note that the main phone number of the Minneapolis PBX is 763 717 5170 The destination pattern 17637175 means that all calls to Minneapolis employees will stay within the suburban Minneapolis PBX and will not reach or be carried on the local PSTN Similarly the Inbound Phone Book for the Baltimore VOIP shown first below generally matches the Outbound Phone Book of the Minneapolis VOIP shown second below Inbound Phone Book Remove Prefix Add Pretix Forward Address 1410 9 Not Used 14103257 7 Not Used 1443 9443 Not Used Number of Entries 3 Details Add Es Channel No Hunting
75. a install screen During the installation you must specify which browser you ll use in the Select Browsers screen InstallShield Wizard Ea Select Browsers T Netscape 6 You may change the default in the Java TM Plug in Control Panel installshield lt Back te Cancel Figure 6 44 Browser choice When installation is complete the Java program runs automatically in the background as a plug in supporting the MultiVOIP web interface No user actions are required After the Java program has been installed you can access the MultiVOIP using the web browser interface Close the MultiVOIP Windows interface Start the web browser Enter the IP address of the MultiVOIP unit Enter a password when prompted A password is needed here only if password has been set for the local Windows interface or for the MultiVOIP s FTP Server function See Setting a Password Web Browser interface earlier in this chapter The web browser interface offers essentially the same control over the VOIP as can be achieved using the Windows interface As noted earlier logging functions cannot be handled via the web interface And because network communications will be slower than direct communications over a serial PC cable command execution will be somewhat slower over the web browser interface than with the Windows interface Multi Tech Systems Inc 133 Chapter 6 Using the Software SysLog Server Functions Multi Tech has
76. a packet to the RADIUS server and doesn t Interval receive a response in the retransmit interval it will retransmit that packet Number of 0 255 again and wait the retransmit interval again for a response How many times Retransmissions it does this is determined by the setting in the Number of Retransmissions field Shared Secret alpha numeric Client encryption key for the current VOIP unit Select Attributes Gives access to RADIUS Attributes screen On Attributes screen one can button specify the parameters to be tallied by the RADIUS server for accounting usually billing purposes The RADIUS dialog box has a secondary dialog box RADIUS Attributes 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 The RADIUS Attributes screen lets you pick which aspects will be included in the accounting reports sent to the RADIUS server RADIUS Attributes Definitions Field Description Field Description Select All Log report to include all fields Start Date Time Date and time the phone call began shown Channel Data channel carrying call Call Mode Voice or fax Number Duration Length of call Packets Received Total packets received in call Packets Sent Total packets sent in call Bytes Received Total bytes received in call Bytes Sent Total bytes
77. a value for maximum bandwidth Advanced Features Silence Y N Determines whether silence compression is enabled checked Compression for this voice channel With Silence Compression enabled the MultiVOIP will not transmit voice packets when silence is detected thereby reducing the amount of network bandwidth that is being used by the voice channel default on Echo Y N Determines whether echo cancellation is enabled checked for Cancellation this voice channel Echo Cancellation removes echo and improves sound quality default on Forward Error Y N Determines whether forward error correction is enabled Correction checked for this voice channel Forward Error Correction enables some of the voice packets that were corrupted or lost to be recovered FEC adds an additional 50 overhead to the total network bandwidth consumed by the voice channel default Off Table is continued on next page Multi Tech Systems Inc 37 Chapter 4 Configuring your VOIP Voice Fax Parameter Definitions continued Field Name Values Description AutoCall Offhook Alert Parameters Auto Call Offhook AutoCall The AutoCall option enables the local MultiVOIP to call a remote MultiVOIP Alert Offhook without the user having to dial a Phone Directory Database number As soon as Alert you access the local MultiVOIP voice fax channel the MultiVOIP immediately connects to the remote MultiVOIP identif
78. above Multi Tech Systems Inc 126 Chapter 6 Using the Software FTP Server File Transfers Downloads Multi Tech 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 The terminology of downloads and uploads gets a bit confusing in this context File transfers from a client to a server are typically considered uploads File transfers from a large repository of data to machines with less data capacity are considered downloads In this case these metaphors are contradictory the FTP server is actually housed in the MultiVOIP unit and the controller PC which is actually the repository of the info to be transferred uses an FTP client program In this situation we have chosen to call the transfer of files from the PC to the VOIP downloads Be aware that some FTP client programs may use the opposite terminology i e they may refer to the file transfer as an upload You can download firmware CAS telephony protocols default configuration parameters and phonebook data for the MultiVOIP unit with this FTP functionality These downloads are done over a network not by a local serial port connection Consequently VOIPs at distant locations can be updated from a central control poin
79. actor of higher frequency of pair 3dB to 31dB This applies to the dial ring busy and unobtainable fast busy tones and mute setting that the MultiVOIP outputs as audio to the FXS FXO or E amp M port Default 16dB Cadence n n n n On off pattern of tone durations used to denote phone ringing phone ms On Off four integer time busy connection unobtainable fast busy dial tone 0 indicates values in continuous tone survivability and re order Default values differ for milliseconds zero value for dial tone indicates continuous tone different countries regions Although most cadences have only two parts an on duration and an off duration some telephony cadences have four parts Most cadences then are expressed as two iterations of a two part sequence Although this is redundant it is necessary to allow for expression of 4 part cadences Custom button Click on the Custom button to bring up the Custom Tone Pair Settings screen The Custom button is active only when Custom is selected in the Country Region field This screen allows the user to specify tone pair attributes that are not found in any of the standard national regional telephony toning schemes Table is continued on next page Multi Tech Systems Inc 63 Chapter 4 Configuring your VOIP Regional Parameter Definitions continued Field Name Values Description
80. again If the Interface Type is FXS Loop Start a station device such as an analog telephone fax machine or KTS Key Telephone System is connected to an analog channel The FXS options group is active If the Interface Type is FXO the Dialing Options Regeneration Flash Hook Timer and Ring Count groups are enabled The FXO Ring Count allows you to set the number of rings before the unit answers the incoming call Check with your local in house phone personnel to verify whether your local PBX dial signaling is pulse or tone DTMF The Flash Hook Options Generation setting allows you to enter the time in milliseconds for the duration of the flash hook signal If the Interface Type is E amp M you are connecting to an analog E amp M trunk on your PBX Check with your in house phone personnel to determine the signaling type Dial Tone or Wink and if it is 2 wire or 4 wire The SS and FX series do not support E amp M or DID operation Interface Parameters Select Channel Channel 1 Ad FXS Options FXS Ring Count 8 V Current Loss FXO Options FXO Ring Count 2 No Response Timer 180 secs E amp M Options Signal ei Wink Timer Type TYPE II M Mode le Aire C Awie No Response Timer 60 secs V Disconnect on Call Progress Tone DID Options Start Modes Wink Start v Wink Timer 200 Interface Type FXS Loop Start Dialing Options Regeneration Inter Digit Timer 2 secs E ion Times 100 Inter Digi
81. apter 4 Configuring your VOIP SMTP Setting the SMTP Parameters Log Reports by Email The SMTP Parameters screen is applicable when the VOIP administrator has chosen to receive log reports by email this is done by selecting the SMTP checkbox in the Others screen and selecting Enable SMTP in the SMTP Parameters screen Email Address for VOIP for email call log reporting This is needed only if log reports of VOIP call traffic are to be sent by email Ask Mail Server administrator to set up email account with password for the MultiVOIP unit itself Be sure to give a unique identifier to each individual MultiVOIP unit Get the IP address of the mail server computer as well MultiVOIP as Email Sender When SMTP is used the MultiVOIP will actually be given its own email account with Login Name and Password on some mail server connected to the IP network Using this account the MultiVOIP will then send out email messages containing log report information The Recipient of the log report email is ordinarily the VOIP administrator Because the MultiVOIP cannot receive email a Reply To address must also be set up Ordinarily the Reply To address is that of a technician who has access to the mail server or MultiVOIP or both and the VOIP administrator might also be designated as the Reply To party The main function of the Reply To address is to receive error or failure messages regarding the emailed reports
82. assages below Note that both Diff Serv parameters Call Control PHB and VOIP Media PHB must be set to zero if you enable Packet Prioritization 802 1p Nonzero Diff Serv values negate the prioritization scheme Multi Tech Systems Inc 32 Chapter 4 Configuring your VOIP Ethernet IP Parameter Definitions 802 1p Field Name Values Description Ethernet Parameters Packet Prioritization Y N Select to activate prioritization under 802 1p protocol described below Frame Type Type Il SNAP Must be set to match network s frame type Default is Type II 802 1p A draft standard of the IEEE about data traffic prioritization on Ethernet networks The 802 1p draft is an extension of the 802 1D bridging standard 802 1D determines how prioritization will operate within a MAC layer bridge for any kind of media The 802 1Q draft for virtual local area networks VLANs addresses the issue of prioritization for Ethernet networks in particular 802 1p enacts this Quality of Service feature using 3 bits This 3 bit code allows data switches to reorder packets based on priority level The descriptors for the 8 priority levels are given below 802 1p PRIORITY LEVELS LOWEST PRIORITY 1 Background Bulk transfers and other activities permitted on the network but should not affect the use of network by other users and applications 2 Spare An unused spare value of the user priority 0 Best Effort defa
83. ation functions The accounting function is well suited for billing of VOIP telephony services In the Select Attributes secondary screen accessed by clicking on Select Attributes button the VOIP administrator can select the parameters to be tallied by the RADIUS server RADIUS VE Server Address Accounting Port Retransmission Interval Number of Retransmissions RADIUS Attributes MV Select All Attributes V Channel Number V Start Date Time MV Duration V Call Mode IV Packets Sent IV Packets Received MV Bytes Sent IV Bytes Received V Packets Lost MV Coder V Dutbound Digits V Prefix Matched 4 Call Status From Details To Details Y Gateway Name Y Gateway Name v IP Address v IP Address Y Description Y Description Options Y Options Figure 4 21 RADIUS settings Multi Tech Systems Inc 68 Chapter 4 Configuring your VOIP The fields of the RADIUS screen are described in the table below RADIUS Screen Field Definitions Field Name Values Description Enable Accounting Y N When checked the MultiVOIP will access the accounting functionality of the RADIUS server Server Address n n n n IP address of the RADIUS server that handles accounting billing for the current MultiVOIP unit Accounting Port 1 65535 TDM time slot at which RADIUS accounting information will be transmitted and received Retransmission If the MultiVOIP sends out
84. ble you must change the COM port setting to a COM port that you have confirmed as being available on your PC COM Port Setup Select Port Coma y OK Baud Rate 115200 y Cancel Modem a sie TATSO TREBSGBTISOORDT Help Init Response OK Dial String 0 Connect Response CONNECT Hangup String ATH NOTE Ifthere is a Dial String specified in Modem Setup Configuration programs will try to initialize modem and dial this string Figure 4 25 COM port setup Multi Tech Systems Inc 75 Chapter 4 Configuring your VOIP Troubleshooting Software Issues In the lower left corner of the screen the connection status of the MultiVOIP will be displayed The messages in the lower left corner will change as detection occurs The message MultiVOIP Found confirms that the MultiVOIP is in contact with the MultiVOIP configuration program If the message displayed is MultiVOIP Not Found please try the resolutions below Fixing a COM Port Problem If the MultiVOIP main screen appears but is grayed out and seems inaccessible the COM port that was specified for its communication with the PC is unavailable and must be changed An error message will appear MultiVOIP COM Comm Port Unavailable Figure 4 26 Error pop up To change the COM port setting use the COM Port Setup dialog box by going to the Connection pull down menu and choosing Settings or use the left side control panel In the Select Port fiel
85. boot sag MultiVOIP 410 Web Based Configuration z e pS over IP Networks Mutt Muhi Tech Systems Inc 2205 Woodale Drive Mounds View MN 55112 USA TEL 763 785 3500 FAX 763 785 3702 N Tech rah supper 2 Applet distrut started 8 inteme Figure 6 42 Web interface main page You can control the MultiVOIP unit with a graphic user interface interface based on the common web browser platform Qualifying browsers are Internet Explorer 6 Netscape 6 and Mozilla Firefox 1 0 MultiVOIP Web Browser interface Overview Function Remote configuration and control of MultiVOIP units Configuration Prerequisite Local Windows interface must be used to assign IP address to MultiVOIP Browser Version Requirement Internet Explorer 6 0 or higher or Netscape 6 0 or higher or Mozilla Firefox 1 0 or higher Java Requirement Java Runtime Environment version 1 4 0_01 or higher this application program is included with MultiVOIP The initial configuration step of assigning the VOIP unit an IP address must still be done locally using the Windows interface However all additional configurations can be done via the web interface The content and organization of the web interface is directly parallel to the Windows interface For each screen in the Windows interface there is a corresponding screen in the web interface The fields on each screen are the same as well The Windows interfa
86. built SysLog server functionality into the software of the MultiVOIP units SysLog is a de facto standard for logging events in network communication systems The SysLog Server resides in the MultiVOIP unit itself To implement this functionality you will need a SysLog client program sometimes referred to as a daemon SysLog client programs both paid and freeware can be obtained from Kiwi Enterprises search the Internet for kiwi syslog daemon among other firms Read the End User License Agreement carefully and observe license requirements SysLog client programs essentially give you a means of structuring console messages for convenience and ease of use Multi Tech Systems does not endorse any particular SysLog client program SysLog client programs by qualified providers should suffice for use with MultiVOIP units Before a SysLog client program is used the SysLog functionality must be enabled within the MultiVOIP in the Logs menu under Configuration Logs m Console Message Settings MV Enable Console Messages Filters Cancel Logs T Tum Off Logs 5 SNMP da ory Seaan aa ee SysLog Server IM Enable IP Address Er 51 4 oe ne AE TL Online Statistics Updation Interval 5 Sec Figure 6 45 Enable SysLog 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 Configuring
87. ce gives access to commands via icons and pull down menus whereas the web interface does not The web interface however cannot perform logging in the same direct mode done in the Windows interface However when the web interface is used logging can be done by email SMTP The graphic layout of the web interface is also somewhat larger scale than that of the Windows interface For that reason it s helpful to use as large of a video monitor as possible The primary advantage of the web interface is remote access for control and configuration The controller PC and the MultiVOIP unit itself must both be connected to the same IP network and their IP addresses must be known In order to use the web interface you must also install a Java application program on the controller PC This Java program is included on the MultiVOIP product CD Java is needed to support drop down menus and multiple windows in the web interface 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 Multi Tech Systems Inc 132 Chapter 6 Using the Software InstallShield Wizard Welcome to the InstallShield Wizard for Java 2 Runtime Environment SE 1 4 0_01 The InstallShield Wizard will install Java 2 Runtime Environment SE v1 4 0_01 on your computer To continue click Next Al Cancel Figure 6 43 Jav
88. cel Help Figure 4 14 SIP call signaling Multi Tech Systems Inc 55 Chapter 4 Configuring your VOIP SIP Call Signaling Parameter Definitions Field Name Values Description SIP Proxy Parameters Signaling Port port Port number on which the MultiVOIP UserAgent software module will be waiting for any incoming SIP requests Default 5060 Use SIP Proxy Y N Allows the MultiVOIP to work in conjunction with a proxy server Allow Incoming Y N When selected incoming calls are accepted only if those calls come through the Calls Through proxy SIP Proxy Only Primary Proxy This is the preferred SIP proxy server for controlling the traffic of the current VOIP Alternate Proxy A first and a second alternate SIP proxy server can be specified for use by the 1 and 2 VOIP for situations where the Primary proxy server is otherwise unavailable Proxy Domain n n n n Network address of the proxy server that the VOIP is using Name IP Address Append SIP Y N When checked the domain name of the SIP Proxy serving the MultiVOIP gateway Proxy Domain will be included as part of the User ID for that gateway If unchecked the SIP Name in User Proxy s IP address will be included as part of the User ID instead of the SIP ID Proxy s domain name Port Number port Logical port number for proxy communications Default 5060 Default This is not implemented in the SS series
89. channel or phone station is busy in Omaha Connected Party Allowed Name Type CNI If the home VOIP unit is receiving a call and Connected Party is selected then the identifier from the Caller Id field will tell the originating remote VOIP unit that the attempted call has been completed and the connection is made This field is applicable only when the home VOIP unit is receiving the call Example Suppose a VOIP system has offices in both Denver and Omaha In the Omaha VOIP unit the home VOIP unit in this example Call Name Identification has been enabled Connected Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field of the Supplementary Services screen When channel 2 of the Omaha VOIP completes an attempted call from any other VOIP phone station for example the Denver office the message Connect Party Omaha Sales Office Voipchannel 2 will be sent back and will appear in the Caller Id field of the Statistics Call Progress screen of the Denver VOIP This confirms to the Denver VOIP that the call has been completed to Omaha 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 or Barnesville Factory Default When this button is clicked all Suppleme
90. ched telephone PBX or CO line The IFM communicates with the main processor indicating the status of the telephone line For example it might indicate that a phone is off hook FXS or that an incoming ring is present FXO The IFM receives operating instructions from the VOIP s main processor For example the IFM might be instructed to ring the phone FXS or seize the line FXO The IFM contains a codec coder decoder to convert the incoming audio to a PCM stream pulse code modulation which it sends to the DSP digital signal processor The IFM s codec also converts outgoing PCM to audio The firmware of the IFMs will change from time to time and you may need to upgrade the firmware on your MultiVOIP unit To do so follow these instructions 1 In the System Information screen of the MultiVOIP Configuration software check the version number of the IFM firmware already installed on the MultiVOIP unit Write down the version number 2 Exit the Configuration software program The MultiVOIP Configuration program must be off when invoking the Download IFM Firmware command If it is on the command will not work 3 To use the Download IFM Firmware command go to Start Programs MultiVOIP x xx Download IFM Firmware 4 A warning window will appear Downloading IFM Firmware will reboot the MultiVOIP Do you want to continue Click OK OE hl Downloading IFM Firmware will Reboot the MultivOIP Do you want to continue
91. corrupted or lost to be recovered FEC adds an additional 50 overhead to the total network bandwidth consumed by the voice channel Default Off Logs Log Total Number of Logs 0 lt gt Cal details Voce coder Packets sent Ditcormect Reston Packets recvd DTMF Capabaity Packets lost Ea Outbound Digts Recvd Bytes sent A Outbound Dipts Sert Bytes recvd Help Servet Detak From detads To detads Gateway Name Gateway Name IP Address IP Address Options Options SC Silence Compression Supplementary Services Into FEC Forward Enor Comecton Cal Transtetred To Call Forwarded To Figure 6 4 Log statistics screen Multi Tech Systems Inc 106 Chapter 6 Using the Software Logs Screen Details Field Definitions Field Name Values Description Log column 1 or higher All calls are assigned an event number in chronological order with the most recent call having the highest event number Start Date Time dd mm yyyy The starting time of the call The date is presented as a day and a month column hh mm ss of one or two digits and a four digit year This is followed by a time of day in a two digit hour a two digit minute and a two digit seconds value Duration column hh mm ss This describes how long the call lasted in hours minutes and seconds Type H 323 SIP SPP Indicates the Call Signaling protocol used for the call H 323 SIP or SPP Status column success or fai
92. d Server Details n n n n The IP address etc of the traffic control server if any being used and or other whether an H 323 gatekeeper a SIP proxy or an SPP registrar gateway related will be displayed here if the call is handled through that server descriptions DTMF Capability inband Indicates whether the DTMF dialing digits are carried Inband or Out of out of band Band The corresponding field values differ for the 3 different VOIP Expressions protocols differ slightly For H 323 this field can display Out of Band or Inband For SIP it can for different display either Out of Band RFC2833 or Out of Band SIP INFO to Call Signaling indicate the out of band condition or Inband to indicate the in band protocols condition For SPP it can display Out of Band RFC2833 or Inband H 323 SIP or SPP Table is continued on next page Multi Tech Systems Inc 105 Chapter 6 Using the Software Call Progress Details Field Definitions continued Field Name Values Description Supplementary Services Status Call on Hold alphanumeric Describes held call by its IP address source location gateway identifier and hold duration Location gateway identifiers come from Gateway Name field in Phone Book Configuration screen of remote VOIP Call Waiting alphanumeric Describes waiting call by its IP address source location gateway identifier and hold duration Location gateway iden
93. d select a COM port that is available on the PC if no COM ports are currently available re allocate COM port resources in the computer s MS Windows operating system to make one available Fixing a Cabling Problem If the MultiVOIP cannot be located by the computer three error messages will appear saying Multi VOIP Not Found Phone Database Not Read and Password Phone Database Not Read MultiVOIP MultiVOIP PDD MultiVOIP PDD Figure 4 27 Cabling errors In this case the MultiVOIP is simply disconnected from the network For instructions on MultiVOIP cable connections see the Cabling section of Chapter 3 Multi Tech Systems Inc 76 Chapter 5 Phone Book Configuration Introduction When a VOIP serves a PBX system it s important that the operation of the VOIP be transparent to the telephone end user That is the VOIP should not entail the dialing of extra digits to reach users elsewhere on the network that the VOIP serves On the contrary VOIP service more commonly reduces dialed digits by allowing users served by PBXs in facilities in distant cities to dial their co workers with 3 4 or 5 digit extensions as if they were in the same facility Furthermore the setup of the VOIP generally should allow users to make calls on a non toll basis to any numbers accessible without toll by users at all other locations on the VOIP system Consider for example a company with VOIP equipped offices in New Yor
94. d if they originate from endpoints at accepted domains or IP addresses Accept any specific Defines if registrations to the MVP SS SIP server will be accepted from any domain or Registrations domains only from specified domains Multiple domains can be listed separated by semicolons for The any domains option is intended for private networks not accessible via Internet Domain name Endpoints separated by semicolon from which the MVP SS will accept registrations Names Accept n n n n Determines whether registrations to the MVP SS SIP server will be accepted from any Registrations or IP address or only from specified IP addresses Multiple IP addresses can be listed for any IP separated by semicolon The any IP addresses option is intended for private addresses networks not accessible via Internet or PSTN IP n n n n List of IP addresses separated by semicolon of endpoints from which the MVP SS Addresses will accept registrations Re in seconds The time after which the UserAgent Client is supposed to register with the proxy Registration default is server Expiration of the registration means that the gateway has lost contact with the Time 3600 main SIP server and that the MVP SS unit will enter survivability mode In survivability mode the MVP SS unit will complete calls acting as a backup to the main SIP server Normally the MVP SS will initiate re registration before the interval lapses Multi Tech Systems Inc 57
95. ddress for the MultiVOIP unit has been established you can choose to configure the unit by using the MultiVOIP web browser interface If you want to do configuration work using the web browser interface you must first set it up e Set IP address of MultiVOIP unit using the MultiVOIP Configuration program the Windows interface e Save Setup in Windows interface e Close Windows interface e Install Java program from MultiVOIP product CD on first use only e Open web browser e Browse to IP address of MultiVOIP unit e f username and password have been established enter them when prompted e Set browser to allow pop ups The MultiVOIP Web interface makes use of pop up windows e The configuration screens in the web browser will have the same content as their counterparts in the software only the presentation differs Configuration Information Checklist To assist with the organization of the information needed below is a chart summarizing what is necessary The SS and FX models do not support E amp M or DID Info Info Type of Configuration Info Configuration screen where info is entered Obtained Entered Gathered y IP info for VOIP unit Ethernet IP parameters e IP address e Gateway e DNS IP if used e 802 1p Prioritization if used Interface Type Interface parameters e E amp M In FXS FXO systems channels used for phone fax or key system are FXS channels used for analog
96. e extended DTMF tones which are used for various PBX functions Tone Detection Y N Enables supervision of call disconnection by detecting cessation of a pre specified tone from the PBX Available dial tone List from which tones can be chosen to signal call disconnection Tones ring tone busy tone unobtainable tone fast busy survivability tone re order tone Disconnect any tone from Currently chosen disconnection supervision tone Tones Available Tones list Multi Tech Systems Inc 48 E amp M Parameters Chapter 4 Configuring your VOIP The parameters applicable to the E amp M telephony interface type are shown in the figure below and described in the table that follows Only the analog MVP210 410 810 models support the E amp M interface the SS and FX models do not Interface Parameters Select Channel Channel 1 v Vv Vv E amp M Options Signal DialTon C wink E obo steeeee Wink Timer 25 Type TYPE II y Mode Aire C awie No Response Timer 60 secs V Disconnect on Call Progress Tone Interface Type E amp M Dialing Options Regeneration Inter Digit Timer 2 secs i Inter Digit Regeneration Timer f 00 ms DTI Message Waiting Indication Stutter Dial Tone v Flash Hook Options Generation e00 ms Detection Range Min 100 ms Max 1000 ms Iv CID Manipulation r Pass Through Options V Enable Figure 4 8 E amp M pa
97. e Outbound Phonebook Commas denote a brief pause for a dial tone allowing time for the PBX to get an outside line Multi Tech Systems Inc 96 Chapter 5 Phonebook Configuration The screen below shows Outbound Phone Book entries for the VOIP located in the company s Paris facility Dutbound Phone Book IP Address Protocol Description I 003110 200 2 8 5 H 323 Rotterdam 003120 200 2 8 5 H 323 Amsterdam 00441483 200 210 3 H 323 Guildford 0044207 200 2103 H 323 London Inner 0044208 200 2103 H 323 London Outer 4 200 2 8 5 H 323 Amsterdam company office employees 5 200 2 10 3 H 323 London company office empl ext Number of Entries 7 Details Remove Prefix Add Prefix r not used Edit Delete H 323 Port 1720 Close PUR Help Round Trip Delay 300 ms dad Figure 5 20 Paris example outbound The Inbound Phone Book for the Paris VOIP is shown below Inbound Phone Book Add Prefix Forward Address 00331 9 Not Used 00334 9 0 Not Used 2 2 Not Used Number of Entries 3 Details Channel No Hunting Edit Description Delete Registration Options Close Help FREEE Subscription Options Figure 5 21 Paris example inbound Multi Tech Systems Inc 97 Chapter 5 Phonebook Configuration The screen below shows Outbound Phone Book entries for the VOIP in the company s Amsterdam facility Dutbound Phone Book IP Address 00331 200 2 9 7 H 323 Paris 00334 200 2 9 7 H 323 Lyo
98. e an error message if an invalid password is entered Multi OIP Invalid Password Figure 6 31 Invalid password Web Browser Interface Setting a password is optional when using the MultiVOIP web browser interface Only one password can be assigned and it works for all MultiVOIP software functions Windows interface web browser interface 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 After a password has been set that password is required to access the MultiVOIP web browser interface NOTE Record your user name and password in a safe place If the password is lost forgotten or irretrievable the user must contact Multi Tech Tech Support in order to resume use of the MultiVOIP web browser interface MultiVOIP 410 Configuration Advanced Phone Book Statistics Change Password Save amp Reboot Password Change Logout User Name default Help AA Old Password New Password Reconfirm Password Figure 6 32 Web interface password Multi Tech Systems Inc 125 Upgrading Software Chapter 6 Using the Software As noted earlier the Upgrade Software command transfers from the controller PC to the MultiVOIP unit firmware including the H 323 stack and settings The settings can be either Factory Default Settings or Current Configuration Settings Configuration e Intern
99. e contact information for them is configured Dynamic by the user and not subject to removal from the registration list due to timeouts Dynamic registrations are registered from an external endpoint with the contact information Dynamic entries must re register before the re registration interval expires else they will be removed from the list Endpoints removed from this list can neither make nor receive calls Re Registration integer The time after which the MultiVOIP UserAgent Client is supposed to register with Interval values in the proxy server seconds Expiration of the registration interval means that the gateway has lost contact default is with the main SIP server and that the MVP SS unit will enter its survivability 3600 mode In survivability mode the MVP SS unit will complete calls acting as a backup to the main SIP server Normally however the MVP SS will initiate re registration with some small margin of time before the interval lapses Contact Information Address n n n n The IP address at which this endpoint can be reached Port 0 64000 Digital time slot on which SIP calls will be made Default is 5060 Re Registration Time See Re Registration Interval entry above Multi Tech Systems Inc 58 SPP Chapter 4 Configuring your VOIP Single Port Protocol was developed by Multi Tech to allow for dynamic IP addressing when it is set to Registrar Client mode The other choice
100. e mail server address has to be entered in the SIP Voice Mail Server Parameters Group The Voice Mail server IP Address Port and Re subscription time are configured on the SIP Call Signaling screen When this is configured the Subscribe with Voice Mail Server option is activated in the inbound phone book Only when this option is enabled the subscribe message will be sent to the VMS The following sequence needs to be done to enable all of the Message Waiting Features 1 The Use SIP Proxy must be enabled and the SIP Proxy Parameters and Voice Mail Server Parameters in the SIP Call Signaling Menu must be set and the Interface Type option must be set to FXS Loop Start on the Interface menu s Message Waiting Indication options become active 2 Then the Message Waiting Indication options must be set to light or stutter tone for the Subscribe to Voice Mail Server option to become available in the Inbound phone book entry with that channel selected 3 In order to send Subscriptions for Inbound Phone Book entries all the following four conditions have to be satisfied e The user needs to enter a valid voice mail server domain name or IP address in the Voice Mail Server Domain Name IP Address field on the Call Signaling screen e For an Inbound Phone Book entry a subscription with Voice Mail Server checkbox is enabled on the Add or Edit Inbound Phone Book entries screen e The Channel type corresponding to that Inbound phone bo
101. e required If advance notice is not practical the telephone company will notify the customer as soon as possible The telephone company may make changes in its facilities equipment operation or procedures that could affect the operation of the equipment If this happens the telephone company will provide advance notice to allow you to make necessary modifications to maintain uninterrupted service 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 may request you to remove the equipment form t network until the problem is resolved 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 Manufacturer Multi Tech Systems Inc Trade name MultiVOIP Model number MVP 210 410 810 FCC registration number US AU7DDNAN46050 Modular jack USOC RJ 48C Service center in USA Multi Tech Systems Inc 2205 Woodale Drive Mounds View MN 55112 Tel 763 785 3500 FAX 763 785 9874 Multi Tech Systems Inc 141 Appendix E Waste Electrical and Electronic Equipment WEEE Statement July 2005 The WEEE directive places an obligation on EU based manufacturers distributors retailers and importers to take back elec
102. e unique The VOIP sends Register request to Registrar for each entry with its configured Username and Password When Authentication is enabled for the endpoint then the registrar proxy sends 401 Unauthorized 407 Proxy Authentication Required response when it receives a REGISTER INVITE request Now the endpoint has to send the authentication details in the Authorization header In this header one of the fields is username Generally proxies accept requests even if both Endpoint Name and Username are same But some proxies expect that the Endpoint Name and Username should be different To support these proxies we have the username and password configuration for every inbound phone book entry which gets registered with a proxy If the username and password are not configured in the inbound phone book then the registration will happen with the default username and password that are configured in the SIP Call Signaling Page Phone Book Save and Reboot When your Outbound and Inbound Phonebook entries are completed click on Save Setup in the sidebar menu to save your configuration You can change your configuration at any time as needed for your system Remember that the initial MultiVOIP setup must be done locally or via the built in Remote Configuration Command Modem using the MultiVOIP program After the initial configuration is complete all of the MultiVOIP units in the VOIP system can be configured re configured and updated fr
103. eGsccnnsseetscamtaetee oete een ena 53 ro eee Eee PEN CEE eee EE ee Cree eee ee errr tae ey rte eigets T ere rary Seren ey een eee ereer ee eee er rrrrrer cn Cotes 55 so Ed Bae eevee ee sn le li cl 59 NM id id al lie ll o ll ld 61 Multi Tech Systems Inc 3 SM TE sur ii is cee 65 RADIU Sut tds 68 Logs Trace uc da alias 70 NAT Traversalicnisa tii Saa 71 Supplements o sbi lab iacla cats Sats 72 SOUS Nd cio o are err nent eer ere T 75 AVE SREDO leisna ld ld dl shia hs ies 15 O 75 O O O 75 Troubleshooting Software SSUBS vaciando ileso abiip ana 76 Chapter 5 Phone Book Configuration kaanan E NENA ANNEN AANA 77 INTOUCIOscuisiciiin ai ita 77 Identity Remote VOIPSite to Calida a 77 dently VOIP Protocol to Pe Usad cian 77 Phonebook Staner Configuration sica A AA ativan AA 78 Cutbouna PRONG BCOK sssini a enn o israel elites trece AEE AE aero ice 78 inbound Phone Hookerin aa attests oslaascnctasboctesta sat otc deat onl EA E Aaa 80 Phone Book Des cipal A 81 Outbound Phone Booklist Entt 6S ios a a e a a a ee ee 81 Addi Edit Gutbound PRONG BOOK LO 82 Inbound Phone BookLIStEMMIE S si td ii a ld ORA aE AREE 86 Addi EditiMbOund Phone BOOK rsatan a aa aae Aa a Ea 87 Phone Book Save and ReBOOl chix iesa sabseier Hictaionk a E Er E E A a E E E NSE 89 Phonebook Examples scort ida 90 North ARMOR Cal adminis is 90 EURO PO iii ias 93 Vahiations of Caller ID wasiscceussassncsvaderacswsassuesnjadstuseaarsgancaagretendagsnasaaszeaanaasdeeesaagasden
104. eight channel unit All of these MultiVOIP units have a 10 100Mbps Ethernet interface and a command port for configuration The MVP428 is an expansion circuit card for the four channel MVP410 that turns it into an eight channel MVP810 These MultiVOIPs inter operate with a telephone switch or PBX acting as a switching device that directs voice and fax calls over an IP network The MultiVOIPs have phonebooks directories that determine to who calls may be made and the sequences that must be used to complete calls through the MultiVOIP The phonebooks allow the phone user to interact with the VOIP system just as they would with an ordinary PBX or telco switch When the phonebooks are set special dialing sequences are minimized or eliminated altogether Once the call destination is determined the phonebook settings determine whether the destination VOIP unit must strip off or add dialing digits to make the call appear at its destination to be a local call Feature Comparison Matrix The main differences between the model versions are the line type capabilities and interface options as detailed in the chart below MultiVOIP MultiVOIP SS MultiVOIP FX H 323 e SPP e e SIP e e e SIP Survivability DID e E amp M e FXS FXO o o e Multi Tech Systems Inc 6 Chapter 2 Quick Start Interface While the web interface appears differs slightly its content and organization are essentially the same as
105. el of fax tones To be changed only under the direction of Multi Default to 3 5dB Tech s Technical Support 9 5 dB Jitter Value Fax Default Defines the inter arrival packet deviation in milliseconds for the fax transmission A 400 ms higher value will increase the delay allowing a higher percentage of packets to be reassembled A lower value will decrease the delay allowing fewer packets to be reassembled Mode Fax FRF 11 FRF11 is frame relay FAX standard using these coders G 711 G 728 G 729 T 38 G 723 1 T 38 is an ITU T standard for real time faxing of Group 3 faxes over IP networks It uses T 30 fax standards and includes special provisions to preclude FAX timeouts during IP transmissions Table is continued on next page Multi Tech Systems Inc 36 Chapter 4 Configuring your VOIP Voice Fax Parameter Definitions continued Coder Parameters Coder Manual or Automatic Determines whether selection of coder is manual or automatic When Automatic is selected the local and remote voice channels will negotiate the voice coder to be used by selecting the highest bandwidth coder supported by both sides without exceeding the Max Bandwidth setting G 723 G 729 or G 711 are negotiated Selected Coder SS models only G 711 a u law 64 kbps G 726 16 24 32 40 kbps G 727 nine bps rates G 723 1 5 3 kbps 6 3 kbps G 729 8kbps Net Coder 6 4 7 2 8 8 8 9 6 kb
106. en audio patch not for use with Wink signaling e DID Options not supported by SS and FX series o Start Modes Immediate Wink or Delay Dial o Wink Timer in milliseconds Multi Tech Systems Inc 24 Chapter 3 Software Installation Call Signaling There are three choices for Call Signaling H 323 SIP and SPP the SS models only support SIP and the FX models support SIP and SPP but not H 323 It is best to select one of these as the protocol to be used rather than mixing them Single Port Protocol SPP is a non standard protocol created by Multi Tech that allows dynamic IP allocation Generally the default settings will work for most users and the individual parameters may be changed if the need arises Additional details for all settings are found in Chapter 4 H Use East Stat Signaling Port 1720 9 Register pith Galekeepel I Allow Incoming Cals Through Gatekeeper Only GaleKeeper RAS Parameters IP Addiess PimayGk 192 16a 3 1 1713 Ahemae GE 1 0 0 0 0 1713 Altemate GK 2 0 0 0 0 1713 RAS TTL Yal GateKeeper Discovery Poling Interval eo secs I Use Online Atemae GaleKeepar List H323 Version 4 Options i in TT H 323 Multiplexing Mud T H 245 Turneing Mun etransmission in ms FT Parallel H245 FS Tun TT Annex E AE Max Retransmission SIP Parameters Chert Options Signaling Port 5050 Use SIP Pros F Allow Incaming Calls Through SIP Prop Ony SIP Proxy Parameters
107. en the clients could have different port numbers or the same port number Default port number 10000 Re transmission 50 5000ms If packets are lost as indicated by absence of an acknowledgment then the endpoint will retransmit the lost packets after this designated time duration has elapsed Default value 2000 milliseconds Max Re transmission 0 20 Number of times the VOIP will re transmit a lost packet if no acknowledgment has been received Default value 3 Client Options Client Option fields are active only in registrar client mode and only for client VOIP units Primary Registrar This is the preferred SPP registrar gateway for controlling the traffic of the current VOIP Alternate Registrar 1 and 2 A first and a second alternate SPP Registrar gateway can be specified for use by the current VOIP for situations where the Primary Registrar gateway is busy or otherwise unavailable Registrar IP n n n n This is the IP address of the registrar VOIP to which this client is assigned Address Default value 0 0 0 0 effectively there is no useful default value Registrar Port 10000 or This is the port number of the registrar VOIP to which this client is assigned other Default port number 10000 Polling Interval integer The interval between the VOIP gateway s successive attempts to connect to and 60 300 be governed by a higher level SPP registrar gateway The Primary Regi
108. er to maintain the in connection If the MultiVOIP does not register before the TTL interval expires the seconds MultiVOIP gateway s registration with the proxy server will expire and the proxy server will no longer permit call traffic to or from that gateway Calls in progress will continue to function even if the gateway becomes de registered Multi Tech Systems Inc 56 SIP Server Configuration Chapter 4 Configuring your VOIP The MultiVOIP 210 410 810 SS models have the additional capability of SIP survivability The settings for SIP server mode are detailed below SIP Server Configuration Operating Mode Survivability Status Check C Survivability Standalone Server Register Registrar Options OK Cancel I Allow Undefined Registrations Any Domains C Specific Domains Any IP address Specific IP address 3600 Note Multiple Domain names and IP addresses can be entered by separating with a semicolon Accept Registrations For Domain Names Accept Registrations For IP Addresses Re registration Time Figure 4 15 SIP Server configuration SIP Server Configuration Parameter Definitions Field Name Values Description Operating Survivability In Survivability mode the MVP SS unit can function as a SIP server for other Mode or gateways in its network in case that network loses contact with the network s main SIP stand alone
109. ermissions Read Only If this community needs to change MultiVOIP settings select Read Write Read Write Otherwise select Read Only to view settings Multi Tech Systems Inc 61 Chapter 4 Configuring your VOIP Regional The Regional Parameters are used to set the phone signaling tones and cadences For the country selected the standard set of frequency pairs will be listed for dial tone busy tone unobtainable tone fast busy or trunk busy ring tone and other more specialized tones If you need settings that are not available the Custom selection will let you set the tones to what is necessary The Regional Parameters fields are described in the table below ustom Tone Pair Settings Tone Pair DialTone X Tone Pair Values Erequency1 350 Hz Cadencel 0 ms Frequency2 440 Hz Cadence2 0 ms Regional Parameters Country Region Custom z Standard Tones Frequencyl 350 Frequency2 Cadence secs On Off 440 0 000 0 000 0 000 0 000 o RingTone 480 440 2 000 4 000 2 000 4 000 16 16 oe oe gt ms BusyTone 480 620 0 500 0 500 0 500 0 500 16 16 nido oo BE z Cadenced 0 UnobtainableTone 480 620 0 000 0 000 0 000 0 000 16 16 i ooo Survivability DialTone 650 650 0 000 0 000 0 000 0 000 16 16 peta Gain2 16 y d8 Cadences 0 me ReorderT one 480 620 0 250 0 250 0 000 0 000 16 16 Help adii Add Edit Tone Tone Type Disconnect Frequency
110. ers M Enable SNMP Agent Trap Manager Address 0 0 0 0 OK Community Name Cancel Port Number 1162 Help Community Name 1 public Permissions Read Only Community Name 2 supervisor Permissions Read write Figure 4 18 SNMP parameters screen The SNMP Parameter fields are described in the table below SNMP Parameter Definitions Field Name Values Description Enable SNMP Y N Enables the SNMP code in the firmware of the MultiVOIP This must be Agent enabled for the MultiVOIP to communicate with and be controllable by the MultiVoipManager software Default disabled Trap Manager Parameters Address n n n n IP address of MultiVoipManager PC Community A community is a group of VOIP endpoints that can communicate with each Name other Often public is used to designate a grouping where all end users have access to entire VOIP network However calling permissions can be configured to restrict access as needed Port Number 162 The default port number of the SNMP manager receiving the traps is the standard port 162 Community Length 19 First community grouping Name 1 characters max Case sensitive Permissions Read Only If this community needs to change MultiVOIP settings select Read Write Read Write Otherwise select Read Only to view settings Community Length 19 Second community grouping Name 2 characters max Case sensitive P
111. escription Answer Supervision fields Current Reversal Y N When this option is selected the FXO interface sends notice to make connection upon detecting current reversal from the PBX which occurs when the called extension goes off hook Answer Delay Y N When this option is selected the FXO interface sends the connection notice to the calling party only when the Answer Delay Timer expires The connection notice is sent regardless of whether or not the called extension has gone off hook Answer Delay 1 65535 When Answer Delay is enabled this value determines when the FXO Timer in seconds interface sends the connection notice Tone Detection Y N When selected call disconnection will be triggered by a tone sequence Available Tones dial tone List from which tones can be chosen to signal call answer ring tone busy tone unobtainable tone fast busy survivability tone re order tone Answer Tones any tone from Available Tones list Currently chosen call answer supervision tone Disconnect Supervision fields There are four possible criteria for disconnection under FXO current reversal current loss tone detection and silence detection Disconnection can be triggered by more than one of the three criteria Current Reversal Y N Disconnection to be triggered by reversal of current from the PBX Current Loss Y N Disconnection to be triggered by loss of current That is when Curren
112. esources available in the command PC the Error in Opencomm handle message will appear when the MultiVOIP program is launched If this occurs you must reset the COM port 8 Acompletion screen will appear Multi Tech Systems InstallShield Wizard Complete Setup has finished installing Multi VOIP on your computer Figure 3 4 Completion Click Finish 9 When setup of the MultiVOIP software is complete you will be prompted to run the MultiVOIP software to configure the VOIP MultiVOIP 6 11 2 Do you want to run Multi OIP Configuration Figure 3 5 Configuration Software installation is now complete Multi Tech Systems Inc 18 Chapter 3 Software Installation Setup Overview With the software now installed you are ready to get your MultiVOIP set up and working There are a few necessary settings that need to be entered in the configuration software to achieve this and they are noted in the action lists for the categories below The following chapters will cover all aspects in detail but here we will cover the basic configuration needed to start VOIP communications Below you will find the list of categories requiring information to be set before VOIP communication will be ready Ethernet IP Voice Fax Interface Call Signaling Regional Phone Book This setup process is followed by the Save amp Reboot step which is very important Uuuu MultiVOIP v6 11 Configuration Advanced Phone Boo
113. et Explorer wI MSN gt Windows Media Playe 33 Windows Messenger D Tour Windows XP All Programs fa MultivOIP 6 11 E Configuration a Configuration Port Setup a Date and Time Setup 5 Download Factory Defaults ES Download Firmware 2 Download IFM Firmware 2 Download User Defaults a Set Password Uninstall 5 Upgrade Software Log OFF o Turn OFF Computer Figure 6 33 Upgrade software path NOTE To upgrade a MultiVOIP from software version 6 04 or earlier an ftp primer file must first be sent to the VOIP This file is located in the Software ftp_Primer folder on the CD and the file name is FTP_Primer bin Before uploading this file it must be renamed mvpt1ftp bin The VOIP will only accept files of this name This is a safety precaution to prevent the wrong files from being uploaded to the VOIP Once the primer file has been uploaded upload the FTP firmware file If you accepted the defaults during the software loading process this file is located on your local drive at C Program Files Multi Tech Systems MultiVOIP X NN where the X is the software number and the NN is the version number of the MultiVOIP software on your local drive Of course the firmware file is named mvpttftp bin Important You cannot go back to 6 04 or earlier versions using FTP You must use upgrade software via the serial port Important These ftp upgrade instructions do not apply to software release 6 05 and
114. f digits 1234567890 Call Hold Enable Y N Select to enable Call Hold function in 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 Hold Sequence phone keypad The numbers and or symbols that the caller must press on the phone characters keypad to initiate a call hold The call hold sequence can be 1 to 4 characters in length using any combination of digits or characters or Call Waiting Enable Y N Select to enable Call Waiting function in VOIP unit Retrieve Sequence Phone keypad characters two characters in length The numbers and or symbols that the caller must press on the phone keypad to initiate retrieval of a waiting call The call waiting retrieval sequence can be 1 to 4 characters in length using any combination of digits or characters or This is the phone keypad sequence that a user must press to retrieve a waiting call Customize able Sequence should be distinct from sequence that might be used to retrieve a waiting call via the PBX or PSTN Call Name Identification Enable Enables CNI function Call Name Identification is not the same as Caller ID When enabled on a given VOIP unit currently being controlled by the MultiVOIP interface the home VOIP Call Name Identification
115. figuration An outline of the equipment setup in these three offices is shown below Wren Clothing Co London Office Country Code 44 Area Code 0208 Wren Clothing Co Paris Office Country Code 33 Area Code 01 Wren Clothing Co Amsterdam Office Country Code 31 Area City Code 020 74 71 2981 4801 688 4800 Figure 5 17 Setup example Multi Tech Systems Inc 95 Chapter 5 Phonebook Configuration The screen below shows Outbound Phone Book entries for the VOIP located in the company s London facility Dutbound Phone Book Destination Pattern IP Address Alternate 003110 200 2 8 5 H323 Rotterdam 200 297 H 323 Paris company office emp extensions 200 2 8 5 H 323 Amsterdam company office employees Number of Entries 6 Details Remove Prefix Add Add Prefix a not used Edit Delete ae i Help Gateway H 323 ID Gateway Prefix H 323 Port 1720 Close Delete Round Trip Delay 300 ms Figure 5 18 London example outbound The Inbound Phone Book for the London VOIP is shown below Inbound Phone Book 00441483 9 01483 Not Used 0044207 9 7 Not Used 0044208 9 8 Not Used 00442089795 5 Not Used 5 5 Not Used Number of Entries 5 Details Add Channel No Hunting Edit Description Delete Reaistration Options Close Help Subscription Options Figure 5 19 London example inbound NOTE Commas are allowed in the Inbound Phonebook but not in th
116. files can still be sent to the VOIP administrator via email which requires using the SMTP logging option Logs Screen Definitions Field Name Values Description Enable Console Y N Allows MultiVOIP debugging messages to be 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 IT support personnel Filters button Click to access secondary screen on where console messages can be included excluded by category and on a per channel basis Turn Off Logs Y N Check to disable log reporting function Logs Buttons Only one of these three log reporting methods GUI SMTP or SNMP may be chosen GUI User must view logs at the MultiVOIP configuration program SNMP Log messages will be delivered to the MultiVoipManager application program SMTP Log messages will be sent to user specified email address SysLog Server Y N This box must be checked if logging is to be done in conjunction with a SysLog Server Enable program IP Address n n n n IP address of computer in VOIP network on which SysLog Server program is running Port 514 Logical port for SysLog Server 514 is commonly used Online Statistics integer Set the interval in seconds at which logging information will be updated Updation Interval Multi Tech Systems Inc 70 Chapter 4 Configuri
117. here local calling rates extend beyond a single area code Local rates between Inner and Outer London are one example of this It is also possible in some locations that calls within an area code may be national calls but this is rare _ Wren Clothing Co VOIP PBX Site Wren Clothing Co Bluebird Apper Co London eee VOIP PBX Site ongon d lt gt Hal aprenen Amsterdam Nx AV The Netherlands Wren Clothing Co VOIP PBX Site YX Paris AS Calls at London local rates y Local Calling Area Sa a France Figure 5 14 Local calling area Multi Tech Systems Inc 93 Chapter 5 Phonebook Configuration This next example will have the following features e Employees in all cities will be able to call each other over the VOIP system using 4 digit extensions e Calls to Outer London and Inner London greater Amsterdam and greater Paris will be accessible to all company offices as local calls e Vendors in Guildford Lyon and Rotterdam can be contacted as national calls by all company offices France Country Code 33 Toulouse Marseille The Netherlands Country Code 31 e 050 Groningen Lefguwarden 9 Den Helder 0223 038 Zwolle Beverwijy 0251 9799 Pdrmerend Haarlem 023 lt I Aalsmee A 20 Amsterdam 0294 Weesp ER e 026 e 010 Arnhem Rotterdam e 0118 Middelburg O 040 Eindhoven Figure 5 16 Netherlands codes Multi Tech Systems Inc 94 Chapter 5 Phonebook Con
118. ied in the Phone Number box of this option If the Pass Through Enable field is checked in the Interface Parameters screen AutoCall must be used The Offhook Alert option applies only to FXS channels The Offhook Alert option works like this if a phone goes off hook and yet no number is dialed within a specific period of time as set in the Offhook Alert Timer field then that phone will automatically dial the Alert phone number for the VOIP channel The Alert phone number must be set in the Voice Fax Parameters Phone Number field if the VOIP system is working without a gatekeeper unit there must also be a matching phone number entry in the Outbound Phonebook One use of this feature would be for emergency use where a user goes off hook but does not dial possibly indicating a crisis situation The Offhook Alert feature uses the Intercept Tone as listed in the Regional Parameters screen This tone will be outputted on the phone that was taken off hook but that did not dial The other end of the connection will hear audio from the crisis end as is it would during a normal phone call Both functions apply on a channel by channel basis It would not be appropriate for either of these functions to be applied to a channel that serves in a pool of available channels for general phone traffic Either function requires an entry in the Outgoing phonebook of the local MultiVOIP and a matched setting in the Inbound Phonebook of the remo
119. ield Any Number can be used in addition to one or more Prefixes Remove Prefix dialed digits portion of dialed number to be removed before completing call to destination often a local PBX Add Prefix dialed digits digits to be added before completing call to destination often a local PBX Channel channel or Channel number to which the call will be assigned as it enters the local telephony Number Hunting equipment often a local PBX Hunting directs the call to any available channel Description Describes the facility or geographical location at which the call originated Call Forward Parameters Enable Y N Click the check box to enable the call forwarding feature Forward Unconditional Unconditional When selected all calls received will be forwarded Condition Busy Busy When selected calls will be forwarded when station is busy No Response No Response When selected calls will be forwarded if called party does not answer after a specified number of rings as specified in Ring Count field Forwarding can be conditioned on both Busy and No Response Forward IP address Phone number or IP address to which calls will be directed Destination phone number For H 323 calls the Forward Destination can be either a Phone Number or an IP port number Address ee For SIP calls the Forward Destination can be one of the following a phone number b IP address c IP address port number d phone number IP addres
120. ils Supplementary Services Status Prefx Malched Cal On Hold Outbound Digits Sent Cal Wating Outbound Digts Revd Calles 1d Server Delads DTMF Capabdity Cal Status On Hook Cal Control Status SC Silence Compression FEC Forward Enor Comection Figure 6 3 Call progress screen Multi Tech Systems Inc 104 Chapter 6 Using the Software Call Progress Details Field Definitions Field Name Values Description Channel 1 n Number of data channel or time slot on which the call is carried This is the channel for which call progress details are being viewed Call Details Duration H M S The length of the call in hours minutes and seconds hh mm ss Mode Voice or FAX Indicates whether the call being described was a voice call or a FAX call Voice Coder G 723 G 729 The voice coder being used on this call G 711 etc IP Call Type H 323 SIP or Indicates the Call Signaling protocol used for the call H 323 SIP or SPP SPP The SS and FX series only support SIP IP Call Direction incoming Indicates whether the call in question is an incoming call or an outgoing outgoing call Packet Details Packets Sent integer value The number of data packets sent over the IP network in the course of this call Packets Revd integer value The number of data packets received over the IP network in the course of this call Bytes Sent integer value The number of bytes
121. ing in seconds before the TimeToLive timer expires If the gateway Duration fails to reregister within this time the endpoint is unregistered Status Registered The current status of the gateway either registered or unregistered unregistered No of Entries The number of gateways currently registered to the Registrar This includes all SPP clients registered and the Registrar itself Details Count of If a registered gateway is selected by clicking on it in the screen The Count of Registered Registered Numbers will indicate the number of registered phone numbers for the Numbers selected gateway When a client registers all of its inbound phonebook s phone numbers become registered List of Lists all of the registered phone numbers for the selected gateway Registered Numbers Multi Tech Systems Inc 111 Chapter 6 Using the Software Servers H 323 GateKeepers The SS and FX series of MultiVOIPs do not support H 323 H 323 GateKeepers IP Address Pot GKName 192 168 3 1 1719 Primary Exit Figure 6 8 H 323 Gatekeepers H 323 Gatekeepers Statistics Servers Field Definitions Field Name Values Description Column Headings IP Address n n n n The IP address of the gatekeeper Port n TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it GK Name alpha Identifier for gatekeeper n
122. ings on the VOIP unit you must choose the Save amp Reboot option otherwise all changes made will be lost when the MultiVOIP is reset or shutdown Multi Tech Systems Inc 29 Chapter 4 Configuring Your MultiVOIP Introduction There are two methods of using your MultiVOIP one is through a web interface and the other is through the Windows software interface There are eight necessary parameters that must be set for the MultiVOIP unit to operate properly with some additional settings that are optional You must know the IP address that will be used the IP mask the Gateway IP the Domain Name Server information and the telephone interface type The MultiVOIP must be configured locally at first but changes to this initial configuration can be done locally or remotely Local configuration is done through a connection between the Command port of the MultiVOIP and the COM port of the computer the MultiVOIP configuration software is used for this Alternatively MultiVoipManager is a Simple Network Management Protocol SNMP agent program that extends the capabilities of the MultiVOIP configuration software MultiVoipManager allows the user to manage any number of VOIPs on a network whereas the MultiVOIP configuration software manages only one The MultiVoipManager can configure multiple VOIPs simultaneously MultiVoipManager may reside on the same PC as the MultiVOIP configuration software This chapter will explain the setup por
123. ion will be more accepting of jitter A lower value is less tolerant of jitter Inactive by default When active default 300 ms However value must equal or exceed Dynamic Minimum Jitter Value Call Duration 1 65535 Call Duration defines the maximum length of time in seconds that a call remains connected before the call is automatically disconnected Inactive by default When active default 180 sec This may be too short for some configurations requiring upward adjustment Consecutive Packets 1 65535 Consecutive Packets Lost defines the number of consecutive packets that Lost are lost after which the call is automatically disconnected Inactive by default When active default 30 Network 1 to 65535 Specifies how long to wait before disconnecting the call when IP network Disconnection Default connectivity with the remote site has been lost 30 sec Configurable Payload Type Not available on the SS series The Configurable Payload Type is located on the bottom of the Voice Fax screen The Configurable Payload Type is used when the remote side uses a different payload type for the associated features In previous firmware versions MultiVOIP s used 101 for DTMF RFC2833 If the remote side uses some other dynamic payload type such as 110 it will fail To avoid these failures the payload types are made configurable DTMF RFC2833 Configurable Payload Type is supported only for SIP amp SPP and
124. itted without acknowledgment guaranteed delivery or guaranteed packet sequence integrity TCP is slower connection oriented protocol with greater overhead but having acknowledgment and guarantees delivery and packet sequence integrity SIP Port 5060 or other The SIP Port Number is a UDP logical port number The VOIP will listen Number See RFC 3087 Control of for SIP messages at this logical port If SIP is used 5060 is the default Service Context using SIP standard or well known port number to be used If 5060 is not used then Request URI by the the port number used is that specified in the SIP Request URI Universal Network Working Group Resource Identifier SIP URL sip userphone hostserver Looking similar to an email address a SIP URL identifies a user s where userphone is the address telephone number and In SIP communications each caller or callee is identified by a SIP URL hostserver is the domain sip user_name host_name The format of a sip URL is very similar to an name or an address on the email address except that the sip prefix is used network SPP Fields The SS series of MultiVOIPs do not support SPP Use Y N Select this checkbox to use registrar when VOIP system is operating in Registrar the Registrar Client SPP mode In this mode one VOIP the registrar as set in Phonebook Configuration screen has a static IP address and all other VOIPs clients point to the registrar s I
125. jadaasuanaptuaesiduseeenausaneeniagenmuaeys 99 Chapter 6 Using the SOWA a 102 TO CU CUO acs sexes cngze dzsuasdcazsannaisaenmpacsuauedaisateapacdaceteaiapationsacassneaaawasoudadaseepatanatnneadcesnasbadanenbaddcntanianateesMaseeantaseeeds 102 Software Categories Covered in This Grappler seer narran a la o iaa tii ita 102 A O 104 Call Progress cassete 104 oro E EA E E E E cp vc ace fasta vet EA E E E E tals sbtca cas E E ET 106 VP SSRIS lso 108 Linke Management ii e le tidad 110 Registered Gateway Detalle ido 111 A A eee rete ree ereer tre ree reer rere 112 Foa Gale RCSD CIS aiani AE O E a E Ia 112 SIP POKIG Siscie n errea E E E E EA EE 113 A 114 Advanced ua Re eee Ret AEE ER 115 Packetization Tie oi a es 115 MultvOlP Program Men lemas po alos lios a 116 Updating FITMWAare escindida 117 implementing a Sottware Upgrade ecuatorianas 118 Identihying Gurrent Fimiware V STSION osacaicio oia aiii 118 Downloading UF IGINIWARS 55 525 2055 fie seas ia iio 119 Downloading Factory Defaults c2x 2 5 2 ezine celica tel tecno bacilo lia tot dc ll abel 120 Multi Tech Systems Inc 4 Downloading IEM IFirtnWateincc opi leia eh a e 121 Setting and Downloading User DefaulltS cocoa nina ia bis 123 Setting a PassWord iia aia 124 Windows Inti ao ere ere 124 Wem Browser laiicin tc era el Dee erie 125 WG rac NGM Soares li dla iodo do lll db 126 FTP Server File Transfers Downloads Tesista maana tii 127 Web Browser INtedace nin tar 132 SysLog
126. k Miami and Los Angeles each served by its own PBX When the VOIP phone books are set correctly personnel in the Miami office should be able to make calls without toll not only to the company s offices in New York and Los Angeles but also to any number that s local in those two cities To achieve transparency of the VOIP telephony system and to give full access to all types of non toll calls made possible by the VOIP system the VOIP administrator must properly configure the Outbound and Inbound phone books of each VOIP in the system The Outbound phonebook for a particular VOIP unit describes the dialing sequences required for a call to originate locally typically in a PBX in a particular facility and reach any of its possible destinations at remote VOIP sites including non toll calls completed in the PSTN at the remote site The Inbound phonebook for a particular VOIP unit describes the dialing sequences required for a call to originate remotely from any other VOIP sites in the system and to terminate on that particular VOIP Briefly stated the MultiVOIP s Outbound phonebook lists the phone stations it can call its Inbound phonebook describes the dialing sequences that can be used fo call that MultiVOIP and how those calls will be directed The phone numbers are not literally listed individually but are instead described by rule Identify Remote VOIP Site to Call When you re done installing the MultiVO
127. k Statistics Download Connection Help HE ES Configuration Advanced a l Phone Book Statistics le s Murti Save Setup Voice Fax over IP Networks Connection a Shela Multi Tech Systems Inc 2205 Woodale Drive Mounds View MN 55112 USA TEL 763 785 3500 FAX 763 785 3702 Tech Support 800 972 2439 Web Site http www multitech com FTP Site ftp 2ftp multitech com Tech Support tsupport multitech com 99999999 Rights No Access Figure 3 6 Main Screen Multi Tech Systems Inc 19 Chapter 3 Software Installation Ethernet IP A unique LAN IP address is required for the MultiVOIP unit as well as a subnet mask and Gateway IP for minimal functionality Other settings in this category pertain to specific features and protocols that can be used but are not necessary for basic operation Details for all settings are provided in chapter 4 Ethernet IP Parameters Ethernet Parameters V Packet Prioritization 802 1 p Frame Type TYPE I 802 1p Parameters Priority Call Control 6 Voice ba VoIP Media 3 Excellent Effort v Cancel Others O Best Effort z VLAN ID 1 Help rik IP Parameters Gateway Name MultivolP Enable DHCP Diff Serv Parameters y Call Control PHB 34 IP Address 192 168 3 143 VoIP Media PHB 46 IP Mask 255 255 255 0 FTP Sever V Enable Gateway Enable SRY DNS Server IP Address TDM Routing Option F Use TDM Routing For I
128. l 2 of the Omaha VOIP receives a call from any other VOIP phone station for example the Denver office the message Alerting Party Omaha Sales Office Voipchannel 2 will be sent back and will appear in the Caller Id field of the Statistics Call Progress screen of the Denver VOIP This confirms to the Denver VOIP that the phone is ringing in Omaha Busy Party Allowed Name Type CNI If the home VOIP unit is receiving a call directed toward an already engaged channel or phone station and Busy Party is selected then the identifier from the Caller Id field will tell the originating remote VOIP unit that the channel or called party is busy This field is applicable only when the home VOIP unit is receiving the call Example Suppose a VOIP system has offices in both Denver and Omaha In the Omaha VOIP unit the home VOIP unit in this example Call Name Identification has been enabled Busy Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field of the Supplementary Services screen When channel 2 of the Omaha VOIP is busy but still receives a call attempt from any other VOIP phone station for example the Denver office the message Busy Party Omaha Sales Office Voipchannel 2 will be sent back and will appear in the Caller Id field of the Statistics Call Progress screen of the Denver VOIP This confirms to the Denver VOIP that the
129. ld Password Cancel New Password Reconfirm Password Figure 6 34 Change password As shown above the user name and password can be set in the web interface as well as in the Windows interface Multi Tech Systems Inc 127 Chapter 6 Using the Software 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 unit at a time With an FTP client program multiple VOIPs can receive FTP file transmissions in response to a single command the transfers may occur serially however Although Multi Tech does not provide an FTP client program with the MultiVOIP software or endorse any particular FTP client program we remind our readers that adequate FTP programs are readily available under retail shareware and freeware licenses Read and observe any End User License Agreement carefully Two examples of this are the WSFTP client and the SmartFTP client with the former having an essentially text based interface and the latter having a more graphically oriented interface as of this writing User preferences will vary 4 Enable FTP Functionality Go to the IP Para
130. lide the main circuit board out of the chassis far enough to unplug the power connector IATA E E A UA A power connector Figure C 4 Accessing the power connector 5 Unplug the power connector from the main circuit board 6 Slide the main circuit board completely out of the chassis and place on a non conductive static safe tabletop surface 7 Remove mounting hardware 2 screws 2 nuts and 4 standoffs from its package 8 On the phone jack side of the circuit card three screws attach the circuit card to the back panel Two of these screws are adjacent to the four phone jack pairs Remove these two screws Screw locations 2 at phone jack edge of board Figure C 5 Screws replaced with standoffs 9 Replace these two screws with standoffs Multi Tech Systems Inc 138 Appendix C MVP428 Upgrade Card 10 There are two copper plated holes at the LED edge of the circuit card Place a nut beneath each hole lock washer side should be in contact with board and attach a standoff to each location Standoff locations 2 at LED edge of board top view Stan doff nut attachment rear bottom view Figure C 6 Standoffs at LED edge of board 11 Locate the male 60 pin vertical connector near the LED edge of the main circuit card Check that pins are straight and evenly spaced If not then correct for straightness and spacing Locate the 60 pin female connector on the upgrade circuit card 12 Set
131. lure Displays the status of the call whether the call was completed or not IP Direction incoming Indicates whether the call is incoming or outgoing with respect to the outgoing gateway Mode column voice or FAX Indicates whether the event being described was a voice call or a FAX call From column gateway name Displays the name of the voice gateway that originates the call To column gateway name Displays the name of the voice gateway that completes the call Special Buttons Previous Displays log entry before currently selected one Next Displays log entry after currently selected one First Displays first log entry Last Displays last log entry Delete File Deletes selected log file Call Details Voice coder Coder protocol The voice coder being used on this call Disconnect Reason Normal or Indicates whether the call was disconnected simply because the desired Local conversation was done or some other irregular cause occasioned disconnection disconnection e g a technical error or failure DTMF Capability inband Indicates whether the DTMF dialing digits are carried Inband or Out of out of band Band The corresponding field values differ for the 3 different VOIP Expressions differ slightly for protocols For H 323 this field can display Out of Band or Inband For SIP it can different Call display either Out of Band RFC
132. manipulation User CID the CID received from PSTN will be replaced by this User CID value Prefix the CID received from PSTN will be prefixed with this value Suffix the CID received from PSTN will be suffixed with this value Multi Tech Systems Inc 45 Chapter 4 Configuring your VOIP 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 FXO Supervision Answer Supervision Current Reversal V Answer Delay Answer Delay Timer 12 secs V Tone Detection OK Available Tones Answer Tones Cancel BusyT one A RingT one DialT one 3 gt gt InterceptT one a ReorderT one lt lt Gurviwahilihu NialT ane Mi Disconnect Supervision Current Reversal MV Current Loss Current Loss Timer 500 ms Silence Detection V Enable iis Onewa y Silence Timer 115 secs Y DTMF Tone Disconnect Tone Seguence y None y Available Tones Disconnect Tones DialTone A BusyT one InterceptT one A gt gt ReorderT one RingT one Qu al Survivability DialT one lina ls Tn lt Figure 4 7 FXO Supervision The table below describes the settings for FXO Supervision Multi Tech Systems Inc 46 Chapter 4 Configuring your VOIP FXO Supervision Parameter Definitions Field Name Values D
133. me in seconds Proxy Polling Interval time between proxy server connect attempts e TTL Value in seconds o SPP not supported by SS series Mode Direct Client or Registrar Signaling Port must be unique for any VOIP unit behind same firewall Retransmission time before retransmission of lost packets Max Retransmission number of retransmission attempts Client Options e Enter information for the Primary and Alternate Registrars e Polling Interval time between connect attempts Keep Alive time out for client un registering Behind Proxy NAT device e Enter Public IP of Proxy NAT server Multi Tech Systems Inc 26 Chapter 3 Software Installation Regional Select the country or region that the MultiVOIP unit will operate in or use the custom option if the available settings are not adequate Regional Parameters a alerce E i 350 0 00040 00040 00040 000 RingTone 430 440 2 000 4 000 2 000 4000 16 enced BusyTone 430 620 0 500 0 509 0 500 0 500 16 Lorca UnoblamableT one 490 620 0 000 0 000 0 000 0 000 16 Defaut Survivably DialTone 650 650 0 000 0 000 0 000 0 000 16 RecederT one 490 620 0 25040 25040 00040 000 16 InterceptT one 440 0 0 024 0 02440 00040 000 B _Heo lt gt Ele dit i Figure 3 11 Regional Parameters Actions e Select the choice that matches the location of the MultiVOIP from the Country Region field o If there is not a selection to fit your needs you may select Custom and se
134. meters screen and click on the FTP Server Enable box Ethernet IP Parameters Ethernet Parameters Y Packet Prioritization 802 1 p Frame Type TY PE II y 802 1p Parameters Priority OK Call Control 6 Voice v 7 VolP Media 3Excellent Effort v _ Cancel Others 0 Best Effort v Help VLAN ID 1 IP Parameters Gateway Name Multi olP M Enable DHCP Diff Serv Parameters Call Control PHB IP Address 192 168 3 143 34 ae WolP Media PHB IP Mask 255 255 255 0 FTP Server V Enable Gateway Figure 6 35 Enable FTP server Multi Tech Systems Inc 128 Chapter 6 Using the Software 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 So for example the file mvpt1Ftp bin can be transferred to update the bin file firmware residing in the MultiVOIP Similarly the file fxo_loopFtp cas could be transferred to enable use of the FXO Loop Start telephony interface in one of the analog VOIP units and the file r2_brazilFtp cas could be transferred to enable a particular telephony protocol used in Brazil Note however that before any CAS file can be used as an update it must be renamed to CASFILE CAS so that it overwrites and replaces the default CAS file
135. n 00441483 200 210 3 H 323 Guildford 0044207 200 210 3 H 323 London Inner 0044208 200 210 3 H 323 London Outer 2 200 2 9 7 H 323 Paris company office employee ext 5 200 210 3 H 323 London company office empl ext Number of Entries 7 Details Remove Prefix Add Add Prefix not used Edit Delete H 323 Port 1720 Close dad Help Round Trip Delay 300 ms Figure 5 22 Amsterdam example outbound The Inbound Phone Book for the Amsterdam VOIP is shown below Inbound Phone Book Add Prefix Forward Address ot Use Not Used 4 Not Used Number of Entries 4 Details Add Channel No Hunting Edit Description Delete Registration Options Close dado Help Subscription Options Figure 5 23 Amsterdam example inbound Multi Tech Systems Inc 98 Chapter 5 Phonebook Configuration Variations of Caller ID The Caller ID feature has dependencies on both the telco central office and the MultiVOIP phone book See the diagram series below CID Flow Call originates here at 1 42pm May 31 Call is received here CID Generating Central Office VoIP with standard telephony Caller ID service phone of dico n Melvin Jones CID Number 763 555 8794 763 555 8794 CID Name Melvin Jones Time Stamp Date 05 31 Time 1 42pm Display shows In x 06 release when SIP protocol is used CID Name field will duplicate value in CID Number field Figure 5 24 Caller
136. n n IP address of the VOIP gateway from which the call was received Options FEC SC Displays VOIP transmission options used by the VOIP gateway originating the call These may include Forward Error Correction or Silence Compression TO Details Gateway Name alphanumeric Identifier for the VOIP gateway that completed terminated this call IP Address n n n n IP address of the VOIP gateway at which the call was completed Options Displays transmission options used by VOIP gateway terminating the call Supplementary Services Info Call Transferred To phone number Number of party called in transfer Call Forwarded To phone number Number of party called in forwarding Multi Tech Systems Inc 107 Chapter 6 Using the Software IP Statistics IP Statistics Total Packets Cl Transmitted 0 Received 0 _ Clear UDP Packets Exit Transmitted 0 Received 0 Help Received with Errors 0 TCP Packets Transmitted 0 Received 0 Retransmitted 0 Received with Errors 0 RTP Packets Transmitted 0 Received 0 Received with Errors 0 RTCP Packets Transmitted 0 Received 0 Received with Errors 0 Figure 6 5 IP statistics screen UDP versus TCP User Datagram Protocol versus Transmission Control Protocol UDP provides unguaranteed connectionless transmission of data across an IP network By contrast TCP provides reliable connection oriented transmission of data Both TCP and UDP
137. n the FTP client program window drag and drop files from the local browser pane to the pane for the MultiVOIP FTP server FTP client interface operations vary In some cases you can choose between immediate and queued transfer In some cases there may be automated capabilities to transfer to multiple destinations with a single command 9 Verify Transfer The files transferred will appear in the directory of the MultiVOIP P ftp voip1 192 168 2 200 Microsoft Internet Explorer A File Edit View Go Fe ae z Address Tip 92 jac A A A aAa casfile cas factdef cnf H323 pdl myptlftp bin QutPhBk tmr InPhBk tmr Figure 6 41 Verify transfer 10 Log Out of FTP Session Whether the file transfer was done with a web browser or with an FTP client program you must log out of the FTP session before opening the MultiVOIP Windows interface Multi Tech Systems Inc 131 Chapter 6 Using the Software Web Browser Interface y MultiVOIP 410 v6 06 Firmware Aug 26 2006 Microsoft Internet Explorer File Edit View Favorites Tools Help El 2929243232935 395 3 J Back fever _ Stop Refresh Home Mail Print Edit Discuss a Address 45 http 204 26 122 105 v Go Links 45 Best of the Web ChannelGuide 4 Customize Links FreeHotMail 45 Internet Stat 45 Microsoft 4 Windows Update z a MultivOIP 410 Configuration Advanced Phone Book Statistics Change Password Save amp Re
138. nection DID Example DID fax system or DID voice phone lines Connect one end of an RJ 11 phone cord to the Channel 1 FXS FXO connector on the back of the MultiVOIP Connect the other end to the DID jack NOTE DID lines are polarity sensitive If during testing the DID line rings busy consistently you will need to reverse the polarity of one end of the connector swap the wires to the two middle pins of one RJ 11 connector 4 Repeat the above step to connect the remaining telephone equipment to the second channel on your MultiVOIP 5 Ensure that the unit is properly connected to earth ground by verifying that it is reliably grounded when mounted within a rack This can be accomplished by connecting a grounding wire between the chassis and a metallic object that will provide an electrical ground 6 Turn on power to the MultiVOIP by placing the ON OFF switch on the back panel to the ON position Wait for the BOOT LED on the MultiVOIP to go off before proceeding This may take a few minutes 7 Proceed to the Software Installation chapter to load the MultiVOIP software Multi Tech Systems Inc 11 Chapter 2 Installing and Cabling the MultiVOIP For DID channels only For any channel on which you are using the DID interface type you must change the jumper on the MultiVOIP circuit card DID is not supported on the SS or FX models 1 2 Disconnect power Unplug the AC power cord from the wall outlet or from the receptacle on
139. ng your VOIP NAT Traversal Setting the NAT Traversal parameters NAT Network Address Translation parameters are applicable only when the MultiVOIP is operating in SIP mode STUN Simple Traversal of UDP through NATs Network Address Translation is a protocol for assisting devices behind a NAT firewall or router with their packet routing This is not available on the SS models NAT Traversal STUN Server Name IP 0 0 0 0 Port 3478 Cancel Timers Keep alive 60 secs Help FEE Figure 4 23 NAT Traversal Descriptions for NAT Traversal screen fields are presented in the table below NAT Traversal Definitions Field Name Values Description Enable STUN Y N Enables STUN client functionality in the MultiVOIP STUN Simple Traversal of UDP through NATs Network Address Translation is a protocol that allows a server to assist client gateways behind a NAT firewall or router with their packet routing Name IP Server n n n n IP address of the STUN server Port Server port The data port TDM time slot at which STUN info will be transmitted and NAT STUN default received 3478 Keep Alive Timers 60 3600 The interval at which the STUN client sends indicator Keep Alive packets to NAT STUN seconds the STUN server to determine whether or not the STUN server is available Multi Tech Systems Inc 71 Chapter 4 Configuring your VOIP Supplementary Services
140. ngle Port Protocol or SPP SPP is a non standard protocol designed by Multi Tech The SS models only support SIP and the FX models do not support H 323 H 323 fields The SS and FX models do not support H 323 Use Gatekeeper Y N Indicates whether or not gatekeeper is used Gateway H 323 ID alpha numeric The H 323 ID assigned to the destination MultiVOIP Only valid if Use Gatekeeper is enabled for this entry Gateway Prefix numeric This number becomes registered with the GateKeeper Call requests sent to the gatekeeper and preceded by this prefix will be routed to the VOIP gateway H 323 Port 1720 This parameter pertains to Q 931 which is the H 323 call signaling protocol Number for setup and termination of calls aka ITU T Recommendation 1 451 H 323 employs only one well known port 1720 for Q 931 signaling If Q 931 message oriented signaling protocol is used 1720 must be chosen as the H 323 Port Number Table is continued on next page Multi Tech Systems Inc 83 Chapter 5 Phonebook Configuration Add Edit Outbound Phone Book Field Definitions continued Field Name Values Description SIP Fields Use Proxy Y N Select if proxy server is used Transport TCP or VOIP administrator must choose between UDP and TCP transmission Protocol UDP protocols UDP is a high speed low overhead connectionless protocol where data is transm
141. nt VOIP Alternate GK A first and a second alternate gatekeeper can be specified for use by the current 1 and 2 VOIP for situations where the Primary GK is busy or otherwise unavailable IP Address n n n n IP address of the GateKeeper RAS Port 1719 Well known port number for GateKeepers Must match port number 1719 Gatekeeper alpha Optional The name of the GateKeeper with which this MultiVOIP is trying to Name numeric register A primary gatekeeper and two alternate units are listed RAS TTL Value seconds The H 323 Gatekeeper Time to Live value As soon as a MultiVOIP gateway registers with a gatekeeper a countdown timer begins The RAS TTL Value is the interval of the countdown timer Before the TTL countdown expires the MultiVOIP gateway needs to register with the gatekeeper in order to maintain the connection If the MultiVOIP does not register before the TTL interval expires the MultiVOIP gateway s registration with the gatekeeper will expire and the gatekeeper will no longer permit call traffic to or from that gateway Calls in progress will continue to function even if the gateway becomes de registered Gatekeeper integer The interval between the VOIP gateway s successive attempts to connect to and Discovery Polling 60 300 be governed by a higher level gatekeeper The Primary GK is the highest level Interval gatekeeper Alternate GK1 is second Alternate GK2 is the lowest Use Online Alternate Gatekee
142. ntary Service parameters are set to their default values Copy Channel Copies the Supplementary Service attributes of one channel to another channel Attributes can be copied to multiple channels or all channels at once Multi Tech Systems Inc 74 Chapter 4 Configuring your VOIP Save Settings Save amp Reboot Saving the MultiVOIP Configuration When values have been set for all of the MultiVOIP s various operating parameters click on Save Setup in the sidebar then Save 8 Reboot Creating a User Default Configuration When a Setup complete grouping of parameters is being saved you will be prompted about designating that setup as a User Default setup A User Default setup may be useful as a baseline of site specific values to which you can easily revert Establishing a User Default Setup is optional Connection Settings This is also accessible from the Start menu in the MultiVOIP software folder Set Baud Rate The Connection option in the sidebar menu has a Settings item that includes the baud rate setting for the COM port of the computer running the MultiVOIP software First it is important to note that the default COM port established by the MultiVOIP program is COM1 Do not accept the default value until you have checked the COM port allocation on your PC To do this check for COM port assignments in the system resource manager of your Windows operating system If COM1 is not availa
143. ntifier of the party that is originating the call occurring on a specific channel This field is applicable only when the home VOIP unit is originating the call Example Suppose a VOIP system has offices in both Denver and Omaha In the Omaha VOIP unit the home VOIP in this example Call Name Identification has been enabled Calling Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field When channel 2 of the Omaha VOIP is used to make a call to any other VOIP phone station for example the Denver office the message Calling Party Omaha Sales Office Voipchannel 2 will appear in the Caller Id field of the Statistics Call Progress screen of the Denver VOIP Alerting Party Allowed Name Type CNI If the home VOIP unit is receiving the call and Alerting Party is selected then the identifier from the Caller Id field will tell the originating remote VOIP unit that the call is ringing This field is applicable only when the home VOIP unit is receiving the call Example Suppose a VOIP system has offices in both Denver and Omaha In the Omaha VOIP unit the home VOIP unit in this example Call Name Identification has been enabled Alerting Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field of the Supplementary Services screen When channe
144. ntra Gateway calls Figure 3 7 IP settings Actions e Select Packet Prioritization if used o Set 802 1p Priority Parameters as needed The Priority levels can be from 0 7 where 0 is lowest priority details in Chapter 4 VLAN ID identifies a virtual LAN by a number 1 to 4094 e Set the Frame Type to match the network that the MultiVOIP is attached to o TYPE ll or SNAP e Enter Gateway Name o Check to enable DHCP if used Enter IP Address for the MultiVOIP unit Enter Subnet IP Mask for the MultiVOIP unit Enter Gateway IP Enable DNS if desired o Enter DNS Server IP Address Enable SRV support if needed e Diff Serv Parameters are for routers that are Diff Serv compatible o Setting both values to 0 effectively disables Diff Serv e FTP Server Enable is only needed for firmware and software updates to the MultiVOIP e TDM Routing can be used if necessary Multi Tech Systems Inc 20 Chapter 3 Software Installation Voice Fax The individual channels must be set up before use The Copy Channel button can save a lot of time during this step if channels are to be set with the same parameters Some options should be noted for future changes if necessary but the defaults are likely to work without adjustment Voice Fax Parameters Select Channel Channel 1 X Voice Gain r Fax Modem Parameters OK V Fax Relay Enable 5 Cancel inex j0 te Ove 10 V Modem Relay Enable min Max Baud Rate 14400 Gain AS Copy Channel
145. of data sent over the IP network in the course of this call Bytes Revd integer value The number of bytes of data received over the IP network in the course of this call Packets Lost integer value The number of voice packets from this call that were lost after being received from the IP network From To Details Description Gateway Name alphanumeric Identifier for the VOIP gateway that handled the origination of this call from string IP Address from n n n n IP address from which the call was received Options SC FEC Displays VOIP transmission options in use on the current call These may include Forward Error Correction or Silence Compression Gateway Name to alphanumeric Identifier for the VOIP gateway that handled the completion of this call string IP Address to n n n n IP address to which the call was sent Options SC FEC Displays VOIP transmission options in use on the current call These may include Forward Error Correction or Silence Compression DTMF Other Details Prefix Matched specified Displays the dialed digits that were matched to a phonebook entry dialing digits Outbound Digits Sent 0 9 The digits transmitted by the MultiVOIP to the PBX telco for this call Outbound Digits 0 9 Of the digits transmitted by the MultiVOIP to the PBX telco for this call Received these are the digits that were confirmed as being receive
146. oice packets which the MultiVOIP interprets as a failure of the IP network Upon detecting an excessive delay in transmission of voice packets overly high latency in the network the MultiVOIP diverts the call to another IP address which itself is connected to the PSTN for example via an FXO port on the self same MultiVOIP could be connected to the PSTN PSTN Failover Feature The MultiVOIP can be programmed to divert calls to the PSTN temporarily in case the IP network fails See Figure 5 4 below for example Call completed Call diverts to Alt IP address in voip PSTN Line via PSTN ing PSTN line accessing ine ore vo Reng VOIP exe IP network fails FEX Call originates Figure 5 4 PSTN failover Multi Tech Systems Inc 85 Chapter 5 Phonebook Configuration Inbound Phone Book List Entries The Details and Registration Options sections will display information based on the setup and protocols chosen The Subscription Options area is used in conjunction with a Voice Mail Server Inbound Phone Book Add Prefix 30 Not Used lt m lw Number of Entries 1 Details Channel No Hunting Edit Description Delete Registration Options Close Help dada Subscription Options Figure 5 5 Inbound phonebook entries Multi Tech Systems Inc 86 Chapter 5 Phonebook Configuration Add Edit Inbound Phone Bo
147. ok Add Edit Inbound Phone Book M Accept Any Number Remove Prefix OK Add Prefix Cancel Channel Number Hunting Help Description Call Forward Y Enable ieis Forward Condition F Unconditional Busy M No Response Forward Destination H323 call Phone or IP address SIP call Phone or IP address or IP address port or Phone HIP address port or SIP URL or Ph IP address SPP call Phone or IP address port or Phone IP address port __ Registration Options Figure 5 6 Add Edit Inbound Phone Book Multi Tech Systems Inc 87 Chapter 5 Phonebook Configuration Enter Inbound Phone Book data for your MultiVOIP The fields of the Add Edit Inbound Phone Book screen are described in the table below Add Edit Inbound Phone Book Field Definitions Field Name Values Description Accept Any Y N 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 H 323 protocol Proxy for SIP protocol Registrar for SPP protocol When no external routing device is used If Any Number is selected calls received from phone numbers not matching a listed Prefix shown in the Remove Prefix column of the Inbound Phone Book will be admitted into the VOIP on the channel listed in the Channel Number f
148. ok entry has to be FXS on the Interface screen e The Message Waiting Indication has to be either Light or Stutter Dial Tone on the Interface Parameters screen The password on the Interface screen is used for that particular channel when a SUBSCRIBE request is sent i e if the MultiVOIP gets a 401 407 response from a subscribe request Then it will take the configured password calculate the response and resend the SUBSCRIBE request Multi Tech Systems Inc 43 Chapter 4 Configuring your VOIP FXO Parameters The parameters applicable to the FXO telephony interface type are shown in the figure below and described in the table that follows Interface Parameters Select Channel Channel 1 y Interface Type FXO y Dialing Options B Regeneration Inter Digit Timer 2 secs c El Inter Digit Regeneration Timer 100 ms E DTMF FXO Options F lt 0 Ring Count 2 Flash Hook Options No Response Timer 1180 secs Generation 1600 ms foo Default Help Cancel Caller ID Type Pen gt BellCore v V Enable Supervision Copy Channel adds I M Disable CID Manipulation eo CID Mode Prefix And Suffix v Prefix Suffix g G Figure 4 6 FXO parameters Multi Tech Systems Inc 44 Chapter 4 Configuring your VOIP FXO Interface Parameter Definitions Field Name Values Description Interface Type FXO Enables FXO functionality Dialing Options Regeneration
149. olarity for a specified time 140 290 ms a wink and then becomes ready to receive dial digits For Delay Dial the VOIP detects the off hook condition Then the VOIP reverses battery polarity for a specified time reverse polarity duration has wider acceptable range than for Wink Start and then becomes ready to receive dial digits Wink Timer Integer values This is the length of the wink for Wink Start and Delay Dial signaling in ms in milliseconds modes Applicable only when Start Mode parameter is set to Wink Start or Delay Dial Dialing Options Inter Digit Timer Integer values This is the length of time that the MultiVOIP will wait between digits in seconds When the time expires the MultiVOIP will look in the phonebook for the number entered Default 2 Message Waiting Not applicable to DID DPO interface Indication Inter Digit Integer values This parameter is applicable when digits are dialed onto a DID DPO Regeneration in milliseconds channel after the connection has been made The length of time Timer between the outputting of DTMF digits Default 100 ms Multi Tech Systems Inc 52 Call Signaling Chapter 4 Configuring your VOIP There are three types of Call Signaling available H 323 SIP and SPP Each type has some individual features that may make it more appealing to use than the others depending on your needs The SS and FX models do not support H 323
150. om one location using the MultiVOIP web interface software program or the MultiVOIP program in conjunction with the built in modem Multi Tech Systems Inc 89 Chapter 5 Phonebook Configuration Phonebook Examples North America The following example demonstrates how Outbound and Inbound Phonebook entries work in a situation of multiple area codes Consider a company with offices in Minneapolis and Baltimore Notice first the area code situation in those two cities Minneapolis s local calling area consists of multiple adjacent area codes Baltimore s local calling area consists of a base area code plus an overlay area code Company l VOED e Baltimore Outstate MD i Overlay i 443 St Paul amp Suburbs 651 VOIP PBX Slte SW Suburbs 952 Baltimore I I I I I I I I I Company 1 1 I Figure 5 7 North America example An outline of the equipment setup in both offices is shown below Local Call Area Codes 612 651 952 H Company HQ Minneapolis North Sub area 763 Overlay O Baltimore area 410 7002 PO 325 7001 Figure 5 8 Equipment setup example Multi Tech Systems Inc 90 Chapter 5 Phonebook Configuration The screen below shows Outbound Phonebook entries for the VOIP located in the company s Baltimore facility Alternate IP Address i Dutbound Phone Book Destination Pattern
151. on Pattern will be directed to the IP Address in the Add Edit Outbound Phone Book screen Any Number can be used in addition to one or more Destination Patterns When external routing device is used If Any Number is selected calls to phone numbers not matching a listed Destination Pattern will be directed to the external routing device used Gatekeeper for H323 protocol Proxy for SIP protocol Registrar for SPP protocol The IP Address of the external routing device must be set in the Phone Book Configuration screen Destination prefixes Defines the beginning of dialing sequences for calls that will be connected to Pattern area codes another VOIP in the system Numbers beginning with these sequences are exchanges diverted from the PSTN and carried on Internet or other IP network line numbers extensions Total Digits as needed Number of digits the phone user must dial to reach specified destination This field not used in North America Remove Prefix dialed digits Portion of dialed number to be removed before completing call to destination Add Prefix dialed digits Digits to be added before completing call to destination IP Address n n n n The IP address to which the call will be directed if it begins with the destination pattern given Description alpha numeric Describes the facility or geographical location at which the call will be completed Protocol Type SIP or H 323 Indicates protocol to be used in outbound transmission Si
152. onnection exists While link is up this LED will flash off to indicate data activity Channel Operation LEDs one set for each channel XMT Transmit This indicator blinks when voice packets are being transmitted to the local area network RCV Receive This indicator blinks when voice packets are being received from the local area network Transmit Signal This indicator lights when the FXS configured channel is off hook the FXO XSG configured channel is receiving a ring from the Telco or the M lead is active on the E amp M configured channel That is it lights when the MultiVOIP is receiving a ring from the PBX Receive Signal This indicator lights when the FXS configured channel is ringing the FXO RSG configured channel has taken the line off hook or the E lead is active on the E amp M configured channel Computer Requirements The computer on which the MultiVOIP s configuration program is installed must meet these requirements e must be IBM compatible PC with MS Windows operating system e must have an available COM port for connection to the MultiVOIP However this PC does not need to be connected to the MultiVOIP permanently It only needs to be connected when local configuration and monitoring are done Nearly all configuration and monitoring functions can be done remotely via the IP network Multi Tech Systems Inc 7 Specifications Chapter 2 Quick Start MVP210 models MVP
153. onto the PC this is Software Installation and is discussed in this chapter Setting values for telephony and IP parameters that will fit your system details are in Chapter 4 Establishing phonebooks that contain the various dialing patterns for VOIP calls made to different locations a detailed discussion of this is found in Chapter 5 Loading MultiVOIP Software onto the PC The software loading procedure does not present every screen or option in the loading process It is assumed that someone with a thorough knowledge of Windows and the software loading process is performing the installation 1 Be sure that your MultiVOIP has been properly cabled and that the power is turned on 2 Insert the MultiVOIP CD into your CD ROM drive The CD starts automatically It may take a few moments for the Multi Tech CD installation window to display Analog MultiVOIP User Documentation Welcome to the Multi Tech Systems Inc aos Analog MultiVOIPs Models MVP 210 410 810 FX 3 Release Notes Install MVP Manager ave software version arlier please read the stalls Notes for ftp Upgrade cet Adobe Reader Figure 3 1 Analog MVP splash screen 3 When the Multi Tech Installation CD dialog box appears click the Install Software icon 4 Asecondary screen appears Click on the button that matches the model you have purchased The installation wizard will start Multi Tech Systems Inc 16 Chapter 3 Software Ins
154. p 161 snmp tray 162 gatekeeper registration 1719 H 323 1720 SIP 5060 SysLog 514 Multi Tech Systems Inc 136 Appendix C Installation Instructions for MVP428 Upgrade Card Installing the MVP428 Upgrade Card In this procedure you will install an additional circuit board into the MVP410 improving it from a 4 channel VOIP to an 8 channel VOIP Summary A Attach four standoffs to main circuit card B Mate the 60 pin connectors male connector on main circuit card female on upgrade card C Attach upgrade card to main circuit card 4 screws A Replace main card screws with standoffs here 2 places Add standoffs here 2 places C Attach upgrade card y screws into standoffs Se oat 4 places B Mate 60 pin connectors Figure C 1 MVP 248 installation Procedure in Detail 1 Power down and unplug the MVP410 unit 2 Using a Phillips driver remove the blank cover plate at the rear of the MVP410 chassis Save the screws screws on blank cover plate 2 Figure C 2 Remove screws from cover plate 3 Using a Phillips driver remove the three screws that secure the main circuit board and back panel assembly to the chassis Important Follow standard ESD precautions to protect the circuit board from static electricity damage Multi Tech Systems Inc 137 Appendix C MVP428 Upgrade Card back panel screws 3 Figure C 3 Remove screws from back panel 4 S
155. per List When selected VOIP will seek an alternate gatekeeper when none of the 3 gatekeepers shown on this screen are available from a list The list will reside on the Primary gatekeeper or one of the Alternate gatekeepers The gatekeeper holding the list would download that list onto the VOIP gateways within the system H 323 Version 4 Options H 323 Multiplexing Y N Signaling for multiple phone calls can be carried on a single port rather than opening a separate signaling port for each This conserves bandwidth resources H 245 Tunneling Tun Y N H 245 messages are encapsulated within the Q 931 call signaling channel Among other things the H 245 messages let the two endpoints tell each other what their technical capabilities are and determine who during the call will be the client and who the server Tunneling is the process of transmitting these H 245 messages through the Q 931 channel The same TCP IP socket or logical port already being used for the Call Signaling Channel is then also used by the H 245 Control Channel This encapsulation reduces the number of logical ports sockets needed and reduces call setup time Parallel H 245 FS Tun Y N FS Fast Start is a Q 931 feature of H 323v2 to hasten call setup as well as pre opening the media channel before the CONNECT message is sent This pre opening is a requirement for certain billing activities Under Parallel H 245 FS Tun this Fas
156. phone equipment 1 Connect the power cord supplied with your MultiVOIP to the power connector on the back of the MultiVOIP and to a live AC outlet as shown in the figure below The SS and FX models do not have the E amp M jacks as shown Voice Fax Channel 1 2 Connections Ay A Ground iL Figure 2 3 Cabling for MVP210 2 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 3 Connect a network cable to the ETHERNET 10 100 connector on the back of the MultiVOIP Connect the other end of the cable to your network a Foran FXS or FXO connection SS and FX series FXS Examples analog phone fax machine FXO Examples PBX extension POTS line from telco central office Connect one end of an RJ 11 phone cord to the Channel 1 FXS FXO connector on the back of the MultiVOIP Connect the other end to the device or phone jack b Foran E amp M connection E amp M Example trunk line from telephone switch Connect one end of an RJ 45 phone cord to the Channel 1 E amp M connector on the back of the MultiVOIP Connect the other end to the trunk line Verify that the E amp M Type in the E amp M Options group of the Interface dialog box is the same as the E amp M trunk type supported by the telephone switch See Appendix B for an E amp M cabling pin out c Fora DID con
157. ps Select from a range of coders with specific bandwidths The higher the bps rate the more bandwidth is used The channel that you are calling must have the same voice coder selected Default G 723 1 6 3 kbps as required for H 323 Here 64K of digital voice is compressed to 6 3K allowing several simultaneous conversations over the same bandwidth that would otherwise carry only one To make selections from the Selected Coder drop down list the Manual option must be enabled Selected Coder G 711 G 729 Or G 729 G 711 Coder Priority has two options G 711 G 729 or G 729 G711 on the Selected Coder listing of the Coder group on the Voice Fax screen If G 711 is the higher priority i e G 711 is preferred to G729 on the sending side then G 711 G 729 option is selected Similarly if G 729 has the higher priority then G 729 G 711 option is selected It is used whenever a user wants to advertise both G 711 and G 729 coders with higher preference to a particular coder It is useful when the calls are made from a particular channel on the VOIP to two different destinations where one supports G 711 and the other supports G 729 Max bandwidth coder 11 128 kbps This drop down list enables you to select the maximum bandwidth allowed for this channel The Max Bandwidth drop down list is enabled only if the Coder is set to Automatic If coder is to be selected automatically Auto setting then enter
158. quency 1 Frequency in Hertz Frequency of lower tone of pair This outbound tone pair enters the MultiVOIP at the input port Frequency 2 Frequency in Hertz Frequency of higher tone of pair This outbound tone pair enters the MultiVOIP at the input port Gain 1 3dB to 31dB Amplification factor of lower frequency of pair This figure describes and mute setting amplification that the MultiVOIP applies to outbound tones entering the MultiVOIP at the input port Default 16dB Gain 2 3dB to 31dB Amplification factor of higher frequency of pair This figure describes and mute setting amplification that the MultiVOIP applies to outbound tones entering the MultiVOIP at the input port Default 16dB Cadence 1 integer time value in On off pattern of tone durations used to denote phone ringing phone busy milliseconds zero value dial tone 0 indicates continuous tone survivability and re order for dial tone indicates Cadence 1 is duration of first period of tone being on in the cadence of continuous tone the telephony signal Cadence 2 duration in milliseconds Cadence 2 is duration of first off period in signaling cadence Cadence 3 duration in milliseconds Cadence 3 is duration of second on period in signaling cadence Cadence 4 duration in milliseconds Cadence 4 is duration of second off period in the signaling cadence Multi Tech Systems Inc 64 Ch
159. rameters Password OK Cancel Default Help Copy Channel Multi Tech Systems Inc 49 Chapter 4 Configuring your VOIP E amp M Interface Parameter Definitions Field Name Values Description Interface E amp M Enables E amp M functionality Type I V Type of E amp M interface being used the individual types are detailed below Default Type II Mode 2 wire or 4 wire Each E amp M interface type can be either 2 wire or 4 wire audio Signal Dial Tone or When Dial Tone is selected no wink is required on the E lead or M Wink lead in the call initiation or setup When Wink is selected a wink is required during call setup Wink Timer 100 350 This is the length of the wink for wink signaling Applicable only when milliseconds Signal parameter is set to Wink No Response Timer 1 65535 The value here denotes the time in seconds after which the call in seconds attempt would be disconnected by the FXO Interface because there was no answer Disconnect on Call Y N Allows call on FXO port to be disconnected when a PBX issues a call Progress Tone progress tone denoting that the phone station on the PBX that has been involved in the call has been hung up Pass Through Y N When enabled Y this feature is used to create an open audio path Enable for 2 or 4 wire The E amp M leads are passed through the VOIP transparently Applicable only for E amp M Signaling with Dial
160. rea answer enter 903305 as Destination Pat tern in code is dropped when combined Destination Pattern in Outbound Phone book of with national dialing access Outbound Phonebook of Seattle VOIP code Such practices vary by Rotterdam VOIP country Answer enter 90121 as Destination Pattern in Outbound Phonebook of London VOIP Not 900121 6 In the Remove Prefix field enter the initial PBX access digit 8 or 9 North America Euro National Call Example Euro International Call Long Distance Example Example Seattle Chicago system London Birmingham system Rotterdam Bordeaux system Answer enter 8 in Remove Prefix Answer enter 9 in Remove Prefix Answer enter 9 in Remove Prefix field of Seattle Outbound field of London Outbound field of Outbound Phonebook for Phonebook Phonebook Rotterdam VOIP Note Some PBXs will not hand off the 8 or 9 to the VOIP But for those PBX units that do it s important to enter the 8 or 9 in the Remove Prefix field in the Outbound Phonebook This precludes the problem of having to make two inbound phonebook entries at remote VOIPs one to account for situations where 8 is used as the PBX access digit and another for when 9 is used 7 In the Protocol Type field group select the VOIP protocol that you will use H 323 SIP or SPP Use the appropriate screen under Configuration Call Signaling
161. rn OK fo Cancel Remove Prefix FO Add Prefix IP Address Description Protocol Type SF C H 323 SPP H 323 Jos Help Advanced LED m H 323 Port Number f 720 SIP r Transport Protocol ICP f UDP SIP Port Number 5060 SPUR SPP 5 Port Number 10000 Alternate Phone Number Remote Device is MultWolP 110 120 200 400 800 Figure 5 2 Add Edit screen Enter Outbound Phone Book data for your MultiVOIP unit Note that the Advanced button gives access to the Alternate IP Routing feature if needed Alternate IP Routing can be implemented in a secondary screen as described after the primary screen field definitions below The SS will only allow SIP settings and the FX models will not allow H 323 The fields of the Add Edit Outbound Phone Book screen are described in the table below Multi Tech Systems Inc 82 Chapter 5 Phonebook Configuration Add Edit Outbound Phone Book Field Definitions Field Name Values Description Accept Any Y N When checked Any Number appears as the value in the Destination Number Pattern field The Any Number feature works differently depending on whether or not an external routing device is used Gatekeeper for H323 protocol Proxy for SIP protocol Registrar for SPP protocol When no external routing device is used If Any Number is selected calls to phone numbers not matching a listed Destinati
162. rts 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 e inthe MultiVOIP program interface e via email SMTP or e atthe MultiVoipManager remote VOIP system management program SNMP Logs Console Messages Filter Settings Console message Settings Trace Off for Functions Trace On for Functions IV Enable Console Messages OK Functions A Fiters Cancel gt Logs gt OK Help SS F Tum Off Logs _ Hop gt v Cancel GUI C SMTP C SNMP lt gt Trace Off for Channels Trace On for Channels Help SysLoa Server Channels ra Enable Channel 1 Server IP addiess 0 0 0 0 gt Port Number 514 F lt lt Onine Statistics Updation Interval 10 Sec Figure 4 22 Logs and Filters screens If you enable console messages you can customize the types of messages to be included excluded in log reports by clicking on the Filters button and using the Console Messages Filter Settings screen If you use the logging function select the logging option that applies to your VOIP system design If you intend to use a SysLog Server program for logging click in that Enable check box The common SysLog logical port number is 514 If you intend to use the MultiVOIP web browser interface for configuration and control of MultiVOIP units be aware that the web browser interface does not support logs directly However when the web browser interface is used log
163. s port number e SIP URL or f phone IP address For SPP calls the Forward Destination can be one of the following a phone number b IP address port or c phone number IP address port Ring Count integer When No Response is condition for forwarding calls this determines how many unanswered rings are needed to trigger the forwarding Registration In an H 323 VOIP system gateways can register with the system using one of these identifiers an Option E 164 identifier a Tech Prefix identifier or an H 323 ID identifier This section not available for Parameters the FX and SS series models In a SIP VOIP system gateways can register with the SIP Proxy This is the only area available to the SS series In an SPP VOIP system gateways can register with the SPP Registrar VOIP unit Multi Tech Systems Inc 88 Chapter 5 Phonebook Configuration Authorized User Name and Password for SIP To enable the Registration Options on the Add Edit Inbound Phone Book you have to activate Use SIP Proxy Option on the Call Signaling SIP Parameters Screen Then add the IP address for the Primary Proxy in the SIP Proxy Parameters This allows you to add a Username and Password to the Inbound Phone Book entry The SS models will only have a password option available This feature is used when the MultiVOIP registers with the proxies that support authorization and need the username password and the endpoint name to b
164. s of the gatekeeper Address Port port TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it Type Primary This field describes the type of gateway as which the MultiVOIP is defined with Predefined respect to the gatekeeper Status registered not The current status of the gateway either registered or unregistered Multi Tech Systems Inc 114 Advanced Packetization Time Chapter 6 Using the Software You can use the Packetization Time screen to specify definite packetization rates for coders selected in the Voice FAX Parameters screen in the Coder Options group of fields The Packetization Time screen is accessible under the Advanced options entry in the sidebar list of the main VOIP software screen In dealing with RTP parameters the Packetization Time screen is closely related to both Voice FAX Parameters and to IP Statistics It is located in the Advanced group for ease of use Packetization Time Parameters Select Channel Channel 1 v Packetization Ratelmsec per packet G711 A law 64 Kbps 80 x G727 40 16Kbps s0 E G711 U law 64Kbps s0 G727 40 24 Kbps s0 Cancel G726 16 Kbps 80 v G727 40 32 Kbps 80 Cop Charnel G726 24 Kbps s0 v G7231 5 3Kbps 90 y G726 32 Kbps 80 y G723 1 6 3Kbps 30 pa G 7266540 Kbps 80 72988 Kbps s0 Help G727 16 Kbps s0 y NetCoder 6 4Kbps a9 G727 24 16 Kbps so y NetCoder 7 2 Kbps
165. sends an identifier and status information to the administrator of the remote VOIP involved in the call The feature operates on a channel by channel basis each channel can have a separate identifier If the home VOIP is originating the call only the Calling Party field is applicable If the home VOIP is receiving the call then the Alerting Party Busy Party and Connected Party fields are the only applicable fields and any or all of these could be enabled for a given VOIP channel The status information confirms back to the originator that the home VOIP is either busy or ringing or that the intended call has been completed and is currently connected The identifier and status information are made available to the remote VOIP unit and appear in the Caller ID field of its Statistics Call Progress screen This is how MultiVOIP units handle CNI messages in other VOIP brands H 450 may be implemented differently and then the message presentation may vary Table is continued on next page Multi Tech Systems Inc 73 Chapter 4 Configuring your VOIP Supplementary Services Definitions continued Field Name Description Calling Party If the home VOIP unit is originating the call and Calling Party is selected then the identifier from Allowed Name the Caller Id field will be sent to the remote VOIP unit being called The Caller Id field gives the Type CNI remote VOIP administrator a plain language ide
166. sent in call Coder Voice Coder Compression Rate used for call will be listed in log Packets Lost Packets lost in call Prefix Matched When selected the phonebook prefix matched in processing the call will be listed in log Outbound Digits Sent Band DTMF digits received by this gateway from remote gateway if that DTMF set to Out of Call Status Successful or unsuccessful Server Details The IP address of the traffic control server being used will be displayed here if the call is handled through that server The Options field refers to non mandatory server features that might be activated For example with H 323 various H 323 Version 4 options might be listed From Details To Details Gateway Originating gateway Gateway Completing or answering gateway Number Name IP Address IP address where call originated IP Address IP address where call was completed answered Descript Identifier of where call originated Descript Identifier of where call was completed answered Options When selected log will not use Options When selected log will not use Silence Silence Compression and Forward Compression and Forward Error Correction by Error Correction by call originator party answering call Multi Tech Systems Inc 69 Chapter 4 Configuring your VOIP Logs Traces The Logs Traces screen lets you choose how the VOIP administrator will receive log repo
167. ser name A User Name is not needed to access the local Windows interface the web browser interface or the commands in the Program group Type your password in the Password field of the Password screen Type this same password again in the Confirm Password field to verify the password you have chosen NOTE Be sure to write down your password in a convenient but secure place If the password is forgotten contact Multi Tech Technical Support for advice ho m Password User Name New Password Reconfirm Password Cancel Help Figure 6 29 Password screen Click OK 5 A message will appear indicating that a password has been set successfully After the password has been set successfully the MultiVOIP will re boot itself and in so doing its BOOT LED will light up 6 After the password has been set the user will be required to enter the password to gain access to the web browser interface and any part of the MultiVOIP software listed in the Program group menu User Name and Password are both needed for access to the FTP Server residing in the MultiVOIP Multi Tech Systems Inc 124 Chapter 6 Using the Software Password Verification Enter Configuration Password Password AT Cancel Help Figure 6 30 Password verification When MultiVOIP program asks for password at launch of program the program will simply shut down if CANCEL is selected The MultiVOIP program will produc
168. set Generally updated firmware must be downloaded from the Multi Tech website to the PC before it can be loaded from the PC to the MultiVOIP Updating Firmware Generally updated firmware must be downloaded from the Multi Tech website to the user s PC before it can be downloaded from that PC to the MultiVOIP Note that the structure of the Multi Tech website may change without notice However firmware updates can generally be found using standard web techniques For example you can access updated firmware by doing a search or by clicking on Support If you choose Support you can select MultiVOIP in the Product Support menu and then click on Firmware to find MultiVOIP resources MultiVOIP Support Product Support MultivOIP Manuals are available on line Firmware is available Product Tour H 323 Upgrade Where to Buy Solutions Read the FAQs App Stories Software is available Figure 6 13 Web locations Once the updated firmware has been located it can be downloaded from the website using normal PC Windows procedures 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 It is usually best to click the Browse button and select a folder that is easy to get to and remember WinZip Self Extractor M YP301f EXE x To unzip all files in MYP301 EXE to
169. 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 To copy a set of Voice FAX parameters to all channels select Copy to All and click Copy The majority of the settings should be left at their default settings as changes often introduce problems with signal quality In each field enter the values that fit your particular setup The SS models do not have Configurable Payload Type available Voice Fax Parameters Select Channel Channel 1 y Voice Gain Fax Modem Parameters OK V Fax Relay Enable ja Cancel Input fo 7 dB DutputjO dB Modem Relay Enable _ Cancel Dtmf Max Baud Rate 14400 y E B Copy Channe High 6 y dB Low 8 v dB Fax Volume 95 v dB Default Jitter Value 400 ms Duration fioo ms Help DTMF Out Of Band Fixed Duration v Mode FRF 11 Out Of Band Mode Rfc2833 Coder Advanced Features Manual C Automatic V Silence Compression Selected Coder 6 71 1 6 729 s IV Echo Cancellation Forward Error Correction Auto Call OffHook Alert Auto Call OffHook Alert Auto Call y I Generate Local Dial Tone Phone Number Dynamic Jitter Buffer Minimum Jitter Value 60 ms Maximum Jitter Value 300 ms Optimization Factor 7 Automatic Disconnection IV Jitter Value 350 ms V Consecutive Packets Lost 30 V Call Duration fi 80 secs M Network Disconnection 30
170. signaling H 323 H 323 is an ITU T recommended set of standards for audio and video communications The fields for this screen are defined in the table below H 323 WV Use Fast Start Signaling Port 11720 F Allow Incoming Calls Through Gatekeeper Only GateKeeper RAS Parameters IP Address Primary GK 192 168 3 1 Alternate GK 1 0 0 0 0 Altemate GK 2 0 0 0 0 RAS TTL Value GateKeeper Discovery Polling Interval Use Online Alternate GateKeeper List H323 Version 4 Options H 323 Multiplexing Mux Parallel H 245 FS Tun oK RAS Port GateKeeper Name ML Cancel 1719 Help 1719 M H 245 Tunneling Tun Annex E AE Figure 4 13 H 323 call signaling 53 Multi Tech Systems Inc Chapter 4 Configuring your VOIP H 323 Call Signaling Parameter Definitions Field Name Values Description Use Fast Start Y N Enables the H 323 Fast Start procedure May need to be enabled disabled for compatibility with third party VOIP gateways Signaling Port port Default 1720 H 323 Register with Y N Check this field to have traffic on current VOIP gateway controlled by a Gatekeeper gatekeeper Allow Incoming Y N When selected incoming calls are accepted only if those calls come through the Calls Through Gatekeeper Only gatekeeper GateKeeper RAS Parameters Primary GK This is the preferred gatekeeper for controlling the traffic of the curre
171. sing Save Setup This command loads the saved user defaults into the MultiVOIP Set Password Select this to create a password for access to the MultiVOIP software programs Program group commands Windows interface web browser interface amp FTP server Only the FTP Server function requires a password for access The FTP Server function also requires that a username be set along with the password Uninstall Select this to uninstall the MultiVOIP software most but not all components are removed from computer when this command is used Upgrade Software Loads firmware including H 323 stack and settings from the controller PC to the MultiVOIP unit User can choose whether to load Factory Default Settings or Current Configuration settings Multi Tech Systems Inc 116 Chapter 6 Using the Software 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 Multi Tech s website Although transferring updated program files from the Multi Tech website to the user s PC can generally be considered a download from the perspective of the PC this type of download cannot be initiated from the MultiVOIP software s Program menu command
172. sion currently loaded on it The firmware version appears in the MultiVOIP Program menu Go to Start Programs MultiVOIP x xx The final expression x xx is the firmware version number When a new firmware version is installed the MultiVOIP software can be upgraded in one step using the Upgrade Software command or piecemeal using the Download Firmware command and the Download Factory Defaults command Download Firmware transfers the firmware including the H 323 protocol stack in the PC s MultiVOIP directory into the nonvolatile flash memory of the MultiVOIP Download Factory Defaults sets all configuration parameters to the standard default values that are loaded at the Multi Tech factory Upgrade Software implements both the Download Firmware command and the Download Factory Defaults command Multi Tech Systems Inc 118 Chapter 6 Using the Software Downloading Firmware 1 The MultiVOIP Configuration program must be off when invoking the Download Firmware command If it is on the command will not work 2 To use the Download Factory Defaults command go to Start Programs MultiVOIP x xx Download Firmware 3 If a password has been established the Password Verification screen will appear Password Verification Enter Configuration Password Password yq e Figure 6 15 Password verification Type in the password and click OK 4 The MultiVOIP x xx Firmware screen appears saying MultiVOIP model n
173. ssuming here that the MultiVOIP will operate in conjunction with a PBX You must configure both the Outbound Phonebook and the Inbound Phonebook A starter configuration only means that two VOIP locations will be set up to begin the system and establish VOIP communication Once this is accomplished you can easily add other VOIP sites to the network Outbound Phonebook 1 Open the MultiVOIP program Start MultiVOIP xxx Configuration 2 Go to Phone Book Outbound Phonebook Add Entry 3 On a sheet of paper write down the calling code of the remote VOIP area code country code city code etc that you ll be calling Follow the example that best fits your situation North America Long Distance Example Euro National Call Example Euro International Call Example Technician in Rotterdam country 31 city 010 to set up one VOIP there another in Bordeaux country 33 area 05 Technician in central London area 0207 to set up VOIP there another in Birmingham area 0121 Answer Write down 0121 Technician in Seattle area 206 must set up one VOIP there another in Chicago area 312 downtown Answer Write down 312 Answer Write down 3305 Suppose you want to call a phone number outside of your building using a phone station that is an extension from your PBX system if present What digits must you dial Often a 9 or 8 must be dialed to get an outside line through the P
174. strar is the highest level registrar gateway Alternate Registrar 1 is second Alternate Registrar 2 is the lowest order SPP registrar gateway Registrar Options Registrar Option fields are active only in registrar client mode and only for registrar VOIP units Keep Alive 30 300 seconds Time out duration before a registrar will un register a client that does not send its Pm here signal Client normally sends its I m here signal every 20 seconds Timeout default 60 seconds Proxy NAT Device Parameters Behind Y N Enables MultiVOIP running in SPP Registrar mode to operate behind a Proxy NAT proxy NAT device NAT Network Address Translation device Proxy NAT n n n n The public IP address of the proxy NAT device which the MultiVOIP is behind Device Parameters Public IP Address Multi Tech Systems Inc 60 Chapter 4 Configuring your VOIP SNMP If you intend to manage your MultiVOIP remotely using the MultiVoipManager software you will need to set the Simple Network Management Protocol parameters To make the MultiVOIP controllable by a remote PC running the MultiVoipManager software check the Enable SNMP Agent box on the SNMP Parameters screen The SS and FX series MultiVOIPs only have limited SNMP functionality available If this is something you wish to use on those models please contact Multi Tech Support for assistance SNMP Paramet
175. t The phonebook downloading feature greatly reduces the data entry required to establish inbound and outbound phonebooks for the VOIP units within a system Although each MultiVOIP unit will require some unique phonebook entries most will be common to the entire VOIP system After the phonebooks for the first few VOIP units have been compiled phonebooks for additional VOIPs become much simpler you copy the common material by downloading and then do data entry for the few phonebook items that are unique to that particular VOIP unit or VOIP site To transfer files using the FTP server 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 2 Establish User Name and Password You must establish a user name and optionally a password for contacting the VOIP over the IP network When connection is made via a local serial connection between the PC and the VOIP unit no user name is needed 3 MultiVoIP MultiVOIP y6 11 0S Firmware Sep 17 2007 Microsoft Internet Explorer File Edit View Favorites Tools Help Q 0 ba a Address El http 192 168 3 143 Multi OIP Configuration o Advanced o Phone Book Statistics Change Password Save amp Reboot Logout Help Current Permission Read vrite Password Change User Name O
176. t Loss is enabled Y the MultiVOIP will hang up the call at a specified interval after it detects a loss of current initiated by the attached device Current Loss Timer 200 to 2000 Determines the interval after detection of current loss at which the call in milliseconds will be disconnected Silence Detection Enable Y N Enables disables silence detection method of supervising call disconnection Silence Detection One Way or Disconnection to be triggered by silence in one direction only or in Type Two Way both directions simultaneously Silence Timer in integer value Duration of silence required to trigger disconnection seconds Table is continued on next page Multi Tech Systems Inc 47 Chapter 4 Configuring your VOIP FXO Supervision Parameter Definitions continued Field Name Values Description Disconnect Supervision fields DTMF Tone Enables supervision of call disconnection using DTMF tones DTMF Tone Pairs Low Tones 1 2 3 A 697Hz 4 5 6 B 770Hz 7 8 9 Cc 852Hz 0 D 941Hz High Tones 1209Hz 1336Hz 1447Hz 1633Hz Disconnect 1 tone pair These are DTMF tone pairs Tone Values for first tone pair are 0 1 9 and A D Sequence 2 tone pair Values for second tone pair are none 0 1 9 A D and The tone pairs 1 9 0 and are the standard DTMF pairs found on phone sets The tone pairs A D ar
177. t Connect feature can operate simultaneously with H 245 Tunneling Annex E AE Y N Multiplexed UDP call signaling transport Annex E is helpful for high volume VOIP system endpoints Gateways with lesser volume can afford to use TCP to establish calls However for larger volume endpoints the call setup times and system resource usage under TCP can become problematic Annex E allows endpoints to perform call signaling functions under the UDP protocol which involves substantially streamlined overhead this feature should not be used on the public Internet due to potential problems with security and bandwidth usage Multi Tech Systems Inc 54 Chapter 4 Configuring your VOIP SIP Session Initiation Protocol is the second option available for application layer control of the MultiVOIP The fields are detailed in the table below SIP Parameters Signaling Port 5060 Allow Incoming Calls Through SIP Proxy Only m SIP Proxy Parameters Proxy Domain Name IPAddress Port Number Primary Proxy 5060 Alternate Proxy 1 5060 Alternate Proxy 2 5060 Append SIP Proxy Domain Name in User ID Default Subscriber Default Username Password Re RegistrationT ime 3600 secs Proxy Polling Interval 60 secs TTL Value 60 secs SIP Voice Mail Server Parameters J Voice Mail Server Domain Name IP Address Patt 5060 Re Subscription time 3600 secs OK Can
178. t Regeneration Timer ms e Message Waiting Indication Light pe Password Flash Hook Options OK Generation 600 ms a Detection Range Cancel Min 100 ms Default Max 1000 ms Help Caller ID Type BellCore X Copy Channel MV Enable CID Manipulation F Disable CID Manipulation CID Mode TransParent v jae es E Pass Through Options MV Enable Figure 3 9 Interface Parameters Multi Tech Systems Inc 23 Actions e Select Channel Chapter 3 Software Installation o Select Interface Type FXS FXO E amp M or DID FXS FXO only for SS and FX series o Regeneration Choose how signal is regenerated as Pulse or DTMF o Inter Digit Timer Time the MultiVOIP waits between digits o Message Waiting Indication is for E amp M only Choose Light or None o Inter Digit Regeneration Timer Length of time between sent DTMF digits Flash Hook Options o Generation used in conjunction with FXO E amp M o Detection Range used in conjunction with FXS E amp M Caller ID o Bellcore is the only option available o CallerlD Manipulation is available if needed o CID Manipulation is not available in the SS models Pass Through opens an audio path through the MultiVOIP FXS Options o Set Ring Count the number of rings allowed before call abandoned default is 8 o Use Current Loss MultiVOIP interrupts current to disconnect o Generate Current Reversal activates Answer Disconnect Supervision to FXO FXO Options o Ring
179. t cause harmful interference and 2 this device must accept any interference that may cause undesired operation Warning Changes or modifications to this unit not expressly approved by the party responsible for compliance could void the user s authority to operate the equipment Industry Canada This Class A digital apparatus meets all requirements of the Canadian Interference Causing Equipment Regulations Cet appareil num rique de la classe A respecte toutes les exigences du Reglement Canadien sur le mat riel brouilleur Canadian Limitations Notice Notice The Industry Canada label identifies certified equipment This certification means that the equipment meets certain telecommunications network protective operational and safety requirements The Department does not guarantee the equipment will operate to the user s satisfaction Before installing this equipment users should ensure that it is permissible to be connected to the facilities of the local telecommunications company The equipment must also be installed using an acceptable method of connection The customer should be aware that compliance with the above conditions may not prevent degradation of service in some situations Repairs to certified equipment should be made by an authorized Canadian maintenance facility designated by the supplier Any repairs or alterations made by the user to this equipment or equipment malfunctions may give the telecommunications company cause
180. t the tones manually o User Defined tones can be created for use in conjunction with FXO Supervision with the Add button Multi Tech Systems Inc 27 Phone Book Chapter 3 Software Installation Without a populated phone book the VOIP unit is unable to translate call traffic You will need the information for both a local and any remote sites that are to be used Detailed descriptions and examples are available in chapter 5 Add Edit Outbound Phone Book Phone Number Details I Accept Any Number Destination Pattern DK jo Cancel Remove Prefix Help Add Prefix Advanced IP Address Description Protocol 1 C H 323 SPP r SIP r Transport Protocol E ICP UDP SIP Port Number 5060 SIP URL r r Figure 3 12 Add Edit Inbound Phone Book I Accept Any Number Remove Prefix OK Add Prefs iti s S Cancel Channel Number Hunting a Help Description BT Call Forward For ondition I Unconditional I Busy I No Response H323 call Phone or IP address SIP call Phone or IP address or IP address port or Phone IP address port or SIP URL or Ph IP address SPP call Phone or IP address port or Phone IP address port Forward Destination lt Registration Options r r ai P E r r r Phone Book screens Multi Tech Systems Inc 28 Chapter 3 Software Installation Actions e Select Outbound Phone Book o
181. tallation Multi Tech Systems Multi VOIP 6 11 Installation Thank you for choosing MultiOIP from Multi Tech Systems Click Next to continue installation Cancel Figure 3 2 Welcome screen Press Enter or click Next to continue 5 Follow the on screen instructions to install your MultiVOIP software The first screen asks you to choose the destination for the MultiVOIP software Multi Tech Systems Multi OIP 6 11 Installation Setup will install Multi OIP in the following folder To install to this folder Click Next To install to another folder Click Browse and select another folder CA AMultiTech Systems MultiVOIP 6 11 Browse Destination Folder InstallShield lt Back Cancel Figure 3 3 Destination Choose a location and click Next 6 At the next screen you must select a program folder location for the MultiVOIP software program icon Click Next Transient progress screens will appear while files are being copied Multi Tech Systems Inc 17 Chapter 3 Software Installation 7 On the next screen you can select the COM port that the command PC will use when communicating with the MultiVOIP unit After software installation the COM port can be re set in the MultiVOIP Software from the sidebar menu select Connection Settings to access the COM Port Setup screen or use keyboard shortcut Ctrl G Note If the COM port setting made here conflicts with the actual COM port r
182. tatic or temporary depending on the needs of the computer IP Address n n n n The unique LAN IP address assigned to the MultiVOIP IP Mask n n n n Subnetwork address that allows for sharing of IP addresses within a LAN Gateway n n n n The IP address of the device that connects your MultiVOIP to the Internet Table is continued on next page Multi Tech Systems Inc 33 Chapter 4 Configuring your VOIP Ethernet IP Parameter Definitions continued Field Name Values Description Diff Serv Diff Serv PHB Per Hop Behavior values pertain to a differential prioritizing system for IP packets as Parameter handled by Diff Serv compatible routers There are 64 values each with an elaborate technical fields description These descriptions are found in TCP IP standards RFC2474 RFC2597 and for present purposes in RFC3246 which describes the value 34 34 decimal 22 hex for Assured Forwarding behavior default for Call Control PHB and the value 46 46 decimal 2E hexadecimal for Expedited Forwarding behavior default for VOIP Media PHB Before using values other than these default values of 34 and 46 consult these standards documents and or a qualified IP telecommunications engineer To disable Diff Serv configure both fields to O decimal Call Control 0 63 Value is used to prioritize call setup IP packets PHB default 34 Setting this parameter to 0 in conjunction with VOIP Media PHB below will disable
183. te VOIP Generate Local Dial Y N Used for AutoCall only If selected dial tone will be generated locally while the Tone call is being established between gateways The capability to generate dial tone locally would be particularly useful when there is a lengthy network delay Offhook Alert Timer 0 3000 The length of time that must elapse before the off hook alert is triggered and a seconds call is automatically made to the phone number listed in the Phone Number field Phone Number Phone number used for Auto Call function or Offhook Alert Timer function This phone number must correspond to an entry in the Outbound Phonebook of the local MultiVOIP and in the Inbound Phonebook of the remote MultiVOIP unless a gatekeeper unit is used in the VOIP system Table is continued on next page Multi Tech Systems Inc 38 Chapter 4 Configuring your VOIP Voice Fax Parameter Definitions continued Field Name Values Description Dynamic Jitter Dynamic Jitter Buffer Dynamic Jitter defines a minimum and a maximum jitter value for voice communications When receiving voice packets from a remote MultiVOIP varying delays between packets may occur due to network traffic problems This is called Jitter To compensate the MultiVOIP uses a Dynamic Jitter Buffer The Jitter Buffer enables the MultiVOIP to wait for delayed voice packets by automatically adjusting the length of the Jitter Buffer between
184. ted Number of pings sent to target endpoint No of Pings as listed Number of pings received by target endpoint Received Round Trip Delay as listed Displays how long it took from time ping was sent to time ping response Min Max Avg in milliseconds was received Last Error as listed Indicates when last data error occurred Multi Tech Systems Inc 110 Chapter 6 Using the Software Registered Gateway Details The Registered Gateway Details screen presents a real time display of the special operating parameters of the Single Port Protocol SPP These are configured in the Call Signaling screen and in the Add Edit Outbound Phone Book screen Registered Endpoints IP Address Register Duration Status Description pS No of Entries Details Count of Registered Numbers List of Registered Numbers v hn Help Exit OS Y Figure 6 7 Registered endpoints Registered Gateway Details Field Definitions Field Name Values Description Column Headings Description alphanumeric This is a descriptor for a particular VOIP gateway unit This descriptor should generally identify the physical location of the unit e g city building etc and perhaps even its location in an equipment rack IP Address n n n n The RAS address for the gateway Port n Port by which the gateway exchanges H 225 RAS messages with the gatekeeper Register The time remain
185. the E amp M interface and the MVP210 410 810 models support them all but Type IV is largely unused and will not be detailed in this section The figures below will show the pin assignments for the MVP RJ48 connector when used in the E amp M jacks on the back of the unit as well as how the signals are used for types one two three and five Common ground between the MultiVOIP and PBX is required for all E amp M Types except Type II Two and four wire audio is available for all E amp M Types and is shown in figure 4 9 below M INPUT E OUTPUT T1 4 WIRE OUTPUT R 4 WIRE INPUT 2 WIRE T 4 WIRE INPUT 2 WIRE R1 4 WIRE OUTPUT SG SIGNAL GND OUTPUT SB SIGNAL BATTERY OUTPUT PIN NO on oan ft O N Figure 4 9 MultiVOIP E amp M Pin assignments and RJ48 Jack Type I E amp M Type II E amp M Open is On hook _ On hook N E E al Off hoo 48v 48v Ground is Off hook L q On hook i Detection M it Detection Off hook gt On hook i 48v MVP PBX 1 MVP PBX Type III E amp M Type V E amp M On hook On hook fe Off hook SG Orejak p z On hook etection M 48v MVP L sB Off hook By PBX Figure 4 10 E amp M Line Types MVP la aA Two Wire Four Wire Figure 4 11 Audio wiring 48v On hook Off hook PBX Multi Tech Systems Inc 51 Chapter 4 Configuring your VOIP DID Parameters The parameters applicable
186. the Software System Information screen This screen presents system information at a glance It is found under the Configuration section and its primary use is in troubleshooting The information presented in figure 6 1 is for reference only and is not meant to be an exact match of your system System Information Wersion Information Boot Version 204c Firmware Version 6 11 05 Configuration Version 6 11 00 00 Phone Book Version 4 04 IFM Version le MAC Address 000800517858 Uptime 00 00 00 59 Hardware ID MY P410 32M Rev B F98F Exit Figure 6 1 System information System Information Parameter Definitions Field Name Values Description Boot Version nn nn Indicates the version of the code that is used at the startup booting of the VOIP alpha The boot code version is independent of the software version numeric Firmware Version nn nn nn Indicates the version of the MultiVOIP firmware alpha numeric Configuration Version nn nn Indicates the version of the MultiVOIP configuration software nn nn alpha numeric Phone Book Version nn nn Indicates the version of the MultiVOIP phone book being used alpha numeric IFM Version nn Indicates version of the IFM module the device that performs the transformation alpha between telephony signals and IP signals numeric Mac Address numeric Denotes the number assigned as the VOIP unit s unique Ethernet address Up Time days Indicates ho
187. 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 Multi Tech Systems Inc 134 Appendix A Cable Pin outs Command Cable RJ 45 Connector End to End Pin Info RJ 45 DB9F Pin No Pin No Clear to Send To DTE To Command Transmt Data Device Port Connector Receive Data 0 Y O Mm who O 0 MN 0 00 y e Signal Ground RJ 45 connector plugs into Command Port of MultiVOIP DB 9 connector plugs into serial port of command PC which runs MultiVOIP configuration software Ethernet Connector The functions of the individual conductors of the MultiVOIP s Ethernet port are shown on a pin by pin basis below RJ 45 Ether TD Data Transmit Positive net Connector Pin Circuit Signal Name jj 1 ia mE 2 TD Data Transmit Negative 3 RD Data Receive Positive 6 RD Data Receive Negative Voice Fax Channel Connectors 12345678 Figure B 1 RJ 48 amp RJ 11 Connectors Pin Functions E amp M Interface Pin Description Function 1 M Input 2 E O
188. the following U S Patent Numbers 6151333 5757801 5682386 5 301 274 5 309 562 5 355 365 5 355 653 5 452 289 5 453 986 Other Patents Pending Trademark Registered trademarks of Multi Tech Systems Inc are MultiVOIP Multi Tech and the Multi Tech logo Windows is a registered trademark of Microsoft World Headquarters Multi Tech Systems Inc 2205 Woodale Drive Mounds View Minnesota 55112 Phone 763 785 3500 or 800 328 9717 Fax 763 785 9874 http www multitech com Technical Support Country By Email By Phone Europe Middle East Africa support multitech co uk 44 118 959 7774 U S Canada all others support multitech com 800 972 2439 or 763 717 5863 Warranty Please visit www multitech com for valuable warranty information for your product Multi Tech Systems Inc 2 CONTENTS Chapter 1 Description and SpecificationS ooommnnonnonennennrnnnnosccccnnennnnnnnncn cnn c nn anno nnc nn nn nn nn rre 6 TOJU HO Ra taa 6 Feature Comparison Malik usual ii dia 6 ime haco e ere Preee ester ne Ce nee e Creare eee Wer ers Seer eer ee errr ete rte Sect eenr ee reenter ey aren Gere erererrnern Ce errr ear eer erre re 7 A o rene tren errr 7 Computer Regui MEMS cantada iaa aia Aaa ia anidan ae tetera mate 7 SCN eNO Sy a eaate nus tanannnrasanoe saat antes alas dna pia 8 Chapter 2 Installing and Cabling the MultiVOIP cccccsssseeeeeseeeeeeeseeeeeeenseeeeeenseeeeeenseeeeeenseseeesnseeeeeeneesenens
189. the upgrade circuit card on top of the main circuit card Align the upgrade card s 4 pairs of phone jacks with the 4 pairs of holes in the backplane of the main card Slide the phone jacks into the holes 13 Mate the upgrade card s 60 pin female connector with the main card s 60 pin male connector These screws 4 places attach upgrade card to main card 60 pin connectors Figure C 7 Attaching upgrade card to main circuit card 14 There are four copper plated attachment holes two each at the front and rear edges of the upgrade card Attach the upgrade card to the main card using 4 Phillips screws The upgrade card should now be firmly attached to the main card 15 Slide the main circuit card back into the chassis far enough to allow re connection of power cable 16 Re connect power cable 17 Slide the main circuit card fully into the chassis 18 Re attach the backplane of the main circuit card to the chassis with 3 screws Multi Tech Systems Inc 139 Appendix D Regulatory Information EMC Safety and R amp 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 and Council Directive 73 23 EEC of 19 February 1973 on the harmonization of the laws of Member States relating to electrical eq
190. tiation Advanced Features e Silence Compression when enabled will not send silence packets e Echo Cancellation removes echo to improve voice quality e Forward Error Correction allows some bad packets to be recovered Choose Auto Call OffHook Alert settings e For automatically calling a remote VOIP without dialing details in Chapter 4 Change Dynamic Jitter values if necessary details in Chapter 4 Select any Automatic Disconnection options needed to ensure lines are not left open Configurable Payload Types are best left at their defaults Not in the SS models o The Copy Channel button is available for easily transferring these settings to the other channels e Repeat for all channels to be used Multi Tech Systems Inc 22 Interface Chapter 3 Software Installation The Interface Parameters are the telephony settings that are to be applied to the individual MultiVOIP channels Configure each channel for the type of interface you are using Channel 1 is selected by default Note Feature options are enabled or unavailable depending on the selected interface type The one option available for all interface types is the inter digit timer option This option defines the maximum amount of time that the unit will wait before mapping the dialed digits to an entry in the phone book database If too much time elapses between digits and the wrong numbers are mapped you will hear rapid busy signal If this happens hang up and dial
191. tifiers come from Gateway Name field in Phone Book Configuration screen of remote VOIP Caller ID Calling Party identifier Alerting Party identifier Busy Party identifier Connected Party identifier This field shows the identifier and status of a remote VOIP which has Call Name Identification enabled with which this VOIP unit is currently engaged in some VOIP transmission The status of the engagement Connected Alerting Busy or Calling is followed by the identifier of a specific channel of a remote VOIP unit This identifier comes from the Caller Id field in the Supplementary Services screen of the remote VOIP unit Call Status fields Call Status hangup active Shows condition of current call Call Control Status Tun FS Tun Displays the H 323 version 4 features in use for the selected call These AE Mux include tunneling Tun Fast Start with tunneling FS Tun Annex E multiplexed UDP call signaling transport AE and Q 931 Multiplexing Mux Silence Compression SC SC stands for Silence Compression With Silence Compression enabled the MultiVOIP will not transmit voice packets when silence is detected thereby reducing the amount of network bandwidth that is being used by the voice channel Forward Error FEC FEC stands for Forward Error Correction Forward Error Correction Correction enables some of the voice packets that were
192. tion of the software pertaining to the list below while Chapter 5 will cover the Phone Book setup and Chapter 6 will discuss the Statistics options and overall maintenance of the MultiVOIP Software Categories Covered in This Chapter Ethernet IP Voice Fax Interface V VW V WV Call Signaling o H 323 SIP SPP SNMP Regional SMTP RADIUS Logs Traces NAT Traversal Supplementary services Save Setup VV VV VV VV WV Connection o Settings Multi Tech Systems Inc 30 Chapter 4 Configuring your VOIP How to Navigate Through the Software The MultiVOIP software is launched from the Start button and is found in the All Programs area under the title of MultiVOIP x xx where x represents version number The top option is Configuration choose this Within the software there are several ways to arrive at the parameter that you want to use through the left hand panel from the drop down menu clicking a taskbar icon if available or a keyboard shortcut if available Once the initial settings are entered you may choose to configure the MultiVOIP through a Web browser instead Web Browser Interface The MultiVOIP web browser interface gives access to the same commands and configuration parameters as are available in the MultiVOIP Windows interface except for logging functions When using the web browser interface logging can be done by email the SMTP option Set up the Web Browser interface Optional After an IP a
193. tones relate to answering and values in disconnection of calls Although most cadences have only two parts an milliseconds on duration and an off duration some telephony cadences have four zero value parts Most cadences then are expressed as two iterations of a two indicates part sequence Although this is redundant it is necessary to allow for continuous tone expression of 4 part cadences Setting Custom Tones and Cadences optional The Regional Parameters dialog box has a secondary dialog box that allows you to customize DTMF tone pairs to create unique ring tones dial tones busy tones or unobtainable tones or re order tones or survivability tones for your system This screen allows the user to specify tone pair attributes that are not found in any of the standard national regional telephony toning schemes To access this customization feature click on the Custom button on the Regional Parameters screen The Custom button is active only when Custom is selected in the Country Region field Custom Tone Pair Settings Definitions Field Name Values Description Tone Pair dial tone busy tone ring tone unobtainable tone survivability tone re order tone Identifies the type of telephony signaling tone for which frequencies are being specified Tone Pair Values About Defaults US telephony values are used as defaults on this screen Fre
194. tronics products at the end of their useful life A sister Directive ROHS Restriction of Hazardous Substances complements the WEEE Directive by banning the presence of specific hazardous substances in the products at the design phase The WEEE Directive covers all Multi Tech products imported into the EU as of August 13 2005 EU based manufacturers distributors retailers and importers are obliged to finance the costs of recovery from municipal collection points reuse and recycling of specified percentages per the WEEE requirements Instructions for Disposal of WEEE by Users in the European Union The symbol shown below is on the product or on its packaging which indicates that this product must not be disposed of with other waste Instead it is the user s responsibility to dispose of their waste equipment by handing it over to a designated collection point for the recycling of waste electrical and electronic equipment The separate collection and recycling of your waste equipment at the time of disposal will help to conserve natural resources and ensure that it is recycled in a manner that protects human health and the environment For more information about where you can drop off your waste equipment for recycling please contact your local city office your household waste disposal service or where you purchased the product Multi Tech Systems Inc 142 Appendix F C ROHS HT TS Substance Concentration KRP ETP ENASAS MS AHP AA B
195. ts will be sent Subject text User specified Subject line that will appear for all emailed log reports for this MultiVOIP unit Reply To Address email address User specified This email address functions as a source email identifier for the MultiVOIP which of course cannot usefully receive email messages The Reply To address provides a destination for returned messages indicating the status of messages sent by the MultiVOIP esp to indicate when log report email was undeliverable or when an error has occurred Recipient Address email address Email address where VOIP administrator will receive log reports Criteria for sending log summary by email The log summary email will be Mail Criteria sent out either when the user specified number of log messages has accumulated or once every day or multiple days whichever comes first Number of Records integer This is the number of log records that must accumulate to trigger the sending of a log summary email Number of Days integer This is the number of days that must pass before triggering the sending of a log summary email Multi Tech Systems Inc 66 Chapter 4 Configuring your VOIP The SMTP Parameters dialog box has a secondary dialog box accessed by the Select Fields button that allows you to customize email logging The MultiVOIP software logs data about many aspects of the call traffic going through the MultiVOIP The Custom Fields
196. tted in Realtime Transport Protocol packets RTP packets are a type or subset of UDP packets Transmitted integer Number of RTP packets transmitted by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received integer Number of RTP packets received by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received with integer Number of error laden RTP packets received by this VOIP gateway since the last Errors value clearing or resetting of the counter within the MultiVOIP software RTCP Packets Realtime Transport Control Protocol packets convey control information to assist in the transmission of RTP voice packets RTCP packets are a type or subset of UDP packets Transmitted integer Number of RTCP packets transmitted by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received integer Number of RTCP packets received by this VOIP gateway since the last clearing or value resetting of the counter within the MultiVOIP software Received with integer Number of error laden RTCP packets received by this VOIP gateway since the last Errors value clearing or resetting of the counter within the MultiVOIP software Multi Tech Systems Inc 109 Chapter 6 Using the Software Link Management The Link
197. uipment designed for use within certain voltage limits and Council Directive 1999 5 EC of 9 March 1999 on radio equipment and telecommunications terminal equipment and the mutual recognition of their conformity FCC Part 15 Class A Statement This equipment has been tested and found to comply with the limits for a Class A digital device pursuant to 47 CFR Part 15 regulations The stated limits in this regulation are designed to provide reasonable protection against harmful interference in a commercial environment This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications However there is no guarantee that interference will not occur in a particular installation If this equipment does cause harmful interference to radio or television reception which can be determined by turning the equipment off and on the user is encouraged to try to correct the interference by one or more of the following measures Reorient or relocate the receiving antenna Increase the separation between the equipment and receiver Plug the equipment into an outlet on a circuit different from that to which the receiver is connected Consult the dealer or an experienced radio TV technician for help This device complies with Part 15 of the CFR 47 rules Operation of this device is subject to the following conditions 1 This device may no
198. ult Normal priority for ordinary LAN traffic 3 Excellent Effort The best effort type of service that an information services organization would deliver to its most important customers 4 Controlled Load Important business applications subject to some form of Admission Control such as preplanning of Network requirement characterized by bandwidth reservation per flow 5 Video Traffic characterized by delay lt 100 ms 6 Voice Traffic characterized by delay lt 10 ms 7 Network Control Traffic urgently needed to maintain and support network infrastructure HIGHEST PRIORITY Call Control Priority 0 7 where O is Sets the priority for signaling packets lowest priority VOIP Media Priority 0 7 where Ois Sets the priority for media packets lowest priority Others Priorities 0 7 where O is Sets the priority for SMTP DNS DHCP and other packet types lowest priority VLAN ID 1 4094 The 802 1Q IEEE standard allows virtual LANs to be defined within a network This field identifies each virtual LAN by number IP Parameter fields Gateway Name alphanumeric Descriptor of current VOIP unit to distinguish it from other units in system Enable DHCP Y N Dynamic Host Configuration Protocol is a method for assigning IP address and disabled by other IP parameters to computers on the IP network in a single message with default great flexibility IP addresses can be s
199. umber is up Reboot to Download Firmware Click OK to download the firmware The Boot LED on the MultiVOIP will light up and remain lit during the file transfer process 5 The program will locate the firmware bin file in the MultiVOIP directory Highlight the correct newest bin file and click Open Look in jmutvop x Ej l ex nyvptt bin File name mvptt Files of type Code Files bin y Cancel La Figure 6 16 Firmware file 6 Progress bars will appear at the bottom of the screen during the file transfer fy m SA nr emICC _ P e rn vnr e o nro ce nen o Downloading Configuration Packets Sent 2 Acks received 2 Errors 0 inn Figure 6 17 Progress bars The MultiVOIP s Boot LED will turn off at the end of the transfer 7 The Download Firmware procedure is complete Multi Tech Systems Inc 119 Chapter 6 Using the Software Downloading Factory Defaults 1 The MultiVOIP Configuration program must be off when invoking the Download Factory Defaults command If it is on the command will not work 2 To use the Download Factory Defaults command go to Start Programs MultiVOIP x xx Download Factory Defaults 3 If a password has been established the Password Verification screen will appear Password Verification Enter Configuration Password Cancel Help Figure 6 18 Password verify Type in the password and click OK 4 The
200. umeric string Type Primary This field describes the type of gateway as which the MultiVOIP is defined with Predefined respect to the gatekeeper Priority n Priority level given Status registered The current status of the gateway either registered or unregistered not registered Multi Tech Systems Inc 112 Chapter 6 Using the Software SIP Proxies SIP Proxies IP Address Port Type A e Exit Figure 6 9 SIP proxies SIP Proxies Statistics Servers Field Definitions Field Name Values Description Column Headings IP Address n n n n The IP address of the SIP proxy by which the MultiVOIP is governed Port port TDMA time slot used for communication between MultiVOIP unit and the SIP Proxy that governs it Type Primary This field describes the type of gateway as which the MultiVOIP is defined with Alternate respect to the gatekeeper Status registered The current status of the MultiVOIP gateway with respect to the SIP proxy either not registered registered or unregistered Multi Tech Systems Inc 113 SPP Registrars The SS models do not support the SPP signaling protocol SPP Registrars Chapter 6 Using the Software IP Address Port Type t Exit Figure 6 10 SPP registrars SPP Registrars Statistics Servers Field Definitions Field Name Values Description Column Headings registered IP n n n n The IP addres
201. under supervision of Multi Tech Tones 31dB 8 Technical Support mute DTMF Parameters Duration DTMF 60 3000 When DTMF Out of Band is selected this setting determines how long each DTMF ms digit sounds or is held Default 100 ms DTMF Out of When DTMF Out of Band is selected the MultiVOIP detects DTMF tones at its input In Out of Band Band or and regenerates them at its output When DTMF Inband is selected the DTMF Inband digits are passed through the MultiVOIP unit as they are received Out of Band RFC 2833 RFC2833 method Uses an RTP mode defined in RFC 2833 to transmit the DTMF Mode SIP Info digits SIP Info method Generates dual tone multi frequency DTMF tones on the telephony call leg The SIP INFO message is sent along the signaling path of the call You must set this parameter per the capabilities of the remote endpoint with which the VOIP will communicate The RFC2833 method is the more common of the two methods FAX Parameters Fax Enable Y N Enables or disables fax capability for a particular channel Modem Relay Y N When enabled modem traffic can be carried on VOIP system When disabled Enable modem traffic will bypass the VOIP system Modem Bypass mode Max Baud Rate 2400 Set to match baud rate of fax machine connected to channel see Fax machine s user Fax 4800 manual 7200 Default 14400 bps 9600 12000 14400 bps Fax Volume 18 5 dB Controls output lev
202. utput 3 T1 4 Wire Output 4 R 4 Wire Input 2 Wire Input 5 T 4 Wire Input 2 Wire Input 6 R1 4 Wire Output 7 SG Signal Ground Output 8 SB Signal Battery Output Pin Functions FXS FXO Interface FXS Pin Description FXO Pin Description 2 N C 2 N C 3 Ring 3 Tip 4 Tip 4 Ring 5 N C 5 N C Multi Tech Systems Inc 135 Appendix B TCP UDP Port Assignments Well Known Port Numbers The following description of port number assignments for Internet Protocol IP communication is taken from the Internet Assigned Numbers Authority IANA web site www iana org The Well Known Ports are assigned by the IANA and on most systems can only be used by system or root processes or by programs executed by privileged users Ports are used in the TCP RFC793 to name the ends of logical connections which carry long term conversations For the purpose of providing services to unknown callers a service contact port is defined This list specifies the port used by the server process as its contact port The contact port is sometimes called the well known port To the extent possible these same port assignments are used with the UDP RFC768 The range for assigned ports managed by the IANA is 0 1023 Well known port numbers especially pertinent to MultiVOIP operation are listed below Port Number Assignment List Function Port Number telnet 23 tftp 69 snm
203. w long the VOIP has been running since its last booting hours mm ss Hardware ID alpha Indicates version of the MultiVOIP circuit board assembly being used numeric Multi Tech Systems Inc 103 Chapter 6 Using the Software The frequency with which the System Information screen is updated is determined by a setting in the Logs Traces screen which is under the Configuration section Logs Console message Settings MV Enable Console Messages Filters Cancel Logs F Tum Off Logs PRE Help GUI C SMTP C SNMP SysLog Server l Enable m Ea Online Statistics Updation Interval 10 Sec Figure 6 2 Logs Traces screen Statistics Section Ongoing operation of the MultiVOIP whether it is in a MultiVOIP PBX setting or MultiVOIP telco office setting can be monitored for performance using the Statistics functions of the MultiVOIP software The following screens are examples of what can be shown and are followed by detailed descriptions of the categories involved The model and signaling used will affect what is available for display Call Progress Call Progress Detads Channel Charre y r Cal Delads Packet Detads Duration Packets Sent Mode Packets Received Voice Coder Bytes Sert IP Cal Type Bytes Recened IP Call Direction Packets Lost Fiom gt To Detal From gt To Cees _Disconnect Gateway Name IP Address 0 0 0 0 0 0 0 0 _En Options r DIMF Othe Deta
204. without standard telephony mE Ch3 Caller ID service Gha phone of Edwin Smith 763 743 5873 Display shows CID Number 423 l E GID Name Anoka Whse VP3 Phone Book Configuration Time Stamp Date 11 15 Gotowy Nao a Time 6 17pm Q 931 Parameters Inbound Phone Book Channel 2 In x 06 release when SIP protocol is used Remove Prefix Add Prefix Forward Addr CID Name field will duplicate value in Gatekeeper RAS Para CID Number field Figure 5 28 Caller ID example 5 Figure 5 29 VOIP Caller ID Case 5 Call through telco central office without standard CID enters DID channel in H 323 VOIP system Multi Tech Systems Inc 101 Chapter 6 Using the Software Introduction This chapter will primarily cover the day to day operation and maintenance sections of the MultiVOIP software How to update the firmware and software are also covered here should either be needed This section will mainly focus on the Statistics section of the configuration software but there are references to a few of the other sections as they are used more in the daily operations than in a setup situation Software Categories Covered in This Chapter System Information Call Progress Logs IP Statistics Link Management Registered Gateway Details VV VV VV WV Servers o H 323 GateKeepers o SIP Proxies o SPP Registrars gt Advanced o Packetization Time Multi Tech Systems Inc 102 Chapter 6 Using

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