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Multi-Tech Systems MVP210-SS User's Manual

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1. 00 313 IP Address to Ping 0 0 0 eee 312 Least Erroreari eataa 313 No of Pings Received 313 No of Pings Sent eee 313 Ping Size in Bytes eee 312 Pings per Test wo irene 312 Response Timeout 0 312 Round Trip Delay eee 313 Start Now command button 312 Timer Interval between Pings 312 401 Index Link Management Statistics screen field definitions 312 313 Link Status fields Link Management Statistics SCTCOM sci vc sadscedevtsdes vovesstesceess 313 List of Registered Numbers field Registered Gateway Details 316 lithium battery caution 0 0 0 62 ENK LEDaren cenit 18 loading of weight in rack 67 local configuration 0 0 0 cece 92 local configuration procedure detailed analog wee eee 97 SUIMMALY sa ros niiina 96 local voip configuration 4 89 local Windows GUI vs web GUI COMPAFISON ese eeeeeeeseeeeeneeeeeeee 362 local rate calls to remote voip sites BI EE AEE T 244 Log statistics logs field 301 log report email customizing 165 167 log report email triggering 164 log reporting method setting 169 LOG TEPOTS aneii atii 94 log reports amp SMTP 162 log reports by email 162 logging Options 170 logging update interval 170 logging web GUI and 363 Login Name SMTP field 163 Logs Statistics fields Bytes
2. 176 410 MultiVOIP User Guide Call Waiting 174 Call Waiting Enable 0 177 Caller Name Identification Enable TE E EE E EE 178 Calling Party oesie 179 Enable Call Hold eee 177 Enable Call Transfer 176 Enable Call Waiting 177 Enable Caller Name Identification aa e sans ese aaa eee 178 Hold Sequence ou eee 177 Retrieve Sequence 177 Select Channel eee 176 Transfer Sequence see 176 Supplementary Services Info lOSS f6P ERT Sivas 305 Supplementary Services Parameter buttons Copy Channel ee 183 Default haearn isie a 183 Supplementary Services Parameter Definitions 176 177 178 179 180 181 182 183 Supplementary Services Parameter fields Call Waiting Enable 177 Hold Sequence ou eee 177 Retrieve Sequence 177 Supplementary Services Parameter fields Call Hold Enable 177 Call Transfer Enable 176 Select Channel ccceee 176 Supplementary Services Parameter fields Call Name Identification Enable 78 Supplementary Services Parameter fields Calling Party o oo 179 Supplementary Services Parameter fields Allowed Name Types 179 Supplementary Services Parameter fields Alerting Party c eceseeeeeee 180 Supplementary Services Parameter fields Allowed Name Types 180 MultiVOIP User Guide Supplementary Services P
3. 51 phonebook icons BLD ER EA EE ES 249 Ted E E AE TAS 208 phonebook keyboard shortcuts Bl EER E RETES 250 Tl E ER EA E sae 209 phonebook objectives amp considerations Bl R A APTES 247 phonebook pulldown menu Bl PE E oie nae ik 250 T ie oi aan nai ee 209 phonebook sidebar menu Bp E E E niin eet 250 Tl esiie ethane ae 209 phonebook starter configuration Quick Start Instructions 40 phonebook tips Quick Start Instructions 47 phonebook objectives amp considerations Bl ea saccth teniitva cae 242 Threats ara tiveness nate 206 phonebooks inbound vs outbound Bil EE E EEO 247 a A PE EEA EE E 206 Ping Size in Bytes Link Management field 312 Pings per Test Link Management field ennienni seiniin rss 312 pinout BRI connector eee 383 405 Index command cable ccccceeees 380 ethernet cable cc cece 380 TI E1 connector 381 Voice FAX connector 5 381 placement of voip Quick Start Instructions 30 polarity sensitivity DID lines and MVP210 SS 76 DID lines and MVP 410SS 810SS 0 ee eeeeeereees 72 pop ups allowing with Web GUI 111 Port Contact Info SIP Server Predefined Endpoint Parameters iccse 199 Port H 323 Gatekeepers Statistics Servers field cceesceeseeeseees 318 Port SIP Proxies Statistics Servers feldsssc28 i ttecescusenhtiees 319 Port SPP Re
4. 6 289 System Information screen for op amp maint 288 System Information screen accessing E EE E EO 200 System Information update interval SENE hedern an eaer 200 for op amp maint 290 T1 E1 connector pinout 381 table top voip models 0 62 TCP UDP compared Bid EE E datas 255 IP Statistics context 307 308 A a S E EET 214 TDM Routing Option Ethernet IP Parameters field c0cc0 110 technical configuration prerequisites CO eee eee eeeees 92 SUIMMALY ois cecis ieai 89 technical configuration procedure detailed ioeie ies 97 SUMMATY issi eeii a Saa 96 technical support eee eeeeeee 373 telecom safety warnings 62 telephony interface parameters 93 telephony interface parameters Seti Gs cheese ciis bens 126 telephony interfaces USES Of ceeeeeseeeeeeee 71 72 75 76 telephony signaling cadences 153 telephony signaling tones 153 telephony toning schemes 159 temperature operating ee eeeeesecereeseeereeeeeees 67 timeout interval voips under SIP proxy server 152 Timer Interval between Pings Link Management field 312 To gateway statistics logs field 301 toll call savings B leer Pa eer ear eee ea 242 a hte a 206 Tone Detection FXO answer supervision criteria field 140 Tone Detection FXO disconnection SUPCLVISION cecceeseeereeeeeeee
5. 180 181 182 Calling Party o s 179 Allowed Name Types Call Name ID Alerting Party cece ceeeeeeeeeee 180 Busy Party scssccssscsesissssveocseseves 181 Calling Party ooeec 179 Connected Party eee 182 allowing pop ups with Web GUI 111 Alternate IP Address field Bi EEES 257 A A EAT 216 Alternate IP Routing Eliapi binnnen inns 252 A A EE EEA A T 211 Alternate Proxy 1 and 2 SIP Call Signaling fields eee 151 Alternate Routing PSTN failover feature and 216 Alternate Routing field definitions Bil testis E eheietibiei ds es 257 Vist ny sidedistetl Aone aed 216 Alternate Routing field definitions El Alternate IP Address 257 Round Trip Delay ee 257 Alternate Routing field definitions T1 Alternate IP Address 216 Round Trip Delay ee 216 analog voip product family 10 Answer Delay FXO answer supervision field eee 140 Answer Delay Timer FXO answer supervision field eee 140 answer supervision criteria FXO 140 Answer Tones FXO answer supervision field eee 140 Index Append SIP Proxy Domain Name in User ID proxy server 151 Auto Disconnect field group 125 AUtOC All eresze 120 AutoCall Voice Fax Params and Pass Through Enable FXS Loop StA e eeni eiai 120 AutoCall Offhook Alert field 720 121 Automatic Disconnection field 125 Available Tones FXO
6. Shortcut Sidebar Multi oIP Multi Configuration Ctrl H 112 Configuration Ethernet f IP Voie ej ax Interface MultiVOIP User Guide Technical Configuration In each field enter the values that fit your particular network m Voice Fax Parameters Select Channel A Voice Gain Fax Modem parameters IV Fax Relay Enable S Cancel Input jO ZhdB OutputjO Zh dB J Modem Relay Enable p Dtmf Max Baud Rate 14400 7 Copy Channel Gain Hah 4 Low 7 Fax Volume 35 dB itter Val faco ms Duration 100 a Jitter Value GA Help DTMF Out Of Band Fixed Duration Mode Ma Out Of Band Mode Rfc2833 gi Info m Coder Advanced Features Manual Automatic IV Silence Compression Selected Coder G 723 1 6 3 kbps 7 IV Echo Cancellation Max bandwidth fio kbps J Forward Error Correction Auto Call 7 OffHook Alert Auto Call OffHook Alert ne IV Generate Local Dial Tone OffHook Alert Timer 10 secs Phone Number en Default m Dynamic Jitter Buffer Minimum Jitter Value eo ms Maximum Jitter Value 300 ms Optimization Factor p m Automatic Disconnection J Jitter Value so ms J Consecutive Packets Lost 20 E Call Duration 180 secs M Network Disconnection 300 secs 113 Technical Configuration MultiVOIP User Guide Note that Voice FAX parameters are a
7. 763 555 4071 iguration Anoka Whse VP3 m Q 931 Parameters Inbound Phone Book Channel 2 Gatekeeper RAS Paral Remove Prefix Add Prefix Forward Addr 423 748 Figure 5 2 Voip Caller ID Case 2 Call through telco central office without standard CID enters H 323 voip system 136 MultiVOIP User Guide Technical Configuration lt CID Flow Call i received Call originates here ere Chi at 5 47pm Sept 27 Terminating Generating _ Central Office O Q VolP a cho __ FXO without standard telephony d Clock Network Ch3 TN Caller ID service AEN 15 26 5 31 Ch4 D phone of Display shows SPP Protocol Henry Brampton 763 555 4077 CID Number 423 CID Name Shipping Dept Time Stamp Date 0927 Time 1747 Inbound Phone Book Channel 2 Remove Prefix Add Prefix Forward Addr Phone Book Configuration if Description field in Add Edit Inbound Phone Book is used Gateway Name Anoka Whse VP3 OR Add Edit Inbound Phone Book CID Number 423 O Use as default entry CID Name Anoka Whse VP3 Remove Eres r Add Prefix Time Stamp Date 0927 Channel Number Channel Time 1747 Description Shipping Dept if Description in Add Edit Inbound Phone Book is blank Figure 5 3
8. 41 QS Phonebook Starter Config MultiVOIP User Guide 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 Long Distance Example Example Seattle Chicago system London Birming system Answer enter 81312 as Leading zero of Destination Pat Birmingham area code is tern in Outbound dropped when combined Phone book of with national dialing Seattle voip access code Such practices vary by country Answer enter 90121 as Destination Pat tern in Outbound Phonebook of London voip Not 900121 Euro International Call Example Rotterdam Bordeaux system Answer enter 903305 as Destination Pattern in Outbound Phonebook of Rotterdam voip 42 MultiVOIP User Guide QS Phonebook Starter Config 6 In the Remove Prefix field enter the initial PBX access digit 8 or 9 North America Euro National Call Long Distance Example Example Seattle Chicago system London Birming system Answer enter 8 in Remove Answer enter 9 in Remove Prefix field of Prefix field of Seattle Outbound London Outbound Phonebook Phonebook Euro International Call Example Rotterdam Bordeaux system Answer enter 9 in Remove Prefix field of Outbound Phonebook for Rotterdam voip Some PBXs will not hand off the 8 or 9 to the voip But for thos
9. 116 Duration RADIUS Attributes field e t e A eisai as ae 190 Duration SMTP logs field 165 Duration statistics logs field 301 Dynamic Jitter Buffer field 123 Dynamic Jitter field group 123 Dynamic Jitter fields 124 dynamic registration 198 E amp M interface MVP210 SS matching telco trunk line 76 USES OF recie 76 E amp M interface MVP 410SS 810SS matching telco trunk line 72 USCS Of rh ae aa ER 72 E amp M Interface Parameter fields Detection Range flash hook 146 Disconnect on Call Progress Tone EE EE EE 144 Flash Hook eee eee eeeeeee 146 Inter Digit Timer dialing 145 Interface aiino eie 144 Message Waiting Indication 145 Pass Through eeseeeeseeeseeeee 144 Index Regeneration dialing 145 Signal sorires 144 Typene 144 Wink TimMer 144 E amp M Parameter definitions 144 145 146 E amp M Parameters 0 00 0 eeeeeeeeeeee 143 Echo Cancellation field 119 echo removing se eeeeeseeeeeeetees 119 Edit selected Inbound Phonebook Entry icon BV series ust idee 249 A i PE E ESE 208 Edit selected Outbound Phonebook Entry icon Bd EER E EAS 249 TL a ar rasera ak Eaa 208 email account for voip unit 163 email address for voip 94 162 email log reports ossessi 162 email logs illustration 168 EMC Safety R amp TTE Direc
10. FXS Ring Count FXS 1 99 Maximum number of rings that the MultiVOIP will issue before giving 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 Reversal Y N When selected this option implements Answer Supervision and Disconnect Supervision to the FXO interface using current reversal to indicate events Applicable only when FXS and FXO interfaces are connected back to back 130 MultiVOIP User Guide Technical Configuration FXS Loop Start Interface Parameter Definitions cont d 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 curren
11. MEMO To LT Department Ask Mail Server ra Ghia ascoan for VOTE 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 voip unit2 biggytech com Get the IP address of the vine mail server computer as well 27 QS Gathering Phone IP Details MultiVOIP User Guide Config Info CheckList Type of Config Info MultiVOIP Gathered Configuration screen on which to enter Config Info Info Obtained 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 Choices E amp M FXS FXO In FXO FXS systems DIP DPO channels used for phone fax or key system are FXS channels used for analog PBX extensions or analog telco lines are FXO E amp M info only if E amp M is used Interface Parameters e Type 1 5 2 or 4 wires e Dial Tone or Wink Country Code Regional Parameters Email address for voip optional SMTP Parameters SIP Operating Mode SIP Server Configuration e Survivability e Stand Alone Network Locations of Alternate SIP Call Signaling SIP Proxy units if used IP Address or Domain Name Alt 1 Alt 2 Endpoint Info SIP Server Predefined Device Name Regist Type Endpoints IP Address Device Name Regist Type IP Address Port a
12. L0p uuo IT CH Oda Id 40JD9UUOD T I P uoisuajyxa Xd bojpuv 10 wajshs auoyd fay io uonpinbyuoo ajowad iof aul auoyd Bojpun o sp uuos xof auoyd 0 sp uuos oul SLOq 0 Buyqnod 4ojaauu0s TI FJ OXA 10P uuoD IT CH SKA aoUffo 0913 40 Xgd wo1f aul Yuni WWW 0 SJaaUUOD 10P UUOI Gh CA NOT quawdinba auoyd o buyqvog WEPOW ox sx4 wga OXa SX4 WES oxsisxswea OXd SX4 W3 Sn aap DE OCI E3 EJEA LOJDDUUOD GH PA y4omjau dI 4noh o Bbuyqog young HO UO UB LULUO youlauy gt P O puewuoo oaea O NS OCS A punoly yuo 0 JoouUOD 6 dq 4a3ndwuoo yo 40JoaUUOD awy D 7D pasn smasog CZ d dIOAUIN 10 L0pauuop s nod fun fo sojoauuos g juo Burpunosy adoNYOs qIOAVINA sarod 10j2aUUod p SPY SSOTbdAW apnjda0eN Gururunu sayndwos 0 Buyqvo sarod 10J2aUUod Z SOY SSOTSdAN 1q09 4amo0g SS OL8dAW 8 SS OLPdAIN dnyooH dIOAINW Bojeuy 31 QS Quick Hookups MultiVOIP User Guide Quick Hookup for MVP210 SS CH1 CH2 E amp M FXO E amp M FXSFXO Z ETHERNET RS232 lir 7 rd ir oai e TILI S 10 100_COMMAND POWER od rr aAA Connections Voice Fax Channel 1 2 E amp M FXO FXS ae Power Connection i Command Port Con
13. 4 The dialed string matches an inbound phonebook entry at the Lincoln voip namely 1402263742 5 The Lincoln voip rings one of the three FXS ports connected to the Lincoln key phone system 6 The call will be routed to extension 7424 either by a human receptionist operator or to an auto attendant which allows the caller to specify the extension to which they wish to be connected 239 T1 Phonebook Configuration MultiVOIP User Guide Site F calling Site D A voip call from a Lincoln key extension to extension 3117 on the PBX in Pierre South Dakota A The required entry in the Lincoln Outbound Phonebook to facilitate origination of the call would be 31 The string 1615492 would have to be added as a prefix The call would be directed to the Pierre voip s IP address 200 2 9 9 B The corresponding entry in the Pierre Inbound Phonebook to facilitate completion of the call would be 1615492 1 Caller at Lincoln picks up phone receiver presses button on key phone set This button has been assigned to a particular voip channel any one of the three FXS ports 2 The caller at Lincoln hears dial tone from the Lincoln voip 3 The caller at Lincoln dials 3117 4 The Lincoln voip adds the prefix 1615492 and sends the entire dialing string 16154923117 to the Pierre voip at IP address 200 2 9 9 5 The Pierre voip matches the called digits 16154923117 to its Inbound Phonebook entry 1615492 6
14. Frequency Frequency Cadencelvecs0v0ll Gant Gard These tone pairs are used in T 7 DINNI 2 2 conjunction with the FXO 5 470 0 N 2000 6HV0 15 A ae N Supervision secondary These tone pairs are used to supervise pai of ae INTERFACE answering and disconnection of calls 5 configuration screen 4 Add Edt Tone Country Selection For Buin Modem Urted StarewUS EKAR Frequency 1 Fimquerncy 2 Custom Tene Pas Semings Iepa Distoe Tore Par Vaes Coderce 1 Evequency 30 Hz Cadercel 0 These tone Hie entrees f pairs serve the same PEE RC Gain is B Cadences standard tones Cadence but use different frequencies Remote Configuration Command Modem Each MVP410 and MVP810 MultiVOIP unit contains a built in modem This modem allows the MultiVOIP to be configured remotely when a standard POTS line is connected to the Command Modem connector on the back panel of the MultiVOIP In the Country Selection for Built In Modem field drop down list select the country that best fits your situation This may not be the same as your selection for the Country Region field The selections in the Country Selection for Built In Modem field entail more detailed groupings of telephony parameters than do the Country Region values In each field enter the values that fit your particular system 154 MultiVOIP User Guide Technical Configuration
15. Some PBX units can also insert digits automatically when they receive certain dialing strings from a phone station For example a PBX may be programmable to insert automatically the three digit VOIP identifier strings into calls to be directed to analog VOIPs The MVP3010 offers complete flexibility for inter operation with PBX units so that a coherent dialing scheme can be established to connect a company s multiple sites together in a way that is convenient and intuitive for phone users When working together with modern PBX units the presence of the MVP3010 can be completely transparent to phone users within the company 280 MultiVOIP User Guide E1 PhoneBook Configuration International Telephony Numbering Plan Resources Due to the expansion of telephone number capacity to accommodate pagers fax machines wireless telephony and other new phone technologies numbering plans have been changing worldwide Many new area codes have been established new service categories have been established for example to accommodate GSM personal numbering corporate numbering etc Below we list several web sites that present up to date information on the telephony numbering plans used around the world While we find these to be generally good resources we would note that URLs may change or become nonfunctional and we cannot guarantee the quality of information on these sites URL Description http phonebooth interoci
16. well Get the IP address of the mail server computer as MEMO To LT Department t re email account for VOIP O Fa AE voip unit2 biggytech com a 94 MultiVOIP User Guide Config Info CheckList Technical Configuration Type of Configuration Info Gathered MultiVOIP Configuration screen on which to enter the Info Info ane Info el IP Info for voip unit Ethernet IP sP address Parameters e Gateway e DNS IP if used e802 1p Prioritization if used Interface Type Interface Choices E amp M Parameters FXS FXO DIP DPO In FXO EXS systems channels used for phone fax or key system are FXS channels used for analog PBX extensions or analog telco lines are FXO E amp M info Interface only if E amp M is used Parameters e Type 1 5 e 2 or 4 wires e Dial Tone or Wink Country Code Regional Parameters Email address for voip SMTP Parameters optional Reminder Be sure to Save Setup after entering configuration values 95 Technical Configuration MultiVOIP User Guide Local Configuration Procedure Summary After the MultiVOIP configuration software has been installed in the Command PC which is connected to the MultiVOIP unit several steps must be taken to configure the MultiVOIP to function in its specific setting Although the summary below includes all
17. Register The time remaining in seconds before the Duration TimeToLive timer expires If the gateway fails to reregister within this time the endpoint is unregistered Status The current status of the gateway either registered or unregistered No of The number of gateways currently Entries registered to the Registrar This includes all SPP clients registered and the Registrar itself Details Count of If a registered gateway is selected by Registered clicking on it in the screen The Count of Numbers Registered Numbers will indicate the number of registered phone numbers for the selected gateway When a client registers all of its inbound phonebook s phone numbers become registered List of Lists all of the registered phone numbers for Registered the selected gateway Numbers 316 MultiVOIP User Guide Operation amp Maintenance About Alternate Server Statistics Accessing Alternate Server Statistics Pulldown Stabisties Call Progress rltak n Logs Oro IP Raits Cote Regetered Gateway Details Corthakew Link Management Corte HIZI Gatekeepers Core SE Prodes rhe SPP Regras rh Shortcut Sidebar Configuration Advanced Phone Book Statistics Call Progress Logs IP Statistics Link Management DEE FE EF Ctrl Alt 4 Registered Gateway Details H 323 Gatekeepers SIP Proxies SPP Registrars H Save Setup Connection Help 317 H 32
18. Server Client Phonebook MVP r Unit 100 kar 200 2 9 8 Host Holds phonebook for both Series 1 analog VOIPs pad Other extensions x7401 x7429 Ree 7 N S 263 7400 P sa 7 118 943 5632 ite C Reading Area Residential 269 E1 Phonebook Configuration MultiVOIP User Guide The Series I analog VOIP phone book resides in the Host VOIP unit at Site B It applies to both of the Series I analog VOIP units Each of the Series II analog MultiVOIPs the MVP210 and the MVP410 requires its own inbound and outbound phonebooks The MVP3010 digital MultiVOIP requires its own inbound and outbound phonebooks as well These seven phone books are shown below Phone Book for Analog VOIP Host Unit Site B VOIP Dir IP Address Channel Comments OR Destination Pattern 102 200 2 9 8 2 Site B FXS channel Reading UK 101 200 2 9 8 1 Site B FXO channel Reading UK 201 200 2 9 7 1 Site A FXS channel Birmingham 421 200 2 9 6 0 Site E FXS channel Carlisle UK 018226374 200 2 9 5 0 Gives remote voip users access to key phone Note 3 system extensions at Tavistock office Site F The key system might be arranged either so that calls go through a human operator or through an auto attendant which prompts user to dial the desired extension 0182 200 2 9 5 4 Gives remote voip users access to Tavistock PSTN via FXO port 4 at Site F 3xx 200 2
19. button to button 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 157 Technical Configuration MultiVOIP User Guide Regional Parameter Definitions cont d Field Name Values Description Country country name MultiVOIP units operating with the Selection for X 06 software release and above Built In include a built in modem The ee administrator can dial into this modem Modem to configure the MultiVOIP unit not applicable remotely The country name values in to MVP this field set telephony parameters that 130 130EXS allow the modem to work in the listed country This value may be different MVP210 than the Country Region value For MVP410ST or example a user may need to choose MVP810ST 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 of supervisory tone pair name specified Cannot be same as name of any by user standard tone pair 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 3dB to 31dB f
20. Details the traffic control PiSits Sent sent by this gateway server if any to the remote being used gateway presuming whether an H 323 that DTMF is set to gatekeeper a SIP Out ofBanid proxy or an SPP registrar gateway will be displayed here if the call is handled through that server Disconnect Indicates whether the call was disconnected simply Reason because the desired conversation was done or some other irregular cause occasioned disconnection e g a technical error or failure Values are Normal and Local disconnection From Details To Details Gateway Originating Gatew N Completing or Number gateway answering gateway IP Addr IP address where IP Addr IP address where call call originated was completed or answered Descript Identifier of site Descript Identifier of site where call where call was originated completed or answered Options When selected log Options When selected log will not Silence will not use Silence Compression and Compression and Forward Error Forward Error Correction by call Correction by party originator answering call 167 Technical Configuration MultiVOIP User Guide r Logs I Enable Console Messages Cancel To use the SMTP Para meters screen SMTP I Tum Off Logs Reporting Method a gs screen must first be selected in the Lo SMTP Parameters SMTP 2 oc SNMP Login Name VOIP UNIT 3 acmetech com
21. Mail Server IP Address 164 Mail Type tiroe pess 164 Number of Days osese 164 Number of Records 04 164 PaSSWOIG ainena ruines 164 Port Number niscenire ienr 164 Recipient Address 0 164 Reply To Address 00 164 Requires Authentication 163 SUDJOCE tie sii necites 164 SMTP parameters accessing 162 SMTP parameters setting 162 SMTP port standard 164 SMTP prerequisites cee 94 SMTP enabling 0 0 eee eee 162 SNMP log reporting type button 171 SNMP agent program 90 software uninstalling detailed 85 UPAates 2 vessccscessesadeecseueeteeceseeess 90 software MultiVOIP uninstalling 348 software configuration SUMMALY eisi nran i 78 software installation detailled eiren nnion 78 Quick Start Instructions 33 software loading csceeeeteeeee 78 software version numbers 80 software MultiVOIP moving around iN eeee 101 software MultiVOIP Screen SUrfiNg iN ee eee 101 Solving Common Connection Problems i csc stesoedh eseesenees 100 sound quality improving 119 SPP Registrars Statistics Servers IP Address sei cteakiiterse sens 320 Index POPE ees 3cetbe en aeons evoke EEE 320 Typene nee eE 320 SPP Registrarss Statistics Servers SAMUS re nter prore hes 320 SRYV record eeesceceesseceeeeteeeeee
22. NAT Traversal Ctrl Alt Sht n Shortcut Sidebar El Configuration Ethernet IP Ctrl Alt U Yoice Fax Interface SIP Call Signal Regional SMTP RADIUS Logs Traces 187 Technical Configuration MultiVOIP User Guide Radius IV Enable Authentication OK Server Address 192 168 2 10 Cancel Authentication Port 1812 Select Athibutes ms Retransmission Interval 2000 Number of Retransmissions 3 Help Shared Secret Radius Attributes IV Select All Attributes V Channel Number Start Date Time IV Duration Call Mode Packets Sent IV Packets Received Bytes Sent IV Bytes Received Packets Lost IV Coder Outbound Digits IV Prefix Matched Call Status From Details To Details IV Gateway Name M Gateway Name IV IP Address IV IP Address IV Description IV Description IV Options V Options The fields of the RADIUS screen are described in the table below 188 MultiVOIP User Guide Technical Configuration RADIUS Screen Field Definitions Field Name Values Description Server n n nn IP address of the RADIUS server that Address 0 255 handles accounting for the current MultiVOIP unit Accounting numeric TDM time slot at which RADIUS Port 1 65535 accounting information will be transmitted and received Retrans mission If the MultiVOIP sends out a packet to Interval the RADIUS server and doesn t Number of 0 255 receive a response in the retransmit
23. Options When selected log will not use will not use Silence Silence Compression and Compression and Forward Error Forward Error Correction by party Correction by call answering call originator 192 MultiVOIP User Guide Technical Configuration 19 Set Baud Rate The Connection option in the sidebar menu has a Settings item that includes the baud rate setting for the COM port of the computer running the MultiVOIP software First it is important to note that the default COM port established by the MultiVOIP program is COM1 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 dialog box es of your Windows operating system If COM1 is not available you must change the COM port setting to COM2 or some other COM port that you have confirmed as being available on your PC The default baud rate is 115 200 bps 193 Technical Configuration 20 Set SIP Server Configuration parameters MultiVOIP User Guide Accessing SIP Server Configuration Parameters Sidebar MultiVoIP MultiVOIP SS Configuration Advanced Phone Ae amp amp B Configuration Advanced Phone Book Statistics Save Setup Connection Sip Server Predefined Endpoints Endpoint Statistics Logs History Save Setup SIP Server Configuration Operating Mode Survivability Status Check Regi
24. Password em Mail Server IPaddress Port Number r Mail Type C Text Subject ReplyTo Address Recipient Address m Mail Criteria Number of Records T Number of Days m Custom Fields Cancel Help Hora Select Fields a Custom Fields lets the user determine ich technical details ofeach phone call to record in the logs Mail Now I Select All Fields T Channel Number M Duration I Packets Sent I Bytes Sent I Packets Lost T I Start DateTime I Call Mode I Packets Received I Bytes Received T Coder I Prefix Matched M Outbound Digits I Call Status R Email reports can be sentout periodically or when a certain numberof phone calls have been logged From Details I GatewayName M IPAddress r To Details I GatewayName N IPAddress I Description I Description I Options I Options Call Logs Sl No Start Date amp Time Duration Status 01 17 2002 amp 09 43 24 00 01 47 01 17 2002 amp 10 30 33 00 26 46 01 17 2002 amp 11 05 22 00 09 47 01 17 2002 amp 11 16 02 00 01 25 01 17 2002 amp 11 21 02 00 00 33 01 17 2002 amp 11 51 26 00 00 00 Call Mode Success Voice The requested technical details will then appear in the log report that is automatically emailed to the VoIP administrator Success Voice Success Voice Success Voice Success Voice Unsuccess Voice Call Logs From IP Addr To
25. Technical Configuration MultiVOIP User Guide Regional Parameter Definitions Field Name Values Description Country USA Japan UK Name of a country or region that Region Custom uses a certain set of tone pairs for Note dial tone ring tone busy tone and Survivability unobtainable tone fast busy tone indicates a tone survivability tone tone special type of heard briefly 2 seconds after going call routing offhook denoting survivable mode redundancy amp of voip unit and re order tone a applies to tone pattern indicating the need for MultiVantage the user to hang up the phone In voip units only 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 ey xi screen Supervision Tones have been set to default values in Interface Page 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 Standard Tones fields Type column dial tone Type of telephony tone pair for ring tone which frequency gain and busy tone cadence are being presented unobtainable tone fast busy survivability Frequency 1 tone re order tone freq in Hertz Lower frequency of pair Frequency 2 freq in Hertz Hi
26. ccccseciccsesc neeite 165 From Gateway Numbev 167 From IP Address cc008 167 Outbound Digits Received 166 Outbound digits sent 167 Packets LoSt ccccseseseeceeeeees 166 Packets Received c008 165 Packets Sentens nosten 165 Prefix Matched ccceeee 166 Select Alleri socckegfasies netee 165 Server Details 0cccccceeeeeee See Start Date Time 0 cce 165 Index To Gateway Numbet 167 To IP Address eee 167 Custom Tone Pair Settings definitions sesh E A etd ame ESS 160 161 Custom Tone Pair Settings fields Cadence Ls eccsscueedpedasories 161 Cadence Zanosi eein 161 Cadence 3ra 161 Cadence 4 nhao 161 Frequency Desierta 160 Frequency 2 yesenpiniis pirin 160 G nn Teana ipana 160 Gain Dorea esrara oieri 160 Tone Pair snoii iscritte 160 customized log email 165 167 customized RADIUS Accounting 190 customized RADIUS accounting par meterS isien a 192 data Capacity isise serria 13 data capacity analog voips 10 data capacity digital voips 9 data capacity ISDN BRI voips 11 data compression eeceeeeeeeeee 14 Date amp Time Setup program menu option command eee 327 Date and Time Setup option description MultiVOIP program MEHU Jn pecs arrir iare 324 debugging messages sss sesee 171 Default Supplementary Services fieldanase ninta
27. or resetting of the counter within the MultiVOIP software Received integer Number of error laden TCP packets with value received by this VOIP gateway since the Errors last clearing or resetting of the counter within the MultiVOIP software 309 Operation amp Maintenance MultiVOIP User Guide IP Statistics Field Definitions cont d RTP Packets Voice signals are transmitted in Realtime Transport Protocol packets RTP packets are a type or subset of UDP packets Transmit integer Number of RTP packets transmitted by ted value this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of RTP packets received by this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of error laden RTP packets with value received by this VOIP gateway since the Errors last clearing or resetting of the counter within the MultiVOIP software value 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 Transmit integer Number of RTCP packets transmitted ted value by this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of RT
28. r 1 Flagstaff Office Area 520 204 16 49 75 8 Channel Analog VolP MVP810 r MultiVOIP User Guide One Common Situation Voip Example This company has offices in three different cities The PBX units all operate alike Notably they all give access to outside lines using 9 They all are smart enough to identify voip calls without using a special access digit 8 is used in some systems Finally the system operates so that employees in any office can dial employees in any other office using only three digits Here are the phonebooks needed for that system Santa Fe Office Area 505 204 16 49 74 8 Channel Analog VolP MVP810 PBX System Main Number 444 3200 Each Outbound Phonebook contains two pairs of entries two entries for each remote site Whenever an out of town employee dials a 12 digit number beginning with the listed 5 digit destination pattern 9 1 area code of another company location the PBX hands the call to the voip system The local voip strips off the 9 and directs the call to the IP address of the remote PBX System Main Number voip Theremote voip receives the call and hands it to its PBX The PBX then completes the call to the PSTN l I The one digit Outbound destination I patterns pertain to 3 digit calling i between company employees I a 51 QS Phonebook Example Voip Sites w
29. received over the IP network in the course of this call 303 Operation amp Maintenance MultiVOIP User Guide Logs Screen Details Field Definitions cont d Field Name Values Description Call Details cont d FROM Details Gateway Name alphanumeric Identifier for the VOIP gateway string that originated this call IP Address X X X X IP address of the VOIP gateway where x has a from which the call was range of 0 to 255 _ 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 string that completed terminated this call IP Address X X X X IP address of the VOIP gateway where x has a at which the call was completed range of 0 to 255 terminated Options Displays VOIP transmission options used by the VOIP gateway terminating the call These may include Forward Error Correction or Silence Compression 304 MultiVOIP User Guide Operation amp Maintenance Logs Screen Details Field Definitions cont d Supplementary Services Info Call Transferred phone number Number of party called in To string transfer Call Forwarded phone number Number of party called in To string forwarding 305 Operation amp Maintenance About IP
30. 105 Frame Typen tutapi 104 Ethernet IP Parameter screen fields Enable DNS ceeeeeceeeteeeeeee 109 Ethernet IP Parameters screen fields Call Control Priority 105 Call Control PHB 005 107 Diff SeN s e o 107 DNS Server IP Address 109 Enable DHCP cccceeseeeeee 106 Enable SRV c ccssceeseeeseeeee 109 FTP Server Enable 0 109 Gateway ranee ena i 106 Gateway Name nsee 106 IP Addr ss inira 106 IP Mask e nenranet onnea 106 Others Priorities 105 Packet Prioritization 802 Ip 104 TDM Routing Option 110 Use TDM Routing for Intra Gateway Calls wo 110 VEAN TID cheese sa ieee 105 VoIP Media Priority 105 Voip Media PHB eee 107 Ethernet IP Parameters screen ACCESSING yoa E E 102 European Community Directives 375 factory default software settings 335 factory defaults downloading 335 factory repair for customers U S amp Canada i siccaeceSsctcicis coves descets 371 MultiVOIP User Guide failover PSTN feature 216 FAQ for MultiVOIPS c cece 8 fast busy unobtainable tones 159 fax baud rate default 0 000 116 Fax Enable field cccccseeeee 116 fax machine connecting to analog voip MVP210 SS eee 76 connecting to MVP210 SS voip 75 connecting to voip MVP 410SS 810SS cee 71 72 FAX Parameters 116 fax ton
31. 4094 The 802 1Q IEEE standard allows virtual LANs to be defined within a network This field identifies each virtual LAN by number 105 Technical Configuration MultiVOIP User Guide Ethernet IP Parameter Definitions cont d Field Name Values Description IP Parameter fields Gateway Name alphanumeric Descriptor of current voip unit to distinguish it from other units in system Enable DHCP Y N disabled by default Dynamic Host Configuration Protocol is a method for assigning IP address and other IP parameters to computers on the IP network in a single message with great flexibility IP addresses can be static or temporary depending on the needs of the computer IP Address 4 places 0 255 The unique LAN IP address assigned to the MultiVOIP IP Mask 4 places 0 255 Subnetwork address that allows for sharing of IP addresses within a LAN Gateway 4 places 0 255 The IP address of the device that connects your MultiVOIP to the Internet 106 MultiVOIP User Guide Technical Configuration Ethernet IP Parameter Definitions cont d Field Name Values Description DiffServ DiffServ PHB Per Hop Behavior values Parameter pertain to a differential prioritizing fields system for IP packets as handled by DiffServ compatible routers There are 64 values each with an elaborate technical description These descriptions are f
32. Bytes Sent aossosiioisisiisivss 191 Call Status eissis 191 CODED cessed activi desis 191 393 Index OPtiONS serrio enrere onses 192 OPtiONS aiioe ieee nE 192 Description callee 0 192 Description caller 6 192 Disconnect Reason 0 191 From Gateway Numbev 192 From IP Address ccc00088 192 Outbound Digits sent 191 Packets Lost ccccseseeeceeceeees 191 Prefix Matched 0 ccceeee 191 Server Details 00 cece 191 To Gateway Number 192 To IP Address cccecccceeeeees 192 Custom Fields RADIUS Attributes Call Mode cceccceeeccceeeeecenee 190 Channel Numbet 0008 190 Duration cccccceeccceeeseeeeeneee 190 Packets Received 008 190 Packets Sent cccceeseeeeeeeeees 190 Select All cc cecceeceereee 190 Start Date Time cce 190 Custom Fields SMTP log email Bytes Received cesses 166 Bytes Sent cisssssetssstscesveotsceeves 166 Call Direction 00 ccceeee 166 Call Mode cceceeeccceceseeeeeneee 165 Call Stats sinr n 166 Call Ty pearance s 166 Channel Numbet 008 165 COdeL ann rn sk 166 OPtiONS eee eeeesecseeeeeneeeeeeee 167 Options rimarini 167 Description callee 167 Description caller 0 167 Disconnect Reason 06 167 DTMF Capability 166 DU atlOn coc
33. Gain 2 custom tone field 160 Gain 2 tone pair scheme 157 158 Gateway Ethernet IP Parameters Meld A AE EE Haha sees 106 Gateway Name callee statistics logs field oo ee eeeeeeeeteeeeeeee 304 Gateway Name caller statistics logs Held asa cee eee entices 304 MultiVOIP User Guide Gateway Name Ethernet IP Parameters field c0 106 Gateway Number From Details RADIUS Attributes field 192 Gateway Number From Details SMTP logs field 0 167 Gateway Number To Details RADIUS Attributes field 192 Gateway Number To Details SMTP logs fela a eoir ekis 167 Generate Current Reversal FXS Loop Start ee 130 Generate Local Dial Tone Voice FAX AutoCall Offhook Alert Heldri rinitis 121 Generation Flash Hook Options field E amp M atoae rottene 146 EXO EE T 135 GK Name H 323 Gatekeepers Statistics Servers field 318 grounding in rack installations 67 MV P210 ipoe saira 76 GUI log reporting type button 171 T1323 coder ssesscssccsscisccsssesiadtesseses 118 H 323 Gatekeepers Statistics Servers GK Nametsv ei aed Haake 318 IP Addres Siriei 318 Portae e O EE 318 Priotity eeii dhe ES 318 StAtUs a a 318 Type e Bis 318 H 450 features compatible with SIP EN E E Siseeeehes ea tees 173 H 450 features incompatible with SIP E E E E 14 H 450 functionality logs POP tienda ee as 305 Hardware ID System In
34. If you notice that something important is lacking please let us know Additional Resources The MultiTech web site www multitech com offers both a list of Frequently Asked Questions the MultiVOIP FAQ and a collection of resolutions of issues that MultiVOIP users have encountered these are Troubleshooting Resolutions in the searchable Knowledge Base MultiVOIP User Guide Overview Digital MultiVOIP Products ares MVP MVP MVP MVP 7 Model 2410 24 48 3010 30 60 Function T1 T1 E1 E1 digital digital digital digital VOIP VOIP VOIP VOIP unit add on unit add on card card Capacity 24 24 30 30 channels added channels jadded channels channels Chassis 19 1U circuit 19 1U circuit Mounting rack card rack card mount only mount only Overview MultiVOIP User Guide Analog MultiVOIP Products poemeton MVP MVP MVP MVP MVP lt 7 Modi 810 428 410 210 130 130FXS Function analog add on analog analog analog voip card voip voip voip Capacity 8 4 added 4 2 1 channels channels channels channels channel Chassis 19 1U circuit 191U Table table Mounting rack card rack top top mount only mount ee MVP MVP MVP Meee 810 SS 410SS 210SS Function analog voip acts analog voip acts analog as minimal SIP as minimal SIP voip acts proxy server proxy server as minimal giving SIP giving SIP proxy SIP
35. Out of Band RFC2833 SPP 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 Outbound Digits 0 9 The digits sent by Multi VOIP to Received PBX telco that were acknowledged as having been received by the remote voip gateway Outbound Digits 0 9 The digits transmitted by the Sent MultiVOIP to the PBX telco for this call 302 MultiVOIP User Guide Operation amp Maintenance Logs Screen Details Field Definitions cont d Field Name Values Description Call Details Server Details n n n n When the Multi VOIP is for n 0 255 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 The number of data packets sent over the IP network in the course of this call Packets received integer value The number of data packets received over the IP network in the course of this call Packets loss integer value The number of voice packets from lost this call that were lost after being received from the IP network Bytes sent integer value The number of bytes of data sent over the IP network in the course of this call Bytes received integer value The number of bytes of data
36. Q x A seach Se Favorites Ol A t LA Address http 192 168 41 81 MISE n MultiVOIP 410 Configuration Multi ec i Ethemet IP Systems VoicelFax Interface Call Signaling SNMP rEthernet IP Parameters Regional SMTP Ethernet Parameters RADIUS C Packet Prioritization 802 1p Frame Type Logs Traces NAT Traversal 802 1p Parameters Supplementary Services Priority Systern Information Call Control 3 Excellent Effort Advanced Phone Book VoIP Media 6 Voice Statistics m Change Password Others 0 Best Effort Save amp Reboot Logout Help Cancel VLAN ID IP Parameters Gateway Name MultivolP Diff Serv Parameters Call Control PHB 34 C Enable DHCP P e VolP Media PHB 46 IP Address 192 168 41 81 IP Mask 255 255 255 0 FTP Server pree Z Enable C Enable DNS C Enable DNS SRV DNS Server IP Address The Windows GUI gives access to commands via icons and pulldown menus whereas the web GUI does not 362 MultiVOIP User Guide Operation amp Maintenance Multi oIP Multi OIP 410 v6 08 C Firmware Aug 04 2005 Configuration Advanced Phone Book Statistics Download Connection Help AESSEPTEBES OMA E Configuration VoicejFax p Ethemet IP Parameters Interface E Call Signaling i fve E SNMP ees pi Erame Type TYPE I Regional 8021p Parameters SMTP Priority RADIUS Call Control
37. 0 dB Output Gain 31dB Modifies audio level being output to to the device attached to the voice 31dB channel The default and recommended value is 0 dB DTMF Parameters 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 3dBto Default value 4 dB Not to be High Tones 31dB amp changed except under supervision of mute MultiTech s Technical Support DTMF Gain 3dBto Default value 7 dB Not to be Low Tones 31dB amp changed except under supervision of mute MultiTech s Technical Support 115 Technical Configuration MultiVOIP User Guide Voice Fax Parameter Definitions cont d Field Name Values Description DTMF Parameters Duration 60 3000 When DTMF Out of Band is selected DTMF ms this setting determines how long each DTMF digit sounds or is held Default 100 ms Not supported in 5 02c BRI software DTMF Out of When DTMF Out of Band is selected In Out of Band or the MultiVOIP detects DTMF tones at Band Inband its input and regenerates them at its output When DTMF Inband is selected the DTMF digits are passed through the MultiVOIP unit as they are received In 502c BRI software DTMF Out of Band can be checked or unchecked Out of Band RFC 2833 RFC2833 method Uses an RTP Mode SIP Info mode defined in RFC 2
38. 10 100Mbps Ethernet interface and a command port for configuration SIP Survivability The MVP210SS MVP410SS and MVP810SS have a special capacity that reaches beyond ordinary voip functionality they can direct call traffic for phones connected to their channels or phones connected to channels of other SIP gateways in the network this is basic SIP server functionality The MVP SS unit would normally be located at a remote branch office served by a central SIP server PBX at the organization s main office The MVP SS is intended as a backup in case the network s main SIP server often a PBX fails or loses contact with the group of gateways at the remote branch office If the main SIP server fails the MVP SS allows branch office phone users to call each other and access the PSTN via POTS lines or a key telephone system Main Office Central SIP Server Main PBX Router n an to ap ai a a ab ee le ee eee te Me ee i he ee Si ad POTS Phane 21 POTS Phone 22 POTS Phone 23 Ordinary SIP Gateway SIP Phone 1 POTS SIP Survivability or KTS Server amp I SIP Phone 2 Gateway SIP Phone 2 SIP Phone 3 PSTN Figure 1 3 SIP Survivability MultiVOIP in system A single MVP210SS MVP410SS or MVP810SS can provide SIP server functionality for as many as 500 other voip gateways However the number of phone lines that thes
39. 110 ST interface ISDN BRI description sssr 384 Start Date Time RADIUS Attributes field eee 190 Start Date Time SMTP logs field O EEE 165 Start Date Time statistics logs field si dedashs cons EE EE E 301 Start Modes DID DPO field147 148 Start Now command Link Management button 312 starter configuration phone IP 34 starter configuration phonebook 40 Startup Tasks Quick Start Instructions 24 Static registration 198 Status SIP Server Endpoint Statistics Parameters iesiri 285 Status H 323 Gatekeepers Statistics Servers field ccc ceeceeeseceeees 318 Status SIP Proxies Statistics Servers field cecceeseceseeeeees 319 Status SPP Registrars Statistics Servers field ceeseceteeeees 320 Status statistics logs field 301 Status field Registered Gateway Details 3 cscs sccsvstcedesvcetes asceesend 316 STUN clients and servers 184 STUN support eee eee 14 Subject email logs field 164 supervisory signaling 0 127 supervisory signaling parameters 126 supervisory signaling types MVP210 SS eee 75 76 MVP 410SS 810SS 71 72 Supplementary Services Alerting Party 180 181 182 Call Hold cece i 174 Call Hold Enable 0 177 Call Name Identification 174 Call Transfer 0 0cccccceeeeeeee 174 Call Transfer Enable
40. 191 Call Status SMTP logs field 166 Call Transfer ccccccceeseesteeeee 174 Call Transfer Enable 0 176 Call Transfer music jingle during hold seu os ETE A 176 Call Transferred To logs statistics field 305 Call Type SMTP logs field 166 Call Waiting 174 Call Progress Details statistics I STEA 0 erence eee eee 297 Call Progress Details statistics fieldsi75 Mies fhe ns 297 Call Waiting call progress field 297 Call Waiting Enable 177 Caler ID ana a a n estes 174 Call Progress Details statistics Hela eee ens n a a 297 Caller ID call progress field 297 Caller ID Supplementary Services feldss arer aa 183 Caller ID enable FXO e ar r a i 135 FXS Loop Start 131 132 Caller ID examples 136 137 138 Caller ID fields FXO ern i 135 Caller ID Type FXO yee nr a eee aie 135 FXS Loop Start ee 131 Caller Name Identification Enable 178 Calling Party Supplementary Services 179 Canadian Class A requirements 376 Canadian Limitations Notice regulatory eee eee eee 377 CD MultiVOIP w eee eeeeeeeeeee 21 Channel call progress field 293 channel capacity eeeeeeeseeeeees 13 channel capacity analog voips 10 channel capacity digital voips 9 channel capacity ISDN BRI voips 11 Channel Number inbound field Index Channel Number RADIUS Attributes field cece 190 Channel Numbe
41. 2 3 none none 204 16 3 digit calls to to extensions 49 74 Santa Fe of company s employees PBX system extensions in Flagstaff 200 240 91208 12 none none 204 16 Outgoing calls 49 73 to Boise area T 3 none none 204 16 3 digit calls to 49 73 Boise employees extensions 700 790 53 QS Phonebook Example MultiVOIP User Guide Phonebook Worksheet Voip Location ID Inbound Phonebook Outbound Phonebook Prefix to Prefix Description Destin Total Prefix to IP Description Remove to Add Incoming Calls Pattern Digits Remove Addr Outgoing Calls Other Details Voip Location ID Inbound Phonebook Outbound Phonebook Prefix to Prefix Description Destin Total Prefix to Prefix IP Description Remove to Add Incoming Calls J Pattern Digits Remove to Add Addr Outgoing Calls Other Details Voip Location ID Inbound Phonebook Outbound Phonebook Prefix to Prefix Description Destin Total Prefix to Prefix IP Description Remove to Add Incoming Calls Pattern Digits Remove to Add Addr Outgoing Calls Other Details 54 QS Phonebook Example MultiVOIP User Guide Enlarged Phonebook Worksheet syjep 3uro3mo uondiioseq Ippyv PPY MWy SBIA ured dl Xl g 0 XIJOIg OL uysoq s e 3uruoou uondiosoq d1 U0 e907 dio S11 9q 19470 AUY 0 XIJ
42. 2 Analog MultiVOIP Server Client Phonebook I MVP210 MVP200 I OKO FXS Unit I l CH1 200 cn PS FR i 1 421 200 2 9 7 201 SSS SSS Se Se ee Client l Site F 7 Lincoln NE im 7 Area Code 402 SS 200 2 9 5 Site B Rochester MN Area Code 507 oes es a eS a I 1 Series 1 Analog MultivOIP Ri N Server Client Phonebook 102 I MVP200 CH2 Unit FXS 100 CHI FXO 717 5000 I I l 200 2 9 8 Hi I I lost Holds phonebook for both Series 1 analog VOIPs 507 717 5662 Site C Suburban Rochester 228 MultiVOIP User Guide T1 PhoneBook Configuration The Series I analog VOIP phone book resides in the Host VOIP unit at Site B It applies to both of the Series I analog VOIP units Each of the Series II analog MultiVOIPs the MVP210 and the MVP410 requires its own inbound and outbound phonebooks The MVP2410 digital MultiVOIP requires its own inbound and outbound phonebooks as well 229 T1 Phonebook Configuration MultiVOIP User Guide These seven phone books are shown below Phone Book for Series Analog VOIP Host Unit Site B VOIP Dir IP Address Channel Comments OR Destination Pattern 102 200 2 9 8 2 Site B FXS channel 101 200 2 9 8 1 Site B FXO channel 421 200 2 9 6 0 Site E FXS channel 201 200 2 9 7 1 Site A FXS channel 1615 200 2 9 9 0 Gives remote voip XXX Note 2 users access to local
43. 297 Call Progress Statistics 291 Call Progress Details statistics field definitions 293 294 295 296 297 298 Call Progress Details statistics screen field Call On Hold eee 297 Call Waiting eee 297 Caller IDe nda aeiiae 297 Call Progress Details statistics screen fields Channel e taes 293 Duration ccccecceesceeeseeeeees 293 Modet is e 293 Voice Coder ceeeecsceceseceeees 293 IP Call Type irises esiis 293 IP Call Direction 0 293 Packets Sentesi 294 Packets Received 05 294 Bytes Sentis 294 Bytes Received cee 294 Packets Lost ccceesseeeseees 294 Outbound Digits Sent 296 Outbound Digits Received 296 Prefix Matched 0c008 296 Server Details 296 DTMF Capability 0 296 Call On Hold cece 297 Call Waiting eee 297 Caller ID riaa 297 Call Status eetees 298 Call Control Status 298 Silence Compression 298 Forward Error Correction 298 Gateway Name from and to 295 IP Address from and to 295 Options from and to 295 Gateway Name from sses 295 IP Address from ccccccesseeeee 295 Options from sssr 295 Gateway Name tO eee 295 IP Address t0 cecceeeseeceeereeeees 295 OPTIONS tOst s 295 391 Index Call Status call progress field 298 Call Status RADIUS Attributes field EE EE A E
44. 999 This field determines how many pings will be generated by the Start Now command Response 500 5000 The duration after which a ping Timeout milliseconds will be considered to have failed Ping Size in Bytes 32 128 bytes This field determines how long or large the ping will be Timer Interval 0 or 30 6000 This field determines how long of between Pings minutes a wait there is between one ping and the next Start Now Initiates pinging command button Clear Erases ping parameters in command Monitor Link field group and button restores default values 312 MultiVOIP User Guide Operation amp Maintenance Link Management screen Field Definitions cont d Field Name Values Description Link Status Parameters These fields summarize the results of pinging IP Address a b c d Target of ping column 0 255 No of Pings as listed Number of pings sent to target Sent endpoint No of Pings as listed Number of pings received by Received target endpoint Round Trip as listed Displays how long it took from Delay in milliseconds time ping was sent to time ping Min Max response was received Avg Last Error as listed Indicates when last data error occurred 313 Operation amp Maintenance MultiVOIP User Guide About Registered Gateway Details The Registered Gateway Details screen presents a real time display of the special o
45. 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 206 MultiVOIP User Guide T1 PhoneBook Configuration remote VoIP sites including non toll calls completed in the PSTN at the remote site The Inbound phonebook for a particular VoIP unit describes the dialing sequences required for a call to originate remotely from any other VOIP sites in the system and to terminate on that particular VOIP Briefly stated the MultiVOIP s Outbound phonebook lists the phone stations it can call its Inbound phonebook describes the dialing sequences that can be used to call that MultiVOIP and how those calls will be directed Of course the phone numbers are not literally listed individually but are instead described by rule Consid
46. Defaults option description MultiVOIP program MENU ieee anar 325 Download Firmware program menu option comman4d 331 332 Download Firmware option description Multi VOIP program MENU earr ar niei 325 Download IFM Firmware program menu option command 337 338 Download IFM Firmware option description Multi VOIP program MENU J anren ES 325 Download User Defaults program menu option command 341 Download User Defaults option description Multi VOIP program MENU Joet e ES 325 downloading firmware machine perspective s es 326 351 downloading IFM firmware 337 395 downloading user defaults 341 downloads vs uploads FTP 351 DTMF extended ssiri whites dente ties 142 Standard neenon ineas 142 DTMF Out of Band and Outbound Digits Sentete iness 167 DTMF Capability call progress field E E E EEE 296 DTMF Capability SMTP logs field E EEEE 166 DTMF Capability statistics logs field on r a T sates 302 DTMF frequency chatt 142 DTMF Gain High Tones field 175 DTMF Gain Low Tones field 7 5 DTMF Gain field eee eee 115 DTMF In Out of Band field 116 DTMF inband ceseseseeeeereeee 116 DTMF out of band eee 116 DTMF Tone FXO disconnect criteria field cece cece 142 DTMF custom tone pairs 160 Duration call progress field 293 Duration DTMF field
47. Descriptions iieii rossi ietaises 254 destination pattern 254 IP Address netsesis 254 Protocol Type 254 Remove Prefix cccccceeee 254 SIP Port Numbet 008 255 SIP UR i aaa n i 255 Total Di gitseniz 254 Transport Protocol SIP 255 Use Proxy SIP oo eeeeeeseeeeee 255 Add Edit Outbound Phonebook fields T1 Accept Any Numbet 212 Add Prefix enei eana 213 Advanced button 0cee 214 Description 213 Destination Pattern 213 IP Address ccccccccceceesssseceeeee 213 Protocol Type 213 Remove Prefix cccccceees 213 SIP Port Numbet 000 214 SIP UR oiiscestes cecteneteshoakedivcs 214 Total Digits seese 213 Transport Protocol SIP 214 Use Proxy SIP eee eeeeeeeseeeeee 214 Add Edit Outbound Phonebook screen A E EE E aces 252 licencia tenet ane EA 211 Address Contact Info SIP Server Predefined Endpoint Parametels cccccccccccceesereeee 199 Advanced button Outbound Phonebook Tiarsa an cuss Sunt coon Hoes 211 Advanced Features field group 119 389 Index AFO W aenneren eirinn 67 Alerting Party Supplementary Services 180 181 182 Allow Incoming Calls Through SIP Proxy Only SIP Call Signaling field ses orire a a pi 151 Allow Undefined Registrations field SIP Server Configuration parameters oes sicvscscoeeseseesevers 196 Allowed Name Type Alerting Party
48. Download Firmware command If it is on the command will not work 2 To invoke the Download Factory Defaults command go to Start Programs MVP x xx Download Firmware Download Firmware 00 4 Q Sk gt Multi OIP 332 MultiVOIP User Guide Operation amp Maintenance 3 If a password has been established the Password Verification screen will appear Password Verification Type in the password and click OK 4 The MultiVOIP ___ Firmware screen appears saying MultiVOIP model number is up Reboot to Download Firmware Mult OIP Firmware a Click OK to download the firmware The Boot LED on the MultiVOIP will light up and remain lit during the file transfer process 333 Operation amp Maintenance MultiVOIP User Guide 5 The program will locate the firmware bin file in the MultiVOIP directory Highlight the correct newest bin file and click Open Look in jmutvoep x Bl c Filename mv Files of type Code Files bin 7 Cancel 7 6 Progress bars will appear at the bottom of the screen during the file transfer fy y 1900000090999 9999999999999 Downloading Configuration Packets Sent 2 Acks received 2 Errors 0 The MultiVOIP s Boot LED will turn off at the end of the transfer 7 The Download Firmware procedure is complete 334 MultiVOIP User Guide Operation amp Maintenance Downloadin
49. E E 134 FXO Interface Parameter fields Current LOSS 2 0 cccececseseeeeeeee 134 Current Loss Detect Timer 134 Detection Range flash hook 135 Flash Hook cccccccccessssseeeees 135 Inter Digit Regeneration Timer 134 Inter Digit Timer dialing 134 Message Waiting Indication 134 No Response Timet 04 134 Regeneration dialing 134 Tone Detection ccceeeeeeeee 134 FXO interface MVP210 SS USES Of cceeceseescececessesseessesseeees 75 FXO interface MVP 410SS 810SS USES Of oc cceeeeeeessecccessesseessesseeeees 71 FXO Parameter fields Caller ID enable 00 0 0 135 Caller ID Type ee 135 FXO Current Detect Timert 140 Tone Detection ccccceees 140 FXO Parameters 133 FXO Supervision answer fields Answer Delay eeeeeeeeeeee 140 Answer Delay Timet 140 Answer Tones ccccceeeeeeeeees 140 Available Tones ccccccee 140 Current Reversal c008 140 Tone Detection eee 140 FXO Supervision disconnect fields Available Tones cccceeeeee 142 Current LOSS sitra iiio 141 Current Loss Timer 0 0 141 Current Reversal c cc0 141 Disconnect Tone Sequence 142 Disconnect ToneS c00eee 142 DTMF Tone 142 Silence Detection Enable 141 Silence Detection Type 141 Silence Timer ccceeeceeeeee
50. E R T 261 Tal eeoa aa 221 Registration Type SIP Server Endpoint Statistics Parameters cccccccccsssseeeees 286 SIP Server Predefined Endpoint Parameters cccccccscessceeeeees 198 Remaining Time SIP Server Endpoint Statistics Parameters 286 remote configuration modem MVP410 SS ooeec 73 MVP810 SS oee 73 Remote Configuration Command Modem SCtUP OL a riiai rares 96 154 remote control configuration web GUI and eee 363 remote phonebook configuration 351 remote voip configuration 89 Remove Prefix inbound field Bl A EE 259 o ea E E cceiiss 219 Remove Prefix outbound field Bil EEA E 254 Tharin A as i EE 213 repair procedures for customers U S amp Canada oseese 371 Reply To Address email logs field E E ded eiien ates 164 Requires Authentication SMTP field E E dnl S 163 Re Registration Interval 407 Index SIP Server Predefined Endpoint Parameters 199 Re Registration Time SIP Server Predefined Endpoint Parametei Sintine 199 Re Registration Time proxy server n a Malte ieee rts 152 Re Registration Time field SIP Server Configuration parameters 00 eee 197 Resolutions MultiVOIP troubleshooting 0 0 0 0 eee 8 Response Timeout Link Management field 312 Retransmission Interval RADIUS screen field eeeesecesseeereees 189 Retrieve Sequence 174 177 REG 2782 kn n basis 109 REE 28335508 eain eteeni 116 REC 3087 ccd dsscssc
51. E1 Example The VOIP system will have the following features 1 Employees in all cities will be able to call each other over the VOIP system using 4 digit extensions 2 Calls to Outer London and Inner London greater Amsterdam and greater Paris will be accessible to all company offices as local calls 3 Vendors in Guildford Lyon and Rotterdam can be contacted as national calls by all company offices Note that the phonebook entries for Series II analog MultiVOIPs MVP 210x 410x 810x used in Euro type telephony settings will be the same in format as entries for the MVP3010 262 MultiVOIP User Guide E1 PhoneBook Configuration Toulouse Marseille 263 E1 Phonebook Configuration MultiVOIP User Guide The Netherlands Country Code 31 Groningen Le guwarden 9 0294 Weesp 026 Arnhem Rotterdam 0118 Middelburg 040 Eindhoven 264 MultiVOIP User Guide E1 PhoneBook Configuration An outline of the equipment setup in these three offices is shown below Wren Clothing Co London Office Country Code 44 Area Code 0208 E1 PBX 5174 200 2 10 3 SR ra ONO 5171 A 979 5170 Wren Clothing Co Paris Office Country Code 33 Area Code 01 PBX E1 Digital RNR VoIP Wren Clothing Co Te CJ Fa Amsterdam Office 29 82 Country Code 31 Te E1 Area City Code 020 74 71 29 81 E 688 4800 265 E1 Phonebook Configuration MultiVOI
52. From column voice or FAX gateway name Indicates whether the event being described was a voice call or a FAX call Displays the name of the voice gateway that originates the call 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 301 Operation amp Maintenance MultiVOIP User Guide Logs Screen Details Field Definitions cont d Field Name Values Description Call Details Voice coder G 723 G 729 The voice coder being used on G 711 etc this call Disconnect Values are Indicates whether the call was Reason Normal and disconnected simply because the Local desired conversation was done disconnection or some other irregular cause occasioned disconnection e g a technical error or failure DTMF Capability inband Indicates whether the DTMF dialing out of band digits are carried Inband or Out of Band The corresponding field Expressions A i values differ for the 3 different voip differ slightly for different protocols Call Signaling For H 323 this field can display Out protocols of Band or Inband For SIP it can H 323 SIP or display either
53. IP Addr OutBound Digits Prefix Matched 204 026 122 105 202 054 039 100 4470 204 026 122 105 202 054 039 100 4470 204 026 122 105 202 054 039 100 4470 202 054 039 100 204 026 122 105 763717 204 026 122 105 202 054 039 100 4470 204 026 122 105 202 054 039 100 4470 168 MultiVOIP User Guide Technical Configuration 15 Set Log Reporting Method The Logs screen lets you choose how the VoIP administrator will receive log reports about the MultiVOIP s performance and the phone call traffic that is passing through it Log reports can be received in one of three ways A in the MultiVOIP program GUD B via email SMTP or C at the MultiVoipManager remote voip system management program SNMP Accessing Logs Traces Screen Pulldown Icon Multi oIP Multi OIP SS v3 08 0H Firmw Configuration Ethernet IP Parameters Ctri Alt I Voice Channels Ctrl H Interface Ctrl alt N Regional Parameters Ctrl R SMTP Parameters Ctri Alt 5 Ctri Alt L Supplementary Services Ctri Alt H System Information Ctrl aAlt y SIP CallSignaling Ctrl Alt Sht P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt Sht Shortcut Sidebar El Configuration Ctrl Alt L Ethernet IP Yoice Fax Interface SIP Call Signaling Regional SMTP RADIUS Logs Traces NAT Traversal If you enable console messages you can customize the types of messages to be included excluded in log reports by clic
54. IP Stats field cece eeeeeeeeeeee 309 Received with Errors Total Packets IP Stats field ceeeeeeeeeeteeeee 309 Received with Errors UDP Packets IP Stats field eee eeeeeeeeeee 309 Recipient Address email logs field E T E 164 recovering voice packets 119 Regeneration dialing FXO field 134 Regional Parameter definitions 155 156 157 158 Regional Parameter fields Cadence vsccccccii cate nieeiiedies 157 Country Region tone schemes 155 Custom tones ccceeeseeeeeee 157 Frequency L i 156 Frequency 2 0 0 eeeeeeeseeseeeees 156 Gain Tecin ariei riaa 156 Gall 2a iniiaiee 156 Pulse Generation Ratio 157 type Of tone 156 Regional Parameters fields Country Selection for Built In Modemin inda aisea 157 regional parameters setting 153 Register Duration field Registered Gateway Details cece 316 Register value Survivability Status Check 195 Registered Gateway Details Statistics screen accessing 316 MultiVOIP User Guide Registered Gateway Details Statistics function 315 316 Registered Gateway Details screen316 Registered Gateway Details screen fields D striptio hns enin s 316 IP Addres S neesti ieies 316 No of Entries seee 316 POPE ASEE EA EA SEEN 316 Register Duration 316 Sre kaa CE EE 316 Registered Gateway Details screen Ee E EE NE 316 Registration Option Parameters Inbound Phone Book Bd ER A
55. Inbound Phone Book p Inbound Phone Book 0044207 Not Used Not Used Not Used Not Used Not Used Not Used Not Used paler Ms od Edit Entry E Statistics i Save Setup E Connection Help Number of Entries 11 p Details ChannelNo 1 Description Calls to Inner London r Registration Options m H323 Register as E 164 Tech Prefix H323 1D SIP Register With SIP Proxy SPP Register With SPP Registrar 257 E1 Phonebook Configuration MultiVOIP User Guide 4 The Add Edit Inbound PhoneBook screen appears r Add Edit Inbound Phone Book J Accept AnyNumber Remove Prefix 0044207 OK Add Prefix 9 7 Cancel Channel Number Hunting Help Description Access to Inner London m Call Forward ME kz r Forward Condition I Unconditional J Busy I No Response Forward Destination 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 Ring Count fo r Registration Options H323 Register as T E164 Tech Prefix m SPP I H3231D J Register With SPP Registrar SIP J Register With SIP Proxy Enter Inbound PhoneBook data for your MultiVOIP unit The fields of the Add Edit Inbound PhoneBook screen are described in the table
56. Installation amp Cabling Unpacking Your MultiVOIP When unpacking your MultiVOIP check to see that all of the items shown are included in the box For the various MultiVOIP models the contents of the box will be different Study the particular illustration below that is appropriate to the model you have purchased If any box contents are missing contact MultiTech Tech Support at 1 800 972 2439 63 Mechanical Installation amp Cabling MultiVOIP User Guide Unpacking the MVP 410SS 810SS MUZ Z ii Cabling MultiVOIP Guide O Multiech Figure 3 1 Unpacking the MVP 410SS 810SS 64 MultiVOIP User Guide Mechanical Installation amp Cabling Unpacking the MVP210 SS MultiVOIP MUR Cabling Guide mat Figure 3 2 Unpacking the MVP210 SS 65 Mechanical Installation amp Cabling MultiVOIP User Guide Rack Mounting Instructions for MVP410 SS amp MVP810 SS The MultiVOIPs can be mounted in an industry standard EIA 19 inch rack enclosure as shown in Figure 3 3 LL ee LU Figure 3 3 Rack Mounting MVP410SS or MVP810SS 66 MultiVOIP User Guide Mechanical Installation amp Cabling Safety Recommendations for Rack Installations Ensure proper installation of the unit in a closed or multi unit enclosure by following the
57. Learning About Access The first part of configuration concerns IP parameters Voice FAX parameters Telephony Interface parameters SNMP parameters Regional parameters SMTP parameters Supplementary Services parameters Logs and System Information In the MultiVOIP software these seven types of parameters are grouped together under Configuration and each has its own dialog box for entering values Generally you can reach the dialog box for these parameter groups in one of four ways pulldown menu toolbar icon keyboard shortcut or sidebar 101 Technical Configuration MultiVOIP User Guide 6 Set Ethernet IP Parameters This dialog box can be reached by pulldown menu toolbar icon keyboard shortcut or sidebar Accessing Ethernet IP Parameters Pulldown Icon Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl Alt N Regional Parameters Ctrl R SMTP Parameters Ctrl Alt S Logs Traces Ctrl Alt L Supplementary Services Ctrl Alt H System Information Ctrl Alt SIP CallSignaling Ctrl Alt SFt P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt SFt y amp Multi oIP Multi OIP Configuration BS Configuration Yoice Fax Interface Shortcut Sidebar Ctrl Alt I 102 E Configuration Voice Fax Interface SIP Call Signaling Regional SMTP RADIUS Logs Traces NAT Traversal Suppleme
58. Multi VOIP xxxx yyyy where x and y represent MultiVOIP model numbers and software version numbers 8B2 In the FTP client program window drag and drop files from the local browser pane to the pane for the MultiVOIP FTP server FTP client GUI 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 SmartFTP 1 0 Build 969 ETP Commands View Tools Favorites Window Help ELI Merika AEE EEM Address 9 192 168 2 200 Login voip Password Port 21 192 168 2 200 Mie x E Local Brows 192 168 2 200 Name a myptt ftp bin fdefftp cnf a fxo_ground cas p 38 fxs_groi a fxs_groundFtp a fxs_loop cas a fxs_loopFtp cas h323 pdl 58 192 168 2 200 359 Operation amp Maintenance MultiVOIP User Guide Some FTP client programs are more graphically oriented see previous screen while others like the WS FTP client are more text oriented WS FTP LE 192 168 2 200 olx m Local System Remote Site Jc Progran Files Multi F 7 C Name Date t casfile cas factdef cnf H323 pdl nvptiftp bin r2_argentina r2_argent inaa r2_argent inaF r2_brazil cas r2_brazilani r2_brazilaniF r2_brazilFtp Refresh r2_china cas xj
59. Only the FTP Server function requires a password for access The FTP Server function also requires that a username be established along with the password Uninstall Upgrade Software Select this to uninstall the MultiVOIP software most but not all components are removed from computer when this command is invoked 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 325 Operation amp Maintenance MultiVOIP User Guide Downloading here refers to transferring program files from the PC to the nonvolatile flash memory of the MultiVOIP Such transfers are made via the PC s serial port This can be understood as a download from the perspective of the MultiVOIP unit When new versions of the MultiVoip software become available they will be posted on MultiTech s web or FTP sites Although transferring updated program files from the MultiTech web FTP site to the user s PC can generally be considered a download from the perspective of the PC this type of download cannot be initiated from the MultiVoip software s Program menu command set Generally updated firmware must be downloaded from the MultiTech web FIP site to the PC before it can be loaded from the PC to the MultiVOIP Configuration Option The Configuration option in the MultiVOIP Progra
60. PROCEDURE 39 QS Phonebook Starter Config MultiVOIP User Guide Phonebook Starter Configuration with remote voip If the topic of voip phone books is new to you it may be helpful to read the PhoneBook Tips section page 47 before starting this procedure To do this part of the quick setup you need to know of another voip that you can call to conduct a test It should be at a remote location typically somewhere outside of your building You must know the phone number and IP address for that site We are assuming here that the MultiVOIP will operate in conjunction with a PBX You must configure both the Outbound Phonebook and the Inbound Phonebook A starter configuration only means that two voip locations will be set up to begin the system and establish voip communication 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 Euro National Call Long Distance Example Example Technician in Seattle area Technician in central 206 must set up one voip London area 0207 to set there another in Chicago up voip there another in area 312 downtown Birmingham area 0121 Answer Write down 312 Answer write down 0121 Euro
61. Phone IP Starter Configuration Phonebook Starter Configuration Connectivity Test Troubleshooting QS Startup Tasks MultiVOIP Startup Tasks Summary The MultiVOIP must be configured to interface with your particular phone system and IP network To do so certain details must be known about those phone and IP systems Decide where you ll mount the voip Some modest minimum specifications must be met A COM port must be set up Connect power phone and data cables per diagram This is the configuration program It s a standard Windows software installation You will enter phone numbers and IP addresses You ll use default parameter values where possible to get the system running quickly Use Config Info CheckList page 28 The phonebook is where you specify how calls will be routed To get the system running quickly you ll make phonebooks for just two voip sites You ll find out if your voip system can carry phone calls between two sites That means you re up and running Detect and remedy any problems that might have prevented connectivity 24 MultiVOIP User Guide QS Gathering Phone IP Details Phone IP Details Absolutely Needed Before Starting the Installation The MultiVOIP will interface with both the IP network and the phone system You must gather information about the IP network and about the phone system so that the MultiVOIP can be configured to operate with them
62. Please have your product information available including model and serial number 373 Regulatory Information MultiVOIP User Guide Chapter 10 Regulatory Information 374 MultiVOIP User Guide 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 equipment designed for use within certain voltage limits and Council Directive 1999 5 EC of 9 March 1999 on radio equipment and telecommunications terminal equipment and the mutual recognition of their conformity FCC Declaration NOTE This equipment has been tested and found to comply with the limits for a Class A digital device pursuant to Part 15 of the FCC Rules These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment This equipment generates uses and can radiate radio frequency energy and if not installed and used in accordance with the instructions may cause harmful interference to radio communications Operation of this equipment in a residential area is likely to cause harmful interference in wh
63. RTP Packets 310 Transmitted TCP Packets 309 Transmitted Total Packets 308 Transmitted UDP Packets 309 IP Statistics function 66 306 ISDN BRI voip product family 11 Java MStal lig sccseteavsecet a eis eels 364 web GUI and eeeeeeeeeeeeeeee 363 jitter buffer rsin iesiri outei 123 Jitter Value Fax field 0 117 Jitter Value field cece 125 MultiVOIP User Guide jitter dynamic eects 123 jumper DID MV P21 0 S Siisera 73 MVP 410SS 810SS uo eee eee 70 Keep Alive Timers NAT STUN 186 key system connecting to analog voip MVP ALOSS 810SS seee 71 connecting to MVP210 SS 75 Knowledge Base online for Multi VOIPS eeen 8 Last button Logs Statistics screen 301 Last Error Link Management field E E E 313 LED definitions BOOT sstesssesveneotiessishseastetetisteseeosns 18 Etherneti uinna 18 PDX CERET T 18 ENK eira eaaa 18 POWT eae sorai reia aT R A 18 RCV channel 0 0 18 RSG is aa arer 18 XMT channel eee 18 KG sep ra aa aon 18 LED indicators channel operation 06 17 general operation 0ee 17 LED indicators active 0 0 0 0 17 LED type Seni niea e a 17 lifting precaution about eee 62 limitations notice regulatory Canadians sccccacindvecduain ies 377 limited warranty 371 Link Management Statistics fields Clear command button 312 IP Address column
64. Re transmis interval it will retransmit that packet sions again and wait the retransmit interval again for a response How many times it does this is determined by the setting in the Number of Retransmissions field Shared alpha Client encryption key for the current Secret numeric voip unit Select Gives access to RADIUS Attributes Attributes screen On Attributes screen one can button specify the parameters to be tallied by the RADIUS server 189 Technical Configuration MultiVOIP User Guide The RADIUS Parameters dialog box has a secondary dialog box Custom Fields that allows you to customize accounting information sent to the RADIUS server by the MultiVOIP The MultiVOIP software logs data about many aspects of the call traffic going through the MultiVOIP The Custom Fields screen lets you pick which aspects will be included in the accounting reports sent to the RADIUS server Custom Fields Definitions Field Description Field Description Select All Log report to include all fields shown Channel Data channel Start Date and time the Number carrying call Date phone call began Time Duration Length of call Call Voice or fax Mode Packets Total packets sent Packets Total packets Sent in call Received received in call 190 MultiVOIP User Guide Technical Configuration Custom Fields Definitions cont d Field Descri
65. SHa Block as shipped for non DID interfaces JP8 Ch 2 Jumper sas configured for DID Interface Figure 3 10 MVP210 SS Channel Jumper Settings e Position the jumper for each DID channel so that it does not connect the two jumper posts For DID operation of a voip channel the MultiVOIP will work properly if you simply remove the jumper altogether but that is inadviseable because the jumper might be needed later if a different telephony interface is used for that voip channel f Slide the main circuit card back into the MultiVOIP chassis and replace the screw at the bottom of the unit 74 MultiVOIP User Guide Mechanical Installation amp Cabling 2 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 Figure 3 11 E A Command Port Connection at Ehemet Connection Figure 3 11 Cabling for MVP210 SS 3 Connect the MultiVOIP to a PC by using a RJ 45 male to DB 9 female cable Plug the RJ 45 end of the cable into the Command port of the MultiVOIP and the other end into the PC serial port See Figure 3 11 4 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 5 For an FXS or FXO connection FXS Examples analog phone fax machine Key Telephone System FXO Examples PBX extensio
66. Services are still applicable to the SIP and SPP voip protocols in which cases these features are implemented differently 173 Technical Configuration MultiVOIP User Guide In each field enter the values that fit your particular network m Supplementary Services Parameters Select Channel Channel 1 bs Call Transfer rm Call Name Identification M Enable M Enable Allowed Name Type Transfer Sequence fs 1 I Calling Party T Busy Party Call Hold I Alerting Party I Connected Party MV Enable Caller Id Hold Sequence Call Waiting MV Enable OK Default Retrieve Sequence ical ean Channel 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 invoked 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 managem
67. Statistics MultiVOIP User Guide Accessing IP Statistics Pulldown Icon Statistics Call Progress Logs Registered Gateway Details Link Management Alternate Servers Connection Help Ctrl Alt 4 Ctrl O trl Alt W Ctrl 2 Ctrl alt 4 gt si IP Statistics Details Shortcut Ctrl P Confi Adva E S Save Conn Help 306 Phone Book Statistics Call Progress Logs IP Statistics Link Management Registered Gateway Sidebar guration nced ervers Setup ection MultiVOIP User Guide IP Statistics Screen IP Statistics IP Address Operation amp Maintenance r Total Packets Transmitted Received UDP Packets Transmitted Received Received with Errors m TCP Packets Transmitted Retransmitted Received Received with Errors THI M RTP Packets Transmitted Received Received with Errors m RTCP Packets Transmitted Received Received with Errors Jd IP Statistics Field Definitions Field Values Name Description 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 307 Operation amp Maintenance MultiVOIP User Guide I
68. TEL 763 785 3500 FAX 763 785 3702 Tech Support 800 972 2439 fip site fitech com 22 ee 2 Applet dirstrut started g Intemet You can control the MultiVOIP unit with a graphic user interface GUI based on the common web browser platform Qualifying browsers are InternetExplorer6 Netscape6 and Mozilla FireFox 1 0 MultiVOIP Web Browser GUI Overview Function Remote configuration and control of MultiVOIP units Configuration Local Windows GUI must be used Prerequisite to assign IP address to MultiVOIP Browser Version Internet Explorer 6 0 or higher or Requirement 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 Video Usability large video monitor recommended 361 Operation amp Maintenance MultiVOIP User Guide The initial configuration step of assigning the voip unit an IP address must still be done locally using the Windows GUI However all additional configuration can be done via the web GUI The content and organization of the web GUI is directly parallel to the Windows GUI For each screen in the Windows GUI there is a corresponding screen in the web GUI The fields on each screen are the same as well A Multi oIP Multi OIP 410 v6 08 C Firmware Aug 04 2005 Microsoft Internet Explorer Fie Edit View Favorites Tools Help
69. 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 CID CID at 4 51pm Oct 3 Terminating Generating Jom FXS gt p VoIP Voip 401 IP Ch2f702 Network phone of Nigel Thurston Clock ch3fzo3 g 10 03 4 51pn N 763 555 9401 i 404 Displayishions H 323 Protocol CID Number 423 E FTT CID Name Anoka Whse VP3 E sell kal Anoka Whse VP3 Time Stamp Date 10 03 Gateway Name MANGANA Time 4 51pm m Q 931 Parameters Inbound Phone Book Channel 2 Add Prefix __Forward Addr In x 06 release when SIP protocol is used CID Name field will duplicate value in CID Number field Remove Prefix 423 748 Gatekeeper RAS Para Figure 5 4 Voip Caller ID Case 4 Remote FXS call on H 323 voip system 137 Technical Configuration Call is received lt CID Flow MultiVOIP User Guide Call originates here ere CID CID a4 at 6 17pm Nov 15 Terminating EN Generating engh Central Office O FXS VolP b VoIP DID without N Network es standard telephony a Clock Ch3 A Caller ID service 11 15 6 17pm chal eA fe Display shows H323P fad N P Rag 323 Protocol Edwin Smith CID Number 423 CID Name Time Stamp Date 11 15 Time 6 17pm In x 06 release
70. b C ASCII Binary T Auto 150 Here it comes Received 52 bytes in 1 0 secs 528 99 bps transfer succeeded 226 Transfer OK Closing connection Close Cancel Logw nd Help Options About 9 Verify Transfer The files transferred will appear in the directory of the MultiVOIP P ftp voip1 192 168 2 200 Microsoft Internet Explorer voip 192 168 2 200 A al casfile cas factdef cnt H323 pdl myptlftp bin OutPhBk tmr InPhBk tmr 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 GUI 360 MultiVOIP User Guide Operation amp Maintenance Web Browser Interface MultiVOIP 410 v6 06 Firmware Aug 26 2006 Microsoft Internet Explorer Eile Edit View Favorites Tools Help e gt 0 A Ala a 3l 3 M E I Back Forward Stop Refresh Home Search Favorites History Mail Print Edit Discuss Address hitp 204 26 122 105 Go Links Best ofthe Web 4 Channel Guide 4 Customize Links 4 Free HotMail Intemet Stat Microsoft 4 Windows Update a MultiVOIP 410 Configuration Advanced Phone Book Statistics Change Password Save amp Reboot Logout MultiVOIP 410 Web Based Configuration Help gt Voice Fax over IP Networks s Multi Tech Systems Inc 2205 Woodale Drive Mounds View MN 55112 USA
71. button Click to access secondary screen on where console messages can be included excluded by category and on a per channel basis See the Console Messages Filter Settings screen on subsequent page Turn Off Logs Y N Check to disable log reporting function Logs Buttons Only one of these two log reporting methods GUI or SMTP may be chosen GUI Y N User must view logs at the MultiVOIP configuration program SNMP Y N Log messages will be delivered to the MultiVoipManager application program SMTP Y N Log messages will be sent to user specified email address SysLog Server Y N This box must be checked if logging is to be Enable done in conjunction with a SysLog Server program For more on SysLog Server see Operation amp Maintenance chapter IP Address n n n n IP address of computer connected to voip for n network on which SysLog Server program is 0 255 running Port 514 Logical port for SysLog Server 514 is commonly used Online Statistics integer Set the interval in seconds at which Updation Interval logging information will be updated Technical Configuration MultiVOIP User Guide To customize console messages by category and or by channel click on Filters and use the Console Messages Filters Settings screen ee 1 Console Message Settings M Enable Console Messages Filters Cancel Console Messages Filter Settings Trace Off for Functions Trace On for Functions Functions
72. by local Answer 0207 is prefix to be Seattle voip 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 44 MultiVOIP User Guide QS Phonebook Starter Config 4 In the Add Prefix field enter any digits that must be dialed from your local voip to gain access to the PSTN North America Euro National Call Long Distance Example Example Seattle Chicago system London Birming system On Seattle PBX 9 is used to On London PBX 9 is used get an outside line to get an outside line Answer 9 is prefix to be Answer 9 is prefix to be added by local added by local Seattle voip London voip 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 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 45 QS Phonebook Starter Config MultiVOIP User Guide 6 In the Descriptio
73. can be seen at the bottom of the screen while files are being copied 339 Operation amp Maintenance MultiVOIP User Guide 8 Then a completion screen entitled IFM Test will appear Mest x IFM Download Complete Number of IFMs 4 Packet Size 44 Ch 0 Code 0 Data 1 Ch 1 Code 0 Data 1 Ch 2 Code 0 Data 1 Ch 3 Code 0 Data 1 Click OK 9 The MultiVOIP will reboot itself When the reboot is complete the MultiVOIP Configuration screen will close 10 The IFM firmware downloading process is complete 340 MultiVOIP User Guide Operation amp Maintenance Setting and Downloading User Defaults The Download User Defaults command allows you to maintain a known working configuration that is specific to your VOIP system You can then experiment with alterations or improvements to the configurations confident that a working configuration can be restored if necessary 1 Before you can invoke the Download User Defaults command you must first save a set of configuration parameters by using the Save Setup command in the sidebar menu of the MultiVOIP software 2 Before the setup configuration is saved you will be prompted to save the setup as the User Default Configuration Select the checkbox and click OK PNE E EE ENEN EE E SesuseSeseosereseeussesuq MultiVOIP will be brought down OK Cancel Help A user default file will be created The MultiVOIP unit will reboot itself 341 Opera
74. connectivity test 56 PC settings specs nesese 30 phone IP details gathering 25 phone IP starter configuration 34 phonebook example 51 phonebook starter configuration 40 phonebook tips ssssseesseeeseeeeee 47 placement of voip eee 30 quick hookup diagram 210 32 quick hookup diagram 410 810 31 software installation 0 0 33 Startup tasks 0 eeeceeeeceeeeeeeee 24 troubleshooting eee eeeeeees 60 rack mounting QLOUNMING 0 eee eeeeseeeeeeeeeeeeeees 67 Safety oen irpreni ase 62 67 rack mounting instructions 66 rack mounting procedure 68 rack equipment weight capacity Of eee 67 rack mountable voip models 62 RADIUS accounting parameters CUStOMIZING eeeeeeeeeees 190 192 RADIUS accounting suppott 14 RADIUS screen field Enable Accounting 189 Retransmission Interval 189 RADIUS screen fields Accounting Pott 189 Server Address eee 189 RCV channel LED we 18 Received RTCP Packets IP Stats field ee ee 310 406 MultiVOIP User Guide Received RTP Packets IP Stats field Received Call Count SIP Server Endpoint Statistics Parameters cccsssceeeeeees 286 Received with Errors RTCP Packets IP Stats field cee eeeeeeeeee 310 Received with Errors RTP Packets IP Stats field eee eeeeeeeeeee 310 Received with Errors TCP Packets
75. csei bisa ccn dette dubs s aran ererat se eae e naor anen iranse e aS 25 Gather Telephone Information s ssesessssseststssstsssertststsrststststererertnrsrstsesersrsrsrersteerenrs 26 Obtain Email Address for VOIP for email call log reporting eeeeeeeeeeserereeeereeeeee 27 Config Info Ch cklist onshore e EE es acess E E e A E E E Identify Remote VOIP Site to Call eee Identify MVP SS Unit s Role in SIP VOIP System Placement enna e sass as EE R vies eu aga eng cae vane vas ca EO TEE EAS OREK ORES T Command Control Computer Setup Specs amp Settings aossesesessreserseeessee 30 Quick Hookup for MVP410 SS amp MVP810 SS cccccscsssssscesecesecnecesecesecnseenes 31 Quick Hookup for MVP210 SS w ccccccecccsccesscesecesecnsecnseeseeeaeesseesseeseeesecnsecnasenaeeaes 32 Load MultiVOIP Control Software Onto PC ssosssesesseseeseeeeeereeeesrressrese 33 Phone IP Starter Confi guration ccccccccccccsccescesecesecesecssecssecseeeseeeeeeseesseeseeteees 34 Phonebook Starter Configuration With remote VOIp cccccscseseeeseeeneeeeeetees 40 Outbound Phonebook eenen Hesiod deeded ETENE EE S PETER EERTSE SS 40 Inbound Phonebook senceres tin isani EE EEEE AEE EE 44 Ph nebook Tipsi re eaor eee e eE E EEE a S S 47 Ph n book Example ir renee en ereta apane E e E e AEE aeae Seia caste 5I C nnectivily T OSE oer merran aee o E E Ee EA EEEE EREE ETE EEEa E asta 56 Troubleshootin t eaea e ae A E E AE EE EREE Oea 60 CHAPTER 3 MECHANIC
76. eee 72 telephony interfaces MVP210 76 telephony interfaces MVP 410SS 810SS oo eee eeeeeeeeeee 71 Silence Compression call progress field oeiee eriei eessettileeses 298 Silence Compression RADIUS Attributes oo cccceeeseeseeesteeeeees 192 Silence Compression SMTP logs 167 Silence Compression field 119 Silence Detection Enable FXO disconnect criteria field 141 Silence Detection Type FXO field E ARER 141 Silence Timer FXO field 141 SIP Call Signaling Parameter definitions 150 151 152 SIP Call Signaling screen fields Password proxy server 152 Proxy Domain Name IP Address A AE Seah aca 151 Proxy Polling Interval 152 Re Registration Time proxy SCLVEL moeien i ies 152 Signaling Number proxy server E E E T 51 TTL Value sinnti neie aeiee 152 Use SIP Proxy soesereseeeeeereresesee 150 User Name proxy server 151 SIP compatibility with H 450 Supplementary Services 173 SIP Fields Outbound Phonebook l AE E E E 255 TIa nnne eiA tais 214 SIP incompatibility with H 450 Supplementary Services 14 SIP Port Number field MultiVOIP User Guide BY eos iirc E 255 A M AA E lee betes 214 SIP port number standard j A K EE S 255 DT EE E E 214 SIP Proxies Statistics Servers IPAddress airne 319 POLE oerein eiea 319 SLALUS S OEE EEA 319 TY Pe e E week ER 319 SIP proxy Capacity eee 13 SIP P
77. interface type are shown in the figure below and described in the table that follows Interface Interface Type _ Dialing Options Regeneration Pulse Inter Digit Timer 2 secs Message Waiting Indication Light he Inter Digit Regeneration Timer fi oo ms Caller ID Fx0 Options Type FXO Ring Count ni BellCore MIERES No Response Timer fi 80 secs DTMF Flash Hook Options Generation 600 Detection Hange Min 500 ms Maw ooo ms 133 Technical Configuration MultiVOIP User Guide FXO Interface Parameter Definitions Field Name Values Description Interface Type FXO Enables FXO functionality Dialing Options Regeneration Pulse DIMF Determines whether digits generated and sent out will be pulse tones or DTMF Inter Digit 1 to 10 seconds This is the length of time that Timer the MultiVOIP will wait between digits When the time expires the MultiVOIP will look in the phonebook for the number entered Default 2 Message Not applicable to FXO interface Waiting Indication Inter Digit 50 to 20 000 The length of time between the Regeneration milliseconds outputting of DTMF digits Time Default 100 ms FXO Options FXO Ring 1 99 Numter of rings required Count before the Multi VOIP answers the incoming call No Response 1 65535 Length of time before call Timer in seconds connect
78. is intended for private networks not accessible via Internet or PSTN Domain Names this com List entries separated by semicolon that org of domains of endpoints from which etc the MVP SS will accept registrations Accept any IP Determines whether registrations to Registrations addresses the MVP SS SIP server will be for specific IP accepted from any IP address or only addresses from specified IP addresses Multiple IP addresses can be listed separated by semicolons The any IP addresses option is intended for private networks not accessible via Internet or PSTN IP Addresses a b c d List entries separated by semicolon q r s t of IP addresses of endpoints from for which the MVP SS will accept values registrations 0 255 196 MultiVOIP User Guide Technical Configuration SIP Server Configuration Parameter Definitions Field Name Values Description Registrar Options Re integer Registration values in Time seconds default is 3600 The time after which the MultiVOIP UserAgent Client is supposed to register with the proxy server Expiration of the registration interval means that the gateway has lost contact with the main SIP server and that the MVP SS unit will enter its survivability 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 ini
79. nta 183 Default Voice FAX field 115 default baud rate Multi VOIP software connection 193 default configuration user 204 default values software 335 delay packets eee eee 123 delay versus voice quality 124 Delete File button Logs Statistics screen 301 Description callee location Bl E E 260 o ea E EE E 219 Description callee outbound phonebook Bl EEE EE E 254 Teisi e taa evi E ak 213 394 MultiVOIP User Guide Description field Registered Gateway Details jasoena 316 Description From Details RADIUS Attributes field 192 Description From Details SMTP logs feldir iai et 167 Description To Details RADIUS Attributes field 192 Description To Details SMTP logs field sa ateen ri 167 Destination Pattern outbound field Bil E EE 254 Til EEE E E 213 destination patterns discussion A PEE E E EN 247 A A E EE ET 206 Detection Range Flash Hook Options field E amp M ee 146 rO O RE A E TAS 135 FXS Loop Start ee 131 dial tone custom cc ce eeeeeeeee 160 Dialing Options E amp M fields 145 Dialing Options FXO fields 134 Gial tOneS c ccc cececsesssessesseeeeeeeees 159 DID interface MVP210 SS USES OF E E T ae oe 76 DID interface MVP 410SS 810SS USES Of ae a a aT 72 DID Interface Parameter definitions E E E ESE TE 148 DID Interface Parameter fields Message Waiting Indicat
80. number in the 651 and 952 area codes because number in both of these area codes are local calls in the Minneapolis St Paul area The simplest case is a cal 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 225 T1 Phonebook Configuration MultiVOIP User Guide Similarly the Inbound PhoneBook for the Baltimore VOIP shown first below generally matches the Outbound PhoneBook of the Minneapolis VOIP shown second below Notice the extended prefix to be removed 14103257 This entry allows Minneapolis users to contact Baltimore co workers as though they were in the Minneapolis facility using numbers in the range 7000 to 7999 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 226 MultiVOIP User Guide T1 PhoneBook Configuration The Outbound PhoneBook for the Minneap
81. of these steps some are optional 1 Check Power and Cabling 2 Start MultiVOIP Configuration Program 3 Confirm Connection 4 Solve Common Connection Problems A Fixing a COM Port Problem B Fixing a Cabling Problem 5 Familiarize yourself with configuration parameter screens and how to access them 6 Set Ethernet IP Parameters 7 Set up web browser GUI optional 8 Set Voice Fax Parameters 9 Set Telephony Interface Parameters 10 Set SIP Call Signaling parameters 12 Set Regional Parameters Phone Signaling Tones amp Cadences and setup for built in Remote Configuration Command Modem 13 Set Custom Tones and Cadences optional 14 Set SMTP Parameters applicable if Log Reports are via Email 15 Set Log Reporting Method GUI locally in MultiVOIP Configuration program or SMTP via email 16 Set Supplementary Services Parameters The Supplementary Services screen allows voip deployment of features that are normally found in PBX or PSTN systems e g call transfer and call waiting 17 Set NAT Traversal STUN parameters Optional Applicable only under SIP Call Signaling when the UDP transport protocol is used 18 Set RADIUS parameters Optional Used only if system interfaces with RADIUS server for billing or other accounting functions 19 Set Baud Rate of COM port connection to Command PC 96 MultiVOIP User Guide Technical Configuration 20 Set SIP Server Configuration parameter
82. or premises wiring using a compatible modular jack that is Part 68 compliant See installation instructions for details 4 If this equipment causes harm to the telephone network the telephone company will notify you in advance that temporary discontinuance of service may be required If advance notice is not practical the telephone company will notify the customer as soon as possible 5 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 6 If trouble is experienced with this equipment the model of which is indicated below please contact Multi Tech Systems Inc at the address shown below for details of how to have repairs made If the equipment is causing harm to the network the telephone company 376 MultiVOIP User Guide Regulatory Information may request you to remove the equipment form t network until the problem is resolved 7 No repairs are to be made by you Repairs are to be made only by Multi Tech Systems or its licensees Unauthorized repairs void registration and warranty 8 Manufacturer Multi Tech Systems Inc Trade name MultiVOIP Model number MVP 810 410 210 FCC registration number US AU7DDNAN46050 Modular jack USOC RJ 48C Service center in USA Multi Tech Syste
83. proxy proxy redundancy to server redundancy to WAN giving SIP WAN proxy redundancy to WAN Capacity 8 channels 4 channels 2 channels Chassis 19 1U 19 1U table top Mounting rack rack unit mount mount 10 MultiVOIP User Guide Overview ISDN BRI MultiVOIP Products hee MVP810ST MVP410ST Function ISDN BRI voip ISDN BRI voip Capacity 4 ISDN lines 2 ISDN lines 8 B channels 4 B channels Chassis 19 1U rack mount 19 1U rack mount Mounting 1 BRI means Basic Rate Interface 11 Overview MultiVOIP User Guide Introduction to Analog MultiVOIPs with SIP Survivability Features MVP 210SS 410SS 810SS VOIP The Free Ride We proudly present Multi Tech s MVP 210SS 410SS 810SS MultiVOIP Voice over IP Gateways These three models allow voice fax communication to be transmitted at no additional expense over your existing IP network which has ordinarily been data only To access this free voice and fax communication you simply connect the MultiVOIP to your telephone equipment and your existing Internet connection These analog MultiVOIPs inter operate readily with T1 or E1 MultiVOIP units Figure 1 2 MVP210SS Chassis 12 MultiVOIP User Guide Overview Capacity MultiVOIP model MVP810SS is an eight channel unit the model MVP410SS is a four channel unit and the MVP210SS is a two channel unit All three of these MultiVOIP units have a
84. remote voip unit 297 Operation amp Maintenance MultiVOIP User Guide Call Progress Details Field Definitions cont d Field Name Values Description Call Status fields Call Status hangup active Shows condition of current call Call Control Tun FS Tun Displays the H 323 version 4 Status AE Mux features in use for the selected call These include tunneling Tun Fast Start with tunneling FS Tun Annex E multiplexed UDP call signaling transport AE and Q 931 Multiplexing Mux See Phonebook Configuration Parameters in T1 or E1 chapters for more on H 323v4 features Silence SC SC stands for Silence Compression 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 Correction Forward Error Correction enables some of the voice packets that were corrupted or lost to be recovered FEC adds an additional 50 overhead to the total network bandwidth consumed by the voice channel Default Off 298 MultiVOIP User Guide Operation amp Maintenance About Logs The Logs Accessing Statistics Logs Pulldown Icon Statistics Call Progress Ctri Alt A aN IP Statistics Ctrl P Registered Gateway Details Ctri Alt W F Link Manag
85. remote VOIP locations via FXO port at Site F 1402 740 0 Gives remote voip users access 263740 to extension of key phone 1402 741 0 system at Site F Lincoln 263741 Because call is completed at key 1402 742 0 system abbreviated dialing 4 263742 digits is not workable Human operator or auto attendant is needed to complete these calls 235 T1 Phonebook Configuration MultiVOIP User Guide Outbound Phone Book for MVP210 Analog VOIP Site E Destin Remove Add IP Comment Pattern Prefix Prefix Address 201 200 2 9 7 To originate calls to Site A 1507 1507 101 200 2 9 8 To originate calls Note 3 to any PSTN phone in Rochester area using the FXO channel channel 1 of the Site B VOIP 102 200 2 9 8 To originate calls to phone connected to FXS port channel 2 of the Site B VOIP 1402 200 2 9 5 Calls to Lincoln area PSTN via FXO channel CH4 of the Site F VOIP 7 1402 200 2 9 5 Calls to Lincoln 263 key extensions with four digits 1615 200 2 9 9 Calls to Pierre area PSTN via Site D PBX 31 1615 200 2 9 9 Calls to Pierre PBX 492 extensions with four digits Note 3 The pound sign is a delimiter separating the VOIP number from the standard telephony phone number 236 MultiVOIP User Guide T1 PhoneBook Configuration Inbound Phonebook for MVP210 Analog VOIP Site E Remove Add Channel Comment P
86. should be configured to maximize savings in long distance calling charges To achieve both of these objectives ease of use and maximized savings the VOIP phonebooks must be set correctly NOTE VOIPs are commonly used for another reason as well VOIPs allow an organization to integrate phone and data traffic onto a single network Typically these are private networks 242 MultiVOIP User Guide E1 PhoneBook Configuration Free Calls One VOIP Site to Another The most direct use of the VOIP system is making calls between the offices where the VOIPs are located Consider for example the Wren Clothing Company This company has VOIP equipped offices in London Paris and Amsterdam each served by its own PBX 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 London Wren Clothing Co VOIP PBX Site Amsterdam The Netherlands Wren Clothing Co VOIP PBX Site Paris Free VOIP Calls 243 E1 Phonebook Configuration MultiVOIP User Guide Local Rate Calls Within Local Calling Area of Remote VOIP In the second use of the VOIP system the local calling area of each VOIP location becomes accessible to all of the VOIP
87. sidebar menu Phonebook Icons Description Phone Book leone Phonebook Configuration D ee ee amp B Phone Book Icons a A a py ke amp B Anne Danian a Phonebook E Bi S oSI Edit selected Inbound a Phonebook Entry CIETFEN Outbound Phonebook Entries List Boe A 2 e amp Ee Phone Book Icons A E m B28 2B Bi E Bute Te oi Edit selected Outbound B p B a i Phonebook Entry 249 E1 Phonebook Configuration MultiVOIP User Guide Phonebook Pulldown Menu Phone Book Outbound Phone Book Alt O gt List Entries Ctrl L Inbound Phone Boo Ls Alt gt AddeEntry cCtri 4 Edit Entry Ctrl E Inbound Phonebook Shortcut Outbound Phonebook Shortcut Alt I Alt O Phonebook Sidebar Menu Configuration Advanced Phone Book Outbound Phone Book List Entries Add Entry Edit Entry Inbound Phone Book List Entries Add Entry Edit Entry Statistics 250 MultiVOIP User Guide E1 PhoneBook Configuration Phonebook Configuration Procedure 1 Select Outbound Phone Book List Entries MuliVorP MultiVOLP vb 0U CP Firmware Jul 22 2005 Configuration Advanced Phone Bock Statistics Download Connection inip ANBZOSCHEtS EN 4D Click Add 251 E1 Phonebook Configuration MultiVOIP User Guide 2 The Add Edit Outbound PhoneBook screen appears Multi OIP Enter Outbo
88. 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 which includes both Inner London and Outer London Generally local calling rates apply only within a single area code and for all calls outside that area code national rates apply There are however some European cases where local calling rates extend beyond a single area code Local rates between Inner and Outer London are one example of this It is also possible in some locations that calls within an area code may be national calls But this is rare United Kingdom Wren Clothing Co VOIP PBX Site Bluebird Zipper Co Wren Clothing Co VOIP PBX Site Amsterdam The Netherlands Wren Clothing Co 5 VOIP PBX Site Paris idani Calls at London local rates Local Calling Area France 244 MultiVOIP User Guide E1 PhoneBook Configuration Similarly the VOIP system allows Wren Clothing employees in London and Amsterdam to call anywhere in Paris at local rates it allows Wren Clothing employees in Paris an
89. 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 A sample presentation of SysLog info in the Kiwi daemon is shown below SysLog programs will vary in features and presentation Kiwi Syslog Daemon Registered version 7 0 2 Display 00 Default Time 17 02 08 17 02 07 17 02 06 17 02 06 17 02 04 17 02 03 17 02 02 17 02 01 17 02 00 17 01 59 17 01 58 17 01 57 17 01 56 17 01 55 17 01 54 17 01 53 17 01 52 17 01 51 17 N1 5n Priority Syslog Warning Local0 Debug Local5 Alert System4 Debug Local3 Info Lpr Critical System4 Notice System1 Critical User Warning System2 Info Local6 Critical Local4 Emerg UUCP Debug Local4 Info User Error Local3 Notice Kernel Info News Info Suctem Critical Hostname 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 127 0 0 1 1 77NN1 369 Message This is Syslog test message number 0020 This is Syslog test message number 0019 This is Syslog test message number 0018 This is Syslog test message number 0017 This is Syslog test message number 0016 This is Syslog test message number 0015 This is Syslog test message number 0014 This is Syslog test message numbe
90. the bottom of the screen during the data transfer by CAAA AAARAAEARAAAARAAEAE Downloading Configuration Packets Sent 2 Acks received 2 Errors 0 LLL The MultiVOIP s Boot LED will turn off at the end of the transfer 7 The Download Factory Defaults procedure is complete Downloading IFM Firmware The Interface Module IFM is the telephony interface for analog MultiVOIP units MVP130 MVP130FXS MVP210 MVP410 MVP810 There is one IFM for each channel of the MultiVOIP unit For each channel the IFM handles the analog signals to and from the attached telephone PBX or CO line The IFM communicates with the main processor indicating the status of the telephone line For example it 337 Operation amp Maintenance MultiVOIP User Guide 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 MultiVOI
91. 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 177 Technical Configuration MultiVOIP User Guide Supplementary Services Definitions cont d Field Name Values Description Call Name Enables CNI function Call Name Identification Identification is not the same as Caller Enable ID When enabled on a given voip unit currently being controlled by the MultiVOIP GUI the home voip Call Name Identification sends an identifier and status information to the administrator of the remote voip 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 callee 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 ot
92. 3 MVP410 SS cabling procedure ce 69 remote configuration modem 73 unpacking seii rrisin 64 MVP810 SS cabling procedure eee 69 remote configuration modem 73 Unpacking 64 Name IP Server field 0 186 NAT inter operation support 14 NAT Traversal screen fields Eablet 186 Keep Alive Timers 5 186 Name IP Server 0 c008 186 l a a so E E SEE S 186 Port Servet sipnrccie iarere 186 national rate calls to foreign voip sites A I O EEEE EES 246 Netcoder coders RTP packetization v ice fak jarein eines 322 Network Disconnection field 125 No Response Timer E amp M field 144 No of Entries SIP Server Endpoint Statistics Parameters 286 No of Entries field Registered Gateway Details eee 316 403 Index No of Pings Received Link Management field 313 No of Pings Sent Link Management field eke eree piesne 313 no response amp busy forwarding dual conditions Bl A E E E ES 260 Ta EE E EEE 220 Number of Days email log criteria P E cashes E odiedeceepys 164 Number of Records email log etena enres 164 Numter of Retransmissions RADIUS screen field ceeesceeseeeneees 189 numbering plan resources 281 obtaining updated firmware 327 Offhook alert eee eeeeeeeeeeeees 120 Offhook Alert Voice Fax Params and Intercept Tone Regional Params eneee 120 Offhook Alert Timer Voice FA
93. 3 protocol Proxy for SIP protocol Registrar for SPP protocol When no external routing device is used If Any Number is selected calls received from phone numbers not matching a listed Prefix shown in the Remove Prefix column of the Inbound Phone Book will be admitted into the voip on the channel listed in the Channel Number field Any Number can be used in addition to one or more Prefixes 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 1 24 or T1 channel number to which Number Hunting the call will be assigned as it enters the local telephony 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 219 T1 Phonebook Configuration MultiVOIP User Guide Add Edit Inbound Phone Book Field Definitions cont d Field Name Values Description Call Forward Parameters Forward Uncondit Unconditional When selected Condition Busy all calls received will be No Resp forwarded Busy When selected calls will be forwarded when station is busy No R
94. 3 GateKeepers LIP Address _ Po GKName_ Type Priority Status 65 126 90 143 1719 MYP_LSGK Primary 0 65 126 90 92 1719 MY PGK1 Operation amp Maintenance MultiVOIP User Guide Registered Predef 0 Not Registere H 323 Gatekeepers Statistics Servers Field Definitions Field Values Description Name Column Headings IP Address n n n n The IP address of the gatekeeper for n 0 255 Port TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it GK Name alpha numeric Identifier for gatekeeper string Type Primary This field describes the type of gateway as Predefined which the MultiVOIP is defined with respect to the gatekeeper Priority Priority refers to Status registered not The current status of the gateway either registered registered or unregistered 318 MultiVOIP User Guide Operation amp Maintenance SIP Proxies IP Address Pot Type 65 126 90 112 5060 Primary Registered 65 126 90 110 5060 Alternate 1 Not Registered SIP Proxies Statistics Servers Field Definitions Field Values Description Name Column Headings IP Address n n n n The IP address of the SIP proxy by which for n 0 255 the MultiVOIP is governed Port TDMA time slot used for communication between MultiVOIP unit and the SIP Proxy that governs it Type Primary This field des
95. 300 milliseconds A higher value means voice transmission will be more accepting of jitter A lower value is less tolerant of jitter Inactive by default When active default 300 ms However value must equal or exceed Dynamic Minimum Jitter Value Call Duration 1 65535 seconds 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 most configurations requiring upward adjustment Consecutive Packets Lost Network Discon nection 1 65535 1 to 65535 seconds Default 30 sec Consecutive Packets Lost defines the number of consecutive packets that are lost after which the call is automatically disconnected Inactive by default When active default 30 Specifies how long to wait before disconnecting the call when IP network connectivity with the remote site has been lost 125 Technical Configuration MultiVOIP User Guide 9 Set Telephony Interface Parameters This dialog box can be reached by pulldown menu toolbar icon keyboard shortcut or sidebar Accessing Telephony Interface Parameters Pulldown Icon Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I voice Channels Ctrl H a Regional Parameters Ctrl R n SMTP Parameters Ctrl Alt S Logs Traces Ctr
96. 386 MIN DDEGX GAIE OA E AN E EE EA TE 388 MultiVOIP User Guide Overview Chapter 1 Overview Overview MultiVOIP User Guide About This Manual This manual is about Voice over IP products made by Multi Tech Systems Inc It describes three analog MultiVOIP units with SIP survivability features models MVP810SS MVP410SS and MVP210SS These MultiVOIP units can inter operate with other contemporary analog MultiVOIP units MVP130 MVP130FXS MVP210 MVP410 and MVP810 with contemporary BRI MultiVOIP units MVP410ST amp MVP810ST with contemporary digital T1 E1 ISDN PRI MultiVOIP units MVP2410 and MVP3010 and with the earlier generation of MultiVOIP products MVP200 MVP400 MVP800 MVP120 etc The table below on next page describes the vital characteristics of the various models in the MultiVOIP product family How to Use This Manual In short use the index and the examples When our readers crack open this large manual they generally need one of two things information on a very specific software setting or technical parameter about telephony or IP or they need help when setting up phonebooks for their voip systems The index gives quick access to voip settings and parameters It s detailed Use it The best way to learn about phonebooks is to wade through examples like those in our chapters on T1 North American standard Phonebooks and E1 Euro standard Phonebooks Finally this manual is meant to be comprehensive
97. 3Excellent Effort gt Logs Traces NAT Traversal VolP Media 6 Voice gt Supplementary Services System Information Others O Best Effort X E Advanced Phone Book VLAN ID p E Statistics Save Setup E Connection Help p IP Parameters Gateway Name MultivolP I Enable DHCP p Diff Serv Parameters Call Control PHB 34 VolP Media PHB 46 IPMask 255 255 25 0 TERE Enable IP Address 192 168 41 81 The web GUI however cannot perform logging in the same direct mode done in the Windows GUI However when the web GUI is used logging can be done by email SMTP The graphic layout of the web GUI is also somewhat larger scale than that of the Windows GUI For that reason it s helpful to use as large of a video monitor as possible The primary advantage of the web GUI is remote access for control and configuration The controller PC and the MultiVOIP unit itself must both be connected to the same IP network and their IP addresses must be known In order to use the web GUI 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 GUI 363 Operation amp Maintenance MultiVOIP User Guide To install the Java program go to the Java directory on the MultiVOIP product CD Double click on the EXE file to begin th
98. 4 unconditional forwarding ED Avge EEES 260 TA vhs geitets sete E EEE 220 412 MultiVOIP User Guide Undefined Registrations 196 Uninstall program menu option COMMANA ee eee ceeeeee eens 348 Uninstall option description MultiVOIP program menu 325 uninstalling MultiVOIP software 85 348 unobtainable tone custom 160 unobtainable tones eee 159 Unpacking ee eesesecseesecseeeeceeeeeeees 63 MV P21 0 oirrese eint en 65 MVP410 SS ooer 64 MV P8I0 S Sorio 64 Up Time System Info eee 202 289 update interval logging 170 updated firmware obtaining 327 Upgrade Software option description MultiVOIP program menu 325 upgrade firmware cee eee 331 uploads vs downloads FTP 351 Use Proxy SIP field Bi E EE poses psattitsers 255 A aA EEEE E TTS 214 Use SIP Proxy field eee 150 Use TDM Routing for Intra Gateway Calls senian erie ied 110 user default configuration creating E aride ava Rl a Wowie dvs 204 user defaults downloading 341 user defaults setting cee 341 user name Windows GUL 344 User Name proxy server field 151 user values software saving 341 variations in PBX characteristics Bla E E EE ETE 280 Theronin dunen ai 240 version numbers software 80 version firmWwale cceeeeeees 331 VLAN ID Ethernet IP Parameters field
99. 5 Options SC FEC Displays VOIP transmission 295 options in use on the current call These may include Forward Error Correction or Silence Compression Operation amp Maintenance MultiVOIP User Guide Call Progress Details Field Definitions cont d DTMF Other Details Field Name Values Description Prefix specified Displays the dialed digits that Matched dialing digits were matched to a phonebook entry Outbound 0 9 The digits transmitted by the Digits Sent MultiVOIP to the PBX telco for this call Outbound 0 9 Of the digits transmitted by the Digits MultiVOIP to the PBX telco for Received this call these are the digits that were confirmed as being received Server Details n n n n The IP address etc of the traffic DTMF Capability for n 0 255 and or other server IP related descriptions inband out of band Expressions differ slightly for different Call Signaling protocols H 323 SIP or SPP control server if any being used whether an H 323 gatekeeper a SIP proxy or an SPP registrar gateway will be displayed here if the call is handled through that server Indicates whether the DTMF dialing digits are carried Inband or Out of Band The corresponding field values differ for the 3 different voip protocols For H 323 this field can display Out of Band or Inband For SIP it can display either Out of Band RFC2833 or Ou
100. 5 TECHNICAL CONFIGURATION csccccssssscesssscccssscccsssssccees 88 CONFIGURING THE MULTIVOIP w ccccccssccccececsscsessececececsesssececceecsesenseeeeceeceenesees LOCAL CONRIGURATION y oree a En T r E A Pre R quisite Sorrisi niipea a Ea E raana Eain TP PAararmneters EAE AE EE EO E A AT ST Telephony Interface Parameters c esceeseeseeteees SMTP Parameters for email call log reporting Config Info Check List ie seetscesgeveleessdeedhcescevest ead doh E eE ko Se er Ee ES Local Configuration Procedure Summary sosser Local Configuration Procedure Detailed cccsccsssccssesssesecrseseenseneesecnsseneeaees Modem Relay sisiccsscesssesresssecsacdassesiea ite ciees tausoegidecss deta sasticednestssesceveseedeasdasegoesebest CHAPTER 6 T1 PHONEBOOK CONFIGURATION u sscccssssscssssccccssseees 205 T1 VERSUS E1 TELEPHONY ENVIRONMENTSG cseccscccececsessseceeececsessseeeeececeenens 206 CONFIGURING T1 NAM TELEPHONY MULTIVOIP PHONEBOOKG cc 000008 206 T1 PHONEBOOK EXAMPLEG csessccccececeessaececececsessaaececcceceessseeeceeeesesennsaeeeeeesenes 222 SSites ALA TL Example seeen nesnenin ea EE A N R En 222 Configuring Mixed Digital Analog VOIP Systems ossos 228 Call Completion Summaries ccccsscceseceenseceeseeensecesneeesseceseceeaseceaeecsaeceeneeenaeees 237 Variations in PBX CHArACtEr istics ccccccccccccccecececececececececececececesececeseseseseseseseeens 240 CHAPTER
101. 7 E1 PHONEBOOK CONFIGURATION u ssccccssssecessscccsesseees 241 E1 VERSUS T1 TELEPHONY ENVIRONMENTSG s sessssccececsesssaeceeececsessstsceeececeenens 242 E1 STANDARD INBOUND AND OUTBOUND MULTIVOIP PHONEBOOKS 00006 242 Free Calls One VOIP Site to ANOthe rn cccccccccccccccccccccsccececececesesesssessessssseseseees 243 Local Rate Calls Within Local Calling Area of Remote VOIP 00 000 244 National Rate Calls Within Nation of Remote VOIP Site 246 Inbound versus Outbound PhOn DOOKS ccccccccecececececscscececececececececssscsssssseeseees 247 PHONEBOOK CONFIGURATION PROCEDURE cccessesssceeececsessnsececeeeceesenseeeceeeceees 251 E1 PHONEBOOK EXAMPLES sssscsccccccecsessaececececsesseaececececsesseaeaecececsesseaeceeeeeenss 262 3 Sites AU E Example ccscccccsccuscceesecasusesenceuvsoensoedvennscteceusiccbanevssnaesanivevieavacodes 262 Configuring Digital amp Analog VOIPs in Same System 269 Call Completion S timmiaries senere aerea oversees death E EEE ER E R tes 277 Variations in PBX CHArACtEr istics ccccccccccccccecececececececececesecececesecesesesesesesesesenens 280 International Telephony Numbering Plan Resources 281 CHAPTER 8 OPERATION AND MAINTENANCE eessseesseceessececssecssssessoceessee 283 OPERATION AND MAINTENANCE c cccccecssssseceeececeessaeceeececsensaececeeeeeesenssaeeeeeeeenes 284 SIP Server Endpoint Statistics SCr N 1 csccsccsseesseeseeeeee
102. 8 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 3 9 and a metallic object that will provide an electrical ground 9 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 Proceed to Chapter 4 to load the MultiVOIP software Cabling Procedure for MVP210 SS Cabling involves connecting the MultiVOIP to your LAN and telephone equipment 1 For DID channels only If both channels of your MVP210 SS MultiVOIP will be using either FXS FXO or E amp M telephony interfaces skip to step 2 For any channel on which you are using the DID interface type you must change the jumper on the MultiVOIP circuit card a Disconnect power Unplug the AC power cord from the wall outlet or from the receptacle on the MultiVOIP unit b 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 SS c Pull the main circuit card out about half way 73 Mechanical Installation amp Cabling MultiVOIP User Guide d Identify the channels on which the DID interface will be used as configured for DID Interface Ploy 1Ch 1 Jumper
103. 833 to transmit the DTMF digits SIP Info method Generates dual tone multi frequency DTMF tones on the telephony call leg The SIP INFO message is sent along the signaling path of the call You must set this parameter per the capabilities of the remote endpoint with which the voip will communicate The RFC2833 method is the more common of the two methods FAX Parameters Fax Enable Y N Enables or disables fax capability for a particular channel Modem Y N When enabled modem traffic can be Relay carried on voip system When disabled Enable modem traffic will bypass the voip system Modem Bypass mode Max Baud 2400 4800 Set to match baud rate of fax machine Rate 7200 9600 connected to channel see Fax machine s Fax 12000 user manual 14400 bps Default 14400 bps 116 MultiVOIP User Guide Technical Configuration Voice Fax Parameter Definitions cont d Field Name Valuee Description FAX Parameters cont d Fax Volume 18 5 dB Controls output level of fax tones To Default to 3 5 dB be changed only under the direction of 9 5 dB Multi Tech s Technical Support Jitter Value Default Defines the inter arrival packet Fax 400 ms deviation in milliseconds for the fax transmission A higher value will increase the delay allowing a higher percentage of packets to be reassembled A lower value will decrease the delay allowing fe
104. 9 9 0 Allows remote voip users Note 1 to call all PBX extensions at Site D Inner London using only three digits 270 MultiVOIP User Guide E1 PhoneBook Configuration Phone Book for Analog VOIP Host Unit Site B continued VOIP Dir IP Address Channel Comments OR Destination Pattern 0207 200 2 9 9 0 Gives remote voip users XXX Note 2 access to phone numbers XXXX in 0207 area code Inner London in which Site D is located 0208 200 2 9 9 0 Gives remote voip users XXX Note 2 access to phone numbers XXXX in 0208 area code Outer London for which calls are local from Site D Inner London Note 1 The x is a wildcard character Note 2 By specifying Channel 0 we instruct the MVP3010 to choose any available data channel to carry the call Note 3 Note that Site F key system has only 30 extensions x7400 7429 This destination pattern 018226374 actually directs calls to 402 263 7430 through 402 263 7499 into the key system as well This means that such calls which belong on the PSTN cannot be completed In some cases this might be inconsequential because an entire exchange fully used or not might have been reserved for the company or it might be unnecessary to reach those numbers However to specify only the 30 lines actually used by the key system the destination pattern 018226374 would have to be replaced by three other destinati
105. AL INSTALLATION AND CABLING 0000 61 INTRODUCTION iieciciecbethetcescescscutestecuneetcoucbuco weenie cvecbtebvecnrus lt Vevbeustvonastechedadustoatence de 62 SAFETY WARNINGS soren en un a E dep tevusesteedbesdveecus AEN 62 Lithium Battery Caution thesia cate evesdeobsvansectawrsueceestesicbeneeusaveresesvsees 62 SAfety Warnings TeleCOMK scssecesscscevecesstbeseveocstnletcabevengssVaresuetveceasechoeustssuepoogecteets 62 UNPACKING YOUR MULTIVOIP cccccccscssesssescsseceseecessscsesesessesesssessessseesereseees 63 Unpacking the MVP 41OSS 810SS ccsccescesecsseesseeeeeaeeeseeseeeseeeseeeaeeseeeneensees 64 Unpacking the MVP21 028S 0 cssisessscsieeisbeiscucs weesesstevenss chin EE aE REEE E 65 Safety Recommendations for Rack Installations cccsccscsssseesecseensesecneeenenseeneees 67 19 Inch Rack Enclosure Mounting Procedure 68 CABLING PROCEDURE FOR MVP 410SS 810SS 0 0 cceececeeseseeescssecseecseesseneseeeneees 69 Cabling Procedure for MVP210 SS ccccccsccesscesecesecsseesecusecsecaeeeseeeseeseeeneensees 73 Contents MultiVOIP User Guide CHAPTER 4 SOFTWARE INSTALLATION ccsssssscssssecesssccccssssccssssseecees 77 INTRODUCTION a a tegeecsyes ventas devs vue de vacdegdavsdesten eas ce setaucsiceseesusasseeees 78 LOADING MULTIVOIP SOFTWARE ONTO THE PC ce cccccececsessscecececeesssnsceeeceesenenseee 78 UN INSTALLING THE MULTIVOIP CONFIGURATION SOFTWARE scs0ccsceceeeesenteee 85 CHAPTER
106. Alternate Routing Common Printfs FTP H 323 H450 HUNTING SYSLOG IGK T 38 LOGS WEB Cancel DIFFSERY OK DSP Trace On for Channels Channel 1 Channel 6 Channel 2 Channel 8 Channel 3 Channel 4 Channel 5 Channel 7 Channel 9 Channel 10 172 MultiVOIP User Guide Technical Configuration 16 Set Supplementary Services Parameters This dialog box can be reached by pulldown menu keyboard shortcut or sidebar Accessing Supplementary Services Parameters Pulldown Icon Multi oIP Multi OIP SS v3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl alt N Regional Parameters Ctrl R SMTP Parameters Ctri Alt 5 Ctri Alt L Supplementary Services Ctrl 4lt H System Information Ctrl aAlt ey SIP CallSignaling Ctrl Alt SFt P RADIUS Ctrl Alt U NAT Traversal Ctrl alt Srt y Shortcut Sidebar E Configuration Ethernet IP Ctrl Alt H Yoice Fax Interface SIP Call Signaling Regional SMTP RADIUS Logs Traces NAT Traversal Supplementary Services System Information Supplementary Services features derive from the H 450 standard which brings to voip telephony functionality once only available with PSTN or PBX telephony Supplementary Services features can also be used under SIP but they are implemented differently in SIP than in H 323 Even though the H 450 standard refers only to H 323 Supplementary
107. CP packets received by this value VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of error laden RTCP packets with value received by this VOIP gateway since the Errors last clearing or resetting of the counter within the MultiVOIP software 310 MultiVOIP User Guide Operation amp Maintenance About Link Management The Link Management screen is essentially an automated utility for pinging endpoints on your voip network This utility generates pings of variable sizes at variable intervals and records the response to the pings Accessing Link Management Pulldown none Shortcut Icon Sidebar Ctrl 2 none Statistics Call Progress Logs IP Statistics Link Management T1j E1 Statistics Registered Gateway r Link Management r Monitor Link IP Address to Ping O05 0 20 Pings per Test p Ping Size in Bytes 22 Response Timeout fiooa Time Interval between Tests ja min ms Start Now Clear r Link Status IP Address Round Trip Delay Min Ma Last Error 311 Operation amp Maintenance MultiVOIP User Guide Link Management screen Field Definitions Field Name Values Description Monitor Link fields IP Address to a b c d This is the IP address of the target Ping 0 255 endpoint to be pinged Pings per Test 1
108. DN BRI service An ISDN Basic Rate BRI U Loop consists of two conductors from the telco central office to the customer premises The equipment on both sides of the U loop accommodates the extensive length of the U loop and the noisy environment in which it may operate At the customer premises the U loop is terminated by an NT1 network termination 1 device An NT1 device makes an end user s 4 wire terminal equipment compatible with the telco s 2 wire twisted pair ISDN BRI line The NT1 drives an S T bus The S T bus is usually made up of 4 wires but in some cases may be 6 or 8 wires S and T refer to connection points in the ISDN specification When a PBX is present S refers to the connection between the PBX and the terminal Terminal can mean any sort of end user ISDN device data terminals telephones FAX machines voip units etc Point T refers to the connection between the NT1 device and customer supplied equipment Terminals can connect directly to the NT1 device at point T or there may be a PBX private branch exchange i e a customer owned telephone exchange The figure below shows S and T connection points in an ISDN network Point S T 4 8 Wires Point T 1 N 48 Wires aN PBX Terminal Point S Terminal NT1 Point U 2 Wires Point S Terminal 384 MultiVOIP User Guide TCP UDP Port As
109. 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 Uncondit Unconditional When selected Condition Busy all calls received will be No Resp forwarded Busy When selected calls will be forwarded when station is busy No Response When selected calls will be forwarded if called party does not answer after a specified number of rings as specified in Ring Count field Forwarding can be conditioned on both Busy and No Response 260 MultiVOIP User Guide E1 PhoneBook Configuration Add Edit Inbound Phone Book Field Definitions cont d Field Name Values Description Forward Phone number or IP address to which calls Destination will be directed IP address For H 323 calls the Forward Destination can phone number be either a Phone Number of an IP Address port number For SIP calls the Forward Destination can be etc one of the following a phone number b IP address c IP address port number d phone number IP addr 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 man
110. E 255 T e ra EA 214 wink signaling DID DPO 148 wink signaling E amp M eee 144 Wink Timer DID DPO field 148 Wink Timer E amp M field 144 XMT channel LED ee 18 XSG LEDien anaa 18 Multi lech Systems S000393B
111. Endpoint Statistics Parameters 286 contacting technical support 373 coordinated phonebook entries Biases EEEE E EESE 248 A D E EE E E Stees he 207 Copy Channel command Interface Parameters c cccccccccessssseeeees 128 Copy Channel command Voice Fax Parameters cc cccccccccessssseeeeeee 114 Copy Channel field ee 115 Copy Channel Supplementary Services command 0 175 Copy Channel Supplementary Services field tees eeeeeeeeee 183 Count of Registered Numbers field Registered Gateway Details 316 Country Selection for Built In Modem Peldi Pak ine tact te aes 158 Country Region tone schemes field stateside ahs age meee sh 155 156 Creating a User Default Configuration Ee rome eco Toe 204 Current Loss FXO disconnect criteria field oo cece eeee 141 Current Loss field FXS Loop Start 130 Current Loss Timer FXO disconnect criteria field oo eee eee 141 Current Reversal FXO answer supervision field eee 140 Current Reversal FXO disconnect criteria field eee eens 141 Custom tones Regional field 157 custom cadences cseeceeeeeteeees 161 custom DTMBP ccsscceeeseeees 160 Custom Fields RADIUS Attributes definitions eee ceeteeeeeeeee 190 Custom Fields RADIUS definitions PEE EEE A ESEE EES 191 Custom Fields SMTP definitions E EAA ATES 165 166 Custom Fields RADIUS Accounting Attributes Bytes Received s s s 191
112. Guide Call Progress Details Field Definitions Field Name Values Description Packet Details Packets Sent integer value The number of data packets sent over the IP network in the course of this call Packets 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 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 294 MultiVOIP User Guide Operation amp Maintenance Call Progress Details Field Definitions cont d From To Details Description Gateway alphanumeric Identifier for the VOIP gateway Name from string that handled the origination of this call IP Address X X X X IP address from which the call from where xhasa was received range of 0 to 255 Options SC FEC Displays VOIP transmission options in use on the current call These may include Forward Error Correction or Silence Compression Gateway alphanumeric Identifier for the VOIP gateway Name to string that handled the completion of this call IP Address X X X X IP address to which the call was to where x has a sent range of 0 to 25
113. Hele 9001 Registered with Alternate 1 00 00 36 0 5001 Registered with 5 00003 0 6 601 Registered with SS 00 00 20 0 61 602 Registered with SS 00002 0 6 z No of Endpoimts 15 Detalls Re registr ation Type Dynamic Endpoint Type Undefined Contact Details Contact Adgress Pon Number RemainingTime 2002927 080 00000428 a e ee es Of 02 03 o z OC Er TO O enet Mstar A A comi prec typerter fe Amartac Festurecal L E Jawor Mukivorr s BDvGm e om 287 Operation amp Maintenance MultiVOIP User Guide System Information screen This screen presents vital system information at a glance Its primary use is in troubleshooting This screen is accessible via the Configuration pulldown menu the Configuration sidebar menu or by the keyboard shortcut Ctrl Alt Y r System Information Boot Version 1 04 Version Information Firmware Version 6 07 C Configuration Version 6 07 00 01 Phone Book Yersion 4 04 IFM Version ee MAC Address 000800501820 Uptime 00 00 00 18 Hardware ID MYP410 32M Rev B F998 Exit 288 MultiVOIP User Guide Operation amp Maintenance System Information Parameter Definitions Field Name Values Description Boot nn nn Indicates the version of the code that Version alpha is used at the startup booting of the numeric voip The boot code version is independent of the sof
114. IF 0 MasterSlaveStatus Slave 00033675 H323IF 0 FastStart Setup Not Used 00033690 CAS 0 TX ABCD 1 1 1 1 00033755 H323IF 0 Coder used g7231 00033810 PSTN pstn call connected on 0 58 MultiVOIP User Guide QS Connectivity Test Console Messages from Terminating VOIP The voip unit connected to the phone where the call is answered will send back messages like that shown below 00170860 00170860 00170885 00171095 00171105 00171105 00171110 00171110 00171110 00171315 00172275 00172285 00172995 00173660 00173760 H323 0 New incoming call PSTNIF Placing call on channel O Outbound digit 7175662 CAS 0 TX ABCD 1 1 1 1 H323 IF 0 MasterSlaveStatus Master CAS 0 RX ABCD 1 1 1 1 Pstn State 7 TimeStamp 171105 H323IF 0 Coder used g7231 H323IF 0 FastStart Setup Not Used H323IF 0 Already opened the outgoing logical channel H323IF 0 Coder used g7231 CAS 0 RX ABCD 0 0 0 0 Pstn State 9 TimeStamp 171315 PSTN dialing digit ended on 0 PSTN pstn proceeding indication on 0 CAS 0 RX ABCD 1 1 1 1 Pstn State 12 TimeStamp 172995 CAS 0 TX ABCD 1 1 1 1 PSTN pstn call connected on 0 9 When you see the following message end to end voip connectivity has been achieved PSTN pstn call connected on X where x is the number of the voip channel carrying the call 10 If the HyperTermin
115. International Call Example Technician in Rotterdam country 31 city 010 to set up one voip there another in Bordeaux country 33 area 05 Answer write down 3305 40 MultiVOIP User Guide QS Phonebook Starter Config 4 Suppose you want to call a phone number outside of your building using a phone station that is an extension from your PBX system if present What digits must you dial Often a 9 or 8 must be dialed to get an outside line through the PBX i e to connect to the PSTN Generally 1 or 11 or 0 must be dialed as a prefix for calls outside of the calling code area long distance calls national calls or international calls On a sheet of paper write down the digits you must dial before you can dial a remote area code North America Euro National Call Long Distance Example Example Seattle Chicago system London Birming system Seattle voip works with London voip works with PBX that uses 8 for all PBX that uses 9 for all voip calls 1 must out of building calls immediately precede area whether by voip or by code of dialed number PSTN 0 must immediately precede area anwar e ag code of dialed number Answer write down 90 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 Answer write down 90
116. MultiVOIP SS Survivable SIP Gateway amp Server User Guide for Voice IP Gateways Models MVP210 SS MVP410 SS MVP810 SS Multi lech Systems User Guide 000393B Analog MultiVOIPs with SIP Survivability Models MVP210 SS MVP410 SS amp MVP810 SS 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 2006 by Multi Tech Systems Inc Multi Tech Systems Inc makes no representations or warranties 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 Description A Doc re organization Follows S000249K 12 09 05 Describes 3 08 software release B Add full details to Quick Start Instructions chapter Ch 2 10 05 06 Patents This Product is covered by one or more of the following U S Patent Numbers 6151333 5757801 5682386 5 301 274 5 309 562 5 355 365 5 355 653 5 452 289 5 453 986 Other Patents Pending Trademark Trademark of Multi Tech Syst
117. MultiVOIP User Guide Technical Configuration 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 Interface Type Dialing Options Inter Digit Timer 2 secs Message Waiting Indication Light Inter Digit Regeneration Timer 100 Pass Through m FXS Options T Enable FXS Ring Count fe I Current Loss Caller ID Generate Current Reversal ae m Flash Hook Options bod BellCore E Generation eo0 ms I Enable Detection Range Min 500 ms Max 1000 ms FXS Loop Start Interface Parameter Definitions Field Name Values Description FXS Loop Y N Enables FXS Loop Start Start interface type 129 Technical Configuration MultiVOIP User Guide FXS Loop Start Interface Parameter Definitions cont d Field Name Values Description Dialing Options fields Inter Digit 1 10 seconds This is the length of time that Timer 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 Not applicable to FXS Loop Waiting Start interface Indication Inter Digit in milliseconds The length of time between the Regeneration outputting of DTMF digits Time Default 100 ms FXS Options fields
118. 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 122 MultiVOIP User Guide Technical Configuration Voice Fax Parameter Definitions cont d 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 configurable minimum and maximum values An Optimization Factor adjustment controls how quickly the length of the Jitter Buffer is increased when jitter increases on the network The length of the jitter buffer directly effects the voice delay between MultiVOIP gateways Minimum Jitter Value 60 to 400 ms The minimum dynamic jitter buffer of 60 milliseconds is the minimum delay that would be acceptable over a low jitter network Default 150 msec 123 Tech
119. OK To do logging with a SysLog client program click on SysLog Server Enable in the Logs screen To implement this function you must install a SysLog client program For more info see the SysLog Server Functions section of the Operation amp Maintenance chapter of the User Guide 37 QS Phone IP Starter Config MultiVOIP User Guide Phone IP Starter Configuration continued 12 Enable premium H 450 telephony features 13 14 15 16 Go to Supplementary Services Select any features to be used For Call Hold Call Transfer amp Call Waiting specify the key sequence that the phone user will press to invoke the feature For Call Name Identification specify the allowed name types to be used and a caller id descriptor If Call Forwarding is to be used enable this feature in the Add Edit Inbound Phone Book screen After making changes click on OK in the current configuration screen before moving on to the next configuration screen RADIUS Support If you intend to use a RADIUS server for billing or other accounting purposes enter the server information in the RADIUS screen STUN Support If you are using the SIP protocol with the UDP transmission protocol and if you want the MultiVOIP to operate behind a NAT Network Address Translation server using the STUN protocol Simple Traversal of UDP through NAT enable this feature in the NAT Traversal screen You must also specify the IP address e
120. P 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 invoke the Download IFM Firmware command go to Start Programs MVP 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 OOOO 6 hl Downloading IFM Firmware will Reboot the Multi OIP Do you want to continue Cancel 4 The Boot LED on the front panel of the Multi VOIP will come on 5 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 Multi VOIP or if you want to overwrite the same version of firmware click Open 338 MultiVOIP User Guide Operation amp Maintenance O MultiVOIP 6 07 e ngayv111 ifm 6 The IFM Firmware Download screen will appear Select Copy to All IFMs and click OK Only in very special circumstances would different IFMs in the same voip be loaded with different IFM firmware IFM Firmware Download M Iv M M M orms M oiFMe IFM7 D IFM8 7 The main MultiVOIP Configuration screen will appear Progress bars
121. P Statistics Field Definitions Field Name Values Description UDP versus TCP continued Both TCP and UDP split data into packets called datagrams However TCP includes extra headers in the datagram to enable retransmission of lost packets and reassembly of packets into their correct order if they arrive out of order UDP does not provide this Lost UDP packets are unretrievable 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 appear as static IP Address n n n n 0 255 IP address of the MultiVOIP For an IP address to be displayed here the Multi VOIP must have DHCP enabled Its IP address in such a case is assigned by the DHCP server Clear button Clears packet tallies from memory Transmit ted Received integer value integer value Total Packets Sum of data packets of all types Total number of packets transmitted by this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Total number of packets
122. P User Guide Custom Fields Definitions cont d Field Description Field Description Bytes Total bytes sent in Bytes Total bytes received Sent call Received in call Packets Packets lost in Coder Voice Coder Lost call Compression Rate used for call will be listed in log Outbound The DTMF dialing Prefix When selected the By ed digits received by Matched phonebook prefix this gateway from matched in the remote processing the call gateway will be listed in log presuming that DTMF is set to Out of Band Call Successful or Call Type Indicates the Call Status unsuccessful Signaling protocol used for the call H 323 SIP or SPP Call Indicates call s DTMF Indicates whether the Capability DTMF dialing digits are carried Inband or Out of Band The corresponding field values differ for the 3 different voip protocols For H 323 this field can display Out of Band or Inband For SIP it can display either Out of Band RFC2833 or Out of Band SIP INFO to indicate the out of band condition or Inband to indicate the in band condition For SPP it can display Out of Band RFC2833 or Inband Direction originating party 166 MultiVOIP User Guide Technical Configuration Custom Fields Definitions cont d Field Description Field Description Server The IP address of Outbound The dialing digits
123. P User Guide The screen below shows Outbound PhoneBook entries for the VOIP located in the company s London facility 200 002 009 007 Lyon 200 002 008 005 Amsterdam 200 002 008 005 Rotterdam 200 002 009 007 Paris company office empl extensions 200 002 008 005 Amsterdam company office employees The Inbound PhoneBook for the London VOIP is shown below NOTE Commas are allowed in the Inbound Phonebook but not in the Outbound Phonebook Commas denote a brief pause for a dial tone allowing time for the PBX to get an outside line 266 MultiVOIP User Guide E1 PhoneBook Configuration The screen below shows Outbound PhoneBook entries for the VOIP located in the company s Paris facility 200 002 008 005 Rotterdam 200 002 010 003 London Inner 200 002 010 003 London Outer 200 002 010 003 Guildford 200 002 010 003 London company office empl extensions 200 002 008 005 Amsterdam company office employees The Inbound PhoneBook for the Paris VOIP is shown below 267 E1 Phonebook Configuration MultiVOIP User Guide The screen below shows Outbound PhoneBook entries for the VOIP in the company s Amsterdam facility 200 002 010 003 London inner 200 002 010 003 Guildford 200 002 009 007 Paris 200 002 009 007 Lyon 200 002 010 003 London company office employ ext 200 002 009 007 Paris company office employee ext The Inbound PhoneBook for the Amsterdam VOIP is sho
124. P810 can be operated in either a North American telephony standards environment potentially operating with T1 digital MultiVOIPs or in a European telephony standards environment potentially operating with E1 digital MultiVOIPs The configuration of the phonebook is the same in either case However because the telephony environment is different in each case and the examples used here must reflect those differences we have separate chapters for phonebook configuration in North American T1 environments Chapter 6 this chapter and for that in European E1 environments Chapter 7 Consult the chapter that best fits the needs of your voip system Configuring T1 NAM Telephony MultiVOIP Phonebooks When a VoIP serves a PBX system it s important that the operation of the VoIP be transparent to the telephone end user That is the VoIP should not entail the dialing of extra digits to reach users elsewhere on the network that the VoIP serves On the contrary VOIP service more commonly reduces dialed digits by allowing users served by PBXs in facilities in distant cities to dial their co workers with 3 4 or 5 digit extensions as if they were in the same facility Furthermore the setup of the VoIP generally should allow users to make calls on a non toll basis to any numbers accessible without toll by users at all other locations on the VoIP system Consider for example a company with VOIP equipped offices in New York Miami and Los
125. Password Reconfirm Password 347 Operation amp Maintenance MultiVOIP User Guide Un Installing the MultiVOIP Software 1 To un install the MultiVOIP configuration software go to Start Programs and locate the MultiVOIP entry Select Uninstall MVP Vx xx versions may vary Windows98 S By Exploring mvp30 dwg C hE Uninstal 2 Two confirmation screens will appear Click Yes and OK when you are certain you want to continue with the uninstallation process Confirm File Deletion Confirm File Deletion 348 MultiVOIP User Guide Operation amp Maintenance 3 A special warning message similar to that shown below may appear for the MultiVOIP software s bin file Click Yes ReadOnly File Detected x An option that you selected requires that files be installed to your system or files be uninstalled from your system or both A read only file C ProgramFiles MVP3000 v4 00a mvpt1 bin was found while performing the needed file operations on your system To perform the file operation click the Yes button otherwise click No Yes No Cancel 4 A completion screen will appear InstallShield Wizard Maintenance Complete InstallShield Wizard has finished performing maintenance operations on MuRiVOIP model version Click Finish 349 Operation amp Maintenance MultiVOIP User Guide Upgrading Software As noted earlier see the section Im
126. Selected Coder drop down list the 8 8 9 6 Manual option must be enabled kbps Max 11 128 This drop down list enables you to bandwidth kbps select the maximum bandwidth coder allowed for this channel The Max Bandwidth drop down list is enabled only if the Coder is set to Automatic If coder is to be selected automatically Auto setting then enter a value for maximum bandwidth 118 MultiVOIP User Guide Technical Configuration Voice Fax Parameter Definitions cont d Field Name Values Description Advanced Features Silence Y N Determines whether silence Compression compression is enabled checked for this voice channel With Silence Compression enabled the MultiVOIP will not transmit voice packets when silence is detected thereby reducing the amount of network bandwidth that is being used by the voice channel Default on Echo Y N Determines whether echo cancellation is Cancellation enabled checked for this voice channel Echo Cancellation removes echo and improves sound quality Default on Forward Y N Determines whether forward error Error correction is enabled checked for this Correction 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 119 Technical Configura
127. Serer nen eer er eee 208 Add Prefix inbound field Eleroane nreno a aesa 259 d BI LE EE ere ee cure eres 219 Add Prefix outbound field ED EA 254 a O S SA 213 Add Edit Inbound Phonebook field definitions Eid REEE E 259 260 261 a EE E a EEAS 219 220 221 Add Edit Inbound Phonebook screen DP rich AE N AEA EEA 259 CE 6 E PE EEA ES 219 Add Edit Inbound Phonebook screen fields E1 Accept Any Number 259 Add Prefix 259 Channel Number 260 Description callee location 260 Enable Call Forwarding 260 Forward Condition 0 0000 260 Forward Destination 0 261 Registration Option Parameters 261 Remove Prefix cc0ecccceeeeee 259 Ring Count ee eeeeeeeseeeeees 261 Add Edit Inbound Phonebook screen fields T1 Accept Any Number 219 Add Prefi eenean este 219 Channel Number 219 Description callee location 219 Enable Call Forwarding 219 Forward Condition 0 0 000 220 MultiVOIP User Guide Forward Destination 220 Registration Option Parameters 221 Remove Prefix ceceee 219 Ring Count 221 Add Edit Outbound Phonebook field definitions BEV AOON SOE 253 254 255 TT EE 212 213 214 Add Edit Outbound Phonebook fields El Accept Any Numbert 253 Add Prefix ccccccceccsceceseeeees 254 Advanced button 0 cee 255
128. The Pierre PBX dials extension 3117 in the office at Pierre Variations in PBX Characteristics The exact dialing strings needed in the Outbound and Inbound Phonebooks of the MVP2410 will depend on the capabilities of the PBX Some PBXs require trunk access codes like an 8 or 9 to access an outside line or to access the VOIP network Other PBXs can automatically distinguish between intra PBX calls PSTN calls and VOIP calls Some PBX units can also insert digits automatically when they receive certain dialing strings from a phone station For example a PBX may be programmable to insert automatically the three digit VOIP identifier strings into calls to be directed to analog VOIPs The MVP2410 offers complete flexibility for inter operation with PBX units so that a coherent dialing scheme can be established to connect a company s multiple sites together in a way that is convenient and intuitive for phone users When working together with modern PBX units the presence of the MVP2410 can be completely transparent to phone users within the company 240 MultiVOIP User Guide E1 Phonebook Configuration Chapter 7 E1 Phonebook Configuration European Telephony Standards 241 E1 Phonebook Configuration MultiVOIP User Guide E1 Versus T1 Telephony Environments Phonebooks for Series II analog MultiVOIP units MVP130 MVP130FXS MVP210 MVP410 MVP810 MVP210SS MVP410SS and MVP810SS can be operated in either an en
129. The Regional Parameters fields are described in the table below Regional Parameter Definitions Field Name Values Description Country USA Japan UK Name of a country or region that Region Custom uses a certain set of tone pairs for dial tone ring tone busy tone unobtainable tone fast busy tone survivability tone tone heard briefly 2 seconds after going offhook 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 Note 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 155
130. VOIP The call passes through the IP network in this case the Internet The call arrives at the Site B VOIP This analog VOIP receives this dialing string from the MVP2410 101 7175662 The analog VOIP seeing the 101 prefix uses its own channel 1 an FXO port to connect the call to the PSTN Then the analog VOIP dials its local phone number 7175662 to complete the call 238 MultiVOIP User Guide T1 PhoneBook Configuration Site D calling Site F A voip call from Pierre PBX to extension 7424 on the key telephone system in Lincoln Nebraska A The required entry in the Pierre Outbound Phonebook to facilitate origination of the call would be 1402263742 The call would be directed to the Lincoln voip s IP address 200 2 9 5 Generally on such a call the caller would have to dial an initial 9 But typically the PBX would not pass the initial 9 to the voip If the PBX did pass along that 9 however its removal would have to be specified in the local Outbound Phonebook B The corresponding entry in the Lincoln Inbound Phonebook to facilitate completion of the call would be 1402263742 for calls within the office at Lincoln 1402 for calls to the Lincoln local calling area PSTN Call Event Sequence 1 Caller at Pierre dials 914022637424 2 Pierre PBX removes 9 and passes 14022637424 to voip 3 Pierre voip passes remaining string 14022637424 on to the Lincoln voip at IP address 200 2 9 5
131. VOIP software to configure the VOIP r MultiVOIP SS 3 08 Software installation is complete at this point You may proceed with Technical Configuration now or not at your convenience Technical Configuration instructions are in the next chapter of this manual 84 MultiVOIP User Guide Software Installation Un Installing the MultiVOIP Configuration Software 1 To un install the MultiVOIP configuration software go to Start Programs and locate the entry for the MultiVOIP program Select Uninstall an Accessories IF Macromedia FreeHand 9 E Jasc Software Mozilla Firefox Set Program Access and Defaults MultivOIP 6 06 SigmaTel MSCN Audio Player eS Windows Catalog an Snaglt 7 4b TEO Microsoft Office I Acrobat Distiller 7 0 WinZip O Adobe Acrobat 7 0 Professional IM FaxFinder CallFinder Manager fi Multivo AE Configuration T FaxFinder Client Software gt Configuration Port Setup HEU y B Date and Time Setup 2 Download Factory Defaults 5 Download Firmware Settings 2 Download IFM Firmware 2 Download User Defaults B Set Password Launch RealOne Player Documents Search Help and Support 2 Upgrade Software CER IE T EE Ln 26 Col1 REC TRK EXT ove QX A Start CS SnagIt Inbox Microso kA Jasc Paint Shop d New Brochure Telephony Em Windows XP Professional 2 Two confirmation screens will appear Click Yes and OK when you are certain you want to contin
132. VP SS unit can function as a back up SIP server that performs SIP server functions when if the network s main SIP server fails or loses contact with the subnetwork in which the MVP SS unit is placed 29 QS Voip Placement amp PC Settings MultiVOIP User Guide Placement Mount your MultiVOIP in a safe and convenient location where cables for your network and phone system are accessible Rack mounting instructions are in Chapter 3 Mechanical Installation amp Cabling of the User Guide Command Control Computer Setup Specs amp Settings The computer used for command and control of the MultiVOIP a must be an IBM compatible PC b must use a Microsoft operating system c must be connected to your local network Ethernet system and d must have an available serial COM port The configuration tasks and control tasks the PC will have to do with the MultiVOIP are not especially demanding Still we recommend using a reasonably new computer The computer that you use to configure your MultiVOIP need not be dedicated to the MultiVOIP after installation is complete COM port on controller PC You ll need an available COM port on the controller PC You ll need to know which COM port is available for use with the MultiVOIP COM1 COM2 etc 30 QS Quick Hookups MultiVOIP User Guide Quick Hookup for MVP410 SS amp MVP810 SS EE M l M l M l e l l l l l l M l M l M l aofo 0913 wolf aul GIG 0 SP uuo
133. Windows NT4 details will differ slightly in other MS operating systems In the upper toolbar of the HyperTerminal screen click on the Properties button In the Connect To tab of the Connection Properties dialog box click on the Configure button In the next dialog box on the General tab set Maximum Speed to 115200 bps On the Connection tab set connection preferences to Data bits 8 Parity none Stop bits 1 Click OK twice to exit settings dialog boxes 57 QS Connectivity Test MultiVOIP User Guide 7 Make VOIP call Make call on a local phone line accessing PSTN directly or through key system 8 Read console messages recorded on HyperTerminal Console Messages from Originating VOIP The voip unit that originates the call will send back messages like that shown below 00026975 CAS 0 RX ABCD 1 1 1 1 Pstn State 1 TimeStamp 26975 00027190 CAS 0 TX ABCD 1 1 1 1 00027190 PSTN cas seizure detected on 0 00027440 CAS 0 TX ABCD 0 0 0 0 00033290 PSTN call detected on 0 num 17637175662 00033290 H323IF 0 destAddr TA 200 2 10 5 1720 NAME Mounds View TEL 17637175662 17637175662 00033290 H323IF 0 srcAddr NAME New York TA 200 2 9 20 00033440 H323IF 0 cmCallStateProceeding 00033500 H323 0 Remote Information Q931 MultiVOIP T1 00033565 CAS 0 TX ABCD 1 1 1 1 00033675 H323
134. X AutoCall Offhook Alert field 122 Online Statistics Updation Interval field LOgS ceeccesceeteeeteeeeeeee 171 Operating Mode field SIP Server Configuration parameters 0 eee 195 Operating system cece 19 operating temperature 0 67 Operating voltage 20 Optimization Factor field 124 Options callee statistics logs field sophia een ER EE bin 304 Options caller statistics logs field E E E EE ET 304 Options value Survivability Status Check 195 Options From Details RADIUS Attributes field eee 192 Options From Details SMTP logs Meld ise suds ian eee 167 Options To Details RADIUS Attributes field 192 Options To Details SMTP logs field E E Sheet a 167 Others Priorities Ethernet IP params 802 1p feld hyrisee 105 out of band DTMF 00 116 Index Outbound Digits Received call progress field oo eee eee 296 Outbound Digits Received statistics logs field eee ee eeeeeeteeeeeeeee 302 Outbound Digits Received SMTP Joss field aini pna k 166 Outbound Digits Sent call progress field or nan a e rien 296 Outbound Digits Sent RADIUS Attributes field cece 191 Outbound Digits Sent SMTP logs fieldy oee er rer ee 167 Outbound Digits Sent statistics logs field iiss rose esias repot airas 302 Outbound Digits Sent and DTMF Out of Band eee 167 Outbound Phonebook Entries List BLD PRE S 249 TAD RE E ES 208 Outb
135. XXXX PSTN of Site D Pierre SD area code 615 3xxx 200 2 9 9 0 Allows remote voip users to call all PBX extensions at Site D Note Pierre SD using only four digits 1402 200 2 9 5 0 Gives remote voip users access to local PSTN of Site F Lincoln NE area code 402 140226374 200 2 9 5 0 Gives remote voip Note 1 users access to key Note 3 phone system extensions at Site F Lincoln 230 MultiVOIP User Guide T1 PhoneBook Configuration Note 1 The x is a wildcard character Note 2 By specifying Channel 0 we instruct the MVP2400 2410 to choose any available data channel to carry the call Note 3 Note that Site F key system has only 30 extensions x7400 7429 This destination pattern 140226374 actually directs calls to 402 263 7430 through 402 263 7499 into the key system as well This means that such calls which belong on the PSTN cannot be completed In some cases this might be inconsequential because an entire exchange fully used or not might have been reserved for the company or it might be unnecessary to reach those numbers However to specify only the 30 lines actually used by the key system the destination pattern 140226374 would have to be replaced by three other destination patterns namely 1402263740 1402263741 and 1402263742 In this way calls to 402 263 7430 through 402 263 7499 would be properly directed to the PSTN In the Site D outbound phonebook the 30 lin
136. ails see the Technical Configuration chapter of the User Guide 3 Do you want to configure and operate the MultiVOIP unit using the web browser GUI It has the same functionality as the local Windows GUI but offers remote access If NO skip to step 5 If YES continue with step 4 34 MultiVOIP User Guide QS Phone IP Starter Config 4 Web Browser GUI Setup Optional To do configuration and operation procedures using the web browser GUI you must first set it up To do so follow these steps The browser used must be Internet Explorer 6 0 or above or Netscape 6 0 or above or FireFox 1 0 or above A Be sure an IP address has been assigned to the MultiVOIP unit this must be done in the MultiVOIP Windows GUI E Open web browser Note The PC being used must be connected to and have an IP address on the same IP network that the voip is on B Save Setup in Windows GUI F Browse to IP address of MultiVOIP unit C Close the MultiVOIP Windows GUI G If username and password have been established enter them when prompted by voip D Install Java program from MultiVOIP product CD Must be Java Runtime Environment 1 4 2_01 or above NOTE Required on first use of Web Browser GUI only H Use web browser GUI to configure or operate voip Need more info See Web Browser Interface in Operation amp Maintenance chapter of User Guide on CD On
137. al messages do not confirm connectivity go to the Troubleshooting procedure below 59 QS Troubleshooting MultiVOIP User Guide Troubleshooting If you cannot establish connectivity between two voips in the system follow the steps below to determine the problem 1 Ping both MultiVOIP units to confirm connectivity to the network mmand Prompt C gt ping 204 26 122 2 Pinging 264 26 122 2 with 32 bytes of data TIL 254 TTL 254 Reply from 264 26 122 2 i TTL 254 Reply from 264 26 122 2 bytes 32 time lt i ms TTL 254 C gt ping 204 26 122 2 Pinging 264 26 122 2 with 32 bytes of data Reply from 264 TTL 254 204 TIL 254 204 26 122 2 i TTL 254 Reply from 204 26 122 2 bytes 32 time lt i ms TTL 254 Ci gt 2 Verify the telephone connections Check cabling Are connections well seated To correct receptacle Are telephone Interface Parameter settings correct 3 Verify phonebook configuration 4 Observe console messages while placing a call Look for error messages indi cating phonebook problems network problems voice coder mismatches etc 60 MultiVOIP User Guide Mechanical Installation amp Cabling Chapter 3 Mechanical Installation and Cabling 61 Mechanical Installation amp Cabling MultiVOIP User Guide Introduction When MVP410 SS or MVP810 SS units are to be installed into a rack two able bodied persons should participate The MVP210 SS is a table top unit that can genera
138. answer supervision field 0 0 0 0 eee 140 Available Tones FXO disconnection SUPCLVISION 0ceeeeseeeteeeeeeeee 142 bandwidth code 0cceeeee 118 battery caution oe eects eeeeeeeees 62 baud rate default Multi VOIP software connection 193 baud rate fax eee eeeeeeeeerees 116 baud rate setting 0 0 eee 193 Boot LED siria erita 18 MV P210 S Siiri iiaa 76 MVP 410SS 810SS oo eee 73 Boot Version System Info eee 201 289 booting CHIME eee eee eeeetecteeeeeeeeee 18 box contents VETIFYIN gak irrien 63 BRI connector pinott 383 BRI interface types ST and Urian ansan 384 built in modem setup in Regional Parameters busy amp no response E1 forwarding dual conditions 260 busy amp no response T1 forwarding dual conditions 220 busy tone Custom eee 160 DUSY tONES 0 ee ee eee este ceeeteeeeee 159 Bytes Received call progress field wield stieseonuer E E EEE 294 Bytes Received RADIUS Attributes field 191 Bytes Received SMTP logs field 166 Bytes received statistics logs field a E ENEA 303 Bytes Sent call progress field 294 390 MultiVOIP User Guide Bytes Sent RADIUS Attributes field EE E sued cestcseeste tones 191 Bytes Sent SMTP logs field 166 Bytes sent statistics logs field 303 cabling diagram quick 210 32 cabling diagram quick 410 810 31 cabling proble
139. ar Dialog m IP Parameters rm Diff Serv Parameters Call Control PHB 34 ie Frame Type TYPE4I 46 VolP Media PHB m IP Parameters I Enable DHCP _ IP Address 192 168s 3 TH Cancel IP Mask 255 255 265 0 Help Gateway i 6 Set the IP values per your particular VOIP system Click OK Progress bars will appear as the MultiVOIP reboots itself 343 Operation amp Maintenance MultiVOIP User Guide Setting a Password Windows GUI After a user name has been designated and a password has been set that password is required to gain access to any functionality of the MultiVOIP software Only one user name and password can be assigned to a voip unit The user name will be required when communicating with the MultiVOIP via the web browser GUI NOTE Record your user name and password in a safe place If the password is lost forgotten or unretrievable the user must contact MultiTech 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 invoke the Set Password command go to Start Programs MVP x xx Set Password iy Multi VOIP 800 v3 01E gt B Configuration 53 MultiVOIP 100 v7 01E gt E Date and Time Setup H MultiVOIP 100 v7 514 gt Z5 Download Factory Defaults RouteFinder Manager 7 28 gt 25 Download F
140. arameter fields Busy Party resres eiee 181 Supplementary Services Parameter fields Allowed Name Types 181 Supplementary Services Parameter fields Connected Party eee 182 Supplementary Services Parameter fields Allowed Name Types 182 Supplementary Services Parameter fields Caller TD oro arataa 183 Supplementary Services Parameters fields Transfer Sequence e 176 Supplementary Services Parameters SCreen ACCESSING eeeeeeeeceeeee 173 Supplementary Services parameters SOU Oe reae e E 173 Supplementary Services compatible With SIP oyni ire ies 173 Supplementary Services incompatible With SIP eieae an 14 support technical eee 373 Survivability Status Check field SIP Server Configuration Parameters eee 195 SysLog Client 0 0 cee eeeeteereeeeeteeees 16 SysLog client programs availability ossee 367 features amp presentation types 369 SysLog functionality eee 16 SySLOg Server oo eee eee eee ceeeeeee 16 SysLog Server Enable field 171 SysLog Server function as added feature eee 367 capabilities Of 369 enabling su ciciadiuditee at 368 location Of eee eeeeeeseeeeeeeeee 367 SysLog Server IP Address field 171 SysLog Server enabling 170 System Information Parameters Boot Version 289 Configuration Version 289 IFM Version seese 289 411 Index Phone Book Version
141. ation Similarly the VOIP system allows Wren Clothing employees in London and Amsterdam to call anywhere in France at French national rates it allows Wren Clothing employees in Paris and Amsterdam to call anywhere in the United Kingdom at its national rates Wren Clothing Co VOIP PBX Site London Wren Clothing Co VOIP PBX Site Amsterdam The Netherlands Wren Clothing Co VOIP PBX Site Oe aed Calls at French one National Rates KIAN Calls at UK y France National Rates Inbound versus Outbound Phonebooks To make the VOIP system transparent to phone users and to allow all possible free and reduced rate calls the VOIP administrator must 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 calls terminating at points beyond the remote VOIP 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 lists the dialing sequences that can be used t
142. ation Protocol Data Compression amp Quality of Service The analog MultiVOIP unit comes equipped with a variety of data compression capabilities including G 723 G 729 and G 711 and features DiffServ quality of service QoS capabilities PSTN Failover Feature The MultiVOIP can be programmed to divert calls to the PSTN temporarily in case the IP network fails RADIUS Support Inter operation with a RADIUS server allows for call accounting especially for billing on a voip system The MultiVOIP supports inter operation with RADIUS servers for the RADIUS accounting function but not the RADIUS authentication function STUN Support The STUN protocol Simple Traversal of UDP through NATs Network Address Translation assists with the packet routing functions of devices behind NAT firewalls or routers The MultiVOIP supports inter operation with STUN servers and NATs SIP based environment only Management Configuration and system management can be done locally with the MultiVOIP configuration software After an IP address has been assigned locally other configuration can be done remotely using the MultiVOIP web browser GUI Remote system management can be done with the MultiVoipManager SNMP software or via the 14 MultiVOIP User Guide Overview MultiVOIP web browser GUI All of these control software packages are included on the Product CD While the web GUI s appearance differs slightly its content and organization a
143. bar menu select Connection Settings to access the COM Port Setup screen or use the keyboard shortcut Ctrl G Multi Tech Systems MultiVOIP SS Setup has detected the following Command Port s Enter the Command Part to be used COMI C COM2 Ali configured COM ports in the command PC will be displayed lt Back Cancel The COM port setting can be changed after installation in the COM Port Setup dialog box Connection COM Port Setup Connect Select Pot COM1 7 Baud Rate 115200 x Modem Setup 13200 Init String SB 19200801 Int Response OK Dial String Connect Response CONNECT Hangup String ATHO NOTE If there is a Dial String specified in Modem Setup Configuration programs wil try to initialize modem and dial this string NOTE If the COM port setting made here conflicts with the actual COM port resources available in the command PC this error message will appear when the MultiVOIP program is launched If this occurs you must reset the COM port Error in Opencomm handle 83 Software Installation MultiVOIP User Guide 8 Transient screens will flash by as files are being copied Then a completion screen will appear Multi Tech Systems InstallShield Wizard Complete Setup has finished installing Multi OIP SS on your computer Y Cancel Click Finish 9 When setup of the Multi VOIP software is complete you will be prompted to run the Multi
144. below 258 MultiVOIP User Guide E1 PhoneBook Configuration Add Edit Inbound Phone Book Field Definitions Field Name Values Description Accept Any Number Y N When checked Any Number appears as the value in the Remove Prefix field The Any Number feature of the Inbound Phone Book does not work when an external routing device is used Gatekeeper for H323 protocol Proxy for SIP protocol Registrar for SPP protocol When no external routing device is used If Any Number is selected calls received from phone numbers not matching a listed Prefix shown in the Remove Prefix column of the Inbound Phone Book will be admitted into the voip on the channel listed in the Channel Number field Any Number can be used in addition to one or more Prefixes 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 259 E1 Phonebook Configuration MultiVOIP User Guide Add Edit Inbound Phone Book Field Definitions cont d Field Name Values Description Channel 1 30 or E1 channel number to which Number Hunting the call will be assigned as it enters the local telephony equipment often a local PBX Hunting directs the call to any available channel Description
145. between the chassis and a metallic object that will provide an electrical ground 8 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 Proceed to the Software Installation chapter to load the MultiVOIP software 76 MultiVOIP User Guide Software Installation Chapter 4 Software Installation 77 Software Installation MultiVOIP User Guide Introduction Configuring software for your MultiVOIP entails three tasks 1 loading the software onto the PC this is Software Installation and is discussed in this chapter 2 setting values for telephony and IP parameters that will fit your system this is Technical Configuration and it is discussed in Chapter 5 and 3 establishing phonebooks that contain the various dialing patterns for VOIP calls made to different locations this is Phonebook Configuration and it is discussed in Chapter 6 for North American T1 telephony standards and in Chapter 7 for European E1 telephony standards 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 The MultiVOIP software and User Guide are contained on the MultiVOIP pr
146. bligation under this warranty shall be limited at MTS s option to repair or replacement of any products which prove to be defective within the warranty period or at MTS s option issuance of a refund of the purchase price Defective products must be returned by Customer to MTS s factory transportation prepaid MTS WILL NOT BE LIABLE FOR CONSEQUENTIAL DAMAGES AND UNDER NO CIRCUMSTANCES WILL ITS LIABILITY EXCEED THE PURCHASE PRICE FOR DEFECTIVE PRODUCTS Repair Procedures for U S and Canadian Customers In the event that service is required products may be shipped freight prepaid to our Mounds View Minnesota factory Multi Tech Systems Inc 2205 Woodale Drive Mounds View MN 55112 Attn Repairs Serial A Returned Materials Authorization RMA is not required Return shipping charges surface will be paid by MTS Please include inside the shipping box a description of the problem a return shipping address it must be a street address not a P O Box number your telephone number and if the product is out of warranty a check or purchase order for repair charges 371 Warranty Service amp Tech Support MultiVOIP User Guide For out of warranty repair charges go to www multitech com documents warranties Extended two year overnight replacement service agreements are available for selected products Please call MTS at 888 288 5470 extension 5308 or visit our web site at www multitech com pro
147. board operator Of course one DID line can handle only one call at atime 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 r Interface Interface Type DID DPO 7 m DID Options r Dialing Options Start Modes wink Start Inter Digit Timer fe secs Wink Timer po Message Waiting Indication Light Inter Digit Regeneration Timer fi 00 ms 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 Wink Start detects the off hook condition Delay Dial initiated by the telco central office call and becomes ready to receive dial digits immediately 147 Technical Configuration MultiVOIP User Guide DID Interface Parameter Definitions cont d Field Name Values Description DID Options cont d Start Modes Immediate Start Wink Start Delay Dial For Wink Start the voip detects the off hook condition Then the voip reverses battery
148. bound Phonebook Prefix to Prefix Description Destin Total Prefix to Prefix Description Remove to Add Incoming Calls Pattern Digits Remove to Add Outgoing Calls 91208 9 Outgoing calls to Santa Fe Boise Area area 7 7 i ncoming calls 3 digit calls to to extensions Santa Fe of company s employees PBX system extensions in Boise 200 to 240 Outgoing calls to Flagstaff area 3 digit calls to Flagstaff employees extensions 600 630 Santa Fe Voip Santa Fe Voip Inbound Phonebook Outbound Phonebook Prefix to Prefix Description Destin Total Prefix to Prefix IP Description Remove to Add Incoming Calls Pattern Digits Remove to Add Addr Outgoing Calls 91505 9 Incoming calls 91208 12 none none 204 Outgoing calls to PSTN 6 49 to Boise area Santa Fe local 73 calls 2 2 Incoming calls 7 3 none none 204 3 digit calls to to extensions 6 49 Boise of company s 73 employees PBX system extensions in Santa Fe 700 790 Outgoing calls to Flagstaff 3 digit calls to Flagstaff employees extensions 600 630 Flagstaff Voip Flagstaff Voip Inbound Phonebook Outbound Phonebook Prefix to Prefix Description Destin Total Prefix to Prefix IP Description Remove to Add Incoming Calls Pattern Digits Remove to Add Addr Outgoing Calls 91520 9 Incoming calls Outgoing calls to PSTN to Santa Fe Flagstaff local area calls 6 6 Incoming calls
149. btainable tone fast busy survivability tone re order tone Disconnect any tone from Currently chosen disconnection Tones Available Tones supervision tone list 142 MultiVOIP User Guide Technical Configuration E amp M Parameters The parameters applicable to the E amp M telephony interface type are shown in the figure below and described in the table that follows r Interface Interface Type E amp M E amp M Options m Dialing Options Inter Digit Timer 2 secs Message Waiting Indication Signal lr DialTone Wink Timer 250 ms lint z Type TYPE Il x Inter Digit Regeneration Timer fi oo ms Mode Wie wie Flash Hook Options Generation eoo ms No Response Timer 60 secs Detection Range Min 500 ms Max fi 000 ms I Disconnect on Call Progress Tone T Enable Pass Through 143 Technical Configuration MultiVOIP User Guide E amp M Interface Parameter Definitions Field Name Values Description Interface E amp M enables E amp M functionality Type Types 1 5 Refers to the type of E amp M interface being used 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 wink is required on the E lead or M lead in the call initiation or setup When Wink is se
150. c These codes are used when making non local calls They always precede the phone number that would be dialed when making a local call 47 QS Phonebook Tips MultiVOIP User Guide b access codes There are digits PSTN access codes that must be dialed to gain access to an operator to access the publicly switched long distance calling system North America to access the publicly switched national calling system Europe and elsewhere or to access the publicly switched international calling system worldwide There are digits PBX access codes that must be dialed by phones connected to PBX systems or key systems Often a 9 must be dialed ona PBX phone to gain access to the PSTN to get an outside line Sometimes 8 must be dialed on a PBX phone to divert calls onto a leased line or to a voip system However sometimes PBX systems are smart enough to route calls to a voip system without a special access code so that 9 might still be used for all calls outside of the building There are also digits special access codes that must be dialed to gain access to a particular discount long distance carrier or to some other closed or proprietary telephone system c local exchange numbers Within any calling area there will be many local exchange numbers A single exchange may be used for an entire small town In cities an exchange may be used for a particular neighborhood although exchange
151. cal B I calls l 2 2 Tncomin 7 3 none none 204 3 digit calls to PBX System to extensions 16 49 Boise Main Number of company s B employees th y extensions 444 3200 I in Santa Fe 700 790 I 91520 12 none none 204 Outgoing calls R I 16 49 to Flagstaff R 40 extensions 75 aea eee ee ee Il 6 3 none none 204 3 digit calls to 16 49 Flagstaff 75 employees extensions 600 630 rOle nmn Ai Flagstaff Voip Flagstaff Voip I Flagstaff Office I Inbound Phonebook Outbound Phonebook I Prefix to Prefix Description Destin Total Prefixto Prefix IP Description I Area 520 E l Remove to Add Incoming Calls Pattern Digits Remove t0Add Addr Outgoing Calls l 204 16 49 75 g 91520 9 Incoming calls 91505 12 none none 204 16 Outgoing calls I l to PSTN 49 74 to Santa Fe I 8 Channel l Flagstaff local area I Analog VoIP I calls MVP810 l 6 6 Incoming calls 2 3 none none 204 16 3 digit calls to l I to extensions 49 74 Santa Fe l I of company s employees l l PBX system extensions I PBX System in Flagstaff 200 240 I Main Number l 91208 12 none none 204 16 Outgoing calls j A Nee l 4973 to Boise area 7 3 none none 204 16 3 digit calls to I i Re I 49 73 Boise 30 extensions we l employees I I extensions es a i cao a 700 790 52 MultiVOIP User Guide QS Phonebook Example Sample Phonebooks Enlarged Boise Voip In
152. ce 104 free calls Bil feiaieetee teksten eaten 243 Thtsetig iss E E E 206 frequencies touch tone 142 Frequency 1 custom tone field 160 Frequency 1 tone pair scheme 156 158 Frequency 2 custom tone field 160 Frequency 2 tone pair scheme 156 158 frequency POWELL eeeeeeeeeeeeeeee 20 PREVI r n antn 117 From gateway statistics logs field a a ae Oa S A A Epa o EE EARS 301 front p nels soii is tssssveetteseesbecte 18 FTP client program s s s 351 FTP client program obtaining 353 FTP client programs graphic vs textual orientation 360 FTP file transfers using FTP client program 353 using web browsert s s s 353 FTP Server Enable Ethernet IP Parameters field c0 cc0 109 FTP Server function as added feature cee 351 Enabling arrati Resia 353 FTP Server contacting 0 355 FTP Server invoking download transfer using FTP client program 359 using web browser s s s 357 FTP Server logging in 356 FTP Server logging out 360 FTP transfers filetype Seisis iii 351 354 phonebooks 0 eee eee 351 Server location e ceseeereeeees 351 Index function tracing on off logging 172 FXO Disconnect On fields 141 FXO disconnection criteria 134 FXO disconnection triggering of 140 141 FXO Interface Parameter definitions EOE T EOE
153. ce you ve begun using the web browser GUI you can go back to the MultiVOIP Windows GUI at any time However you must log out of the web browser GUI before using the MultiVOIP Windows GUI 35 QS Phone IP Starter Config MultiVOIP User Guide Phone IP Starter Configuration continued 5 Go to Configuration Voice Fax Select Coder Automatic At the right hand side of the dialog box click OK If you know any specific parameter values that will apply to your system enter them Click Copy Channel Select Copy to All Click Copy At main Voice Fax Parameters screen click OK to exit from the dialog box 6 Enter telephone system information Go to Configuration Interface Enter parameters obtained from phone company or PBX administrator 7 Go to Configuration Regional Parameters Select the Country Region that fits your situation Click OK and confirm Click OK to exit from the dialog box 8 Go to Configuration Regional Parameters In the Country Selection for Built In Modem field drop down list select the country that best fits your situation This may not be the same as your selection for the Country Region field The selections in the Country Selection for Built In Modem field entail more detailed groupings of telephony parameters than do the Country Region values 9 Do you want the phone call logs produced by the MultiVOIP to be sent out by email to your Voip Administrator or someone else If NO ski
154. cribes the type of gateway as Alternate which the MultiVOIP is defined with respect to the gatekeeper Status registered The current status of the MultiVOIP not registered gateway with respect to the SIP proxy either registered or unregistered 319 Operation amp Maintenance MultiVOIP User Guide r SPP Registrars IP Address Pot Type 65 126 90 24 10000 Primary Registered 65 126 90 81 10000 Predef 1 Not Registered gt SPP Registrars Statistics Servers Field Definitions Field Values Description Name Column Headings IP Address n n n n The IP address of the gatekeeper for n 0 255 Port TDMA time slot used for communication between MultiVOIP unit and the gatekeeper that serves it Type Primary This field describes the type of gateway as Predefined which the Multi VOIP is defined with respect to the gatekeeper Status registered not The current status of the gateway either registered registered or unregistered 320 MultiVOIP User Guide Operation amp Maintenance About Packetization Time You can use the Packetization Time screen to specify definite packetization rates for coders selected in the Voice FAX Parameters screen in the Coder Options group of fields The Packetization Time screen is accessible under the Advanced options entry in the sidebar list of the main voip software screen In dealing with RTP paramete
155. d London to call anywhere in Amsterdam at local rates United Kingdom Wren Clothing Co VOIP PBX Site Amsterdam Wren Clothing Co a VOIP PBX Site London pe The Netherlands Wren Clothing Co VOIP PBX Site Paris AW 4 Calls at Amsterdam local rates Calls at Paris local rates France Local Calling Areas 245 E1 Phonebook Configuration MultiVOIP User Guide National Rate Calls Within Nation of Remote VOIP Site In the third use of the VOIP system the national calling area of each VOIP location becomes accessible to all of the VOIP system s users As a result international calls can be made at national calling rates Again significant savings are possible For example suppose that the Wren Clothing Company buys its buttons from the Chickadee Button Company in the Dutch city of Rotterdam In that case Wren Clothing personnel in both London and Paris could call the Chickadee Button Company without paying international long distance rates only Dutch national calling rates would be charged This applies to calls completed anywhere in The Netherlands United Kingdom The Netherlands Wren Clothing Co VOIP PBX Site n Amsterdam Wren Clothing Co VOIP PBX Site y K oricadee Button Co Rotterdam Wren Clothing Co VOIP PBX Site Paris lt _ Calls at Dutch National Rates France 246 MultiVOIP User Guide E1 PhoneBook Configur
156. de If you conduct a search for example on the word MultiVoip you will be directed to a list of firmware that can be downloaded a MuoltiVOIP Firmware Find all the firmware a for the MultiVOIP here 5 13 01 se If you choose Support you can select MultiVoip in the Product Support menu and then click on Firmware to find MultiVOIP resources Product Support MultiVOIP MultivOIP MultiFRAD Related Links Support MultiModemDSVD zi N ases Manuals are available on line A Firmware is available Product Product Tour 4 amp H 323 Upgrade Where to Buy Solutions Read the FAQs App Stories Software is available 328 MultiVOIP User Guide Operation amp Maintenance Once the updated firmware has been located it can be downloaded from the web ftp site using normal PC Windows procedures While the next 3 screens below pertain to the MVP3010 similar screens will appear for any MultiVOIP model described in this manual File Download x You have chosen to download a file from this location MVP3000x EXE from ftp multitech com What would you like to do with this file C Bun this program from its current location M Always ask before opening this type of file Cancel More Info 781 KB of M P301f EXE Copied e x g Saving MVP3000x EXE from ftp multitech com Estimated time left Not known Opened so far 781 KB Download to C VoipSys
157. de Operation amp Maintenance Chapter 8 Operation and Maintenance 283 Operation amp Maintenance MultiVOIP User Guide Operation and Maintenance Although most Operation and Maintenance functions of the software are in the Statistics group of screens an important summary appears in the System Information of the Configuration screen group Also the SIP Server Endpoint Statistics screen presents statistical information unique to the MVP SS MultiVOIP units SIP Server Endpoint Statistics screen This screen shows values previously entered in the Add Predefined Endpoint screen as well as various measures of the IP phone traffic that have occurred on each endpoint in the SIP system This is a screen whereupon settings may be read and performance data may be read However no parameter values are set on this screen Accessing Endpoint Statistics screens Pulldown Icon Sip Server Configuration Ctrl Alt 8 Predefined Endpoints Ctrl Alt 9 Endpoint Statistics Ctrl Alt 1 Logs History Ctrl Alt 0 Ms Shortcut Sidebar Sip Server Configuration Ctrl Alt 1 Predefined Endpoints Endpoint Statistics Logs History 284 MultiVOIP User Guide Operation amp Maintenance r Endpoint Statistics warehouse motorpool2 quardsh testlab4 No of Entries Registered with SS Registered with s55 Registered with R egistered with SS m Details Registration Ty
158. ded 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 6 08 where the X is the software number and the 08 is the version number of the MultiVOIP software on your local drive Of course the firmware file is named mvptlftp bin Important You cannot go back to 6 04 or earlier versions using FTP You must use upgradesoftware via the serial port Important These ftp upgrade instructions do not apply to software release 6 05 and above 350 MultiVOIP User Guide Operation amp Maintenance FTP Server File Transfers Downloads MultiTech has built an FTP server into the MultiVOIP unit Therefore file transfers from the controller PC to the voip unit can be done using an FTP client program or even using a browser e g Internet Explorer Netscape or FireFox used in conjunction with Windows Explorer 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 pro
159. down gt Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl Alt N Regional Parameters Ctrl R SMTP Parameters Ctrl Alt S Logs Traces Ctri Alt L Supplementary Services Ctri Alt H System Information Ctrl Alt SIP CallSignaling Ctrl Alt SFt P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt SFt Shortcut Sidebar E Configuration Ethernet IP Ctrl Alt Shft P Yoice Fax Interface SIP Call Signaling Regional 149 Technical Configuration MultiVOIP User Guide SIP Parameters Signaling Port 5060 IV Use SIP Proxy IV Allow Incoming Calls Through SIP Proxy Only r SIP Proxy Parameters Proxy Domain Name IPAddress Port Number Primary Proxy 5060 Alternate Proxy 1 5060 Alternate Proxy 2 5060 I Append SIP Proxy Domain Namein User ID UserName sda Password baa Re ReaistrationT ime 2600 secs Proxy Polling Interval jeo Ssi TTL Value feo secs OK Cancel Help The tables below describes all fields in the general SIP Call Signaling screen SIP Call Signaling Parameter Definitions Field Name Values Description SIP Proxy Parameters Signaling Port Port number on which the MultiVOIP UserAgent software module will be waiting for any incoming SIP requests Use SIP Proxy Y N Allows the MultiVOIP to work in conjunction with a
160. e 6 119 Compression Silence RADIUS Attributes cccceecceeseeeeteeeeees 192 Compression Silence SMTP logs P e E EEEE 167 computer requirements 0 004 19 Config Info Checklist Quick Start Instructions 28 configuration of voip local versus remote 89 90 Configuration option description MultiVOIP program menu 324 Configuration Parameter Groups ACCESSING E 101 Configuration Port Setup option description MultiVOIP program MMU foe vesca a Eee ART 324 configuration procedure local Getatled ineine 97 SUMMALY is sscsusishsstssssscssviecseeeaee 96 Configuration Version System Info sses 202 Configur ation Version System Information 0 289 configuration local ceee 92 configuration phonebook Blasi dines E 248 Divi shied ti he rin 207 configuration SaVING ecee 203 S EE eaves dvcesstets 341 configuration user default 204 Configuring MultiVOIP phonebooks general Bla E E Sides 242 Pl E E E axeecetea cs 206 conflicts COM portisson nsss 83 Connection Problems Solving 100 connectivity test Quick Start Instructions 56 Consecutive Packets Lost field 125 Console Message Settings Filters for E VE N E te oes ssh gus S EESE 172 console messages enabling 170 console parameters tracked 172 Contact Address MultiVOIP User Guide SIP Server
161. e offhook Answer Delay integer values When Answer Delay is enabled re order tone Timer in seconds this value determines when the Range 1 65535 FXO interface sends the connection notice Tone Detection Y N When selected call disconnection will be triggered by a tone sequence Available dial tone List from which tones can be Tones ring tone chosen to signal call answer busy tone unobtainable tone fast busy survivability tone Answer Tones any tone from Available Tones list Currently chosen call answer supervision tone 140 MultiVOIP User Guide Technical Configuration FXO Supervision Parameter Definitions Field Name Values Description 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 Current Loss is enabled X 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 in milliseconds Determines the interval after detection of current loss at wh
162. e Inbound PhoneBook of the MVP3010 is shown below Inbound Phone Book for MVP3010 Digital VOIP Site D Remove Add Channel Comments Prefix Prefix Number 0207 9 7 0 Allows phone users at remote voip sites Note 4 to call local numbers those within the Note 5 Site D area code 0207 Inner London over the VOIP network 0208 9 8 0 Allows phone users at remote voip sites Note 4 to call local numbers those in Outer Note 5 London over the VOIP network 0207 3 0 Allows phone users at remote voip sites 39883 to call extensions of the Site D PBX using three digits beginning with 3 Note 4 9 gives PBX station users access to outside line Note 5 The comma represents a one second pause the time required for the user to receive a dial tone on the outside line PSTN Commas can be used in the Inbound Phonebook but not in the Outbound Phonebook 273 E1 Phonebook Configuration MultiVOIP User Guide Outbound Phone Book for MVP410 Analog VOIP Site F Destin Remove Add IP Comment Pattern Prefix Prefix Address 201 200 2 9 7 To originate calls to Site A Birmingham 01189 0118 101 200 2 9 8 To originate calls Note 3 to any PSTN phone in Reading area using the FXO channel channel 1 of the Site B VOIP 102 200 2 9 8 To originate calls to phone connected to FXS port channel 2 of the Site B VOIP Reading 421 200 2 9 6 Calls to Site E Carlis
163. e Office Area 208 I PBX System Main Number 333 2700 FOB 2 90 extensions 204 16 49 73 24 Channel Digital VolP I I I I I I I I I I f MVP2410 Inbound Phonebook Each Inbound Phonebook contains two entries The first entry 4 digits specifies how incoming calls from the other voip sites will be handled if they go out onto the local PSTN Essentially all those calls come to the receiving voip with a pattern beginning with 1 area code The local voip removes those four digits because they aren t needed when dialing locally The local voip attaches a 9 at the beginning of the number to get an outsideline The PBX then completes the call to the PSTN The second Inbound Phonebook entry 1 digit is for receiving calls from company employees in the other two cities The out of town employee simply dials 3 digits The first of the three digits is uniquely used at each site and so acts as a destination pattern Boise extensions are 7xx Santa Fe extensions 2xx Flagstaff extensions 6xx The local voip sees the pattern in its inbound phone book and notes the first digit here either 2 5 or 6 To make the match this first digit 2 5 or 6is put in the Remove Prefix field This first digit must then be added back once again so that the voip will send all three digits to the PBX The PBX can then dial the specific extension identified by the three digit number
164. e 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 SIP field group select Use Proxy and specify the Transport Protocol to be used TCP or UDP Use the default SIP Port Number 5060 8 Click OK to exit from the Add Edit Outbound Phonebook screen 43 QS Phonebook Starter Config MultiVOIP User Guide 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 Euro National Call Long Distance Example Example Seattle Chicago system London Birming system Seattle is area 206 Chicago Inner London is 0207 area employees must dial 81 Birmingham employees must before dialing any Seattle dial 9 before dialing any number on the voip system London number on the voip Answer 1206 is prefix to be system removed
165. e installation Follow the instructions on the Install Shield screens InstallShield Wizard x 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 364 MultiVOIP User Guide Operation amp Maintenance During the installation you must specify which browser you ll use in the Select Browsers screen InstallShield Wizard Select Browsers istellStiela When installation is complete the Java program becomes accessible in your Start Programs menu Java resources are readily available via the web However the Java program runs automatically in the background as a plug in supporting the MultiVOIP web GUI No overt user actions are required 365 Operation amp Maintenance MultiVOIP User Guide thew Wier erer FA dan Setter T LzenUlh e LU currert 2 TAINAS a verns leses Lad hrir Aali 2 Tons Frito ay Lumens H Fisdy darase OIC Eh Setirg 3 Cries T me lor Wiican Lh Staap Gy Hisce acos R Miter Lxzel E EN e zaklat Gereh o Welcome to the Java 2 Platform Fe ecome wo the Java 2 Standard Editior Purim Evomris Thie zo7idee c nplete tuntime s gt sort fe Faraz applications The mrt tarinmi ine des t Dee Y Pisia andat whia rawta nimrir g ta dawal sau JEU Site WEY IUW IF References Res tar Dasr Trizin piarchi
166. e units support 4 for the MVP4105S 8 13 Overview MultiVOIP User Guide for the MVP810SS constitutes a practical limitation on their capacity to support PSTN access for other gateways Systems must be scaled to match required capacity by including additional MultiVOIP SS units Mounting Mechanically the MVP410SS and MVP810SS MultiVOIPs are designed for a one high industry standard EIA 19 inch rack enclosure The product must be installed by qualified service personnel in a restricted access area in accordance with Articles 110 16 10 17 and 110 18 of the National Electrical Code ANSI NFPA 70 Phone System Transparency 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 Voip Protocol The MVP SS units use the SIP protocol only SIP means Session Initi
167. ed 92 MultiVOIP User Guide Technical Configuration Write down the values for these IP parameters You will need to enter these values in the IP Parameters screen in the Configuration section of the MultiVOIP software You must have this IP information about every VOIP in the system Telephony Interface Parameters The following parameters must be known about the PBX or telco central office equipment to which the analog MultiVOIP will connect Phone Parameters Ask phone company or telecom manager Telephony Interface Parameters s9 Record for this VOIP Site e Which interface type is to be used E amp M FXS FXO DIP DPO e If FXS determine whether the line will be used for a phone fax or KTS key telephone system e If FXO determine if line will be an analog PBX extension or an analog line from a telco central office e If E amp M determine these aspects of the E amp M trunk line from the PBX e What is its Type 1 2 3 4 or 5 e Is it 2 wire or 4 wire e Is it Dial Tone or Wink 93 Technical Configuration MultiVOIP User Guide SMTP Parameters for email call log reporting VOIP call traffic are to be sent by email required if log reports of Optional SMTP Parameters Preparation Task 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
168. ee ee a ae a a cae Reminder Be sure to Save Setup after entering configuration values 28 MultiVOIP User Guide QS Gathering Phone IP Details Identify Remote VOIP Site to Call When you re done installing the MultiVOIP you ll want to confirm that it is configured and operating properly To do so it s good to have another voip that you can call for testing purposes You ll want to confirm end 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 MVP SS Unit s Role in SIP VOIP System The MVP210 SS 410 SS 810 SS unit always uses the SIP protocol However the MVP SS units are equipped to play an additional role in the voip system the role of a SIP server And as a SIP server the MVP SS unit can operate in either stand alone mode or SIP survivability mode Stand Alone Mode The MVP SS unit can function as a stand alone SIP server that controls the flow of phone traffic to lines connected to gateways that are registered with the MVP SS unit This stand alone capability allows the MVP SS to operate 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 SIP Survivability Mode The M
169. ee nimentaNen fre aed ifa irr eo ite gta i prudish Se tas Cave 2 Plates web ex foz r12 sZortnes 92 mthe Jara 3 Fiann Coptic 2203 Pav ITO raras Ex L Som Seon Boe Pak Ake Calas WLU A alagera My amp ier After the Java program has been installed you can access the MultiVOIP using the web browser GUI Close the MultiVOIP Windows GUI Start the web browser Enter the IP address of the MultiVOIP unit Enter a password when prompted A password is needed here only if password has been set for the local Windows GUI or for the MultiVOIP s FTP Server function See Setting a Password Web Browser GUI earlier in this chapter The web browser GUI offers essentially the same control over the voip as can be achieved using the Windows GUI As noted earlier logging functions cannot be handled via the web GUI 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 GUI than with the Windows GUI 366 MultiVOIP User Guide Operation amp Maintenance SysLog Server Functions MultiTech has built SysLog server functionality into the software of the MultiVOIP units SysLog is a de facto standard for logging events in network communication systems The SysLog Server resides in the MultiVOIP unit itself To implement this functionality you will need a SysLog client program sometimes referred to as a daemon Sy
170. eesesssesceseesceseeecsesaeeseesceeeeeeeeseeaes 331 Downloading Firmware csceseeseseeeeee Downloading Factory Defaults Downloading IFM Firtmwdre scssccscesccssesceseeseesecnseeecnseeseseeseesecasesecneeeeeaeeasees Setting and Downloading User Defaults ccccsccesceeeceeeeereeenseeeeeseeenseeneenseenaes 34 Setting a Password Windows GUI 0 sccsscceseeeseesseesseeseeeseeeeeeeeeneeeecnseenaeenaeenes 344 Setting a Password Web Browser GUI 0 scccsscceseceensecesneeensecesneeenseceeneeeeeeses 347 Un Installing the MultiVOIP Software 0 1cccsccesesseesseeseeeeeeeeeeeeeeeeseeeseeeeenaes 348 Uperading SOfEWATE ci cosoisreicts ced hoe eeecke te in poe etna coeds coset eat E a eaa S 350 FTP SERVER FILE TRANSFERS DOWNLOADS cc ssccesssseeceeseeeceseeeeeesneeeesenaeees 351 WEB BROWSER INTERFACE s ccescceescecesrcessceceseeeescecaceeeneeceseeeeneeceaceeenaeceeneeeneeeee 361 SYSLOG SERVER FUNCTIONS c cccsssssssccssseesssccensecnssccensesnsceosnsesnssesnsesnecesensers 367 CHAPTER 9 WARRANTY SERVICE AND TECH SUPPORT ss00008 370 LIMITED W ARRANTY senine eorn tan aR E ER REE NE E 371 REPAIR PROCEDURES FOR U S AND CANADIAN CUSTOMERS ccceseceeseeceseeeenees 371 TECHNICAL SUPPORT oot TELE eke R E E E antes vane cee ean ees 373 Contacting Technical SUPPOrt 1 cccccscesecesecnseesseeeeeeeeeeeeseceseceseensecnaecnaeenaeens 373 CHAPTER 10 REGULATORY INFORMATION ccsscssss
171. eld 164 password lost forgotten 344 347 Password setting 344 web browser GUIL cee eeeeeee 347 patents iirc ea ER meee 2 PBX characteristics variations in BY EE ET 280 Tl ausyn ei nS 240 PBX interaction ceccesseeeeereees 14 PC Settings Specs Quick Start Instructions 30 PC command COM port assignment detailed 83 personnel requirement for rack installation 0 67 to lift during installation 68 to lift unit during installation 62 Phone Book Version System Info sesser 202 System Information 0 289 Phone Number Voice FAX AutoCall Offhook Alert field 722 Phone Signaling Tones amp Cadences a T re e E E ees 153 phone IP details importance of writing down 92 Phone IP details gathering Quick Start Instructions 25 phone IP starter configuration Quick Start Instructions 34 phonebook FTP remote file transfers 351 MultiVOIP User Guide phonebook configuration 89 phonebook configuration remote 351 Phonebook Configuration icon BV E EEE 249 Ts hs E EE EE E E 208 Phonebook Configuration Procedure E rere e besten traits 248 g i KOE E TEE EES 207 Phonebook Configuration screen j A E E EEE 248 A U P E S ated E 207 phonebook entries coordinating BL a E E EEE 248 TL RE E ETRS 207 phonebook example Quick Start Instructions
172. elect Port field select a COM port that is available on the PC If no COM ports are currently available re allocate COM port resources in the computer s MS Windows operating system to make one available B Connection m COM Port Setup Ctrl G Connection Help sonnect ot Hs Disconnect Ctrl D Settings Ctrl G Connect Select Port com 7 Disconnect Settings gt Baud Rate 115200 z Modem Setup pim Init String MARA 5 619200401 Init Response Koo Diisi SS Connect Response JcONNECT assis Hangup Sting aT oo NOTE If there is a Dial String specified in Modem Setup Configuration programs will try to initialize modem and dial this string 100 MultiVOIP User Guide Technical Configuration 4B Fixing a Cabling Problem If the MultiVOIP cannot be located by the computer four error messages will appear saying MultiVOIP SS Not Found Phone Database Not Read SIP Endpoint Database Not Read and Password Phone Database Not Read Multivolp ss MultiVOIP SS PDD MultiVOIP SS Not Found Phone Database not Read MultiVOIP SS SIP Endpoint MultiVOIP SS PDD SIP Endpoint Database not Read Password Phone Database not Read In this case the MultiVOIP is simply disconnected from the network For instructions on MultiVOIP cable connections see the Cabling section of Chapter 3 5 Configuration Parameter Groups Getting Familiar
173. els Cancel Vv gg ss EES Help g 8 Hf S BH 8 8 8 we HB 8 8 ww He Bg 8 8 wf ff E a E E J 323 Operation amp Maintenance MultiVOIP User Guide MultiVoip Program Menu Items After the MultiVoip program is installed on the PC it can be launched from the Programs group of the Windows Start menu Start Programs MultiVOIP__ In this section we describe the software functions available on this menu len Programs Ay Acrobat Reader 5 0 z A FullShot93 gt _ Favorites gt RouteFinder Manager 7 25 gt N Daen gt Gun Wear HK ScreenSaver gt C MultivOIP 200 v2 51 gt Eh Settings gt _MultivOIP 100 v7 50C gt a md MultivOIP 100 v7 01C e Multvoip 100 v7 50 r BD Configuration 2 Help amp MultiVOIP 100 v7 50 gt a Configuration Port Setup ee MultivolP 800 v3 01E BD Date and Time Setup Bun C MultVOiP 100 v7 01E gt 23 Download Factory Defaults A MultiVOIP 100 v7 514 gt pA Download Firmware E amp PEN ele RouteFinder Manager 7 28 gt Z9 Download IFM Firmware RP Shut Down Fal ey ee 7 26 2 Download User Defaults A Paint Shop Pro gt a Set Password Aster A E a MuktivOIP 100 v7 51B gt 5 Uninstall GYExploring mvp24 REIT N Z5 Upgrade Software Several basic software functions are accessible from the MultiVoip software menu as shown below MultiVOIP Program Menu Menu Selection Description Configuration Select this to ent
174. ement Ctrl 2 inme Logs Details Alternate Servers Ctrl Alt 4 gt Shortcut Sidebar Statistics Call Progress Ctrl O Reports IP Statistics T1ZE1 Statistics 299 Operation amp Maintenance MultiVOIP User Guide The Logs Screen 300 MultiVOIP User Guide Operation amp Maintenance 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 column dd mm yyyy hh mm ss The starting time of the call event The date is presented as a day expression of one or two digits a month expression of one or two digits and a four digit year This is followed by a time of day expression presented as a two digit hour a two digit minute and a two digit seconds value statistics logs field Duration column hh mm ss This describes how long the call event lasted in hours minutes and seconds Type H 323 SIP or SPP Indicates the Call Signaling protocol used for the call H 323 SIP or SPP Status column success or Displays the status of the call i e failure whether the call was completed successfully or not IP Direction incoming Indicates whether the call is outgoing incoming or outgoing with respect to the gateway Mode column
175. ems MultiVOIP xxxx yyyy where x and y represent MultiVOIP model numbers and software version numbers 8A2 Drag and drop files from the local Windows browser e g Windows Explorer to the web browser Exploring Multi VOIP 2410 4 03 Aa Danah iniiai i PA ftp 77voip1 192 168 2 2007 Microsoft Intemet Explorer Fie Edit View Go Favorites Tools Help eae Fonverd Up Cut Copy dress C Program Files Multi Tech Systems MultiVOl x Name casfile cas factdef cnf H323 pdl myptl ftp bin a fxs_loopFtp cas icrosoft Office a h323 pdl Address ry tp M MultiTech System is InPhBk tmr into browser window Multi Tech Systen a Lhsstrs Mutivoir T a LogsMaster log A New file from PC E MultivOIP 24i D MultiVOIP 24 E MultivoIP 30i E MultivOIP 30 E MultivolP 5 91 MulVoIP 6 0 E MultiVoip3000 will overwrite old file Drag this a porate on Voip unit 357 Operation amp Maintenance MultiVOIP User Guide You may be asked to confirm the overwriting of files on the MultiVOIP Do so Confirm File Replace File transfer between PC and voip will look like transfer within voip directories 358 MultiVOIP User Guide Operation amp Maintenance 8B Download with FTP Client Program 8B1 In the local directory browser of the FTP client program locate the directory holding the MultiVOIP program files The default location will be C Program Files Multi Tech Systems
176. ems Inc is the Multi Tech logo Windows and NetMeeting are registered trademarks of Microsoft Multi Tech Systems Inc 2205 Woodale Drive Mounds View Minnesota 55112 763 785 3500 or 800 328 9717 U S Fax 763 785 9874 Technical Support 800 972 2439 http www multitech com CONTENTS CHAPTER 1 OVERVIEW scsccssssscssscscescssccscescsssssssssnessessessesssssescsssnessssneseeses 7 ABOUT THIS MANUAL eaaa a EAEE a ara EE e EESE E EAEE EEE E E EEEE TERRE 8 INTRODUCTION TO ANALOG MULTIVOIPS WITH SIP SURVIVABILITY FEATURES MVP 210SS 410SS 810SS oo eeeeeescecsseceeeeceseeeeneeceeeeeaeceeeeenaeceeeeenaeceneeenaeeeee 12 MultiVOIP Front Panel LEDS cccsccessseceseceenceenseceeneecsecesneececeeaeeceeeeeaeecseeeens 17 COMPUTER REQUIREMENTS cssssssssececeessssecesececeeseaececececsessaseeeeeceesensasaeeeeeceenenaees 19 SPECIFICATIONS oa ius csuctcevesvec ity stedaecteske odesuesid ousesvcodtesbusnaws evs E O ctytatetee 20 INSTALLATION AT A GLANCE aai ire codussatebcenteviedlesbadscotestcelepsnsiacedessdodgysiodeses 21 RELATED DOCUMENTATION inre ir a n AE O a Eia 21 CHAPTER 2 QUICK START INSTRUCTIONS sessesesececsesoroesesececoosorsesesecoeeesoe 22 INTRODUCTION PAE E EEE EEES EEE OE 23 MULTIVOIP STARTUP TASKS ccessssvececteecsntenssseesstsesseestesevenesheveneseesesvedsnessaecehsesstere 24 Phone IP Details Absolutely Needed Before Starting the Installation 25 Gather TP Tinformation s 4
177. en 253 E1 Phonebook Configuration MultiVOIP User Guide Add Edit Outbound Phone Book Field Definitions Field Name Values Description Destination prefixes Defines the beginning of Pattern area codes dialing sequences for calls exchanges that will be connected to line another VOIP in the system numbers Numbers beginning with extensions these sequences are diverted from the PTSN and carried on Internet or other IP network Total Digits as needed number of digits the phone user must dial to reach specified destination 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 nn n n the IP address to which the for 0 255 call will be directed if it begins with the destination pattern given Description alpha Describes the facility or numeric geographical location at which the call will be completed 254 MultiVOIP User Guide E1 PhoneBook Configuration Add Edit Outbound Phone Book Field Definitions cont d Field Name Values Description SIP Fields Use Proxy Y N Select if proxy server is used Transport TCP or Voip administrator must choose Protocol UDP between UDP and TCP transmission protocols UDP is a high speed low overhead connectionless protocol where data is transmitted
178. ent function Invoked by keypad sequence 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 Invoked by keypad sequence Call Name Identification When enabled for a given voip unit the home voip this feature gives notice to remote voips involved in calls Notification goes to the remote voip administrator not to individual phone stations When the home voip is the caller a plain English descriptor will be sent to the remote callee voip identifying 174 MultiVOIP User Guide Technical Configuration 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 paramete
179. er 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 Ethernet Channel Ope 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 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 connection exists While link is up this LED will flash off to indicate data activity ration 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 XSG Transmit Signal This indicator lights when the FXS configured channel is off hook the FXO 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 RSG Receive Signal This indicator lights when the FXS configured channel is ringing the FXO configured channel
180. er Guide Technical Configuration This screen presents vital system information at a glance Its primary use is in troubleshooting r System Information m Version Information Boot Version 2 03b Firmware Version 3 08 0H Configuration Version 3 08 09 03 Phone Book Version 4 04 IFM Version 5 BS MAC Address 000800510356 Uptime 00 04 10 03 Hardware ID MYP410 32M RevB F918 System Information Parameter Definitions Field Name Values Description Boot nn nn Indicates the version of the code that Version is used at the startup booting of the voip The boot code version is independent of the software version Firmware alpha Indicates version of MultiVOIP Version numeric firmware 201 Technical Configuration MultiVOIP User Guide System Information Parameter Definitions cont d Field Name Values Description Configur nn nn nn Indicates version of MultiVOIP ation nn Configuration software which Version alpha includes screens for IP Parameters numeric SMTP Parameters Regional Parameters etc Phone Book numeric Indicates the version of the inbound Version and outbound phonebook portion of the MultiVOIP software IFM Version numeric Indicates the version of the firmware running on the MultiVOIP s Interface Module which is its analog telephony hardware Mac alpha Denotes the number assigned as the Address num
181. er 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 324 MultiVOIP User Guide Operation amp Maintenance MultiVOIP Program Menu cont d Menu Selection Description Download Factory Defaults Select this to return the configuration parameters to the original factory values Download Firmware Select this to download new versions of firmware as enhancements become available Download IFM Firmware Select this to download new versions of IFM firmware as enhancements become available The Interface Module IFM is the telephony interface for analog MultiVOIP units MVP130 MVP130EFXS MVP210 MVP410 MVP810 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 using 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 GUI web browser GUI amp FTP server
182. er two types of calls in the three city system described above 1 calls originating from the Miami office and terminating in the New York Manhattan office and 2 calls originating from the Miami office and terminating in New York City but off the company s premises in an adjacent area code an area code different than the company s office but still a local call from that office e g Staten Island The first type of call requires an entry in the Outbound PhoneBook of the Miami VOIP and a coordinated entry in the Inbound phonebook of the New York VOIP These entries would allow the Miami caller to dial the New York office as if its phones were extensions on the Miami PBX The second type of call similarly requires an entry in the Outbound PhoneBook of the Miami VOIP and a coordinated entry in the Inbound Phonebook of the New York VOIP However these entries will be longer and more complicated Any Miami call to New York City local numbers will be sent through the VOIP system rather than through the regular toll public phone system PSTN But the phonebook entries can be arranged so that the VOIP system is transparent to the Miami user such that even though that Miami user dials the New York City local number just as they would through the public phone system that call will still be completed through the VOIP system This PhoneBook Configuration procedure is brief but it is followed by an example case For many people the example ca
183. ered Gateway Details 2c 2ieiactanisiaidcdes 316 IP Address to Ping Link Management field 312 IP Address From Details RADIUS Attributes field eee 192 IP Address From Details SMTP logs field arrossit 167 400 MultiVOIP User Guide IP address SysLog Server 171 IP Address To Details RADIUS Attributes field 192 IP Address To Details SMTP logs field eesriie etso rtis i 167 IP Addresses acceptable for registration field SIP Server Configuration parameters 0 eee 196 IP Call Direction call progress field E EE EREE 293 IP Call Type call progress field 293 IP datagram and DiffServ 108 IP Direction statistics logs field 301 IP Mask field 106 IP Statistics field IP Address iiris essesi 308 IP Statistics field definitions 308 309 IP Statistics fields SA EE T E A ea citiece 308 Received RTCP Packets 310 Received RTP Packets 310 Received TCP Packets 309 Received Total Packets 308 Received UDP Packets 309 Received with errors RTCP Packets lt 2 s sssiescsciecsataitenieds 310 Received with errors RTP Packets Udine welediteae Ns Ri aicoee eigen 310 Received with errors TCP Packets E EA EA EEE 309 Received with errors Total Packets lt lt s cess iccdekesbeeceieis 309 Received with errors UDP Packets s si cecsedeciccstsicaieedeteis 309 Transmitted RTCP Packets 310 Transmitted
184. eric voip unit s unique Ethernet address Up Time days Indicates how long the voip has been hours running since its last booting mm ss Hardware alpha Indicates the version of the ID numeric MultiVOIP unit s circuit board and components 202 MultiVOIP User Guide Technical Configuration The frequency with which the System Information screen is updated is determined by a setting in the Logs screen 23 Saving the MultiVOIP Configuration When values have been set for all of the MultiVOIP s various operating parameters click on Save Setup in the sidebar E Configuration H Phone Book onnection elp C l H 203 Technical Configuration MultiVOIP User Guide 24 Creating a User Default Configuration When a Setup complete grouping of parameters is being saved you will be prompted about designating that setup as a User Default setup A User Default setup may be useful as a baseline of site specific values to which you can easily revert Establishing a User Default Setup is optional Mult OIP __ will be brought down OK Cancel Help 204 MultiVOIP User Guide T1 Phonebook Configuration Chapter 6 T1 Phonebook Configuration North American Telephony Standards 205 T1 Phonebook Configuration MultiVOIP User Guide T1 Versus E1 Telephony Environments Phonebooks for Series II analog MultiVOIP units MVP130 MVP130FXS MVP210 MVP410 and MV
185. ermines whether the MVP SS needs to 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 195 Technical Configuration MultiVOIP User Guide SIP Server Configuration Parameter Definitions Field Name Values Description Registrar Options Allow Y N If undefined registrations are allowed Undefined value Y then gateways other than Registrations those listed in the PreDefined Endpoints list can register with the MVP SS voip unit as it functions in its SIP server mode If undefined registrations are allowed then incoming registrations will be allowed if they originate from endpoints at accepted domains or accepted IP addresses specified below in this software screen Accept any Determines whether registrations to Registrations domains the MVP SS SIP server will be for specific accepted from any domain or only domains from specified domains Multiple domains can be listed separated by semicolons The any domains option
186. ers may occur serially however Although MultiTech 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 Examples here show use of both programs 4 Enable FTP Functionality Go to the IP Parameters screen and click on the FTP Server Enable box m Ethernet IP Parameters m Ethernet Parameters JV Packet Prioritization 802 1p Frame Type TYPE I hd 802 1p Parameters Priority Call Control 3 Excellent Effort Y VoIP Media BVoice h Cancel Others O Best Effort z Help VLAN ID 1 r IP Parameters Gateway Name MultolP I Enable DHCP Diff Serv Parameters 192 a L E Call Control PHB faa VoIP Media PHB 46 IP Mask 255 255 255 O FTP Server IV Enable i Gateway I Enable SRY DNS Server IP Address k x IP Address 353 Operation amp Maintenance MultiVOIP User Guide 5 Identify Files to be Updated Determine which files you want to update Six
187. es output level 117 Fax Volume field 0 ccceeeeee 117 FCC Declaration 0cccccceeeeeeee 375 FCC Part 68 Telecom rules 376 FCC registration number 377 FCC rules Part 15 0 0 375 FDX LED a a R 18 Filters Console Message Settings 172 Filters button Console Message S C t PS ees Gres n 171 firmware upgrade implementing 331 Firmware Version System Information 4 289 Firmware Version System Info 201 firmware version identifying 331 firmware downloading 332 firmware obtaining updated 327 Flash Hook Options fields E amp M senean 146 FXO e nia 135 forgotten password 344 347 Forward Condition Call Fwdg Eleran haw bie iia 260 Thc Si nae eve Ale 220 Forward Destination Inbound PhBk Bl E EE EEE 261 Thrais es iian a 220 Forward Error Correction call progress field oo tees 298 Forward Error Correction RADIUS Attributes iivecedsctossecesteescesdese 192 Forward Error Correction SMTP JOBS arinrin ioeie 167 Forward Error Correction field 719 forward on busy Ts aneanc is phani 220 260 397 Index Forward upon No Response l N E EE E 260 A a EEE E E 220 forwarding dual conditions E1 busy amp no response 4 260 forwarding dual conditions busy amp no response Di EEE E ehaeiea 220 frame relay and fax ceeee 117 Frame Type field ce
188. es 142 Tone Pair custom field 160 tones signaling s s s 153 Index Total Digits outbound field 2 eee eee ee AE ee eee 254 A U KEE E otespesote seed EES 213 touch tone frequencies 142 trace on off logging eee 172 Transfer Sequence 174 176 Transmitted RTCP Packets IP Stats field aoran eneee 310 Transmitted RTP Packets IP Stats field eee ener er ee 310 Transmitted TCP Packets IP Stats field ierit 309 Transmitted Total Packets IP Stats field EREE 308 Transmitted UDP Packets IP Stats field irean in taia 309 Transport Protocol SIP field BL ARR ATS 255 TED ERE ERE TSS 214 triggering log report email 164 troubleshooting Quick Start Instructions 60 Troubleshooting Resolutions for Multi VOIPS ooeec 8 TTL Value SIP Call Signaling field pian EEEE aca ees 152 Turn Off Logs field eee 171 Type call statistics logs field 301 Type E amp M type field 144 Type H 323 Gatekeepers Statistics Servers field ceeceesseeeeees 318 Type of tone Regional Parameters field miiie eee 156 Type SIP Proxies Statistics Servers field inasahan ia 319 Type SPP Registrars Statistics Servers field cceceeeseeeees 320 Type of Service IP header field amp DiffServ sissies yen aori 108 U interface ISDN BRD description sssr 384 UDP TCP compared A R E E E E 255 IP Statistics context 307 308 DP E 21
189. es Description Password alpha Login password for MultiVOIP numeric unit s email account Mail Server IP n n n n This is the mail server s IP address Address for n 0 to This mail server must be accessible 255 on 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 reports will be sent Subject text User specified Subject line that will appear for all emailed log reports for this MultiVOIP unit Reply To email address User specified This email address Address functions as a source email identifier error has occurred email address receive log reports Mail Criteria 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 User specified Email address at which VOIP administrator will Criteria for sending log summary by email The log summary email will be sent out either when the user specified number of log messages has accumulated or once every day or multiple days which ever 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
190. es are defined exactly that is without making any adjacent phone numbers unreachable through the voip system 231 T1 Phonebook Configuration MultiVOIP User Guide Outbound Phone Book for MVP2410 Digital VOIP Site D Destin Remove Add IP Comment Pattern Prefix Prefix Address 201 200 2 9 7 To originate calls to Site A Bismarck 1507 1507 101 200 2 9 8 To originate calls Note 3 to Rochester local PSTN using the FXO channel channel 1 of the Site B VOIP 102 200 2 9 8 To originate calls to phone connected to FXS port channel 2 of the Site B VOIP 421 200 2 9 6 Calls to Site E Cheyenne 1402 200 2 9 5 Calls to Lincoln area local PSTN via FXO channel CH4 of the Site F VOIP 1402 200 2 9 5 Calls to extensions 263 thirty of key 740 system at Site F 1402 200 2 9 5 Lincoln Human 263 operator or auto 741 attendant is 1402 200 2 9 5 needed to 263 complete these 742 calls Note 3 The pound sign is a delimiter separating the VOIP number from the standard telephony phone number 232 MultiVOIP User Guide T1 PhoneBook Configuration Inbound Phonebook for MVP2410 Digital VOIP Site D Remove Add Channel Comment Prefix Prefix Number 1615 9 0 Allows phone users at remote Note 4 voip sites to call non toll Note 5 numbers within the Site D area code 615 Pierre SD over t
191. esiess ar 214 REC 3489 20h earn neire 184 REGATA a aean ieaie 107 RC 25 97 casata neea 107 REC2833 eeccssbistetiastetns 166 296 302 RE C3246 cies sssscnsitststussensapentesins 107 REC768 320 0205 citi oi Sil 386 REC793 sccsscecniniai ei nai 386 ring cadences custoM 6 161 Ring Count field FXS Loop Start ee 130 Ring Count forwarding condition Blithe diy iii sie arlens 261 A a EE Gini aes aied 221 TING tone CUSCOM ee eeeeeeeeeeeee 160 TING CONES ooo eeeeceseeeeeeeceeeeeeeeeeeee 159 Round Trip Delay Link Management field 313 Round Trip Delay field Bl A E E Raabe 257 Pi sane kickin m 216 RSG LED ices sscnsitieshssvesseiotiaheth 18 RTP packetization ranges amp INCFEMENS s es 322 RTP Parameters screen 322 Safety Recommendations for Rack Installations see eeeeeeeeeeeeeees 67 Safety Warnings esseere 62 Safety Warnings Telecom 62 Index Save Setup command 203 saving configuration cece 203 USED N EEE OEE 341 Saving the MultiVOIP Configuration er eee ee 203 savings on toll calls j a AE csi E E 242 TM EEE E E E 206 Select All RADIUS Attributes field PEE E EE EEE 190 Select All SMTP logs field 165 Select Attributes RADIUS button EE E E E T 189 Select Channel field 0 000 115 Select Channel Supplementary Services field ceeeeseeteeeee 176 Selected Coder field 00 00 118 Server Address RADIUS screen f
192. esponse 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 Phone number or IP address to which calls Destination will be directed IP address For SIP calls the Forward Destination can be phone number one of the following port number a phone number b IP address c IP address port number d phone number IP addr port number e SIP URL or f phone IP address etc 220 MultiVOIP User Guide T1 PhoneBook Configuration Add Edit Inbound Phone Book Field Definitions cont d Field Name Values and Description Ring Count 0 1 2 3 etc When No Response is condition for forwarding calls this determines how many unanswered rings are needed to trigger the forwarding Registration In a SIP voip system gateways can register Option with the SIP Proxy Parameters 5 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
193. etails section describe various SIP parameters m Outbound Phone Book Destination Pattern IP Address Protocol _ Description Ss Alternal Any Number SIP voip remotel Number of Entries 1 m Details Remove Prefix Add Prefix SIP Pott Edi Delete Transport Protocol SIP URL Close Round Trip Delay 300 ms Help Click Add 210 MultiVOIP User Guide T1 PhoneBook Configuration 2 The Add Edit Outbound PhoneBook screen appears m Add Edit Outbound Phone Book r Phone Number Details I Accept Any Number Destination Pattern 365 Total Digits jo Cancel Remove Prefix Add Pes ss lt s IP Address 2 Description Help Advanced Transport Protocol l TCP C UDP SIP Port Number 5060 SIP URL Enter Outbound PhoneBook data for your MultiVOIP unit Note that the Advanced button gives access to the Alternate IP Routing feature if needed Alternate IP Routing can be implemented in a secondary screen as described after the primary screen field definitions below 211 T1 Phonebook Configuration MultiVOIP User Guide The fields of the Add Edit Outbound Phone Book screen are described in the table below Add Edit Outbound Phone Book Field Definitions Field Name Values Description Accept Any Y N When checked Any Number Number appears as the value in the Destination Pattern field The A
194. figuration The controller PC and the MultiVOIP unit itself must both be connected to the same IP network and their IP addresses must be known Once you ve begun using the web browser GUI you can go back to the MultiVOIP Windows GUI at any time However you must log out of the web browser GUI before using the MultiVOIP Windows GUI Logging of System Events MultiTech has built SysLog Server functionality into the software of the MultiVOIP units SysLog is a de facto standard for logging events in network communication systems m Logs IV Enable Console Messages Logs I Turn Off Logs GUI 2 maaana AATE Tia Par M SysLog Server M Enable IP Address Seena Port 514 dan fee oo a Aes wee mang ean eene en Online Statistics Updation Interval 5 Sec 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 among other firms See www kiwisyslog com SysLog client programs essentially give you a means of structuring console messages for convenience and ease of use MultiTech Systems does not endorse any particular SysLog client program SysLog client programs by any qualified provider should suffice for use with MultiVOIP units Kiwi s brief description of their SysLog program indica
195. finitions Field Values Description Name Alternate n n n n Alternate destination for outbound data traffic IP where in case of excessive delay in data transmission Address n 0 255 Round milliseconds The Round Trip Delay is the criterion for Trip judging when a data pathway is considered Delay blocked When the delay exceeds the threshold specified here the 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 voice packets which the MultiVOIP interprets as a failure of the IP network Upon detecting an excessive delay in transmission of voice packets overly high latency in the network the MultiVOIP diverts the call to another IP address which itself is connected to the PSTN for example via an FXO port on the self same MultiVOIP could be connected to the PSTN Call completed Call diverts to P Alt IP address in voip PSTN Line via PSTN accessing PSTN line FXO X vor Qed or PBX AA IP network fails Call originates PSTN Failover Feature The MultiVOIP can be programmed to divert calls to the PSTN temporarily in case the IP network fai
196. fo 202 System Information 289 Hold Sequence 174 177 hold caller on musical jingle for eee 176 hookup diagram quick 210 32 hookup diagram quick 410 810 31 TANA ine a no thee atiea 386 399 Index icons phonebook Bl ES 249 Ti e E EE 208 identifying current firmware version David Sted R A Eaa ESE r EERS 331 IFM interface module analog voips only description cece 337 IFM firmware downloading 337 338 IFM Version System Info sses 202 System Information 0 0 289 implementing firmware upgrade 331 in band DTMF 116 Inbound Phonebook Entries List icon BD AETS 249 A N E EREE es SS 208 Inbound Phonebook entries list BI i saeeastiees ni kutana 257 A N EEEE EA 217 inbound vs outbound phonebooks l A EE ARE 247 A A REA TEAT 206 Industry Canada requirements 376 info sources IPidetailS ei nanai 92 SMTP details eee eeeeeeeeeeeee 94 telephony interface details 93 voip email account 0 0 eee 94 Initiated Call Count SIP Server Endpoint Statistics Parameters 286 Input Gain field eee 115 installation AITPLOW ssevstee ies adeno Aiko 67 in a nutshell oo eee cree 21 INLACK osteo eet aie ai ay 66 log reports by email 94 software detailed 00 000 78 voip email account 00 0 eee 94 installation prerequisites 92 93 installation mechanical 14 installing Java vis a v
197. formation 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 378 MultiVOIP User Guide Appendix A Cable Pinouts Appendix A Cable Pinouts 379 Cable Pinouts MultiVOIP User Guide Appendix A Cable Pinouts Command Cable RJ 45 Connector Ethernet Connector End to End Pin Info RJ 45 DB9F PIN NO PIN NO CLEARTOSEND To DTE TRANSMIT DATA Device Receive pata 9 PC To Command Port Connector SIGNAL GROUND RJ 45 connector plugs into Command Port of Multi VOIP DB 9 connector plugs into serial port of command PC which runs MultiVOIP configuration software The functions of the individual conductors of the MultiVOIP s Ethernet port are shown on a pin by pin basis below RJ 45 Ethernet Connector Pin Circuit Signal Name 1 TD Data Transmit Positive C2230 TS 2 TD Data Transmit Negative 3 RD Data Receive Positive 6 RD Data Receive Negative 380 MultiVOIP User Guide Cable Pinouts T1 E1 Connector T1 E1 Connector e gt Receive Pair from line Transmit Pair to line Voice Fax Channel Connectors Pin Functions E amp M Interface Pin Descr Function 1 M Input 2 E Output 3 TI 4 Wire Output 4 4 Wire Input 2 Wire Inp
198. g Factory Defaults 1 The MultiVoip Configuration program must be off when invoking the Download Factory Defaults command If it is on the command will not work 2 To invoke the Download Factory Defaults command go to Start Programs MVP x xx Download Factory Defaults Download Factory Defaults oO P d z Multi OIP 335 Operation amp Maintenance MultiVOIP User Guide 3 If a password has been established the Password Verification screen will appear Password Verification Type in the password and click OK 4 The MVP___ Firmware screen appears saying MultiVOIP model number is up Reboot to Download Firmware Multi OIP Firmware oR cancel _ Click OK to download the factory defaults The Boot LED on the MultiVOIP will light up and remain lit during the file transfer process 336 MultiVOIP User Guide Operation amp Maintenance 5 After the PC gets a response from the MultiVOIP the Dialog IP Parameters screen will appear Dialog IP Parameters r Diff Serv Parameters 134 Call Control PHB Frame Type TYPE I SF E VolP Media PHB m IP Parameters I Enable DHCP _ IP Address 192 e E 149 Cancel IP Mask 20n 200 2 0 Gateway 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
199. g for MVP 410SS 810SS 3 Connect the Multi VOIP 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 3 8 4 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 5 For an FXS or FXO connection FXS Examples analog phone fax machine Key Telephone System 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 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 amp M connector on the back of the MultiVOIP 71 Mechanical Installation amp Cabling MultiVOIP User Guide 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 pinout For a DID connection 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 sensiti
200. gher frequency of pair 156 MultiVOIP User Guide Technical Configuration Regional Parameter Definitions cont d Field Name Values Description Standard Tones fields cont d Gain 1 gain in dB Amplification factor of lower 3dB to 31dB frequency of pair and mute This applies to the dial ring busy setting and unobtainable tones that the MultiVOIP outputs as audio to the FXS FXS or E amp M port Default 16dB Gain 2 gain in dB Amplification factor of higher 3dB to 31dB frequency of pair and mute This applies to the dial ring busy setting and unobtainable fast busy tones 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 msec On Off four integer time used to denote phone ringing values in phone busy connection milli seconds unobtainable fast busy dial tone zero value for 0 indicates continuous tone dial tone survivability and re order Default indicates values differ for different continuous tone 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 Click on the Custom
201. gistrars Statistics Servers field cceeseeeseeereees 320 Port field Registered Gateway Details ccccssecesscessseseneeeeees 316 Port field SysLog Server 171 Port Number SIP Server Endpoint Statistics Parameters cceeeeesseceeeee 286 Port Number proxy server field 151 Port Number SMTP field 164 power CONSUMPTION ee eeeeeeeeee 20 power frequency eee 20 Power LED neni cieee nienie 18 Prefix Matched call progress field e arses a E a aa 296 Prefix Matched RADIUS Attributes field operne neiaa 191 Prefix Matched SMTP logs field 166 prerequisites for technical configuration 92 Primary Proxy SIP Call Signaling field iss cits canvest ash heesoueed 151 Priority H 323 Gatekeepers Statistics Servers field 318 Priority Levels 802 1p 104 105 product CD eee eeeeeeeeeees 21 use in software installation 78 Program Menu items 0006 324 Protocol Type outbound phonebook Index TDs E E 213 Proxy Domain Name IP Address field o eeann ei 151 Proxy Polling Interval SIP Call Signaling field oe 152 PSTN failover feature Alternate Routing and 216 quality Of ServiCe ceeeseereereees 14 quick hookup diagram 210 Quick Start Instructions 32 quick hookup diagram 410 810 Quick Start Instructions 31 Quick Start Instructions config info checklist 0 0 28
202. gita PBX X e 7003 LU 200 2 9 7 7002 325 7001 223 T1 Phonebook Configuration MultiVOIP User Guide The screen below shows Outbound PhoneBook entries for the VOIP located in the company s Baltimore facility 200 002 010 003 St Paul 200 002 010 003 Minneapolis N Suburbs 200 002 010 003 Minneapolis S Suburbs The entries in the Minneapolis VOIP s Inbound PhoneBook match the Outbound PhoneBook entries of the Baltimore VOIP as shown below 224 MultiVOIP User Guide T1 PhoneBook Configuration To call the Minneapolis St Paul area a Baltimore employee must dial eleven digits In this case we are assuming that the Baltimore PBX does not require an 8 or 9 to seize an outside phone line If a Baltimore employee dials any phone number in the 612 area code the call will automatically be handled by the company s voip system Upon receiving such a call the Minneapolis voip will remove the digits 1612 But before the suburban Minneapolis voip can complete the call to the PSTN of the Minneapolis local calling area it must dial 9 to get an outside line from the PBX and then a comma which denotes a pause to get a PSTN dial tone and then the 10 digit phone number which includes the area code 612 for the city of Minneapolis which is different than the area code of the suburb where the PBX is actually located 763 A similar sequence of events occurs when the Baltimore employee calls
203. gram 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 point The phonebook downloading feature greatly reduces the data entry required to establish inbound and outbound phonebooks for the voip units within a system Although each MultiVOIP unit will require some unique phonebook entries most will be common to the entire voip system After the phonebooks for the first few voip units have been compiled phonebooks for additional voips become much simpler you copy the common material by downloading and then do data entry for the few phonebook items that are unique to that particular voip unit or voip site 351 Operation amp Maintenance MultiVOIP User Guide To transfer files using the FTP server functionality in the MultiVOIP follow these directions 1 Establish Network Connection and IP Addresses Both the controller PC and the MultiVOIP unit s must be connected to the same IP network An IP address must be assigned for each IP Add
204. grams orc for details on rates and coverages Please direct your questions regarding technical matters product configuration verification that the product is defective etc to our Technical Support department at 800 972 2439 or email tsupport multitech com Please direct your questions regarding repair expediting receiving shipping billing etc to our Repair Accounting department at 800 328 9717 or 763 717 5631 or email mtsrepair multitech com Repairs for damages caused by lightning storms water power surges incorrect installation physical abuse or used caused damages are billed on a time plus materials basis 372 MultiVOIP User Guide Warranty Service amp Tech Support Technical Support Multi Tech Systems has an excellent staff of technical support personnel available to help you get the most out of your Multi Tech product If you have any questions about the operation of this unit or experience difficulty during installation you can contact Tech Support via the following Contacting Technical Support Country By E mail By telephone France support multitech fr 33 1 64 61 09 81 India support 91 124 340778 multitechindia com U K support 44 118 959 7774 multitech co uk US amp tsupport 800 972 2439 Canada multitech com Rest of support 763 785 3500 World multitech com Internet http www multitech com _forms email_tech_support htm
205. has taken the line off hook or the E lead is active on the E amp M configured channel 18 MultiVOIP User Guide Overview 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 19 Overview MultiVOIP User Guide Specifications Parameter MVP410SS MVP810SS MVP210SS Model Operating 100 240 VAC 100 240 VAC External Voltage 1 2 0 6 A 1 2 0 6 A transformer Current 3A 5V Mains 50 60 Hz 50 60 Hz 50 60 Hz Frequencies Power 29 watts 46 watts 19 watts Consumption Mechanical 1 75 H x 1 75 Hx 6 2 Wx Dimensions 17 4 Wx 174 Wx 9 Dx 8 5 D 8 5 D 14 H 4 5cm H x 4 5cm H x 15 8cm W x 44 2 cm W x 44 2 cm W x 22 9cm D x 21 6 cm D 21 6 cm D 3 6cm H Weight 7 1 lbs 7 7 lbs 1 8lbs 82kg 3 2 kg 3 5 kg 2 6lbs 1 17kg with transformer 20 MultiVOIP User Guide Overview Installation at a Glance The basic steps of installing your MultiVOIP network involve unpacking the units connecting the cables and configuring
206. he VOIP network 1615 31 0 Allows voip calls directly to 49231 employees at Site D at extensions x3101 to x3199 Note 4 9 gives PBX station users access to outside line Note 5 The comma represents a one second pause the time required for the user to receive a dial tone on the outside line PSTN The comma is only allowed in the Inbound phonebook 233 T1 Phonebook Configuration MultiVOIP User Guide Outbound Phone Book for MVP410 Analog VOIP Site F Destin Remove Add IP Comment Pattern Prefix Prefix Address 201 200 2 9 7 To originate calls to Site A Bismarck 1507 1507 101 200 2 9 8 To originate calls Note 3 to any PSTN phone in Rochester area using the FXO channel channel 1 of the Site B VOIP 102 200 2 9 8 To originate calls to phone connected to FXS port channel 2 of the Site B VOIP Rochester 421 200 2 9 6 Calls to Site E Cheyenne 1615 200 2 9 9 Calls to Pierre area PSTN via Site D PBX 31 1615 200 2 9 9 Calls to Pierre PBX 492 extensions with four digits Note 3 The pound sign is a delimiter separating the VOIP number from the standard telephony phone number 234 MultiVOIP User Guide T1 PhoneBook Configuration Inbound Phonebook for MVP410 Analog VOIP Site F Remove Add Channel Comment Prefix Prefix Number 1402 4 Access to Lincoln local PSTN by users at
207. he setting MultiVOIP applies to outbound tones entering the MultiVOIP at the input port Default 16dB Gain 2 gain in dB Amplification factor of higher 3dB to 31dB frequency of pair This figure and mute describes amplification that the setting MultiVOIP applies to outbound tones entering the MultiVOIP at the input port Default 16dB 160 MultiVOIP User Guide Technical Configuration 161 Custom Tone Pair Settings Definitions Field Name Values Description Cadence 1 integer time On off pattern of tone durations value in used to denote phone ringing milli seconds phone busy dial tone 0 zero value for indicates continuous tone dial tone survivability and re order indicates Cadence 1 is duration of first continuous tone period of tone being on in the cadence of the telephony signal which could be ring tone busy tone unobtainable tone or dial tone Cadence 2 duration in Cadence 2 is duration of first milliseconds off period in signaling cadence Cadence 3 duration in Cadence 3 is duration of second milliseconds on period in signaling cadence Cadence 4 duration in Cadence 4 is duration of second milliseconds off period in the signaling cadence after which the 4 part cadence pattern of the telephony signal repeats Technical Configuration MultiVOIP User Guide 14 Set SMTP Parameters Log Reports by Email The SMTP Parameters scree
208. he Site B VOIP Reading 01822 01822 200 2 9 5 Calls to Tavistock area PSTN via FXO channel of the Site F VOIP 0182 200 2 9 5 Calls to Tavistock 26374 key system operator or auto attendant 0207 0207 200 2 9 9 Calls to London area PSTN via Site D PBX 8 0207 200 2 9 9 Calls to London 398 PBX extensions with four digits Note 3 The pound sign is a delimiter separating the VOIP number from the standard telephony phone number 276 MultiVOIP User Guide E1 PhoneBook Configuration Inbound Phonebook for MVP210 Analog VOIP Site E Remove Add Channel Comment Prefix Prefix Number 421 1 Call Completion Summaries Site A calling Site C Method 1 1 Dial 101 2 Hear dial tone from Site B 3 Dial 9435632 4 Await completion Talk Site A calling Site C Method 2 5 Dial 10149435632 6 Await completion Talk Note Some analog VOIP gateways will allow completion by Method 2 Others will not Site C calling Site A 1 Dial 9436161 2 Hear dial tone from Site B VOIP 3 Dial 201 4 Await completion Talk 277 E1 Phonebook Configuration MultiVOIP User Guide Site D calling Site C 1 Dial 901189435632 2 9 gets outside line On some PBXs an 8 may be used to direct calls to the VOIP while 9 directs calls to the PSTN However some PBX units can be programmed to identify the destination patte
209. her voip brands H 450 may be implemented differently and then the message presentation may vary 178 MultiVOIP User Guide Technical Configuration Supplementary Services Definitions cont d Field Name Values Description Calling If the home voip unit is originating Party the call and Calling Party is selected Allowed then the identifier from the Caller Id Name Type field will be sent to the remote voip CNI unit being called The Caller Id field gives the remote voip administrator a plain language identifier of the party that is originating the call occurring on a specific channel This field is applicable only when the home voip unit is originating the call Example Suppose a voip system has offices in both Denver and Omaha In the Omaha voip unit the home voip in this example Call Name Identification has been enabled Calling Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field When channel 2 of the Omaha voip is used to make a call to any other voip phone station for example the Denver office the message Calling Party Omaha Sales Office Voipchannel 2 will appear in the Caller Id field of the Statistics Call Progress screen of the Denver voip Technical Configuration MultiVOIP User Guide Supplementary Services Definitions cont d Fie
210. hrough a connection between the Command port of the MultiVOIP and the COM port of the computer the MultiVOIP configuration program is used Remote configuration is done through a connection between the MultiVOIP s Ethernet network port and a computer connected to the same network The computer could be miles or continents away from the MultiVOIP itself There are two ways of doing remote configuration and operation of the MultiVOIP unit 1 using the MultiVoipManager SNMP program or 2 using the MultiVOIP web browser interface program MultiVoipManager MultiVoipManager is an SNMP agent program Simple Network Management Protocol that extends the capabilities of the MultiVOIP configuration program MultiVoipManager allows the user to manage any number of VOIPs on a network whereas the MultiVOIP configuration program can manage only the VOIP to which it is directly locally connected The MultiVoipManager can configure multiple VOIPs simultaneously whereas the MultiVOIP configuration program can configure only one at a time MultiVoipManager may but does not need to reside on the same PC as the MultiVOIP configuration program The MultiVoipManager program is on the MultiVOIP Product CD Updates when applicable may be posted at on the MultiTech FTP site To download go to ftp ftp multitech com MultiVoip Web Browser Interface The MultiVOIP web browser GUI gives access to the same commands and configuration parameters a
211. iVOIP User Guide 3 Confirm Connection If the MultiVOIP is set for an available COM port and is correctly cabled to the PC the MultiVOIP main screen will appear If the main screen appears grayed out and seems inaccessible go to step 4 oe Phone Book E Statistics 98 MultiVOIP User Guide Technical Configuration In the lower left corner of the screen the connection status of the MultiVOIP will be displayed The messages in the lower left corner will change as detection occurs The message MultiVOIP Found confirms that the MultiVOIP is in contact with the MultiVOIP configuration program Skip to step 5 Looking for Response from MultiVolP Please Wait C E E E E e E E E E E E Reading Configuration 99090900000909000 Reading Configuration 1999999090900 090900 Multi olP found 1999909090 90990909909 99 Technical Configuration 4 Solving Common Connection Problems A Fixing a COM Port Problem If the MultiVOIP main screen appears but is grayed out and seems inaccessible the COM port that was specified for its communication with the PC is unavailable and must be changed An error message will appear MultiVOIP User Guide Multi OIP COH B Error in Opencomm handle To change the COM port setting use the COM Port Setup dialog box which is accessible via the keyboard shortcut Ctrl G or by going to the Connection pull down menu and choosing Settings In the S
212. iVOIP 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 162 MultiVOIP User Guide Technical Configuration The SMTP Parameters screen is shown below SMTP Parameters V Enable SMTP Password Mail Server IPAddress Port Number Text e Type Subject Reply To Address Login Name MuttivolP Recipient Address MultivolP multitech com Cancel Help Select Fields Mail Now m Mail Criteria Number of Records J Number of Days SMTP Parameters Definitions Field Name Enable SMTP Values Y N Description In order to send log reports by email this box must be checked However to enable SMTP functionality you must also select SMTP in the Logs screen Requires Authentication Y N If this checkbox is checked the MultiVOIP will send Authentication information to the SMTP server The authentication information indicates whether or not the email sender has permission to use the SMTP server Login Name alpha numeric per email domain This is the User Name for the MultiVOIP unit s email account 163 Technical Configuration MultiVOIP User Guide SMTP Parameters Definitions cont d Recipient Address Field Name Valu
213. ich the call will be disconnected Silence Y N Enables disables silence Detection detection method of Enable supervising call disconnection Silence One Way or Disconnection to be triggered Detection Type Two Way by silence in one direction only Silence Timer in seconds integer value 141 or in both directions simultaneously Duration of silence required to trigger disconnection Technical Configuration MultiVOIP User Guide FXO Supervision Parameter Definitions 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 03 852Hz 0 D 941Hz High Tones 1209Hz 1336Hz 1447Hz 1633Hz Disconnect 1st tone pair These are DTMF tone pairs Tone Sequence 4 2nd tone pair Values for first tone pair are 0 1 9 and A D Values for second tone pair are none 0 1 9 A D and The tone pairs 1 9 0 and are the standard DTMF pairs found on phone sets The tone pairs A D are extended DTMF tones which are used for various PBX functions 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 Tones ring tone chosen to signal call busy tone disconnection uno
214. ich case the user will be required to correct the interference at his own expense This device complies with Part 15 of the FCC rules Operation is subject to the following two conditions 1 This device may not cause harmful interference 2 This device must accept any interference that may cause undesired operation 375 Regulatory Information MultiVOIP User Guide 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 FCC Part 68 Telecom 1 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 2 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 3 An FCC compliant telephone cord and modular plug is provided with this equipment This equipment is designed to be connected to the telephone network
215. ield enerig areeni iisas 189 Server Details call progress field 296 Server Details RADIUS Attributes STE Ke E ETET 191 Server Details SMTP logs field 167 Server Details statistics logs field E AET TETA 303 Service Records eeeeeeeeeeees 110 Set Baud Rate oo eee 193 Set Log Reporting Method 169 Set Password program menu option command sesser 344 Set Password web browser GUI command 000 eee ee eee este eee tees 347 Set Password option description MultiVOIP program menu 325 Set Regional Parameters 153 Set SMTP Parameters 0 00 162 Set Supplementary Services Parameters 000 0 eeeeeeeseeeeeeee 173 Set Telephony Interface Parameters AeA Minh a e ao Ea s 126 Set Voice FAX Parameters 112 setting Ethernet IP parameters 102 setting password 344 web browser GUI ee 347 setting RTP Parameters 322 setting user defaults eee 341 Setup SAVING oo eee eeeseeeecseeeeeeeeee 203 USE EERE E EEE EA 341 408 MultiVOIP User Guide setup saving user values 341 Shared Secret RADIUS screen field E EENE E 189 Signal type E amp M field 144 signaling cadences 153 signaling parameters 126 Signaling Port SIP Call Signaling field ee eee ere eee 150 Signaling tones s src 153 signaling types MVP210 SS oseeeseessesesesesreesreees 75 MVP 410SS 810SS
216. ify 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 version 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 In the illustration below the firmware version is 4 00a made for the E1 MultiVOIP MVP3010 Ee MultiVOlP 3000 4 004 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 331 Operation amp Maintenance MultiVOIP User Guide Download Firmware transfers the firmware including the H 323 protocol stack in the PC s MultiVOIP directory into the nonvolatile flash memory of the MultiVOIP Download Factory Defaults sets all configuration parameters to the standard default values that are loaded at the MultiTech factory Upgrade Software implements both the Download Firmware command and the Download Factory Defaults command Downloading Firmware 1 The MultiVoip Configuration program must be off when invoking the
217. il illustration 168 Logs screen definitions 170 Logs screen field definitions 171 Logs screen parameters Enable Console Messages 171 Filters ss vionin iia nes 171 GUL asst ate 171 IP Address SysLog Server 171 Online Statistics Updation Interval SGniia EE E EEES 171 Port SysLog Server 0 0 0 171 SMT Pernet meane ain 171 SNMP oia yah 171 SysLog Server Enable 171 Turn Off Logs oseese 171 logs Screen ACCESSING eee 169 long distance call savings Bib ch tat iokiiee obs R 242 ADD E E 206 lost packets consecutive 125 lost password sssseeseeeseeeeeeee 344 347 Mac Address System Info eee 202 289 402 MultiVOIP User Guide mail criteria SMTP records 164 Mail Server IP Address SMTP field aa EE ENE AAKA ETAL SE SSS 164 Mail Type SMTP logs field 164 Mains frequency 20 Max bandwidth coder 118 Max Baud Rate field 04 116 Max Expiry Time SIP Server Endpoint Statistics Patametei Sissies 285 Maximum Jitter Value field 124 Message Waiting Indication DID DPO ehe et a aria aiik 148 Message Waiting Indication E amp M and DiD irse iii ereat 145 Message Waiting Indication field DID DPO ricine 148 E amp M aeeie as 145 FXO heepra niea 134 FXS Loop Start ee 130 Minimum Jitter Value field 123 Mode call progress field 293 Mode Fa
218. inbound phonebook outbound phonebook InPhBk tmr OutPhBk tmr This file updates the inbound phonebook in the MultiVOIP unit This file updates the outbound phonebook in the MultiVOIP unit 354 MultiVOIP User Guide Operation amp Maintenance 6 Contact MultiVOIP FTP Server You must make contact with the FTP Server in the voip using either a web browser or FTP client program Enter the IP address of the MultiVOIP s FTP Server If you are using a browser the address must be preceded by ftp otherwise you ll reach the web GUI within the MultiVOIP unit A ftp 192 168 2 2007 Microsoft Internet Explorer 4 Back mA Fie Edit View Go F ait lt Bttp 192 168 2 200 355 Operation amp Maintenance MultiVOIP User Guide 7 Log In Use the User Name and password established in item 2 above The login screens will differ depending on whether the FTP file transfer is to be done with a web browser see first screen below or with an FTP client program see second screen below 356 MultiVOIP User Guide Operation amp Maintenance 8 Invoke Download Downloading can be done with a web browser or with an FTP client program 8A Download with Web Browser 8A1 In the local Windows browser locate the directory holding the MultiVOIP program files The default location will be C Program Files Multi Tech Syst
219. ion 148 DID Interface Parameters 147 DID jumper MVPZ210 SS 0 0 ceeccceeesseeceeneeee 73 MVP 410SS 810SS 0 0 70 DID lines MVP210 SS polarity sensitivity and 76 DID lines MVP 410SS 810SS polarity sensitivity and 72 DID DPO Interface Parameter definitions oc eeeeeeeee ee eeeeee 147 DID DPO Interface Parameter fields Inter Digit Timer dialing 148 Start Modes eecececeeeeeeeeeee 147 Wink Timet 147 DID DPO Parameter fields MultiVOIP User Guide Inter Digit Regeneration Timer dialing eeeeeeeseeeseeeeeeeee 148 DID DPO vs DID DPT 147 DiffServ and IP datagram 108 DiffServ PHB Per Hop Behavior VALU sirp reii rokit 107 digital voip product family 9 dimensions ceesseceeeceeseeceteeeeeeees 20 Disconnect on Call Progress Tone E amp M field oo eee eececeseeeeees 144 Disconnect Reason SMTP logs field EOE 167 Disconnect Reason statistics logs PENG EENE E E A E 302 Disconnect Tone Sequence FXO IET e E AE E 142 Disconnect Tones FXO disconnection supervision 142 disconnection criteria FXO 134 141 DNS Server IP Address Ethernet IP Parameters field c c00 109 Domain Names acceptable for registration field SIP Server Configuration PALAMELETS 5 60 s seseceessessveseeess 196 Download Factory Defaults program menu option command 335 Download Factory
220. ion can be handled through the web browser GUI as well see the Operation and Maintenance chapter of this manual In most aspects of configuration the Windows GUI and web browser GUI differ only graphically not functionally For information on SNMP remote configuration and management see the MultiVoipManager documentation Pre Requisites To complete the configuration of the ZS MultiVOIP unit you must know several things about the overall system Before configuring your MultiVOIP Gateway unit you must know the values for several IP and telephone parameters that describe the IP network system and telephony system PBX or telco central office equipment with which the digital MultiVOIP will interact If you plan to receive log reports on phone traffic by email SMTP you must arrange to have an email address assigned to the VOIP unit on the email server on your IP network A summary of this configuration information appears on page 58 Config Info CheckList IP Parameters The following parameters must be known about the network LAN WAN Internet etc to which the MultiVOIP will connect Ask your computer network administrator IP Network Parameters oS M Record for each VOIP Site in System Info needed to operate all MultiVOIP models e IP Address e IP Mask e Gateway e Domain Name Server DNS Info e If SIP protocol is used determine whether or not 802 1p Packet Prioritization will be us
221. ion attempt is abandoned 134 MultiVOIP User Guide Technical Configuration FXO Interface Parameter Definitions cont d Field Name Values Description Flash Hook Options fields Generation 50 1500 Length of flash hook that will milliseconds be generated and sent out when the remote end initiates a flash hook and it is regenerated locally Default 600 ms Detection Not applicable to FXO Range Caller ID fields Caller ID Type Bellcore The Multi VOIP 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 Y N Caller ID information is a enable description of the remote calling party received by the called party The description has three parts name of caller phone number of caller and time of call The time of call portion is always generated by the receiving MultiVOIP unit on FXS channel based on its date and time setup The forms of the Caller Name and Caller Phone Number differ depending on the IP transmission protocol used H 323 SIP or SPP and upon entries in the phonebook screens of the remote CID generating voip unit The CID Name and Number appearing on the phone at the terminating FXS end will come either from a central office switch showing a PSTN phone number or the phonebook of the remote CID sending voip u
222. ionality in STUN the MultiVOIP STUN Simple Traversal of UDP through NATs Network Address Translation is a protocol that allows a server to assist client gateways behind a NAT firewall or router with their packet routing Name IP n n n n IP address of the STUN server Server 0 255 Port numeric The data port TDM time slot at Server default which STUN info will be transmitted NAT STUN 3478 and received Keep Alive 60 3600 The interval at which the STUN client Timers in sends indicator Keep Alive NAT STUN seconds packets to the STUN server to determine whether or not the STUN server is available 186 MultiVOIP User Guide Technical Configuration 18 Set RADIUS parameters In general RADIUS is concerned with authentication authorization and accounting The MultiVOIP SS supports the authentication functions In the Attributes secondary screen accessed by clicking on Select Attributes the voip administrator can select the parameters to be tallied by the RADIUS server Accessing RADIUS Parameters Pulldown Icon gt Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl Alt N Regional Parameters Ctrl R SMTP Parameters Ctrl Alt S Logs Traces Ctrl Alt L Supplementary Services Ctrl Alt H System Information Ctr Alt SIP CallSignaling Ctrl Alt SFt P
223. ip email account eee 163 Voip Media PHB field 107 VoIP Media Priority Ethernet IP parameters field eee 105 voip software host PO 8 avi skies 19 90 voip system example conceptual E1 calls to remote PSTN 244 foreign calls national rates 246 Voip site to voip Site eee 243 voip system example digital amp analog with phonebook details EE E EEE 269 Tl E E EET 228 voip system example digital only with phonebook details Bl EE E E E S 262 Tlirectecirnme Gd ean 222 Index voltage operating eseese 20 warnings safety seee 62 Warranty escscecscecicesevecccsonssvsesecsdeceses 371 web browser GUI and logs 170 web browser GUI enabling 111 analo t kenir see ier 35 web browser interface browser version requirement 361 365 generalise aese 361 Java requirement eee 361 prerequisite local assigning of IP addressees te esyes 362 video useability eee 361 web GUI Java aN sienien irass 363 remote control configuration and E RA 363 web GUI vs local Windows GUI COMPAFISON sesse serete 362 414 MultiVOIP User Guide web GUI logging and 363 WELSNE sisi ec disieuiatat dendbecctn 20 weight loading AD eTe AE E E 67 weight of unit lifting precaution 0 0 cee 62 personnel requirement 62 Well Known Ports eee 386 well known port number SMTP r E S 164 well known port SIP I a E
224. irmware H Broadband Manager 7 26 gt Z5 Download H 323 PDL Paint Shop Pro gt Z5 Download User Defaults H MultiVOIP 100 v7 51B fa Mult OIP 3 You will be prompted to confirm that you want to establish a password which will entail rebooting the MultiVOIP which is done automatically Multi OIP Password x Multi OIP is Up Reboot to set password Click OK to proceed with establishing a password 344 MultiVOIP User Guide Operation amp Maintenance 4 The Password screen will appear If you intend to use the FTP Server function that is built into the MultiVOIP enter a user name A User Name is not needed to access the local Windows GUL the web browser GUI 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 MultiTech Technical Support for advice Password Click OK 5 A message will appear indicating that a password has been set successfully Hult OIP Password aK __ cancel _ After the password has been set successfully the MultiVOIP will re boot itself and in so doing its BOOT LED will light up 345 Operation amp Maintenance MultiVOIP User Guide 6 After the password has been set the user
225. is web GUI 364 integrated phone data networks 242 Inter Digit Regeneration Time E amp M sonete areant 145 PROS etek saatsin eae 134 FXS Loop Start ee 130 Inter Digit Timer dialing field Index DID DPO we esc esses ceseeseees 148 E amp M oret aane etsy 145 EXO earranan ise 134 FXS Loop Start 130 Intercept Tone Regional Params and Offhook Alert Voice Fax Params eo r e trata OERE EE 155 Intercept Tone and required Interface amp Voice Fax settings 0 0 0 0 155 Interface field DID DPO 147 Interface field E amp M 00 144 interface parameters accessing 126 interface parameters setting 126 interface types BRI SD EAE ERAT 384 U 384 inter office dialing Bid E E SATEET 243 TD E E ASS 207 inter operation analog with TI E1 VOIpS cesses 12 inter operation with phone system 14 Helde a ere Mee en oct 304 IP Address caller statistics logs Piel aeina ee ok 304 IP Address Ethernet IP Parameters field v2 cccsinininwaGinae ae 106 IP Address H 323 Gatekeepers Statistics Servers field 318 IP Address IP Statistics field 308 IP Address outbound phonebook BD EE ied Geshe eee 254 TI E E cide dle 213 IP Address ping target Link Management field 313 IP Address SIP Proxies Statistics Servers field asinis ies ts 319 IP Address SPP Registrars Statistics Servers field ceeseesseeeeees 320 IP Address field Regist
226. issco tetas caterers 105 Voice Coder call progress field 293 Voice coder statistics logs field 302 voice delay seere 123 124 Voice Gain field eee eeeeeeee 115 voice packets recovering lost corrupted 119 MultiVOIP User Guide voice packets consecutive lost 125 voice packets delayed 123 124 voice packets re assembling 117 voice quality improving 119 voice quality versus delay 124 Voice FAX connector pinott 381 Voice FAX Parameter definitions 124 125 Voice FAX Parameter Definitions115 116 117 118 119 123 Voice FAX Parameter fields AutoCall Offhook Alert 120 121 AutoCall Offhook Alert fields 120 121 Generate Local Dial Tone 121 Offhook Alert Timer 122 Out of Band Mode DTMF 115 Phone Number Auto Call Offhook Alertan eretia 122 Voice FAX Parameter fields Copy Channel eee 115 Default cc ccecccecssceceseeeees 115 DTMF Gain ee eeeececeeeeeeee 115 DTMF Gain High Tones 115 DTMF Gain Low Tones 115 DTMF In Out of Band 115 Duration DTMF 115 Input Gain eee eeeeeeeeeeee 115 Output Gain oseese 115 Select Channel ccee 115 VOICE GiM moraa en 115 Voice FAX Parameter fields Fax Enable aeeeeeeee 116 Voice FAX Parameter fields Max Baud Rate Fax 116 Voice FAX Parame
227. ith Phonebooks Boise Office Area 208 PBX System Main Number 333 2700 204 16 49 73 24 Channel Digital VoIP MVP2410 MultiVOIP User Guide Boise Voip Boise Voip Inbound Phonebook Outbound Phonebook Prefix to Prefix Description Destin Total Prefixto Prefix IP Description Remove to Add Incoming Calls pattern Digits Remove to Add Addr Outgoing Calls 91208 Incoming calls 91505 12 none none 204 Outgoing calls to PSTN 16 49 to Santa Fe Boise Area he area 7 7 incoming calls 2 3 none none 204 3 digit calls to to extensions 16 49 Santa Fe of company s m4 employees PBX s extensions in Boise 200 to 240 91520 12 none none 204 Outgoing calls 16 49 to Flagstaff 75 area 6 3 none none 204 3 digit calls to 16 49 Flagstaff 15 employees extensions 600 630 A I Santa Fe Office Area 505 I Santa Fe Voip Santa Fe Voip I Inbound Phonebook Outbound Phonebook 204 16 49 74 I Prefix to Prefix Description Destin Total Prefixto Prefix IP Description l Remove to Add Incoming Calls ff Pattern Digits Remove toAdd Addr Outgoing Calls 8 Channel I 91505 9 Incoming calls 91208 12 none none 204 Outgoing calls Analog VoIP I to PSTN 16 49 to Boise area MVP810 l Santa Fe lo
228. iting light on a PBX extension phone Mode codes 53 PBX extension gt turns message light on 53 PBX extension gt 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 milliseconds The length of time between the Regeneration outputting of DTMF digits Timer Default 100 ms 145 Technical Configuration MultiVOIP User Guide E amp M Interface Parameter Definitions cont d Field Name Values Description Dialing Options cont d Flash Hook Options fields Generation integer values in Length of flash hook that will milliseconds be generated and sent out when the remote end initiates a flash hook and it is regenerated locally Default 600 ms Detection for Min and Max For a received flash hook to be Range 50 1500 regarded as such by the milliseconds MultiVOIP its duration must fall between the minimum and maximum values given here 146 MultiVOIP User Guide Technical Configuration DID Parameters The parameters applicable to the Direct Inward Dial DID telephony interface type are shown in the figure below and described in the table that follows The DID interface allows one phone line to direct incoming calls to any one of several extensions without a switch
229. king on the 169 Technical Configuration MultiVOIP User Guide Filters button and using the Console Messages Filter Settings screen see subsequent page If you use the logging function select the logging option that applies to your VoIP system design If you intend to use a SysLog Server program for logging click in that Enable check box The common SysLog logical port number is 514 If you intend to use the MultiVOIP web browser GUI for configuration and control of MultiVOIP units be aware that the web browser GUI does not support logs directly However when the web browser GUI is used log files can still be sent to the voip administrator via email which requires activating the SMTP logging option in this screen r Logs r Console message Settings JV Enable Console Messages Filters Cancel m Logs Hel I Tum Off Logs e GUI SMTP SysLog Server l Enable Server IP address Port Number Online Statistics Updation Interval fi 0 Sec 170 MultiVOIP User Guide Technical Configuration Logs Screen Definitions Field Name Values Description Enable Y N Allows MultiVOIP debugging messages to be Console read via a basic terminal program like Messages HyperTerminal or equivalent Normally this should be disabled because it uses MultiVOIP processing resources Console messages are meant for tech support personnel Filters
230. l Alt L Supplementary Services Ctri Alt H System Information Ctrl alt SIP CallSignaling Ctrl Alt SFt P RADIUS Ctrl alt U NAT Traversal Ctrl Alt SFt Shortcut Sidebar Configuration Ethernet IP Ctrl Alt N Yoice Fax Interface SIP Call Signaling 126 MultiVOIP User Guide Technical Configuration In each field enter the values that fit your particular network Channeli Z FXS Loop Start ZI Wink Stat Ei E Delay Dial E 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 We present here the various parameters grouped and organized by interface type 127 Technical Configuration MultiVOIP User Guide Note that Interface parameters are applied on a channel by channel basis However once you have established a set of Interface parameters for a particular channel you can apply this entire set of Voice FAX parameters to another channel by using the Copy Channel button and its dialog box To copy a set of Interface parameters to all channels select Copy to All and click Copy Interface Parameters Select Channel Channel 1 OK Cancel Default Help Supervision Copy Channel r Copy Channel Copy Channel 1 Interface Parameters to T Copy to All r Channels Bn 6 r Channels nH st 8 G r Channels EuS E DES E ng 69 S EES i 128
231. l Installation amp Cabling MultiVOIP User Guide 19 Inch Rack Enclosure Mounting Procedure Attaching the MultiVOIP to a rack rail of an EIA 19 inch rack enclosure will certainly require two persons Essentially the technicians must attach the brackets to the MultiVOIP chassis with the screws provided as shown in Figure 3 4 and then secure unit to rack rails by the brackets as shown in Figure 3 5 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 Figure 3 4 Bracket Attachment for Rack Mounting MVP410SS amp MVP810SS beee Figure 3 5 Attaching MultiVOIP to Rack Rail MVP410 SS amp MVP810 SS 68 MultiVOIP User Guide Mechanical Installation amp Cabling Cabling Procedure for MVP 410SS 810SS Cabling involves connecting the MultiVOIP to your LAN and telephone equipment 1 For DID channels only If all channels of y
232. ld Name Values Description Alerting If the home voip unit is receiving the Party call and Alerting Party is selected Allowed then the identifier from the Caller Id Name Type field will tell the originating remote CNI voip unit that the call is ringing This field is applicable only when the home voip unit is receiving the call Example Suppose a voip system has offices in both Denver and Omaha In the Omaha voip unit the home voip unit in this example Call Name Identification has been enabled Alerting Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field of the Supplementary Services screen When channel 2 of the Omaha voip receives a call from any other voip phone station for example the Denver office the message Alerting Party Omaha Sales Office Voipchannel 2 will be sent back and will appear in the Caller Id field of the Statistics Call Progress screen of the Denver voip This confirms to the Denver voip that the phone is ringing in Omaha 180 MultiVOIP User Guide Technical Configuration Supplementary Services Definitions cont d Field Name Values Description Busy Party If the home voip unit is receiving a Allowed call directed toward an already Name Type engaged channel or phone station and CNI Busy Party is selected then the identifier from
233. le 0207 200 2 9 9 Calls to Inner London area PSTN via Site D PBX 0208 200 2 9 9 Calls to Inner London area PSTN via Site D PBX 3 0207 200 2 9 9 Calls to Inner 398 London PBX 8 extensions with three digits Note 3 The pound sign is a delimiter separating the VOIP number from the standard telephony phone number 274 MultiVOIP User Guide E1 PhoneBook Configuration Inbound Phonebook for MVP410 Analog VOIP Site F Remove Add Channel Comment Prefix Prefix Number 01822 2 4 Calls to Tavistock local PSTN through FXO port Port 4 at Site F 0182 740 0 Gives remote voip users access 263 to extensions of key phone 740 system atTavistock office 0182 741 0 Because call is completed at key 263 system abbreviated dialing 3 741 digits is not workable 0182 742 0 Human operator or auto 263 attendant is needed to 742 complete these calls 275 E1 Phonebook Configuration MultiVOIP User Guide Outbound Phone Book for MVP210 Analog VOIP Site E Destin Remove Add IP Comment Pattern Prefix Prefix Address 201 200 2 9 7 To originate calls to Site A Birmingham 01189 0118 101 200 2 9 8 To originate calls Note 3 to any PSTN phone in Reading area using the FXO channel channel 1 of the Site B VOIP 102 200 2 9 8 To originate calls to phone connected to FXS port channel 2 of t
234. lected a wink is required during call setup Wink Timer integer values This is the length of the wink in ms in milliseconds for wink signaling No Response Timer integer values in seconds Applicable only when Signal parameter is set to Wink The value here denotes the time in seconds after which the call attempt would be disconnected by the FXO Interface because there was no answer Disconnect on Y N Allows call on FXO port to be Call Progress disconnected when a PBX issues a Tone call 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 Enable feature is used to create an open 144 audio path 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 Tone not applicable for Wink signaling MultiVOIP User Guide Technical Configuration E amp M Interface Parameter Definitions cont d Field Name Values Description Dialing Options Inter Digit integer values This is the length of time that Timer in seconds the MultiVOIP will wait between digits When the time expires the MultiVOIP will look in the phonebook for the number entered Default 2 Message Light or None Allows MultiVOIP to pass Waiting mode code sequences between Indication Avaya Magix PBXs to turn on and off the message wa
235. led digits Digits to be added before completing call to destination IP Address n n n n The IP address to which the for call will be directed if it n 0 255 begins with the destination pattern given Description alpha Describes the facility or numeric geographical location at which the call will be completed Protocol Type SIP or H 323 Indicates protocol to be used in or SPP outbound transmission For the MVP SS units only SIP is used 213 T1 Phonebook Configuration MultiVOIP User Guide Add Edit Outbound Phone Book Field Definitions cont d Field Name Values Description SIP Fields Use Proxy Y N Select if proxy server is used Transport TCP or Voip administrator must choose Protocol UDP between UDP and TCP transmission protocols UDP is a high speed low overhead connectionless protocol where data is transmitted without acknowledgment guaranteed delivery or guaranteed packet sequence integrity TCP is slower connection oriented protocol with greater overhead but having acknowledgment and guarantees delivery and packet sequence integrity SIP Port 5060 or other The SIP Port Number is a Number UDP logical port number See RFC 3087 The voip will listen for SIP Control of messages at this logical port Servi If SIP is used 5060 is the Context using default standard or well SIP Request known port number to be URI by the used If 5060 is not used Ne
236. ll be present in any case Entries organized by model number in the knowledge base and troubleshooting resolutions sections of the MultiTech web site found under Support constitute another source of help for problems encountered in the field 21 Quick Start MultiVOIP User Guide Chapter 2 Quick Start Instructions 22 MultiVOIP User Guide QS Intro Introduction This chapter contains streamlined instructions to get the MultiVOIP up and running quickly These start up instructions include assistance on setting up the MultiVOIP s Inbound and Outbound Phonebooks These sections of the Quick Start Instructions may be particularly useful for phonebook configuration Phonebook Starter Configuration Phonebook Tips Phonebook Example One Common Situation The Quick Start Guide also contains a Phonebook Worksheet section You may want to print out several worksheet copies Paper copies can be very helpful in comparing phonebooks at multiple sites at a glance This will assist you in making the phonebooks clear and consistent and will reduce surfing between screens on the configuration program A printed Cabling Guide is shipped with the MultiVOIP and an electronic copy is included on the Product CD 23 MultiVOIP User Guide Task Collecting Phone IP Details vital Placement Command Control Computer Setup Specs amp Settings Hookup Software Installation
237. ll transfer The call transfer sequence can be 1 to 4 characters in length using any combination of digits or characters or The sequences for call transfer call hold and call waiting can be from 1 to 4 digits in length consisting of any combination of digits 1234567890 176 MultiVOIP User Guide Technical Configuration Supplementary Services Definitions cont d Field Name Values Description Call Hold Y N Select to enable Call Hold function in Enable voip unit Call Hold allows one party to maintain an idle non talking connection with another party while receiving another call Call Waiting while initiating another call Call Transfer or while performing some other call management function Hold phone The numbers and or symbols that the Sequence keypad 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 Y N Select to enable Call Waiting function Enable in voip unit Retrieve phone The numbers and or symbols that the Sequence keypad caller must press on the phone characters keypad to initiate retrieval of a two waiting call characters The call waiting retrieval sequence inlength can be 1 to 4 characters in length using any combination of digits or characters or This is the phone keypad sequence that a
238. lly be handled easily by one person 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 the date and time may be incorrect If the battery fails the board must be sent back to Multi Tech Systems for battery replacement Warning There is danger of explosion if the battery is incorrectly replaced 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 uninsulated telephone wires or terminals unless the telephone line has been disconnected at the network interface 5 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 9 This product must be disconnected from its power source and telephone network interface when servicing 62 MultiVOIP User Guide Mechanical
239. ls 216 MultiVOIP User Guide T1 PhoneBook Configuration 3 Select Inbound PhoneBook List Entries r Inbound Phone Book Forward Address Not Used Not Used Not Used Not Used Not Used Not Used Not Used Number of Entries 7 m Details Channel No Edit Description Delete m Registration Options Close SIP Help Register with SIP Proxy 217 T1 Phonebook Configuration MultiVOIP User Guide 4 The Add Edit Inbound PhoneBook screen appears Add Edit Inbound Phone Book Accept Any Number Remove Prefix OK Add Prefix Cancel Channel Number Hunting X Help Description Call Forward MV Enable Forward Condition F Unconditional l Busy Forward Destination SIP call Phone or IP address or IP address port or Phone 1P address port or SIP URL or Ph IP address Ring Count jo Registration Options m Password 218 MultiVOIP User Guide T1 PhoneBook Configuration Enter Inbound PhoneBook data for your MultiVOIP The fields of the Add Edit Inbound PhoneBook screen are described in the table below Add Edit Inbound Phone Book Field Definitions Field Name Values Description Accept Any Values Y N Number Description When checked Any Number appears as the value in the Remove Prefix field The Any Number feature of the Inbound Phone Book does not work when an external routing device is used Gatekeeper for H32
240. m fixing 101 cabling procedure MYV P210S Sokeria 73 MVP410 SS ooon 69 MVP810 SS ce eeeeeeeseeeeeneeeeeeeeee 69 Cadence 1 custom field 161 Cadence 2 custom field 161 Cadence 3 custom field 161 Cadence 4 custom field 161 Cadence field ceeee 157 158 cadences custom TLE liie raii aaro 161 cadences signaling eee 153 Call Control PHB field 107 Call Control Priority Ethernet IP parameters field eee 105 Call Control Status Call Progress Details statistics Hel dts a ees 298 Call Control Status call progress Piel ds se csisecatiteeiuciielaieeey avis 298 Call Direction SMTP logs field 166 Call Duration field 0 0 cee 125 Call Forward Parameters inbound phonebook Bl aie ihaki E 260 Taa ei Bese EER 220 Call Forwarded To logs statistics field 305 Call Holdeni iran na 174 Call Hold Enable 177 Call Mode RADIUS Attributes field PEE coer tea NAET T 190 Call Mode SMTP logs field 165 Call Name Identification 174 Call Name Identification Calling Party o s 179 Call Name Identification Alerting Party cece 180 Call Name Identification Alerting Party oseese 181 Call Name Identification MultiVOIP User Guide Alerting Party scenes 182 Call On Hold Call Progress Details statistics PIE ci 6 dodec nn na 297 Call on Hold call progress field
241. m menu launches the MultiVOIP Configuration software program Configuration Port Setup The Configuration Port Setup option in the MultiVOIP Program menu brings up the COM Port Setup screen of the MultiVOIP configuration software COM Port Setup Select Port com M Baud Rate 115200 m Modem Setup Init String 4TSO 18E5 SB1152008D1 Init Response O D Dial String Connect Response CONNECT Hangup String ATHO NOTE If there is a Dial String specified in Modem Setup Configuration programs will try to initialize modem and dial this string 326 MultiVOIP User Guide Operation amp Maintenance Date and Time Setup The dialog box below allows you to set the time and date indicators of the MultiVOIP system Date and Time Settings Date mm dd yy iE 9 01 l i smm ss 17 Timefhh mm ss 11 17 25 AM S E Cancel Obtaining Updated Firmware Generally updated firmware must be downloaded from the MultiTech web FTP site to the user s PC before it can be downloaded from that PC to the MultiVOIP Note that the structure of the MultiTech web FTP site 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 multistech online EE Multiycip Support Documents Partners E 327 Operation amp Maintenance MultiVOIP User Gui
242. me of the machine providing the service An example SRV record might look like this _sip _tcp example com 86400 IN SRV 0 5 5060 sipserver example com This expression denotes a server named sipserver example com This server listens on TCP port 5060 for SIP protocol connections The priority given here is 0 and the weight is 5 TDM Routing Option Parameter fields Use TDM Y N Allows calls placed Routing for enabled by between ports on the Intra Gateway default same MultiVOIP voice calls channel board to be routed over internal Time Division Multiplex bus without conversion to IP TDM routing effectively eliminates the delay introduced by IP conversion If you require all calls to be IP routed disable the ase TDM Routing for Intra Gateway Calls option Since this is not normally required we generally recommend leaving TDM Routing enabled 110 MultiVOIP User Guide Technical Configuration 7 Set up the Web Browser GUI Optional After an IP address for the MultiVOIP unit has been established you can choose to do any further configuration of the unit a by using the MultiVOIP web browser GUI or b by continuing to use the MultiVOIP Windows GUI If you want to do configuration work using the web browser GUI you must first set it up To do so follow the steps below A Set IP address of MultiVOIP unit using the MultiVOIP Configuration program the Windows GUI B Save Setup in Window
243. ms Inc 2205 Woodale Drive Mounds View MN 55112 Tel 763 785 3500 FAX 763 785 9874 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 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 inspec
244. n field it is useful to describe the ultimate destination of the calls For example ina 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 40 characters max North America Euro National Call Long Distance Example Example Seattle Chicago system London Birming system Possible Description Possible Description Free Seattle access all Local rate London access employees all empl Euro International Call Example Rotterdam Bordeaux system Possible Description Local rate Rotterdam access all empl 7 Inthe Add Edit Inbound Phonebook screen under Registration Options enter the special password if any that will be used for this inbound phonebook entry If you specify a special password that applies only to this inbound phonebook entry that password will override the general password used by endpoints registering with the SIP server in the SIP Call Signaling screen 8 Repeat steps 2 8 for each inbound phonebook entry When all entries are complete go to step 9 9 Click OK to exit the inbound phonebook screen 10 Click on Save Setup Highlight Save and Reboot Click OK Your starter inbound phonebook configuration is complete 46 MultiVOIP User Guide QS Phonebook Tips Phonebook Tips Preparing the pho
245. n 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 75 Mechanical Installation amp Cabling MultiVOIP User Guide 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 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 pinout For a DID connection 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 connections of the wires to the two middle pins of one RJ 11 connector 6 Repeat the above step to connect the remaining telephone equipment to the second channel on your MultiVOIP 7 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
246. n 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 The SMTP Parameters screen can be reached by pulldown menu keyboard shortcut or sidebar Accessing SMTP Parameters Pulldown Icon gt Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl alt N Regional Parameters Ctrlt R SMTP Parameters Ctrl Alt S Logs Traces Ctri Alt L Supplementary Services Ctrl Alt H System Information Ctrl Alt SIP CallSignaling Ctrl Alt St P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt Sht Shortcut Sidebar Configuration Ethernet IP Ctrl Alt S Yoice Fax Interface SIP Call Signaling Regional RADIUS 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 Mult
247. nager 5 certain telephony attributes that are common to particular nations or regions 6 its operation with a mail server on the same IP network per SMTP parameters such that log reports about VoIP telephone call traffic can be sent to the administrator by email 7 implementing some common premium telephony features Call Transfer Call Hold Call Waiting Call ID Supplementary Services and 8 selecting the method by which log reports will be made accessible The process of specifying values for the various parameters in these seven categories is what we call technical configuration and it is described in this chapter Phonebook Configuration The second type of configuration that is required for the MultiVOIP pertains to the phone number dialing sequences that it will receive and transmit when handling calls Dialing patterns will be affected by both the PBX telephony equipment and the other VOIP devices that the MultiVOIP unit interacts with We call this Phonebook Configuration and for analog MultiVOIP units it is described in Chapter 6 The Quick Start Guide presents additional information on phonebook setup Local Remote Configuration The MultiVOIP must be configured locally at first to establish an IP address for the MultiVOIP unit But changes to this initial configuration can be done either locally or remotely 89 Technical Configuration MultiVOIP User Guide Local configuration is done t
248. nce have eight values 0 7 ranging from normal precedence value of 0 to network control value of 7 When set the D bit requests low delay the T bit requests high throughput and the R bit requests high reliability Routers that support DiffServ can examine the six DSCP bits and prioritize the packet based on the DSCP value The DiffServ Parameters fields in the MultiVOIP IP Parameters screen allow you to configure the DSCP bits to values supported by the router Specifically the Voip Media PHB field relates to the prioritizing of audio packets RTP and RTCP packets and the Call Control PHB field relates to the prioritzing of non audio packets packets concerning call set up and tear down gatekeeper registration etc The MultiVOIP Call Control PHB parameter defaults to 34 decimal 22 hex 100010 binary consider vis a vis TOS field above for Assured Forwarding behavior The MultiVOIP Voip Media PHB parameter defaults to the value 46 decimal 2E hex 101110 binary consider vis a vis TOS field above To disable DiffServ configure both fields to 0 decimal 108 MultiVOIP User Guide Technical Configuration Ethernet IP Parameter Definitions cont d Field Name Values Description FTP Parameter fields FTP Server Enable Y N Default disabled See FTP Server File Transfers in Operation amp Maintenance chapter MultiVOIP unit has an FTP Server function so that firmware a
249. nd other important operating software files can be transferred to the voip via the network DNS Para meter fields Enable DNS Y N Default disabled Enables Domain Name Space System function where computer names are resolved using a worldwide distributed database 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 Address 4 places 0 255 IP address of specific DNS server to be used to resolve Internet computer names 109 Technical Configuration MultiVOIP User Guide About Service Records An SRV record holds the following information Service the symbolic name of the desired service Protocol this is usually either TCP or UDP Domain name the domain for which this record is valid TTL standard DNS time to live field Class standard DNS class field this is always IN Priority the priority of the target host Weight A relative weight for records with the same priority Port the TCP or UDP port on which the service is to be found Target the hostna
250. ndon PBX to extension 7424 on the key telephone system in Tavistock UK A The required entry in the London Outbound Phonebook to facilitate origination of the call would be 90182263742 The call would be directed to the Tavistock voip s IP address 200 2 9 5 Generally on such a call the caller would have to dial an initial 9 But typically the PBX would not pass the initial 9 dialed to the voip If the PBX did pass along that 9 however its removal would have to be specified in the local Outbound Phonebook B The corresponding entry in the Tavistock Inbound Phonebook to facilitate completion of the call would be 0182263742 for calls within the office at Tavistock 01822 for calls to the Tavistock local calling area PSTN Call Event Sequence 1 Caller in Inner London dials 901822637424 2 Inner London voip removes 9 3 Inner London voip passes remaining string 018226374240n to the Tavistock voip at IP address 200 2 9 5 4 The dialed string matches an inbound phonebook entry at the Tavistock voip namely 0182263742 5 The Tavistock voip rings one of the three FXS ports connected to the Tavistock key phone system 6 The call will be routed to extension 7424 either by a human receptionist operator or to an auto attendant which allows the caller to specify the extension to which they wish to be connected 279 E1 Phonebook Configuration MultiVOIP User Guide Site F calling Site D A v
251. nebook for your voip system is a complex task that at first seems quite daunting These tips may make the task easier 1 Use Dialing Patterns Not Complete Phone Numbers You will not generally enter complete phone numbers in the voip phonebook Instead you ll enter destination patterns that involve area codes and other digits If the destination pattern is a whole area code you ll be assigning all calls to that area code to go to a particular voip which has a unique IP address If your destination pattern includes an area code plus a particular local phone exchange number then the scope of calls sent through your voip system will be narrowed only calls within that local exchange will be handled by the designated voip not all calls in that whole area code In general when there are fewer digits in your destination pattern you are asking the voip to handle calls to more destinations 2 The Four Types of Phonebook Digits Used Important Destination patterns to be entered in your phonebook will generally consist of a calling area codes b access codes c local exchange numbers and d specialized codes YS we Although voip phonebook entries may look confusing at first it s useful to remember that all the digits in any phonebook entry must be of one of these four types a calling area codes There are different names for these around the world area codes city codes country codes et
252. nection T Ethernet Connection 32 QS Software Installation MultiVOIP User Guide Load MultiVOIP Control Software onto PC For more details see Chapter 4 Software Installation in User Guide 1 MultiVOIP must be properly cabled Power must be turned on 2 Insert MultiVOIP CD into drive Allow 10 20 seconds for Autorun to start If Autorun fails go to My Computer CD ROM drive Open Click Autorun icon 3 At first dialog box click Install Software 4 At welcome screen click Next 5 Follow on screen instructions Accept default program folder location and click Next 6 Accept default icon folder location Click Next Files will be copied 7 Select available COM port on command control computer 8 At completion screen click Finish 9 At the prompt Do you want to run MultiVOIP Configuration click No Software installation is complete 33 QS Phone IP Starter Config MultiVOIP User Guide Phone IP Starter Configuration This isa summary For full details see Technical Configuration chapter of User Guide 1 Open MultiVOIP program Start MultiVOIP xxx Configuration 2 Go to Configuration Ethernet IP Enter the IP parameters for your voip site Activate Packet Prioritization 802 1p if desired If you use a Domain Name Server DNS specify its IP address If DNS is used you can activate the Service Record SRV feature For det
253. nical Configuration MultiVOIP User Guide Voice Fax Parameter Definitions cont d Field Name Values Description Dynamic Jitter Maximum Jitter Value 60 to 400 ms The maximum dynamic jitter buffer of 400 milliseconds is the maximum delay tolerable over a high jitter network Default 300 msec Optimizat ion 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 Modem Relay To place modem traffic onto the voip network an application called modem relay use Coder G 711 mu law at 64kbps MultiVOIP User Guide Technical Configuration Voice Fax Parameter Definitions cont d Field Name Values Description Auto Disconnect Automatic Disconnect ion The Automatic Disconnection group provides four options which can be used singly or in any combination Jitter Value 1 65535 milli seconds The Jitter Value defines the average inter arrival packet deviation in milliseconds before the call is automatically disconnected The default is
254. nit 135 Technical Configuration MultiVOIP User Guide The Caller ID feature has dependencies on both the telco central office and the MultiVOIP phone book See the diagram series below A CID Flow Call is received Call originates here here CID cD at 1 42pm May 31 Terminating Generating Central Office OrO FXS VoIP IP VolP FXO with Network standard telephony a Clock FN Caller ID service 5 31 7 z Display shows 1 42pm N phone of Hees OE SRE Melvin Jones Protocol CID Number 763 555 8794 CID Name Melvin Jones Time Stamp Date 05 31 Time 1 42pm In x 06 release when SIP protocol is used CID Name field will duplicate value in CID Number field 763 555 8794 Figure 5 1 Voip Caller ID Case 1 Call through telco central office with standard CID enters voip system CID Flow Call i received Call originates here ere CID CID r at 4 19pm July 10 Terminating Generating pmb Central Office 6 O FXS VolP E VoIP 1 FXO without Is Network are standard telephony ZEN S Clock on Caller ID service en z 7 7 10 4 19pm on phone of SP Ey S TONS H 323 Protocol EIEE CID Number 423 CID Name Anoka Whse VP3 Time Stamp Date 7 10 Time 4 19pm In x 06 release when SIP protocol is used CID Name field will duplicate value in CID Number field Gateway Name Phone Book Con
255. ntary Services System Information MultiVOIP User Guide Technical Configuration In each field enter the values that fit your particular network SVoice F 3 Excellent Effort 7 O Best Effort 7 Multi olP 216 133 69 77 103 Technical Configuration MultiVOIP User Guide The Ethernet IP Parameters fields are described in the tables and text passages below Note that both DiffServ parameters Call Control PHB and VoIP Media PHB must be set to zero if you enable Packet Prioritization 802 1p Nonzero DiffServ values negate the prioritization scheme Ethernet IP Parameter Definitions cont d Field Name Values Description Ethernet Parameters Packet Y N Select to activate Prioritization prioritization under 802 1p 802 1p protocol described below Frame Type Type II 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 descriptor
256. nterval expires else they will be removed from the list Endpoints removed from this list can neither make nor receive calls Endpoint pre un Indicates whether the listed endpoint Type defined has been predefined within the SIP system or is an endpoint using the SIP server under rules of open access to endpoints at specified URLs or domain names Contact a b c d The IP address at which this endpoint Address for can be reached values 0 255 Port 0 64000 Indicates the digital time slot on Number which SIP calls will be made Default is 5060 Remaining numeric Indicates the time remaining before Time in sec the endpoint s registration with the SIP server has expired 286 MultiVOIP User Guide Operation amp Maintenance The illustration below shows the SIP Server Endpoint Statistics screen for an active SIP phone system in web GUI format F MultiVoIP MuRtiVOIP SS v3 08 0 e Nav 78 7005 Microsolt Internet Explorer Fle Edt View Favorkes Toots u gt OIl Oh res rede GD ab Address E reto 200 2 9 27 MunVOIP 83 Configuration Ageanced Phone Book Statistics SIP Server Condguraton Predefined Endpoints SIP Server Endpoint Statistics Endpoint Stanstics Endpoint Name Status Max Expiry Initiated _ Recene Change Password 4001 Registered with Alternate 1 000004 274 J Save amp Reboot a ee ey Logout 4002 Registered with Aiternate 1 000004 269
257. ny Number feature works differently depending on whether or not an external SIP Proxy routing device is used When no 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 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 SIP proxy routing device The IP Address of the external routing device must be set in the Phone Book Configuration screen 212 MultiVOIP User Guide T1 PhoneBook Configuration Add Edit Outbound Phone Book Field Definitions cont d Field Name Values Description Destination prefixes Defines the beginning of Pattern area codes dialing sequences for calls exchanges that will be connected to line another VOIP in the system numbers Numbers beginning with extensions these sequences are diverted from the PTSN and carried on Internet or other IP network Total Digits as needed This field currently disabled Number of digits the phone user must dial to reach specified destination Remove Prefix dialed digits Portion of dialed number to be removed before completing call to destination Add Prefix dia
258. o call that MultiVOIP Of course the phone numbers are not literally listed individually The phone stations that can originate or complete calls over the VOIP system are described by numerical rules called destination patterns These destination patterns generally consist of country codes area codes or city codes and local phone exchange numbers 247 E1 Phonebook Configuration MultiVOIP User Guide In order for any VOIP phone call to be made there must be both an Inbound Phonebook entry and an Outbound Phonebook entry that describe the end to end connection The phone station originating the call must be connected to the VOIP system The Outbound Phonebook for that VOIP unit must have a destination pattern entry that includes the called phone that is the phone completing the call The Inbound Phonebook of the VOIP where the call is completed must have a destination pattern entry that includes the digit sequence dialed by the originating phone station The PhoneBook Configuration procedure below is brief but it is followed by an example case For many people the example case may be easier to grasp than the procedure steps Configuration is not difficult but all phone number sequences destination patterns and other information must be entered exactly otherwise connections will not be made 248 MultiVOIP User Guide E1 PhoneBook Configuration Phonebook configuration screens can be accessed using icons or the
259. oduct CD Because the CD is auto detectable it will start up automatically when you insert it into your CD ROM drive When you have finished loading your MultiVOIP software you can view and print the User Guide by clicking on the View Manuals icon 1 Be sure that your MultiVOIP has been properly cabled and that the power is turned on 78 MultiVOIP User Guide Software Installation 2 Insert the MultiVOIP CD into your CD ROM drive The CD should start automatically It may take 10 to 20 seconds for the Multi Tech CD installation window to display gt Release Notes Welcome to the Multi Tech Systems Inc Analog MultiVOIPs i Models MVP410 55 re version X 04 or earlier otes for ftp Upgrade muic If the Multi Tech Installation CD window does not display automatically click My Computer then right click the CD ROM drive icon click Open and then click the Autorun icon 3 When the Multi Tech Installation CD dialog box appears click the Install Software icon 79 Software Installation MultiVOIP User Guide 4 A welcome screen appears 4 Multi Tech Systems Welcome to Multi VOIP SS Multi Tech Systems MultiVOIP SS 3 08 Installation Thank you for choosing MultiVOIP SS from Multi Tech Systems Click Next to continue installation Back Cancel 9 fam 2 Windows u s Multi Tech Sy 4y 8 50PM Press Enter or click Next to continue 80 MultiVOIP User Guide Sof
260. oip call from a Tavistock key extension to extension 3117 on the PBX in Inner London A The required entry in the Tavistock Outbound Phonebook to facilitate origination of the call would be 3 The string 02073988 is added preceding the 3 The call would be directed to the Inner London voip s IP address 200 2 9 9 B The corresponding entry in the Inner London Inbound Phonebook to facilitate completion of the call would be 020739883 1 The caller in Tavistock picks up the phone receiver presses a button on the key phone set This button has been assigned to a particular voip channel 2 The caller in Tavistock hears dial tone from the Tavistock voip 3 The caller in Tavistock dials 02073983117 4 The Tavistock voip sends the entire dialed string to the Inner London voip at IP address 200 2 9 9 5 The Inner London voip matches the called digits 02073983117to its Inbound Phonebook entry 020739883 which it removes Then it adds back the 3 as a prefix 6 The Inner London PBX dials extension 3117 in the office in Inner London Variations in PBX Characteristics The exact dialing strings needed in the Outbound and Inbound Phonebooks of the MVP3010 will depend on the capabilities of the PBX Some PBXs require trunk access codes like an 8 or 9 to access an outside line or to access the VOIP network Other PBXs can automatically distinguish between intra PBX calls PSTN calls and VOIP calls
261. olg 55 QS Connectivity Test MultiVOIP User Guide Connectivity Test The procedures Phone IP Starter Configuration and Phonebook Starter Configuration must be completed before you can do this procedure 1 These connections must be made MultiVOIP to local phone station OR MultiVOIP to extension of key phone system MultiVOIP to command PC MultiVOIP to Internet 2 Inbound Phonebook and Outbound Phonebook must both be set up with at least one entry in each These entries must allow for connection between two voip units 3 Console messages must be enabled If this has not been done already go in the MultiVOIP GUI to Configuration Logs and select the Console Messages checkbox 56 MultiVOIP User Guide QS Connectivity Test 4 You now need to free up the COM port connection currently being used by the MultiVOIP program so that the HyperTerminal program can use it To do this you can either a click on Connection in the sidebar and select Disconnect from the drop down box or b close down the MultiVOIP program altogether 5 Open the HyperTerminal program asdf HyperTerminal File Edit View Call Transfer Help Dis 513 26 2l Print echo Auto detect 1152008 N1 SCROLL CAPS NUM_ Capture Disconnected 6 Use HyperTerminal to receive and record console messages from the MultiVOIP unit To do so set up HyperTerminal as follows setup shown is for
262. olis 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 200 002 009 007 Baltimore overlay 200 002 009 007 Baltimore Office 227 T1 Phonebook Configuration MultiVOIP User Guide Configuring Mixed Digital Analog VOIP Systems Analog MultiVOIP units like the MVP 210 410 810 410SS 810SS are compatible with digital MultiVOIP units like the MVP2410 In many cases digital and analog VOIP units will appear in the same telephony IP system In addition to MVP 210 410xx 810xx MultiVOIP units Series II units legacy analog VOIP units Series I units made by MultiTech may be included in the system as well When legacy VOIP units are included the VOIP administrator must handle two styles of phonebooks in the same VOIP network The diagram below shows a small scale system of this kind one digital VOIP the MVP2410 operates with two Series II analog VOIPs an MVP210 and an MVP410 and two Series I legacy VOIPs two MVP200 units EXAMPLE Digital amp Analog VOIPs Site D s jerre in Same System Area Code 615 yeeros oa ra mA I ZN 1 A LAZ a Ne Z 1 N SSi 615 492 3100 27 ar we Site E ita A aeyenno WY a ND EE EEA i T aa Area Code 701 I 200 2 9 6 posc aere eee Py Series 1 Analog MultiVOIP I l Series
263. on patterns namely 0182263740 0182263741 and 0182263742 In this way calls to 0182 263 7430 through 0182 263 7499 would be properly directed to the PSTN In the Site D outbound phonebook the 30 lines are defined exactly that is without making any adjacent phone numbers unreachable through the voip system 271 E1 Phonebook Configuration MultiVOIP User Guide The Outbound PhoneBook of the MVP3010 is shown below Outbound Phone Book for MVP3010 Digital VOIP Site D Destin Remov Add IP Comment Pattern e Prefix Address Prefix 201 200 2 9 7 To originate calls to Site A Birmingham 901189 901189 101 200 2 9 8 To originate calls to any Note 3 PSTN phone in Reading area using the FXO channel channel 1 of the Site B VOIP Reading UK 421 200 2 9 6 Calls to Site E Carlisle 90182 Calls to Tavistock local PSTN Site F could be arranged by operator or possibly by auto attendant 90182 9 200 2 9 5 Calls to extensions of key 263 phone system at Tavistock 740 office 90182 9 200 2 9 5 263 741 90182 9 200 2 9 5 263 742 102 200 2 9 8 To originate calls to phone connected to FXS port channel 2 of the Site B VOIP Reading Note 3 The pound sign is a delimiter separating the VOIP number from the standard telephony phone number 272 MultiVOIP User Guide E1 PhoneBook Configuration Th
264. or all channels at once 183 Technical Configuration MultiVOIP User Guide 17 Set NAT Traversal parameters NAT Network Address Translation parameters are applicable only when the MultiVOIP is operating in SIP mode The use of STUN Simple Traversal of UDP NATs servers to aid networks with NAT devices is described in RFC 3489 m NAT Traversal m STUN Server Name IP Port Timers Keep alive Cancel Help 184 MultiVOIP User Guide Technical Configuration Accessing NAT Traversal Parameters Pulldown Icon Multi oIP Multi OIP SS 3 08 0H Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl Alt y Regional Parameters Ctrl R SMTP Parameters Ctrl Alt S Logs Traces Ctrl Alt L Supplementary Services Ctrl Alt H System Information Ctrl alt SIP CallSignaling Ctrl Alt SFt P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt SFt Shortcut Sidebar Configuration Ethernet IP Ctrl Alt Sft Yoice Fax VH Interface SIP Call Signaling Regional SMTP RADIUS Logs Traces NAT Traversal Supplementary Descriptions for NAT Traversal screen fields are presented in the table below 185 Technical Configuration MultiVOIP User Guide NAT Traversal Definitions cont d Field Name Values Description Enable Y N Enables STUN client funct
265. ormation Address Pott Revesistration Intervat 3 Cancel Edt Predefined Endpoint a OK Password Registration Type Static 7 Dynamic Cancel p Contact Information Address 23 45 98 1 Endpoint Name warehousel Port 601 Resegistration Intervat SIP Server Predefined Endpoints Parameter Definitions Field Name Values Description Endpoint alpha Identifier for gateway within SIP voip Name numeric system Max length is 33 characters Password alpha This password is for authentication of numeric gateway to SIP server Registration Static Static registrations are fixed and the Type Dynamic contact information for them is configured 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 198 MultiVOIP User Guide Technical Configuration neither make nor receive calls Re Registration Interval integer values in seconds default is 3600 The time after which the MultiVOIP UserAgent Client is supposed to register with the proxy server Expiration of the registration interval means that the gateway has lost contact with the main SIP server and that the MVP SS
266. ound Phonebook 5 Ease of Use The phonebook setup determines how easy the voip system is to use Generally you ll want to make it so dialing a voip call is very similar to dialing any other number on the PSTN or through the PBX 6 Avoid Unintentional Calls to Official Emergency Numbers Dialing a voip call will typically be somewhat different than ordinary dialing Because of this it s possible to set up situations quite unwittingly where phone users may be predisposed to call official numbers without intending to do so Conversely a voip PBX system might also make it difficult to place an official emergency call when one intends to do so Study your phonebook setup and do some test dialing on the system to avoid these pitfalls 7 Inbound Outbound Pattern Matching In general the Inbound Phonebook entries of the local voip unit will match the Outbound Phonebook entries of the remote voip unit Similarly the Outbound Phonebook entries of the local voip unit will match the Inbound Phonebook entries of the remote voip unit There will often be non matching entries but it s nonetheless useful to notice the matching between the phonebooks 8 Simulating Network in lab on benchtop One common method of configuring a voip network is to set up a local IP network in a lab connect voip units to it and perhaps have phones connected on channel banks to make test calls 50 QS Phonebook Example Phonebook Example Bois
267. ound Phonebook entries list Bd ERR EE ESAT 251 TL RE EET 210 outbound vs inbound phonebooks BD ERR R RA RTA 247 Then a e a 206 Out of Band DTMF and Outbound Digits Sent 00 cee eeeeeeseeeeeeeee 167 Output Gain field eee 115 output level fax tones 117 Packet Prioritization 802 1p Ethernet IP parameters 104 packet priority and DiffServ 108 packetization RTP ranges amp INCFEMENHS ee eee eeeeeeeeereeeeeee 322 packetization rates coder options and eee 321 Packets Lost call progress field 294 Packets Lost RADIUS Attributes Packets Lost SMTP logs field 166 Packets lost statistics logs field 303 Packets Received call progress field PEE EEE hate eae Riceeer es 294 Packets Received RADIUS Attributes field cece 190 Packets Received SMTP logs field Perak stabs ice usenet eis 165 404 MultiVOIP User Guide Packets received statistics logs field Packets Sent call progress field 294 Packets Sent RADIUS Attributes Packets Sent SMTP logs field 165 Packets sent statistics logs field 303 packets consecutive lost 125 parameters tracked by console 172 Pass Through Enable FXS Loop Start interface and AutoCall Voice Fax Params cccsesceceestecesseeeeeeeeee 131 Password SIP Server Predefined Endpoint Parameters cceseeeseeeeeeee 198 Password proxy server field 152 Password SMTP fi
268. ound 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 DiffServ configure both fields to 0 decimal The next page explains DiffServ in the context of the IP datagram Call Control 0 63 Value is used to PHB default 34 prioritize call setup IP packets Voip Media 0 63 Value is used to PHB default 46 prioritize the RTP RTCP n audio IP packets 107 Technical Configuration MultiVOIP User Guide The IP Datagram with Header Its Type of Service field amp DiffServ bits gt 0 4 8 16 19 24 31 VERS HLEN TYPE OF TOTAL LENGTH IDENTIFICATION FLAGS FRAGMENT OFFSET TIME TO LIVE PROTOCOL HEADER CHECKSUM SOURCE IP ADDRESS DESTINATION IP ADDRESS IP OPTIONS if any PADDING end of header DATA The TOS field consists of eight bits of which only the first six are used These six bits are called the Differentiated Service Codepoint or DSCP bits The Type of Service or TOS field 0 1 2 3 4 5 6 7 PRECEDENCE D T R unused three precede
269. our MultiVOIP will be using either FXS FXO or E amp M telephony interfaces skip to step 2 For any channel on which you are using the DID interface type you must change the jumper on the MultiVOIP circuit card a Disconnect power Unplug the AC power cord from the wall outlet or from the receptacle on the MultiVOIP unit b 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 Screws 3 holding circuit card assembly to chassis MVP410 810 rear panel A zL Figure 3 6 MVP 410SS 810SS Rear Screw Locations c Pull the main circuit card out about 5 inches the power connection to the board prevents it from being removed entirely from the chassis 69 Mechanical Installation amp Cabling MultiVOIP User Guide d Identify the channels on which the DID interface will be used Jumper Configurations _ enlarged 2 g ford Upper Circuit Card oyo opo i MVP810 only oF be ae For DID aeh Interface type Gio i i a chs chel ch7 chs we oot aE we eye Jumpers 5 8 ae oye i np e a For non DID eyo ea Interface type XER or o i ped deb d ppd pde Main Circuit Card MVP 410 810 Gene
270. p to step 11 If YES continue with step 10 10 Go to Configuration SMTP SMTP lets you send phone call log records to the Voip Administrator by email Select Enable SMTP You should have already obtained an email address for the MultiVOIP itself this serves as the origination email account for email logs that the MultiVOIP can email out automatically Enter this email address in the Login Name field Type the password for this email account Enter the IP address of the email server where the MultiVOIP s email account is located in the Mail Server IP Address field Typically the email log reports are sent to the Voip Administrator but they can be sent to any email address Decide where you want the email logs sent and enter that email address in the Recipient Address field 36 MultiVOIP User Guide QS Phone IP Starter Config Whenever email log messages are sent out they must have a standard Subject line Something like Phone Logs for Voip N is useful If you have more than one MultiVoip unit in the building you ll need a unique identifier for each one select a useful name or number for N In this Subject field enter a useful subject title for the log messages In the Reply To Address field enter the email address of your Voip Administrator 11 Go to Configuration Logs Select Enable Console Messages To allow log reports by email if desired click SMTP Click
271. pe Static Endpoint Type Predefined r Contact Details 23 65 12 45 Contact Address PotNo sd Remaining Ti 00 00 59 59 SIP Server Endpoint Statistics Parameter Definitions Field Name Values Description Endpoint alpha Identifier for gateway within SIP voip Name numeric system Max length is 33 characters Status server Indicates the SIP server that is identifier controlling traffic for this endpoint Max Expiry numeric Indicates the time remaining before Time in sec the endpoint s registration with the SIP server has expired 285 Operation amp Maintenance MultiVOIP User Guide SIP Server Endpoint Statistics Parameter Definitions Field Name Values Description Initiated numeric Indicates how many calls were Call Count initiated by phones connected to this endpoint Received numeric Indicates how many calls were Call Count received by phones connected to this endpoint No of numeric Indicates how many endpoints are Entries included in the system Registration Static Static registrations are fixed and the Type Dynamic contact information for them is configured 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 i
272. perating parameters of the Single Port Protocol SPP These are configured in the Call Signaling screen and in the Add Edit Outbound PhoneBook screen Accessing Registered Gateway Details Pulldown Icon Statistics Call Progress Ctrl alt a Logs Ctrl 0 IP Statistics Ctrl P s Ctrl w Ctrl 2 Alternate Servers Ctrl alt 4 gt Shortcut Sidebar Ctrl Alt W Configuration J Advanced Phone Book Statistics Call Progress Logs IP Statistics Link Management Registered Gateway Details Servers Save Setup Connection Help 314 MultiVOIP User Guide Operation amp Maintenance r Registered Endpoints Description IP Address Port Register Duration_ Status No of Entries ft Details Count of Registered Numbers jo List of Registered Numbers v 315 Operation amp Maintenance MultiVOIP User Guide Registered Gateway Details Field Definitions Field Values Description Name Column Headings Description alphanumeric This is a descriptor for a particular voip gateway unit This descriptor should generally identify the physical location of the unit e g city building etc and perhaps even its location in an equipment rack IP Address n n n n The RAS address for the gateway for n 0 255 Port Port by which the gateway exchanges H 225 RAS messages with the gatekeeper
273. plementing a Software Upgrade above the Upgrade Software command transfers from the controller PC to the MultiVOIP unit firmware including the H 323 stack and settings The settings can be either Factory Default Settings or Current Configuration Settings an Accessories IF Macromedia FreeHand 9 an Jasc Software IF Mozilla Firefox Set Program Access and Defaults Muttivorr 2 06 IF Microsoft Office eS Windows Catalog A Acrobat Distiller 7 0 Adobe Acrobat 7 0 Professional MultiVOIP 6 08 B Configuration Winzip Multi OIP 2410 4 08 d B Configuration Port Setup an Java 2 Runtime Environment gt B Date and Time Setup an Java Web Start gt 25 Download Factory Defaults I TalkAnytime 10 08 gt 25 Download Firmware windows Update Launch RealOne Player man ian Programs y 2 Download IFM Firmware 25 Download User Defaults B Set Password Settings B Uninstall LLA Documents amp Upgrade Software Search _ Help and Support Run Shut Down 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 mvptlftp 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 uploa
274. polarity for a specified time 140 290 ms a wink and then becomes ready to receive dial digits For Delay Dial the voip detects 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 in ms integer values in milliseconds This is the length of the wink for Wink Start and Delay Dial signaling modes Applicable only when Start Mode parameter is set to Wink Start or Delay Dial Dialing Options Inter Digit Regeneration Timer integer values in milliseconds Inter Digit integer values This is the length of time that Timer in seconds the MultiVOIP will wait between digits When the time expires the MultiVOIP will look in the phonebook for the number entered Default 2 Message Not applicable to DID DPO Waiting interface Indication This parameter is applicable when digits are dialed onto a DID DPO channel after the connection has been made The length of time between the outputting of DTMF digits Default 100 ms 148 MultiVOIP User Guide Technical Configuration 10 Set Call Signaling Parameters This dialog box addresses SIP Call Signaling parameters It can be reached by pulldown menu keyboard shortcut or a sidebar menu Accessing Call Signaling Parameters Pull
275. pplied on a channel by channel basis However once you have established a set of Voice FAX parameters for a particular channel you can apply this entire set of Voice FAX parameters to another channel by using the Copy Channel button and its dialog box To copy a set of Voice FAX parameters to all channels select Copy to All and click Copy Voice Fax Parameters Select Channel Channel 1 OK Cancel Copy Channel Default 2Help Copy Channel Copy Channel 1 Yoice Fax Parameters to I Copy to All Copy Cancel Channels J EES E Help Channels BH ss fw a r Channels Bn sf 8 E ny sf Ss S 114 MultiVOIP User Guide Technical Configuration The Voice FAX Parameters fields are described in the tables below Voice Fax Parameter Definitions Field Name Values Description Default When this button is clicked all Voice FAX parameters are set to their default values Select 1 2 210 Channel to be configured is selected Channel 1 4 410 here 1 8 810 Copy Copies the Voice FAX attributes of Channel 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 Modifies audio level entering voice to channel before it is sent over the 31dB network to the remote VOIP The default amp recommended value is
276. properly A summary of this configuration information appears on page 28 Config Info CheckList Gather IP Information Ask your computer network Info needed to operate administrator all MultiVOIP models IP Network Parameters s9 Record for each VOIP Site in System e IP Address e IP Mask e Gateway e Domain Name Server DNS Info optional e Determine whether or not 802 1p Packet Prioritization will be used 25 QS Gathering Phone IP Details MultiVOIP User Guide Phone IP Details Absolutely Needed Gather Telephone Information Telephony Parameters Ask phone company or telecom manager Analog Telephony Interface Parameters aN M Record for this VOIP Site e Which interface type is used E amp M FXS FXO DID DPO e If FXS determine whether the line will be used for a phone fax or KTS key telephone system e If FXO determine if line will be an analog PBX extension or an analog line from a telco central office e If E amp M determine these aspects of the E amp M trunk line from the PBX e What is its Type 1 2 3 4 or 5 e Is it 2 wire or 4 wire e Is it Dial Tone or Wink 26 MultiVOIP User Guide QS Gathering Phone IP Details Phone IP Details Often Needed Wanted Obtain Email Address for VOIP for email call log reporting required if log reports of VOIP call traffic Optional are to be sent by email SMTP Parameters Preparation Task
277. proxy server 150 MultiVOIP User Guide Technical Configuration SIP Call Signaling Parameter Definitions cont d Field Name Values Description SIP Proxy Parameters Allow Y N When selected incoming calls Incoming Calls are accepted only if those calls Through SIP come through the gatekeeper Proxy Only Primary Proxy This is the preferred SIP proxy server for controlling the traffic of the current voip Alternate ae A first and a second alternate SIP Proxy 1 and 2 proxy server can be specified for use by the current voip for situations where the Primary proxy server is busy or otherwise unavailable Proxy Domain n n n n Network address of the proxy Name IP where server that the voip is using Address n 0 255 Append SIP Y N When checked the domain Proxy Domain Name in User ID name of the SIP Proxy serving the MultiVOIP gateway will be included as part of the User ID for that gateway If unchecked the SIP Proxy s IP address will be included as part of the User ID instead of the SIP Proxy s domain name Port Number Logical port number for proxy communications User Name Values alphanumeric Description Identifier used when proxy server is used in network If a proxy server is used in a SIP voip network all clients must enter both a User Name and a Password before being allowed to make a call 151 Technical Configuration M
278. ption Field Description Bytes Total bytes sent in Bytes Total bytes received Sent call Received in call Packets Packets lost in Coder Voice Coder Lost call Compression Rate used for call will be listed in log Outbound The DTMF dialing Prefix When selected the Digits digits received by Matched phonebook prefix Sent this gateway from matched in the remote processing the call gateway will be listed in log presuming that DTMF is set to Out of Band Call Successful or Status unsuccessful Server The IP address etc of the traffic control server if any Details being used whether an H 323 gatekeeper a SIP proxy or an SPP registrar gateway will be displayed here if the call is handled through that server 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 Multiplexing Tunneling etc 191 Technical Configuration MultiVOIP User Guide Custom Fields Definitions cont d Field Description Field Description From Details To Details Gateway Originating Gatew N Completing or Number gateway answering gateway IP Addr IP address where IP Addr IP address where call call originated was completed or answered Descript Identifier of site Descript Identifier of site where call where call was originated completed or answered Options When selected log
279. r SMTP logs field gh Save dh E E E E 165 channel tracing on off logging 172 Checklist of configuration info 28 Clear IP Statistics button 308 Clear command Link Management DUULODG norna 312 coder bandwidth max cc ccceees 118 GT E E E 118 G23 Vite esse E 118 Er REAT 118 1E A E E EET 118 TE Ar S ET 118 Net Codef r eissii 118 Coder RADIUS Attributes field 191 Coder SMTP logs field 166 Coder field isitin rrira 118 coder options packetization rates and 321 Coder Parameters field group 118 coder types voice fax RTP packetization cesses 322 COM port conflict resolving e 100 EITOT MESSAGE eeeeeceeeeeneeeeee 100 on command PC eee 83 COM port allocation eee 193 COM port assignments 193 COM port conflict ETOT MESSAGE 0 ee eeeeceeeeeeseeeeeeee 83 COM Port Setup screen 83 100 command cable pinout 380 command modem and Regional Parameters screen 96 154 Command Modem SCtUP FOF ee eeeeeeeeteeneeeee 96 154 command PC COM port assignment detailed 83 Command PC COM port requirement 19 non dedicated use Of cee 19 Operating system eee 19 compatibility H 450 services with SIP sci Kates a EEEE ath 173 392 MultiVOIP User Guide compatibility H 450 with H 323 not WIth SIP sco aioe aie 14 compression silenc
280. r 0013 This is Syslog test message number 0012 This is Syslog test message number 0011 This is Syslog test message number 0010 This is Syslog test message number 0009 This is Syslog test message number 0008 This is Syslog test message number 0007 This is Syslog test message number 0006 This is Syslog test message number 0005 This is Syslog test message number 0004 This is Syslog test message number 0003 Thie ic Guelnn tect meceane number NNN 100 61MPH 17 02 09 18 2002 Warranty Service amp Tech Support MultiVOIP User Guide Chapter 9 Warranty Service and Tech Support 370 MultiVOIP User Guide Warranty Service amp Tech Support Limited Warranty Multi Tech Systems Inc MTS warrants that its products will be free from defects in material or workmanship for a period of two years from the date of purchase or if proof of purchase is not provided two years from date of shipment MTS MAKES NO OTHER WARRANTY EXPRESSED OR IMPLIED AND ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE HEREBY DISCLAIMED This warranty does not apply to any products which have been damaged by lightning storms water or power surges or which have been neglected altered abused used for a purpose other than the one for which they were manufactured repaired by the customer or any party without MTS s written authorization or used in any manner inconsistent with MTS s instructions MTS s entire o
281. r in order to maintain the connection If the MultiVOIP does not register before the TTL interval expires the MultiVOIP gateway s registration with the proxy server will expire and the proxy server will no longer permit call traffic to or from that gateway Calls in progress will continue to function even if the gateway becomes de registered 152 MultiVOIP User Guide Technical Configuration 12 Set Regional Parameters Phone Signaling Tones amp Cadences This dialog box can be reached by pulldown menu keyboard shortcut or sidebar Accessing Regional Parameters Pulldown Icon gt Multi oIP Multi OIP SS 3 08 0H Firmy Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl Alt N Regional Parameters SMTP Parame Ctrl Alt S Logs Traces Ctrl Alt L Supplementary Services Ctrl Alt H System Information Ctrl Alt SIP CallSignaling Ctrl Alt SFt P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt SFt Shortcut Sidebar El Configuration Ctrl R Ethernet IP Yoice Fax Interface SIP Call Signaling 153 Technical Configuration MultiVOIP User Guide The Regional Parameters screen will appear For the country selected the standard set of frequency pairs will be listed for dial tone busy tone unobtainable tone fast busy or trunk busy ring tone and other more specialized tones User Defined Tones
282. r 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 Par Sotinos Tore Pae Ds Tore 159 Technical Configuration MultiVOIP User Guide The Custom Tone Pair Settings fields are described in the table below Custom Tone Pair Settings Definitions Field Name Values Description Tone Pair dial tone busy tone ring tone unobtainable tone survivability tone re order tone TONE PAIR VALUES Identifies the type of telephony signaling tone for which frequencies are being specified About Defaults US telephony values are used as defaults on this screen However since this dialog box is provided to allow custom tone pair settings default values are essentially irrelevant Frequency 1 frequency in Frequency of lower tone of pair Hertz This outbound tone pair enters the MultiVOIP at the input port Frequency 2 frequency in Frequency of higher tone of pair Hertz This outbound tone pair enters the MultiVOIP at the input port Gain 1 gain in dB Amplification factor of lower 3dB to 31dB frequency of pair This figure and mute describes amplification that t
283. rality Ko For channels using the DID Ch1 Ko interface the jumper must Cna eha ene K not straddle across the Gi cross hatched area between F ae the jumper posts 1 For channels using any non DID interface it is acceptable that the 4Jumpers 1 4 jumper straddles across the sr cross hatched area between the jumper posts mre SEL eel lel ry po P bP SG bP b df J ppro p d CT UEZ Figure 3 7 MVP 410SS 810SS Channel Jumper Settings e Position the jumper for each DID channel so that it does not connect the two jumper posts For DID operation of a voip channel the MultiVOIP will work properly if you simply remove the jumper altogether but that is inadviseable because the jumper might be needed later if a different telephony interface is used for that voip channel f Slide the main circuit card back into the MultiVOIP chassis and replace the three screws 70 MultiVOIP User Guide Mechanical Installation amp Cabling 2 Connect the power cord supplied with your MultiVOIP to a live AC outlet and to the power connector on the back of the MultiVOIP as shown at top right in Figure 3 8 Command Modem connector for remote configuration Voice Fax Channel Connections Channels 1 4 Bottom MVP410 8 10 Channels 5 8 Top MVP810 Only E amp M FXS FXO gja a Command Port Connection Figure 3 8 Cablin
284. re essentially the same as that of the Windows GUI except for logging Multi oIP Multi OIP SS 3 08 0H Firmware Sep 09 2005 File Edit View Favorites Tools Help gt 9 Aa a 9 e a a Stop Refresh Edit Forward Home Search Favorites History Mail Print Discuss MultivoOIP 4108S Configuration IP Voice Fax Interface Current Permission Read Write SIP Call Signalin SNMP IP Parameters Regional r SMTP Enable Diffserv Frame Type Type ll Logs Supplementary IP Parameters System Informat Enable DHCP Phone Book Ok Statistics IP Address 254 25 162 165 SIP Server Save Setup IP Mask 255 255 255 128 Cancel Connection Help d Gateway 207 2941221 Mult VoIP Mult VOTP SS v3 08 0H Firmware Sep 09 2005 Conliguration Advaxed Phone Book Statistics Download Connection Heip ABBR RErS OMA r Ethomat 7 IP Parameters Imerface Ethernet Parameters Ae Aa Packet Pioihizsion 802 Tp Eume Tya vt sump 802 1p Parameters RADIS Prioniy nii Logs Traces Coll Contec Emeteri Etat NAT Traversal Supplemnerkary Services BVoice z System Information Advanced Othors DBest Effort E Phone Book P Saitis wano Po IP Parameters Gateway Name Mutio gt Enotie DHCP IP Address 192 169 4 i 15 Overview MultiVOIP User Guide The primary advantage of the web GUI is remote access for control and con
285. reCvd eee eee eee eeee 303 Bytes Sent e eee eee eeeeeee 301 Call Forwarded to 305 Call Transferred to 0 0 eee 305 Disconnect Reason e0 302 DTMF Capability 302 Duration s icscihegdanies 301 From gateway 301 Gateway Name callee 304 Gateway Name caller 304 H 450 functionality 305 IP Address callee 00 304 IP Address caller 304 IP Direction column 4 301 LOB AEE EE 301 MO de ines incase cose eese segusses sos 301 Options callee eeeeeeeeeee 304 Index Options caller 0 0 eee 304 Outbound digits eee 304 Outbound Digits Recvd 302 Outbound Digits Sent 302 Packets lost cc cecesscceeesreeees 303 Packets recvd cccsecceseereeees 303 Packets Sent c ccccsseceeeeseeeees 303 Packets Sent ccceecceeeeseeeees 301 Server Details cccccceseeeees 303 Start Date Time 005 301 Stats AO ch dednveesties 301 Supplementary Services info 305 To gateway 301 Type call column 301 Voice COE ceeeceesseceeeeeseeeeee 302 Logs Statistics function 299 Logs Statistics screen Delete File button 0 301 field definitions 301 302 303 304 305 First button 301 Last Dutton sses 301 Next button sses 301 Previous button eee 301 logs and web browser GUI 170 logs by ema
286. received by this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software 308 MultiVOIP User Guide Operation amp Maintenance IP Statistics Field Definitions cont d Field Values Description Name Total Packets Sum of data packets of all types cont d Received integer Total number of error laden packets with value received by this VOIP gateway since the Errors last clearing or resetting of the counter within the MultiVOIP software UDP Packets User Datagram Protocol packets Transmit integer Number of UDP packets transmitted by ted value this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of UDP packets received by this value VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of error laden UDP packets with value received by this VOIP gateway since the Errors last clearing or resetting of the counter within the MultiVOIP software TCP Packets Transmission Control Protocol packets Transmit integer Number of TCP packets transmitted by ted value this VOIP gateway since the last clearing or resetting of the counter within the MultiVOIP software Received integer Number of TCP packets received by this value VOIP gateway since the last clearing
287. 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 such as loading heavy equipment in rack unevenly 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 Maximum ambient temperature for the unit is 60 degrees Celsius 140 degrees Fahrenheit at 20 90 non condensing relative humidity This equipment should only be installed by properly qualified service personnel Only connect like circuits In other words connect SELV Secondary Extra Low Voltage circuits to SELV circuits and TN Telecommunications Network circuits to TN circuits 67 Mechanica
288. refix Prefix Number 421 1 Call Completion Summaries Site A calling Site C Method 1 1 Dial 101 2 Hear dial tone from Site B 3 Dial 7175662 4 Await completion Talk Site A calling Site C Method 2 1 Dial 101 7175662 2 Await completion Talk Note Some analog VOIP gateways will allow completion by Method 2 Others will not Site C calling Site A 1 Dial 7175000 2 Hear dial tone from Site B VOIP 3 Dial 201 4 Await completion Talk 237 T1 Phonebook Configuration MultiVOIP User Guide Site D calling Site C 1 Dial 9 15077175662 2 9 gets outside line On some PBXs an 8 may be used to direct calls to the VOIP while 9 directs calls to the PSTN However some PBX units can be programmed to identify the destination patterns of all calls to be directed to the VOIP PBX at Site D is programmed to divert all calls made to the 507 area code and exchange 717 into the VOIP network It would also be possible to divert all calls to all phones in area code 507 into the VOIP network but it may not be desirable to do so The MVP2410 removes the prefix 1507 and adds the prefix 101 for compatibility with the analog MultiVOIP s phonebook scheme The is a delimiter separating the analog VOIP s phone number from the digits that the analog VOIP must dial onto its local PSTN to complete the call The digits 101 7175662 are forwarded to the Site B analog
289. requency of pair and mute setting This applies to any supervisory tones that the MultiVOIP outputs as audio to the FXS FXS or E amp M port Default 16dB Gain 2 gain in dB Amplification factor of higher 3dB to 31dB frequency of pair and mute setting This applies to any supervisory tones 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 msec On Off four integer time to denote supervisory tones specified values in by user Supervisory tones relate to milli seconds zero value for dial tone indicates continuous tone answering and disconnection of calls Although most cadences have only two parts an on duration and an off duration some telephony cadences have four parts Most cadences then are expressed as two iterations of a two part sequence Although this is redundant it is necessary to allow for expression of 4 part cadences 158 MultiVOIP User Guide Technical Configuration 13 Set Custom Tones and Cadences optional The Regional Parameters dialog box has a secondary dialog box that allows you to customize DTMF tone pairs to create unique ring tones dial tones busy tones or unobtainable tones fast busy signal or re order tones telling the user that she must hang up an off hook phone or survivability tones an indication of call routing redundancy for you
290. ress of Control PC IP Address of voip unit 1 IP address of voip unit n 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 ZJ MultiVOIP 410 v6 08 Firmware Sep 06 2006 Microsoft Internet Explorer gt gt File Edit View Favorite Address http 192 168 2 200 acl MultiVOIP 410 O Configuration Multi echi Phone Book Systems Statistics Change Pass Save amp Reboo Logout Current Permission Read Write Help Password Change Old Password lt New Password Reconfirm Password As shown above the username and password can be set in the web GUI as well as in the Windows GUI 352 MultiVOIP User Guide Operation amp Maintenance 3 Install FTP Client Program or Use Substitute You should install an FTP client program on the controller PC FTP file transfers can be done using a web browser e g Netscape or Internet Explorer in conjunction with a local Windows browser a e g Windows Explorer but this approach is somewhat clumsy it requires use of two application programs rather than one and it limits downloading to only one VOIP unit at a time With an FTP client program multiple voips can receive FTP file transmissions in response to a single command the transf
291. rienircas innia 207 Accept Any Number inbound A EE EEE 259 Tlhspirnascas innin taa in as 219 Accept Any Number outbound field AE E Aisne 253 Tlranmirnaranu na uini an 212 Accept Registrations for domains field SIP Server Configuration PaTAMeters ee eee 196 Accept Registrations for IP addresses field SIP Server Configuration PaTAMeters eee 196 accessing Statistics Logs screen PEE pies Woteses EE E 299 accessing Call Progress Statistics SCTCED oieee anea vests 291 accessing configuration parameter TEKO E E 101 accessing Ethernet IP Parameters SCT CD ass espi iri eiss eissis 102 accessing interface parameters 126 accessing IP Statistics screen 306 accessing Logs Statistics screen e r E RE E EAEE a ee 299 accessing logs screen sees 169 accessing Regional Parameters 153 accessing Registered Gateway Details Statistics screen cccceseeeeee 316 accessing Registered Gateway Details screen 0004 315 316 accessing RTP Parameters screen 321 accessing SMTP parameters 162 accessing Supplementary Services accessing System Information screen r eea E T a Shots EET 200 388 MultiVOIP User Guide accessing Voice FAX Parameters SCLEEN E E EE 112 Accounting Port RADIUS screen Helden na eee cA 189 Add Inbound Phonebook Entry icons d A AEE EEEE EE EE AT 249 We cise E LEE N T 208 Add Outbound Phonebook Entry icon BY ake Renita tes a ies 249 d RI eee EEEREN A
292. rns of all calls to be directed to the VOIP 3 PBX at Site D is programmed to divert all calls made to the 118 area code and exchange 943 into the VOIP network It would also be possible to divert all calls to all phones in area code 118 into the VOIP network but it may not be desirable to do so 4 The MVP3010 removes the prefix 0118 and adds the prefix 101 for compatibility with the analog MultiVOIP s phonebook scheme The is a delimiter separating the analog VOIP s phone number from the digits that the analog VOIP must dial onto its local PSTN to complete the call The digits 101 9435632 are forwarded to the Site B analog VOIP 5 The call passes through the IP network in this case the Internet 6 The call arrives at the Site B VOIP This analog VOIP receives this dialing string from the MVP3010 101 9435632 The analog VOIP seeing the 101 prefix uses its own channel 1 an FXO port to connect the call to the PSTN Then the analog VOIP dials its local phone number 9435632 to complete the call NOTE In the case of Reading Berkshire England both 1189 and 1183 are considered local area codes This is ina sense however a matter of terminology It simply means that numbers of the form 9xx xxxx and 3xx xxxx are both local calls for users at other sites in the VOIP network 278 MultiVOIP User Guide E1 PhoneBook Configuration Site D calling Site F A voip call from Inner Lo
293. roxy Parameters 0 00 150 SIP Server Configuration parameters Accept Registrations for domains iia AERE R 196 Accept Registrations for JP Addresses nossos 196 Allow Undefined Registrations 196 Domain Names acceptable for registration oossoo 196 IP Addresses acceptable for TEQISTYALION crisser sa iasi 196 Operating Mode eee 195 Re Registration Time 197 Survivability Status Check 195 SIP Server Endpoint Statistics Contact Address eee 286 SIP Server Endpoint Statistics Parameters Endpoint Name uu eee 285 Endpoint Type sses 286 Initiated Call Count 286 Max Expiry Time eee 285 No of Entries cceeeeseeeee 286 Port Number 0 0 0 0 eeeeeeeeeceeee 286 Received Call Count 286 Registration Type 286 Remaining Time eee 286 Stasia aaea 285 SIP Server Predefined Endpoint Parameters Address Contact Info 199 Endpoint Name cece 198 Pass WO0d inarrit 198 Port Contact Info e 199 Registration Type cece 198 Re Registration Interval 199 Re Registration Time 199 409 Index SIP survivability 13 SIP URL field Bl lattice E E E 255 T ea a iea 214 SMTP log reporting type button 171 SMTP logs by email illustration 168 SMTP Parameters definitions 164 SMTP Parameters fields Enable SMTP eeeseeeeeeeee 163 Login Name sesser 163
294. rs 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 Accessing Packetization Time Pulldown amp MultiVOIP v 04 Shortcut Icon Sidebar none none Configuration Advanced Packetization Time Phone Book Statistics Save Setup Connection Hel 321 Operation amp Maintenance MultiVOIP User Guide Packetization Time Screen ketization Time E Statistics ic Save Setup E Connection Help eld Channel el o sl jeo xl feo zl EW fso zl feo zi E feo l eo l eo feo zi o zi Packetization rates can be set separately for each channel The table below presents the ranges and increments for packetization rates Packetization Ranges and Increments Coder Types Range in Kbps Increments in Kbps default value G711 G726 G727 5 120 5 5 G723 30 120 30 30 G729 10 120 10 10 Netcoder 20 120 20 20 322 MultiVOIP User Guide Operation amp Maintenance Once the packetization rate has been set for one channel it can be copied into other channels Packetization Time Select Channel Packetization Rate J G711 A law 64 Kbps G727 40 16 Kbps x e e Cancel Copy Channel G711 U law 64 Kbps G727 40 24 Kbps Copy Channel Copy Channel 1 Packetization Parameters to T Copy to All Chann
295. rs 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 al OK Default Copy Channel Copy Channel 1 Supplementary Services to Copy T Copy to All Cancel Channels ny ss 8 ws Channels no ss sg sf nH sf 8 E E 175 Technical Configuration MultiVOIP User Guide The Supplementary Services fields are described in the tables below Supplementary Services Parameter Definitions Field Name Values Description Select Channel 1 4 4108S 1 8 8105S The channel to be configured is selected here 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 character The numbers and or symbols that the caller must press on the phone keypad to initiate a ca
296. s 21 Set SIP Server PreDefined Endpoint parameters 22 View System Info screen and set updating interval optional 23 Save the MultiVOIP configuration 24 Create a User Default Configuration optional When technical configuration is complete you will need to configure the MultiVOIP s inbound and outbound phonebooks This manual has separate chapters describing T1 Phonebook Configuration for North American influenced telephony settings and E1 Phonebook Configuration for Euro influenced telephony settings Local Configuration Procedure Detailed You can begin the configuration process as a continuation of the MultiVOIP software installation You can establish your configuration or modify it at any time by launching the MultiVOIP program from the Windows Start menu 1 Check Power and Cabling Be sure the MultiVOIP is turned on and connected to the computer via the MultiVOIP s Command Port DB9 connector at computer s COM port RJ45 connector at MultiVOIP Start MultiVOIP Configuration Program Launch the MultiVOIP program from the Windows Start menu from the folder location determined during installation B Configuration Port Setup B Date and Time Setup 2 Download Factory Defaults 25 Download Firmware Settings 25 Download IFM Firmware 25 Download User Defaults N Documents Search B Set Password Help and Support B UnInstall 2 Upgrade Software 97 Technical Configuration Mult
297. s 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 MultiVOIP User Guide Technical Configuration Voice Fax Parameter Definitions cont d Field Name Values Description AutoCall Offhook Alert Parameters Auto Call AutoCall continued from previous page Offhook Offhook Both functions apply on a channel by Alert Alert channel basis It would not be appropriate for either of these functions to be applied to a channel that serves in a pool of available channels for general phone traffic Either function requires an entry in the Outgoing phonebook of the local MultiVOIP and a matched setting in the Inbound Phonebook of the remote voip Generate Y N Local Dial Tone Used for AutoCall only If selected dial tone will be generated locally while the call is being established between gateways The capability to generate dial tone locally would be particularly useful when there is a lengthy network delay 121 Technical Configuration MultiVOIP User Guide Voice Fax Parameter Definitions cont d Field Name Values Description AutoCall Offhook Alert Parameters Offhook 0 3000 The length of time that must elapse Alert Timer seconds before the offhook alert is triggered and a call is automatically made to the phone number listed in the Phone
298. s 141 Tone Detection c ce eee 142 398 MultiVOIP User Guide FXO Supervision Parameter definitions eee eeeeeeeeeeseeeeees 140 FXS interface MVP210 SS USCS Of A E sccesins eevtdets 75 FXS interface MVP 410SS 810SS USES Of AAE EO 71 FXS Loop Start Interface parameter definitions ee eeeceeseeceteeeeeee 129 FXS Loop Start Interface Parameter fields Caller ID enable 0 132 Caller ID Enable 0 0 0 131 Caller ID Type ee 131 Current LOSS cccccccccecseeeeees 130 Detection Range flash hook 131 Inter Digit Regeneration Timer 130 Inter Digit Timer eee 130 Message Waiting Indication 130 Pass Through Enable 131 Ring Count eee neris 130 FXS Loop Start Parameter fields Generate Current Reversal 130 Inter Digit Timer eee 129 Message Waiting Light 129 FXS Loop Start Parameters 129 FXS FXO connector MVP210 SS oo eeeeeceeeeeeseeeeeeeees 75 MVP 410SS 810SS eee 71 G711 coders RTP packetization VOICE LAX ueniet iai 322 G723 coders RTP packetization VOICE FAX einet ia 322 G726 coders RTP packetization VOICE FAX perintei iai 322 G727 coders RTP packetization VOICE LAX fecissieedescasseeedeiteeden essere 322 G729 coders RTP packetization VOICE LAX tscdasieeshs aos fo dteese ceshed 322 Gain 1 custom tone field 160 Gain 1 tone pair scheme 157 158
299. s GUI C Close Windows GUI D Install Java program from MultiVOIP product CD on first use only E Open web browser F Browse to IP address of MultiVOIP unit G If username and password have been established enter them when when prompted H Set browser to allow pop ups The MultiVOIP Web GUI makes extensive use of pop up windows to access screens and commands I Use web browser GUI to configure or operate MultiVOIP unit The configuration screens in the web browser GUI will have the same content as their counterparts in the Windows GUI only the graphic presentation will be different For more details on enabling the MultiVOIP web GUL see the Web Browser Interface section of the Operation amp Maintenance chapter of this manual 111 MultiVOIP User Guide Technical Configuration 8 Set Voice FAX Parameters This dialog box can be reached by pulldown menu toolbar icon keyboard shortcut or sidebar Accessing Voice FAX Parameters Pulldown Icon gt Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctri Alt I Voice Channels Interface Ctrl alt N Regional Parameters Ctrl R SMTP Parameters Ctrl Alt S Logs Traces Ctrl Alt L Supplementary Services Ctrl Alt H System Information Ctrl Alt SIP CallSignaling Ctrl Alt St P RADIUS Ctrl alt U NAT Traversal Ctrl alt Sht y amp Multi oIP Multi OIP 6 0i Configuration Interface
300. s are available in the MultiVOIP Windows GUI except for logging functions When using the web browser GUI logging can be done by email the SMTP option 90 MultiVOIP User Guide Technical Configuration Functional Equivalence of Interfaces The MultiVOIP configuration program is required to do the initial configuration that is setting an IP address for the MultiVOIP unit so that the VOIP unit can communicate with the MultiVoipManager program or with the web browser GUI Management of the VOIP after that point can be done from any of these three programs since they all offer essentially the same functionality Functionally either the MultiVoipManager program or the web browser GUI can replace the MultiVOIP configuration program after the initial configuration is complete with minor exceptions as noted WARNING Do not attempt to interface the MultiVOIP unit with two control programs simultaneously that is by accessing the MultiVOIP configuration program via the Command Port and either the MultiVoipManager program or the web browser interface via the Ethernet Port The results of using two programs to control a single VOIP simultaneously would be unpredictable 91 Technical Configuration MultiVOIP User Guide Local Configuration This manual primarily describes local configuration with the Windows GUI After IP addresses have been set locally using the Windows GUI most aspects of configuration logging functions are an except
301. s details can be viewed by clicking on an icon one for each channel arranged similarly on the web browser screen Pulldown Icon Statistics Help Ctrl alt 4 Logs hs Ctrl 0 R IP Statistics Ctrl P Registered Gateway Details Ctrl Alt W Link Management Ctrl 2 Alternate Servers Call Progress Details Shortcut Sidebar Statistics Ctrl Call Progress Alt A Logs Reports IP Statistics T1 E1 Statistics 291 Operation amp Maintenance MultiVOIP User Guide The Call Progress Details Screen 292 MultiVOIP User Guide Operation amp Maintenance 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 Hours The length of the call in hours Minutes minutes and seconds hh mm ss Seconds 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 G 711 etc this call IP Call Type H 323 SIP or Indicates the Call Signaling SPP protocol used for the call H 323 SIP or SPP IP Call incoming Indicates whether the call in Direction outgoing question is an incoming call or an outgoing call 293 Operation amp Maintenance MultiVOIP User
302. s 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 default Normal priority for ordinary LAN traffic 3 Excellent Effort The best effort type of service that an information services organization would deliver to its most important customers 104 MultiVOIP User Guide Technical Configuration Ethernet IP Parameter Definitions cont d Field Name Values Description Ethernet Parameters 802 1p 4 Controlled Load Important business continued 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 0 7 where 0 is Sets the priority for Priority lowest priority signaling packets VoIP Media 0 7 where 0 is Sets the priority for media Priority lowest priority packets Others 0 7 where 0 is Sets the priority for SMTP Priorities lowest priority DNS DHCP and other packet types VLAN ID 1
303. s in cities do not always cover easily discernible areas Organizations like businesses governments schools and universities are also commonly assigned exchange numbers for their exclusive use In some cases these organizational assigned exchanges can become non localized because the exchange is assigned to one facility and linked by the organization s private network to other sometimes distant locations d specialized codes Some proprietary voip units assign to sites and phone stations numbers that are not compatible with PSTN numbering This can also occur in PBX or key systems These specialized numbers must be handled on a case by case basis 48 MultiVOIP User Guide QS Phonebook Tips 3 Knowing When to Drop Digits Example When calling area codes and Area code for Inner London is access codes are used in listed as 0207 However in combination a leading 1 or 0 international calls the leading must sometimes be dropped 0 is dropped U K Country Code Phonebook Entry 0044207 International x ies Leading Zero Access Code Dropped from Area Code 49 QS Phonebook Tips MultiVOIP User Guide 4 Using a Comma Detail Commas are used in telephone dialing strings to indicate a pause 1 second pause to allow a dial tone to appear common on PBX and key in many PBX systems systems Commas may be used not needed in all only in the Add Prefix field of the Inb
304. s 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 182 MultiVOIP User Guide Technical Configuration Supplementary Services Definitions cont d Field Name Values Description Caller ID This is the identifier of a specific channel of the home voip unit The Caller Id field typically describes a person office or location for example Harry Smith or Bursar s Office or Barnesville Factory Default When this button is clicked all Supplementary 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
305. sLog client programs both paid and freeware can be obtained from Kiwi Enterprises among other firms Read the End User License Agreement carefully and observe license requirements See www kiwisyslog com SysLog client programs essentially give you a means of structuring console messages for convenience and ease of use MultiTech Systems does not endorse any particular SysLog client program SysLog client programs by qualified providers should suffice for use with MultiVOIP units Kiwi s brief description of their SysLog program is as follows Kiwi Syslog Daemon is a freeware Syslog Daemon for the Windows platform It receives logs displays and forwards Syslog messages from hosts such as routers switches Unix hosts and any other syslog enabled device There are many customizable options available 367 Operation amp Maintenance MultiVOIP User Guide Before a SysLog client program is used the SysLog functionality must be enabled within the MultiVOIP in the Logs menu under Configuration The IP Address used will be that of the MultiVOIP itself In the Port field entered by default is the standard well known logical port 514 368 MultiVOIP User Guide Operation amp Maintenance Configuring the SysLog Client Program Configure the SysLog client program for your own needs In various SysLog client programs you can define where log messages will be saved archived opt for interaction with an SNMP
306. se may be easier to grasp than the procedure steps Configuration is not difficult but all phone number sequences and other information must be entered exactly otherwise connections will not be made 207 T1 Phonebook Configuration MultiVOIP User Guide Phonebook configuration screens can be accessed using icons or the sidebar menu Phonebook Icons Description Phone Book leone Phonebook Configuration D ee ee amp B Phone Book Icons a A a py ke amp B Anne Danian a Phonebook E Bi S oSI Edit selected Inbound a Phonebook Entry CIETFEN Outbound Phonebook Entries List Boe A 2 e amp Ee Phone Book Icons A E m B28 2B Bi E Bute Te oi Edit selected Outbound B p B a i Phonebook Entry MultiVOIP User Guide T1 PhoneBook Configuration Phonebook Pulldown Menu Phone Book Outbound Phone Book Alt O gt Inbound Phone Boo Ls Alt I gt List Entries Ctrl L Add Entry Ctri 4 Edit Entry Ctrl E Inbound Phonebook Shortcut Outbound Phonebook Shortcut Alt I Alt O Phonebook Sidebar Menu Configuration Advanced Phone Book Outbound Phone Book List Entries Add Entry Edit Entry Inbound Phone Book List Entries Add Entry Edit Entry Statistics 209 T1 Phonebook Configuration MultiVOIP User Guide 1 Select Outbound Phone Book List Entries Fields in the D
307. seeseeeseeeeeeseeeneeeseenaes 284 System Information Screens sssini ir eirt E Sa EARE EE EN AEAEE EESE 288 STALISTICS Sereen S seori setete oeeaaeaii ra ar ibresinin ia 291 About Call Progressie issen aar ea Sea AE EESAN EEEE VESNE AEN 291 About LORS sissies abetes sashes a A E EE E AROE EENES Gani R Ea 299 MultiVOIP User Guide Contents ADOUt IP Statisties renn E RE Goan ing E E eee 306 About Link MANA geMeNnt ccccceseccessesesecessceceseceeacecuscceeacecusceeeaeecuseeeeaeecsaeeesaes 311 About Registered Gateway Details cccccssccscsscssesscreseceseesseeceseeseesecneeseceeeees 314 About Alternate Server Statistics cccccccccccccccssececessecesssscecessececusaeeeesssseenssaeees 317 About Packe tization Time ccccccccccccecssscecssccecesssececusscecessseeecseaeeecseaaesesssaeenssaeees 321 MULTIVOIP PROGRAM MENU ITEMS scesscecsseceeeeecsacceeececsaeceneecsaeceeeeecsaeeesees 324 Configuration OPTION osc ceapiet sages spea eee ats eeaeee EE AS esa aoe esas 326 Configuration Port SCtup scccsccescesscsseesseersescesseesseesecesecesecaecnaecaaecaecnaesnaeens 326 Dat and Time Setup r sa casins seats wes dag ssh uae ds egees E ose neue Sanya staged tees 327 Obtaining Updated Firmware csccesccscssscssssecnsesseeseeseesecaeeseesecaeesecueeeeenaeeeeeaees 327 Implementing a Software Upgrade ccccccsccscssceseesessescesecnseseesecaeeecneeeeensseneens 331 Identifying Current Firmware Version ccccccee
308. signments Appendix B TCP UDP Port Assignments 385 TCP UDP Port Assignments MultiVOIP User Guide 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 WwWw 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 Well Known Port Numbers Function Port Number telnet 23 tftp 69 snmp 161 snmp tray 162 gatekeeper registration 1719 H 323 1720 SIP 5060 SysLog 514 386 MultiVOIP User Guide Index Index 387 Index INDEX 802 1p Priority Levels 104 105 abbreviated dialing inter office Ebari n aiaa 243 Tlisp
309. sscssscsscessssseeseeees 374 EMC Safety and R amp TTE Directive Compliance cccccccescceesseceseceencecesneeeseees 375 FCC DECLARATION muirne eacdenis a a a a a 375 Industry CONGUE yoresi E E aes sake aang E E E et a 376 FEC Part 68 Telecom ersinnen en a R a e R 376 Canadian Limitations Notice ccccccccccccccscccccecsseceesescecessececccsaeeessuseesesaeeesneaeeess 377 WEEE StQte ment ccc vices cetscttes n a oases tes Rg sasha 378 APPENDIX A CABLE PINOUTS esseesescossesessoesesoossesoessescssosssesoesossessossesossseso 379 APPENDIX A CABLE PINOUT cccsssscssecesscecnoceesssecnecesssevensessssevonsesssevensessnees 380 Command Cable tics vie cisxccel aches cach anton castes avis ve Baste a oes ae OR 380 Ethernet Onn CtOt osese ech Rec B een Rese eae ea eed 380 TIEI Connect Trosne Recieve heteai hl eee eas Neg eee Ae en 381 Voice Fax Channel Connectors ccccccccscccccecssecesssececensececucscecessueeessssaeeesneaseenesaes 381 ISDN BRI RJ 45 Pinout Information o ceccescces cee ceee ene enee eee eeeeeeeeeeeeeeeneeneeeaes 383 ISDN Interfaces ST Gnd U unega srani doeii iiis haie dese fees 384 Contents MultiVOIP User Guide APPENDIX B TCP UDP PORT ASSIGNMENT cccsccssssssssssssccscccsssscecees 385 WELL KNOWN PORT NUMBERS 00sssesesesesescsesssesesssesesesesssesescsssesesessseseseseeeeeees 386 PORT NUMBER ASSIGNMENT LIST ccccccscscscececececececececesecesesecesececeeeseeeeesseesseees
310. strar Options C Survivability Standalone Server Allow Undefined Registrations OK Any Domains C Specific Domains Cancel Accept Registrations For Domain Names Accept Registrations For IP Addresses Re tegistration Time Note Multiple Domain names and IP addresses can be entered ae by separating with a semicolon 194 Register Any IP address Specific IP address MultiVOIP User Guide Technical Configuration SIP Server Configuration Parameter Definitions Field Name Values Description Operating Mode surviv stnd alone In Survivability mode the MVP SS unit can function as a SIP server for other gateways in its network in case that network loses contact with the network s main SIP server typically a PBX When in Survivability mode the MVP SS unit is essentially a backup SIP server In Stand Alone mode the MVP SS functions as a primary SIP server for other gateways In stand alone 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 Status Check Register Options One of two status check packets is sent to the main SIP Proxy servers to which the MVP SS serves as a backup Regardless of the packet type used this packet det
311. t 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 296 MultiVOIP User Guide Operation amp Maintenance Call Progress Details Field Definitions cont d Field Name Values Description Supplementary Services Status Call on Hold alphanumeric Describes held call by its IP address source location gateway identifier and hold duration Location gateway identifiers comes from Gateway Name field in Phone Book Configuration screen of remote voip Call Waiting alphanumeric Describes waiting call by its IP address source location gateway identifier and hold duration Location gateway identifiers comes from Gateway Name field in Phone Book Configuration screen of remote voip Caller ID There are four values Calling Party identifier Alerting Party identifier Busy Party identifier and 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
312. tc of the STUN server you will use The STUN server could be a local device or it could be a public STUN server accessible on the Internet Network Locations of SIP Servers Primary amp Alternate Go to SIP Call Signaling and enter the IP address or domain name for the primary SIP Server in your system as well as any alternate SIP servers The UserName and Password entered here will be used to authenticate all inbound phonebook entries that do not already have their own unique usernames and passwords Endpoint Info Go to SIP Server Predefined Endpoints For every other endpoint gateway to be registered with the MultiVOIP SS unit enter values for the following parameters The parameters required are different for static registrations than for dynamic registrations as shown in the table below Static Registration Dynamic Registration Endpoint Name Endpoint Name IP Address Password Port Re Registrat Interval sec 38 MultiVOIP User Guide QS Phone IP Starter Config 17 Go to Save Setup Save and Reboot Click OK This will save the parameter values that you have just entered The MultiVOIP s BOOT LED will light up while the configuration file is being saved and loaded into the MultiVOIP Don t do anything to the MultiVOIP until the BOOT LED is off a loss of power at this point could cause the MultiVOIP unit to lose the configuration settings you have made END OF
313. tem MVP3000 MVP301f EXE R Transfer rate 260 KB sec 329 Operation amp Maintenance MultiVOIP User Guide Generally the firmware file will be a self extracting compressed file with zip extension which must be expanded decompressed or unzipped on the user s PC in a user specified directory WinZip Self Extractor M YP301f EXE To unzip all files in M P301fF EXE to the specified folder press the Unzip button N Run WinZip Unzip to folder C Acme Inc MVP3000 firm Browse Close M Overwrite files without prompting Unzip x pea jems Dia Ea About Help 330 MultiVOIP User Guide Operation amp Maintenance Implementing a Software Upgrade MultiVOIP software can be upgraded locally using a single command at the MultiVOIP Windows GUI namely Upgrade Software This command downloads firmware including the H 323 stack and factory default settings from the controller PC to the MultiVOIP unit When using the MultiVOIP Windows GUI firmware and factory default settings can also be transferred from controller PC to MultiVOIP piecemeal using separate commands When using the MultiVOIP web browser GUI 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 piecemeal software upgrade whether from the Windows GUI or web browser GUI follow these steps in order 1 Ident
314. ter fields Fax Volume 0cccccccececeeees 117 Voice FAX Parameter fields Jitter Value Fax ccceeeesees 117 Voice FAX Parameter fields Mode Fax sscccccccecsesseeeee 117 Voice FAX Parameter fields Silence Compression 119 Voice FAX Parameter fields Echo Cancellation 0 06 119 Voice FAX Parameter fields Forward Error Correction 119 Voice FAX Parameter fields 413 Index Dynamic Jitter Buffer 123 Voice FAX Parameter fields Minimum Jitter Value 0 123 Voice FAX Parameter fields Maximum Jitter Value 124 Voice FAX Parameter fields Optimization Factor 0 124 Voice FAX Parameter fields Automatic Disconnection 125 Voice FAX Parameter fields Jitter Value oo cece ceeeeeeeeeeee 125 Voice FAX Parameter fields Call Duration cccceee eee 125 Voice FAX Parameter fields Consecutive Packets Lost 125 Voice FAX Parameter fields Network Disconnection 125 Voice FAX Parameters screen ACCESSING trn a E 112 Voice FAX parameters setting 112 Voip Caller ID Case 1 telco standard CID enters voip system EET 136 Voip Caller ID Case 2 H 323 voip system no telco CID 136 Voip Caller ID Case 3 SPP 137 Voip Caller ID Case 4 Remote FXS call on H 323 voip system 137 Voip Caller ID Case 5 DID channel in H 323 voip system s s s 138 vo
315. tes the typical scope of such programs Kiwi Syslog Daemon is a freeware Syslog Daemon for the Windows platform It receives logs displays and forwards Syslog messages from hosts such as routers switches Unix hosts and any other syslog enabled device There are many customizable options available 16 MultiVOIP User Guide Overview MultiVOIP Front Panel LEDs LED Types The MultiVOIPs have two types of LEDs on their front panels 1 general operation LED indicators for power booting and ethernet functions and 2 channel operation LED indicators that describe the data traffic and performance in each VOIP data channel Active LEDs On both the MVP410SS and MVP810SS there are eight sets of channel operation LEDs However on the MVP4105S only the lower four sets of channel operation LEDs are functional On the MVP8108SS all eight sets are functional C660 6660 0660 e6do Multiech 2660 0660 0660 0660 Figure 1 4 MVP 410SS 810SS LEDs Similarly the MVP210 has the general operation indicator LEDs and two sets of channel operation LEDs one for each channel Multi Veice FaX over IP Netwer lt s Voice Fsx 1 Vo celFiax 2 OOOO OOOO XMI 2CV XSG RSG XMI RZV XSG ESS Figure 1 5 MVP210SS LEDs 17 Overview MultiVOIP User Guide LED Descriptions for MultiVOIP SS Units LED NAME Front Panel LED Definitions DESCRIPTION General Operation LEDs one set on each MultiVOIP model Pow
316. that must pass before triggering the sending of a log summary email 164 MultiVOIP User Guide Technical Configuration The SMTP Parameters dialog box has a secondary dialog box Custom Fields that allows you to customize email log messages for the MultiVOIP The MultiVOIP software logs data about many aspects of the call traffic going through the MultiVOIP The Custom Fields screen lets you pick which aspects will be included in the email log reports m Custom Fields MV Select All m Fields IV Channel Number IV Start Date Time M Duration MV Call Mode IV Packets Sent IV Packets Received M Bytes Sent V Bytes Received Help IV Packets Lost V Coder Cancel I Outbound Digits Received JY Prefix Matched IV Call Status lv Call Type V Call Direction V DTMF Capability V Server Details IV Outbound digits sent JV Disconnect Reason From Details r To Details V Gateway Name V Gateway Name JV IP Address JV IP Address V Description V Description V Options JV Options Custom Fields Definitions Field Description Field Description Select All Log report to include all fields shown Channel Data channel Start Date and time the Number carrying call Date phone call began Time Duration Length of call Call Voice or fax Mode Packets Total packets sent Packets Total packets Sent in call Received received in call 165 Technical Configuration MultiVOI
317. the Caller Id field will tell the originating remote voip unit that the channel or called party is busy This field is applicable only when the home voip unit is receiving the call Example Suppose a voip system has offices in both Denver and Omaha In the Omaha voip unit the home voip unit in this example Call Name Identification has been enabled Busy Party has been enabled as an Allowed Name Type and Omaha Sales Office Voipchannel 2 has been entered in the Caller Id field of the Supplementary Services screen When channel 2 of the Omaha voip is busy but still receives a call attempt from any other voip phone station for example the Denver office the message Busy Party Omaha Sales Office Voipchannel 2 will be sent back and will appear in the Caller Id field of the Statistics Call Progress screen of the Denver voip This confirms to the Denver voip that the channel or phone station is busy in Omaha 181 Technical Configuration MultiVOIP User Guide Supplementary Services Definitions cont d Field Name Values Description Connected If the home voip unit is receiving a Party call and Connected Party is selected Allowed then the identifier from the Caller Id Name Type field will tell the originating remote CNI voip unit that the attempted call has been completed and the connection is made This field is applicable only when the home voip unit i
318. the VOIP system can be configured re configured and updated from one location using the MultiVOIP web GUI software program or the MultiVOIP program in conjunction with the built in modem 221 T1 Phonebook Configuration MultiVOIP User Guide T1 Phonebook Examples The following example demonstrates how Outbound and Inbound PhoneBook entries work in a situation of multiple area codes Consider a company with offices in Minneapolis and Baltimore The system depicted is H 323 However the phonebook entries presented are still applicable for SIP systems 3 Sites All T1 Example 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 cts death tela a at s aa al a cat Company VOIP PBX Site I l l Baltimore i Outstate MD i Overlay i 443 I iy Company VOIP PBX Site SW Suburbs 952 Baltimore 410 222 MultiVOIP User Guide T1 PhoneBook Configuration An outline of the equipment setup in both offices is shown below Pesan Local Call Area Codes 1 612 651 i 952 r Company HQ Minneapolis North Sub area 763 T1 Digital PBX voIP 7 200 2 10 3 5173 mW 717 5170 Overlay Area Code i 443 i g ao Baltimore CJ Sales Ofc area 410 R u T Digital 71 u i
319. the units using management software MultiVOIP Configuration software and confirming connectivity with another voip site This process results in a fully functional Voice Over IP network Related Documentation The MultiVOIP User Guide the document you are now reading comes in electronic form and is included on your system CD It presents in depth information on the features and functionality of Multi Tech s MultiVOIP Product Family The MultiVOIP is shipped with a printed Cabling Guide The CD media is produced using Adobe Acrobat for viewing and printing the user guide To view or print your copy of a user guide load Acrobat Reader on your system The Acrobat Reader is included on the MultiVOIP CD and is also a free download from Adobe s Web Site www adobe com prodindex acrobat readstep html This MultiVOIP User Guide is also available on Multi Tech s Web site at http www multitech com Viewing and printing a user guide from the Web also requires that you have the Acrobat Reader loaded on your system To select the MultiVOIP User Guide from the Multi Tech Systems home page click Documents and then click MultiVOIP Family in the product list drop down window All documents for this MultiVOIP Product Family will be displayed You can then choose User Guide Multi VOIP Product Family to view or download the pdf file Note that the configuration of the MultiTech home page is subject to change The current User Guide wi
320. tiate re registration with some small margin of time before the interval lapses 21 Set 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 Accessing Predefined E ndpoints Parameters ullaown con Sip Server Configuration Ctrl Alt 8 s Ctrl Alt 9 Ctrl Alt 1 Logs History Ctrl Alt 0 Shortcut Sidebar Ctrl Alt 9 Sip Server Configuration Predefined Endpoints Endpoint Statistics Logs History 197 Technical Configuration MultiVOIP User Guide 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 r SIP Server Endpoints Endpoint Name Type Re registration Interval warehousel motorpool2 quardshack3 testlab4 Static Static Static Static Add Delete Edit Save Add Predefined Endpoint Endpoint Name Password Registration Type C Static Dynamic p Contact Inf
321. tion MultiVOIP User Guide Voice Fax Parameter Definitions cont d Field Name Values Description AutoCall Offhook Alert Parameters Auto Call AutoCall The AutoCall option enables the local Offhook Offhook MultiVOIP to call a remote MultiVOIP Alert Alert without the user having to dial a Phone Directory Database number As soon as you access the local MultiVOIP voice fax channel the MultiVOIP immediately connects to the remote MultiVOIP identified in the Phone Number box of this option If the Pass Through Enable field is checked in the Interface Parameters screen AutoCall must be used The Offhook Alert option applies only to FXS channels The Offhook Alert option works like this if a phone goes offhook 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 wa
322. tion amp Maintenance 3 To download the user defaults go to Start Programs MultiVOIP xxx Download User Defaults Windows XP Professional Set Program Access and Defaults w Windows Catalog Windows Update WinZip Launch RealOne Player Programs Documents Settings Search Help and Support Run Shut Down an Accessories an Macromedia FreeHand 9 an Jasc Software fr Mozilla Firefox Mutivorp 2 06 an Microsoft Office OS Acrobat Distiller 7 0 Adobe Acrobat 7 0 Professional fH Multivorp MultivorP 2410 4 08 an Java 2 Runtime Environment IF Java Web Start IF TalkAnytime 10 08 342 MultiVOIP User Guide gt B Configuration gt B Configuration Port Setup gt BB Date and Time Setup Download Factory Defaults 2 Download Firmware 5 Download IFM Firmware ss ad efe B Set Password ey Uninstall Upgrade Software MultiVOIP User Guide Operation amp Maintenance 4 A confirmation screen will appear indicating that this action will entail rebooting the MultiVOIP Multi OIP 410 x Downloading User Defaults will Reboot the Multi OIP 410 Do you want to continue cmi Click OK 5 Progress bars will appear during the file transfer process i SSS 000 Downloading Configuration Packets Sent 2 Acks received 2 Errors 0 TTT 5 When the file transfer process is complete the Dialog IP Parameters screen will appe
323. tion authority or electrician as appropriate 377 Regulatory Information MultiVOIP User Guide WEEE Statement Waste Electrical and Electronic Equipment July 2005 The WEEE directive places an obligation on EU based manufacturers distributors retailers and importers to take back electronics products at the end of their useful life A sister Directive ROHS Restriction of Hazardous Substances compliments the WEEE Directive by banning the presence of specific hazardous substances in the products at the design phase 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 in
324. tive Compliance ee eeeeeeeeeeeeeee 375 Enable Call Fwdg Bd ERE RS 260 Tiera iia cena 219 Enable STUN field 186 Enable Call Hold eee 177 Enable Call Transfer 176 Enable Call Waiting 177 Enable Caller Name Identification 178 Enable Console Messages field 171 Enable DHCP Ethernet IP Parameters field c00 106 Enable DNS Ethernet IP Parameters Pel es ara Gvcests 109 Enable SMTP field c eee 163 Enable SRV Ethernet IP Parameters PIG ds EE AAE AENA 109 enabling SMTP eee eeeeeeeeee 162 enabling web browser GUI 111 analogion sissies essi ee 35 Endpoint Name SIP Server Endpoint Statistics Pa ramet rS arseniat 285 Endpoint Type SIP Server Endpoint Statistics ParameterS sceeeceesseceeeee 286 396 MultiVOIP User Guide Error Correction RADIUS Attributes ccccceeceeeseeeteeeeeee 192 Error Correction SMTP logs 167 error correction forward 119 error message COM port conflict 83 100 MultiVOIP SS Not Found 101 Password Phone Database Not REAd cccesscccsceseaecteese tees eetes 101 Phone Database Not Read 101 SIP Endpoint Database Not Read aaae E E A E E 101 ethernet cable pinout 380 Ethernet interface 0 ccee eee 13 Ethernet IP parameter definitions 04 105 106 107 109 Ethernet IP Parameter fields 802 1p Priority Levels 104
325. tly 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 131 Technical Configuration MultiVOIP User Guide FXS Loop Start Interface Parameter Definitions cont d Field Name Values Description Caller ID fields Enable cont d Y N 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 The Caller ID feature has dependencies on both the telco central office and the MultiVOIP phone book See the diagram series after the FXO Parameters section below 132 MultiVOIP User Guide Technical Configuration FXO Parameters The parameters applicable to the FXO telephony
326. to 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 m Add Edit Outbound Phone Book Phone Number Details Destination Pattern 0334 C Total Digits 42 Remove Prefix 0334 OC Add Prefs IP Address 200 G62 O09 OOF Description Access to Lyon area Alternate Routing Altemate IP Address S OK Round Trip Delay Cancel 256 MultiVOIP User Guide E1 PhoneBook Configuration Alternate Routing Field Definitions Field Values Description Name Alternate nnn Alternate destination for outbound data traffic IP where in case of excessive delay in data transmission Address n 0 255 Round milliseconds The Round Trip Delay is the criterion for Trip judging when a data pathway is considered Delay blocked When the delay exceeds the threshold specified here the data stream will be diverted to the alternate destination specified as the Alternate IP Address 3 Select Inbound PhoneBook List Entries Multi oIP Multi OIP v6 08 CP Firmware Jul 22 2005 Configuration Advanced Phone Book Statistics Download Connection Help EB ET ETESIE Configuration E Advanced E Phone Book E Outbound Phone Book List Entries Add Entry Edit Entry
327. tor net wtng The World Telephone Numbering Guide presents excellent international numbering info that is both broad and detailed This includes info on re numbering plans carried out worldwide in recent years to accommodate new technologies http www oftel gov uk numbers number htm UK numbering plan from the Office of Telecommunications the UK telephony authority http www itu int home index html The International Telecommunications Union is an excellent source and authority on international telecom regulations and standards National and international number plans are listed on this site 281 E1 Phonebook Configuration MultiVOIP User Guide URL Description http kropla com phones htm Guide to international use of modems http www numberplan org National and international numbering plans based on direct input from regulators worldwide Includes lists of telecom carriers per country http www eto dk European Telecommunications Office Primarily concerned with mobile wireless radiotelephony GSM etc http www eto dk ETNS htm European Telephony Numbering Space Resources for pan European telephony services standards etc Part of ETO site http www regtp de en reg_tele start List of European fs_05 html telecom regulatory agencies by country from German telecom authority 282 MultiVOIP User Gui
328. tware Installation 5 Follow the on screen instructions to install your MultiVOIP software The first screen asks you to choose the folder location of the files of the MultiVOIP software r Multi Tech Systems MultiVOIP SS 3 08 Installation Setup will install Multi OIP SS in the following folder To install to this folder Click Next To install to another folder Click Browse and select another folder Destination Folder C4 Multi Tech Systems MultiVOIP SS 3 08 Browse lt Back Cancel Choose a location and click Next 81 Software Installation MultiVOIP User Guide 6 At the next screen you must select a program folder location for the MultiVOIP software program icon m Multi Tech Systems MultiVOIP SS 3 08 Installation Setup will add program icons to the Program Folder listed below You may type a new folder name or select one from the existing folders list Click Next to continue Program Folders MultiVOIP SS 3 08 Existing Folders Accessories Administrative Tools Adobe Acrobat Jasc Software MCS12000 RasFinder 4 07 lt Back Cancel Click Next Transient progress screens will appear while files are being copied 82 MultiVOIP User Guide 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 side
329. tware version Firmware nn nn nn Indicates the version of the Version alpha MultiVOIP firmware numeric Configur nn nn Indicates the version of the ation nn nn Multi VOIP configuration software Version alpha numeric Phone Book nn nn Indicates the version of the Version alpha MultiVOIP phone book being used numeric IFM Version nn Indicates version of the IFM module alpha the device that performs the numeric transformation between telephony signals and IP signals Mac numeric Denotes the number assigned as the Address voip unit s unique Ethernet address Up Time days Indicates how long the voip has been hours running since its last booting mm ss Hardware alpha Indicates version of the MultiVOIP ID numeric circuit board assembly being used 289 Operation amp Maintenance MultiVOIP User Guide The frequency with which the System Information screen is updated is determined by a setting in the Logs screen 290 MultiVOIP User Guide Operation amp Maintenance Statistics Screens Ongoing operation of the MultiVOIP whether it is ina MultiVOIP PBX setting or MultiVOIP telco office setting can be monitored for performance using the Statistics functions of the MultiVOIP software About Call Progress Accessing Call Progress Statistics Channel Icons Main Screen Lower Left Channel icons are green when data traffic is present red when idle In the web GUI call progres
330. twork then the port number used is Working that specified in the SIP roti Request URI Universal Resource Identifier SIP URL sip userphone Looking similar to an email hostserver where aserphone is the telephone number and hostserver is the domain name or an address on the network address a SIP URL identifies a user s address In SIP communications each caller or callee is identified by a SIP url sip user_name host_name The format of a sip url is very similar to an email address except that the sip prefix is used 214 MultiVOIP User Guide T1 PhoneBook Configuration Clicking on the Advanced button brings up the Alternate Routing secondary screen This feature provides an alternate path for calls if the primary IP network cannot carry the traffic Often in cases of failure call traffic is temporarily diverted into the PSTN However this feature could also be used to divert traffic to a redundant backup unit in case one voip unit fails The user must specify the IP address of the alternate route for each destination pattern entry in the Outbound Phonebook Add Edit Outbound Phone Book Phone Number Details Destination Pattern Total Digits jo Remove Prefix Add Prefix IP Address Description Alternate Routing Alternate IP Address Round Trip Delay 215 T1 Phonebook Configuration MultiVOIP User Guide Alternate Routing Field De
331. 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 mvptlFtp bin can be transferred to update the bin file firmware residing in the MultiVOIP Similarly the file fxo_loopFtp cas could be transferred to enable use of the FXO Loop Start telephony interface in one of the analog voip units and the file r2_brazilFtp cas could be transferred to enable a particular telephony protocol used in Brazil Note however that before any CAS file can be used as an update it must be renamed to CASFILE CAS so that it overwrites and replaces the default CAS file File Type File Names Description firmware bin file mvptlFtp 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 em_winkFtp cas 12_brazilFtp cas 12_chinaFtp cas These telephony files are for Channel Associated Signaling The directory contains many CAS files some labeled for specific functionality others for countries or regions where certain attributes are standard Any CAS file used must first be renamed to CASFILE CAS
332. ue with the uninstallation process Confirm File Deletion Are you sure you want to completely remove the selected application and all of its components No 85 Software Installation MultiVOIP User Guide Confirm File Deletion 3 A special warning message similar to that shown below may appear concerning the MultiVOIP software s bin file Click Yes ReadOnly File Detected 86 MultiVOIP User Guide Software Installation 4 A completion screen will appear Maintenance Complete InstallShield Wizard has finished performing maintenance operations on the Multi VOIP Barcel Click Finish 87 Technical Configuration MultiVOIP User Guide Chapter 5 Technical Configuration 88 MultiVOIP User Guide Technical Configuration Configuring the MultiVOIP There are two ways in which the MultiVOIP must be configured before operation technical configuration and phonebook configuration Technical Configuration First the MultiVOIP must be configured to operate with technical parameter settings that will match the equipment with which it interfaces There are eight types of technical parameters that must be set These technical parameters pertain to 1 its operation in an IP network 2 its operation with telephony equipment 3 its transmission of voice and fax messages 4 its interaction with SNMP Simple Network Management Protocol network management software MultiVoipMa
333. ultiVOIP User Guide SIP Call Signaling Parameter Definitions cont d Field Name Values amp Description SIP Proxy Parameters Password Values alphanumeric Description Password for proxy server function See User Name description above Re Values numeric in seconds Registration Description This is the timeout interval for Time registration of the MultiVOIP with a SIP proxy server The time interval begins the moment the MultiVOIP gateway registers with the SIP proxy server and ends at the time specified by the user in the Re Registration Time field this field When if registration lapses call traffic routed to from the MultiVOIP through the SIP proxy server will cease However calls in progress will continue to function until they end Proxy Polling integer The interval between the voip Interval 60 300 gateway s successive attempts to connect to and be governed by a higher level SIP proxy server The Primary Proxy is the highest level gatekeeper Alternate Proxy 1 is second Alternate Proxy 2 is the lowest order SIP proxy server TTL Value The SIP proxy Time to Live value As soon as a in seconds MultiVOIP gateway registers with a SIP proxy server allowing the proxy server to control its call traffic a countdown timer begins The TTL Value is the interval of the countdown timer Before the TTL countdown expires the MultiVOIP gateway needs to register with the gatekeepe
334. und PhoneBook data for your MultiVOIP unit Note that the Advanced button gives access to the Alternate IP Routing feature if needed Alternate IP Routing can be implemented in a secondary screen as described after the primary screen field definitions below 252 MultiVOIP User Guide E1 PhoneBook Configuration The fields of the Add Edit Outbound Phone Book screen are described in the table below Add Edit Outbound Phone Book Field Definitions Field Name Values Description Accept Any Y N When checked Any Number Number appears as the value in the Destination Pattern field The Any Number feature works differently depending on whether or not an external 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 Destination 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 scre
335. unit will enter its survivability 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 a b c d The IP address at which this endpoint for can be reached values 0 255 Port 0 64000 Digital time slot on which SIP calls will be made Default is 5060 Re See Re Registration Interval entry Registration above Time 199 Technical Configuration MultiVOIP User Guide 22 View System Information screen and set updating interval optional This dialog box can be reached by pulldown menu keyboard shortcut or sidebar Accessing System Information Screen Pulldown Icon gt Multi oIP Multi OIP SS 3 08 0H Firmw Configuration Ethernet IP Parameters Ctrl Alt I Voice Channels Ctrl H Interface Ctrl alt N Regional Parameters Ctrl R SMTP Parameters Ctrl alt s Logs Traces Ctrl Alt L Supplementary Services Ctrl Alt H System Information Ctrl Alt SIP CallSignaling Ctrl ailt SFt P RADIUS Ctrl Alt U NAT Traversal Ctrl Alt Sht y Shortcut Sidebar Ctrl Alt Y El Configuration Ethernet IP Voice Fax Interface SIP Call Signaling Regional SMTP RADIUS s Traces T Traversal Supplementary Servi 2m Information 200 MultiVOIP Us
336. ut 5 T 4 Wire Input 2 Wire Input 6 R1 4 Wire Output 7 SG Signal Ground Output 8 SB Signal Battery Output 381 Cable Pinouts MultiVOIP User Guide 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 382 MultiVOIP User Guide Cable Pinouts ISDN BRI RJ 45 Pinout Information The S T interface uses an 8 conductor modular cable terminated with an 8 pin RJ 45 plug An 8 pin RJ 45 jack located on the terminal is used to connect the terminal to the DSL Digital Subscriber Loops using this modular cable The table below shows the Pin Number Terminal Pin Signal Name and Network Pin Signal name for the S T interface Pin TE Signal NT Signal Pin 1 Not used Not used 1 2 Not used C Not used 2 3 Tx Rx 3 4 R Te 4 5 Rx Tx 5 6 T Re 6 7 Not used Not used 7 8 Not used C Not used 8 TE Terminal Equipment NT Network 383 Cable Pinouts MultiVOIP User Guide ISDN Interfaces ST and U The MVP410ST and MVP810ST are ISDN BRI voip units that use an S T outlet interface You will need an NT1 device to connect these units to any network equipment that has the U ISDN interface In the UK and in many European countries the telco supplies an NT1 device for IS
337. ve 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 6 Repeat step 5 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 N If you intend to configure the MultiVOIP remotely using the Multi VOIP Windows GUI connect an RJ 11 phone cable between the Command Modem connector at the rear of the MultiVOIP and a receptacle served by a telco POTS line See Figure 3 9 The Command Modem is built into the MultiVOIP unit To configure the MultiVOIP remotely using its Windows GUI 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 GUI Command Modem connector for remote configuration TS B Slag jus D h MVP 410SS 810SS Rear Panel m E Grounding Screw Telco POTS Line Figure 3 9 MVP 410SS 810SS Voip Connections for GND amp Remote Config Modem 72 MultiVOIP User Guide Mechanical Installation amp Cabling
338. vironment of either North American telephony standards potentially operating with T1 digital MultiVOIPs or of European telephony standards potentially operating with E1 digital MultiVOIPs The configuration of the phonebook is the same in either case However because the telephony environment is different in each case and the examples used here must reflect those differences we have separate chapters for phonebook configuration in North American T1 environments Chapter 6 and for that in European E1 environments Chapter 7 this shapter Consult the chapter that best fits the needs of your voip system E1 Standard Inbound and Outbound MultiVOIP Phonebooks Important The MultiVOIP s Outbound phonebook Definition lists the phone stations it can call its Inbound phonebook describes the dialing sequences that can be used to call that MultiVOIP and how those calls will be directed When a VOIP serves a PBX system the operation of the VOIP should be transparent to the telephone end user and savings in long distance calling charges should be enjoyed Use of the VOIP should not require the dialing of extra digits to reach users elsewhere on the VOIP network 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 More importantly the VOIP system
339. wer packets to be reassembled Mode Fax FRF 11 FRF11 is frame relay FAX standard using T 38 these coders G 711 G 728 G 729 G 723 1 T 38 not T 38 is an ITU T standard for storing currently and forwarding FAXes via email using sup X 25 packets It uses T 30 fax standards ported and includes special provisions to preclude FAX timeouts during IP transmissions 117 Technical Configuration MultiVOIP User Guide Voice Fax Parameter Definitions cont d Coder Parameters Coder Manual or Determines whether selection of Auto coder is manual or automatic matic When Automatic is selected the local and remote voice channels will negotiate the voice coder to be used by selecting the highest bandwidth coder supported by both sides without exceeding the Max Bandwidth setting G 723 G 729 or G 711 are negotiated Selected G 711 a u_ Select from a range of coders with Coder law 64 specific bandwidths The higher the kbps bps rate the more bandwidth is G 726 used The channel that you are 16 24 32 calling must have the same voice 40 kbps coder selected G 727 nine bps Default G 723 1 6 3 kbps as rates required for H 323 Here 64K of G 723 1 digital voice are compressed to 5 3 kbps 6 3K allowing several simultaneous 6 3 kbps conversations over the same G 729 bandwidth that would otherwise 8kbps carry only one Net Coder To make selections from the 6 4 7 2 8
340. when SIP protocol is used CID Name field will duplicate value in CID Number field Anoka Whse VP3 Phone Book Configura ion 763 743 5873 Gateway Name Anoka W hse VP3 m Q 931 Parameters Inbound Phone Book Channel 2 Gatekeeper RAS Paral Remove Prefix Add Prefix Forward Addr 423 748 Figure 5 5 Voip Caller ID Case 5 Call through telco central office without standard CID enters DID channel in H 323 voip system 138 MultiVOIP User Guide Technical Configuration FXO Supervision When the selected Interface type is FXO the Supervision button is active Click on this button to access call answering supervision parameters and call disconnection parameters that relate to the FXO interface type 139 Technical Configuration MultiVOIP User Guide FXO Supervision Parameter Definitions Field Name Values Description 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 offhook 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 gon
341. will be required to enter the password to gain access to the web browser GUI and any part of the MultiVOIP software listed in the Program group menu User Name and Password are both needed for access to the FTP Server residing in the MultiVOIP Password Yerification When MultiVOIP program asks for password at launch of program the program will simply shut down if CANCEL is selected The MultiVOIP program will produce an error message if an invalid password is entered Multi OIP eA 346 MultiVOIP User Guide Operation amp Maintenance Setting a Password Web Browser GUI Setting a password is optional when using the MultiVOIP web browser GUI Only one password can be assigned and it works for all MultiVOIP software functions Windows GUI web browser GUI FTP server and all Program menu commands e g Upgrade Software only the FTP Server function requires a User Name in addition to the password After a password has been set that password is required to access the MultiVOIP web browser GUI NOTE Record your user name and password in a safe place If the password is lost forgotten or unretrievable the user must contact MultiTech Tech Support in order to resume use of the MultiVOIP web browser GUI MultiVOIP 410 Configuration Advanced Phone Book Statistics Change Password Save amp Reboot Password Change Logout Help User Name default OK Old Password New
342. 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 Number UDP logical port number See RFC3087 The voip will listen for SIP Control of messages at this logical port Servi If SIP is used 5060 is the Context using default standard or well SIP Request known port number to be URI by the used If 5060 is not used Network then the port number used is Working that specified in the SIP roti Request URI Universal Resource Identifier SIP URL sip userphone Looking similar to an email hostserver where aserphone is the telephone number and hostserver is the domain name or an address on the network address a SIP URL identifies a user s address In SIP communications each caller or callee is identified by a SIP url sip user_name host_name The format of a sip url is very similar to an email address except that the sip prefix is used 255 E1 Phonebook Configuration MultiVOIP User Guide Clicking on the Advanced button brings up the Alternate Routing secondary screen This feature provides an alternate path for calls if the primary IP network cannot carry the traffic Often in cases of failure call traffic is temporarily diverted in
343. wn below 268 MultiVOIP User Guide E1 PhoneBook Configuration Configuring Digital amp Analog VOIPs in Same System Analog MultiVOIP units like the MVP 210 410 810 are compatible with digital MultiVOIP units like the MVP3010 In many cases digital and analog VOIP units will appear in the same telephony IP system In addition to MVP 210 410 810 MultiVOIP units Series II units legacy analog VOIP units Series I units made by MultiTech may be included in the system as well When legacy VOIP units are included the VOIP administrator must handle two styles of phonebooks in the same VOIP network The diagram below shows a small scale system of this kind one digital VOIP the MVP3010 operates with two Series II analog VOIPs an MVP210 and an MVP410 and two Series I legacy VOIPs two MVP200 units EXAMPLE ien Digital amp Analog VOIPs Inner London UK in Same System Area Code 0207 va 200 2 9 9 Digital VolP IMVP3010 FR otnor extensions R 8301 x8399 af Pa 020 7398 8300 7 Site E ito A Carlisle UK biniien He ieee UK ar ee Oe Area Code 0121 Series 1 Analog MultiVOIP Server Client Phonebook Unit l 1 l 1 1 200 cm PS FO 1 I 200 2 9 7 201 EROAA Client ie i ee A E te Site F Site B ite B a Tavistock UK Reading Berkshire UK A APARO URR N Area Code 0178 200 2 9 5 Sf A a a 1 Series 1 Analog MultiVOIP
344. x field 0 ccceeeeeeeee 117 Mode statistics logs field 301 modem relay eeeeeseeceereeeneeeeee 124 modem traffic on voip network 124 modem command and Regional Parameters Country Selection cece 96 154 modem remote configuration command Setup FOF oe eee eeeeeeeeeenees 96 154 Monitor Link fields Link Management Statistics Sree ea wyatt 312 MOUNt NE 0 0 ceeeeeseeeeeeceeeeeeeeeeeees 14 mounting in rack sses 66 procedure fOF eceeeeeeeeees 68 Salety E ETT 62 67 mounting OPTIONS 0 eect eters 9 MultiVOIP FAQ on MTS web site 8 MultiVOIP Program Menu items 324 MultiVOIP Program Menu options Configuration sses 324 Configuration Port Setup 324 Date amp Time Setup 0 0 324 Download Factory Defaults 325 Download Firmware 325 MultiVOIP User Guide Download IFM Firmware 325 Set Password 325 Uninstall 2 ste orn sipre einsi 325 Upgrade Software eee 325 MultiVOIP program menu option descriptions s s s 324 325 MultiVOIP software Installing oe ee eeeeeeeteeeeeeeeeeees 78 location of files 1 0 0 ee eee 8l program icon location 82 uninstalling eee eeeeeeee 85 348 MultiVOIP software moving around iN ceeee 101 MultiVoipManager ee eee 90 musical jingle during call transfer 176 MVP210 QLOUNCING 0 eo ea 76 UNPACKING iniii tiretta 65 MVP210 SS cabling procedure 7
345. y unanswered rings are needed to trigger the forwarding Registration In an H 323 voip system gateways can Option register with the system using one of these Parameters identifiers a an E 164 identifier b a Tech Prefix identifier or c an H 323 ID identifier In a SIP voip system gateways can register with the SIP Proxy In an SPP voip system gateways can register with the SPP Registrar voip unit 5 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 261 E1 Phonebook Configuration MultiVOIP User Guide Remember that the initial MultiVOIP setup must be done locally or via the built in Remote Configuration Command Modem using the MultiVOIP program However after the initial configuration is complete all of the MultiVOIP units in the VOIP system can be configured re configured and updated from one location using the MultiVOIP web GUI software program or the MultiVOIP program in conjunction with the built in modem E1 Phonebook Examples To demonstrate how Outbound and Inbound PhoneBook entries work in an international VOIP system we will re visit our previous example in greater detail It s an international company with offices in London Paris and Amsterdam In each office a MVP3010 has been connected to the PBX system 3 Sites All

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