5 IP Office 500v2 – overview 2 – 384 users Modular, flexible platform 4 slots built-incan connect to 8 expansion modules, or 12 when using 4-port expansion cardSupports classical and IP telephony:Analog, digital and IP (SIP & H.323) setsAnalog, ISDN and SIP trunksOptional software applications to increase user productivity and customer service
6 IP500v2 – front & back LAN IP: 192.168.42.1 / 255.255.255.0 WAN IP: /
7 Base card with trunk card mounted: Insert base card in to Control Unit Base/Daughter Cards:Base card with trunk card mounted:Insert base card in to Control UnitStation Card is called Base Card.Trunk Card is called Daughter Card.Trunk card is mounted on top of Station card. (except Combo trunk card)
8 BASE AND DAUGHTER CARD OPTIONS IP Office 500v2 cardsBASE AND DAUGHTER CARD OPTIONSRange of interface modules:Base cards8 Port Digital Station (max. 3)8 Port TCM (Norstar/BCM)2 or 8 Port Phone Station32 or 64 Channel VCM Module* (max. 2)4 port expansion card (max. 1)does not take daughter cardTrunk ‘Daughter cards’ fit on top of a base card in the same slot4 Port Analog Trunk Module2 or 4 Port BRI (4 or 8 channels)Single or Dual PRI( 8 channels per PRI enabled, other channels are licensed)Combination cards (max 2) :6 Digital, 2 Phone, 10 VCM, 2 port BRI6 Digital, 2 Phone, 10 VCM, 4 port Analog TrunkC110 Unified Communications module (max 1):Linux server card with Preferred Edition, one-X Portal and IM/Mobility server- SD card for licensing, Embedded Voic and firmware/config filesA closer look at the customization of the IP Office 500v2 system.*IP phones also require an IP Endpoint license, 12 licenses included with VCM32/64
9 Bus architecture Digital phones Analog phones Analog / ISDN trunks TDM BusPacket BusDS portsPOTS portsline portsVoice mailMoHConference bridgeIP extIP lineIP hard/soft phonesIP trunks
10 Card port numbering Base card (extensions) Daughter card (trunks) 1 3 5791124681012
11 IP Office 500 expansion modules Choice of expansion modules:DS 16, DS 30: provide 16 or 30 digital extensionsDS16A, DS30A: provide 16 or 30 BCM/Norstar extensions (RJ21)Phone 16, Phone 30: provide 16 or 30 analog extensionsATM16: provide 16 analog trunksMax 8 connected through ports at the back of the IP500v2, additional 4 with the 4-port expansion base card
12 System SD Card & Software Licensing fits directly on IP Office 500v2always required, also if no licenses are neededhas unique serial number (watermark)License Key (RFA) Code is a 32-Character String Based On:unique serial number of dongle (10 digits)feature / software being enabled
13 Application Trial License. Enables applications for 60 daysTrial operates from license generation dateTrial Packages:Advanced Edition, Preferred Edition, Multisite.VMPro Additions: Voice Networking, TTS (1).Power User (5), Teleworker (5): Mobile Worker (5),Office Worker Profile (5), Receptionist (1)CCR Supervisor(1) and Agent (5).Avaya IP Endpoints (5).3rd Party IP Endpoint (5).SIP Trunk Channels (1).Contact Store™ includes a 45 day trial facility.Ordered as a standard ADI license keyTrial license can be ordered on a certificateTrial will run for 45 days from the generation date of the license NOT the installation dateThe Trial License will not be invoicedUn-expired trail licenses will be transferred as part of a maintenance swap (dongle swap)Once the license expires the applications will cease to operateThis license can only be ordered once per Dongle or replacement DongleLicense key enquiries (print and download) will display expired trial license (i.e.: visible as to who has had a trial)
18 SD card /primary /backup /lvmail /doc /dynamic /temp Contains firmware files, music on hold files, license key files, configThis is the main set of files used by the IP Office system when booting up/backupContains a copy of the primary folder at some previous point/lvmailthe system prompts used by embedded voic ./AAG folder: auto-attendant greetings/docContains initial installation documentation/dynamicContains files used by the IP Office and retained through a reboot of the IP Office system/lvmail: individual user and group mailbox messages, name recordings and announcements (15 hours max)/tempTemporary files, not retained through a reboot
19 Boot processIP Office looks for IP500v2.bin file using the following order:1.System SD card /primary folder.2.System SD card /backup folder.3.Optional SD card /primary folder.4.Optional SD card /backup folder.If all fails, BOOTP is usedConfig and other firmware files are used from the same folderIf system starts from a different folder than 1, this will result in an alarm. Config will be read-only.
24 Install IPO Manager Login: User Name: Administrator Password: AdministratorInstall Admin/User DVD to load IPO Manager application to manage IP Office
25 System Settings Set system Locale to adjust trunk parameters System System Name for IdentificationSet Locale to adjust Trunk ParametersIPO address for IP phone firmware upgradeSet system Locale to adjust trunk parametersSet TFTP & HTTP Server IP to upgrade IP phone firmware
26 LAN Settings Default LAN1 IP: 192.168.42.1 DHCP is ON by default System LAN1 LAN Settings LAN1 IP Address and MaskDHCP PoolDefault LAN1 IP:DHCP is ON by default
27 LAN Settings Option to lease IP to Avaya IP phones only System LAN1 DHCP Pools Avaya IP Phones OnlyDHCP PoolOption to lease IP to Avaya IP phones onlyUp to 8 DHCP Pools can be added
28 Default Configuration User Name:AdministratorPassword:LAN1:IP address – ;Mask – ;DHCP Server: Yes;Pool Size: 200LAN2 / WAN:IP address – ;Default Starting Extension:201 onwardHunt Group:200; where first 10 users are the membersIncoming Call Route:Voice calls are routed to hunt group 200 membersOutgoing Call Route:Calls can be dialed out with out any prefix
29 License – the 32-character key License Key (RFA) Code is a 32-Character String Based On:unique serial number of dongle (10 digits)feature / software being enabledIPOv2 Demo kit comes with Licenses available on a CD.
30 License – the 32-character key License License Tip: Copy Keys from .xls file and Paste in to License menu
31 Manager - directoriesConfig files (.cfg) and backup files (.BAK) automatically stored in Working DirectoryBin file directory used as source for upgrades through Manager
32 IP Office Upgrades - online Shows version available on the PC. Must show 8.0.xCheck to copy system files on to SDCardInstall new CD (ADMIN)Upgrade through Upgrade Wizard or using SD cardUpgrades can be performed over the WAN
33 IP Office Upgrades – using SD card Upgrades can be done by copying all the required files to the SD card, and the rebooting the systemPossible methods:Using System SD card locally on PC (quickest method!)Shutdown System SD card and remove from systemRecreate the SD card from Manager (File -> Advanced -> Recreate SD card).Insert the SD card in the IP Office and reboot the systemUpgrade from Optional SD cardRecreate the Optional SD cardFrom Embedded File Management, select File-> Upgrade Binaries to copy the files to the System SD card. The IP Office will be restarted automatically.Using Manager / Embedded File ManagementSelect File -> Backup SystemSelect File -> Upload System Files
35 Security Settings – Manager Users & Passwords, secure communication with IPO Standard login: security / securitypwdSeveral levels of access to manage the IPO configurationUsers & passwords are stored on IPO, separate from configCan be changed using „security settings“ in ManagerAlso allows setting password complexity and aging rules, as well as secure comms (TLS) to IPO
36 Ebedded File Management View and manage files stored on the SD Card
37 Manager Offline Configuration Prepare config file without connecting to actual system
39 ExtensionsPhysical extensions are created automatically, and cannot be added or deletedIP extensions can be added manually or automatically
40 Extensions (continued) 2 to 9 digits, max 7 digits for IP extensionsBASE EXTENSION = directory number of default associated userafter restart, IPO will try to login the previously logged in user (unless user has „force login“ set).Module / Port number indicate location of physical port (0 / 0 for VoIP extensions). BP= base unit, analog. BD = base unit, digital
41 Digital Extensions Physical extensions are Plug-and-Play Extension Extension BD1: Base, Digital, Module 1BP1: Base, Analog Phone, Module 1Physical portPlug-and-Play AppearPhysical extensions are Plug-and-PlayAutomatically detects Extension and add Users for each
42 IP Extensions DHCP or Static IP Need Own IP address Call Server IP address & portHTTP server (firmware upgrades & settings file download)Extension (stored on the set)IP phones need to register. With “auto-create“ extensions enabled, IP phones can register with any free extension number that the user enters.Static addressing: press “*“ when prompted during startupNote: standard PROCPSWD on 96xx: 27238View settings (16xx) : <MUTE> VIEW #Clear settings (also EXT NUMBER, 16xx): <MUTE> CLEAR #On 96xx: access VIEW/CLEAR procedures through:<MUTE> PROCPSWD #, so by default: <MUTE> #
43 IP Extensions Set TFTP & HTTP Server IP to upgrade IP phone firmware System System IPO address for IP phone firmware upgradeFirmware is stored on SD CardRequired by Soft Video phoneSet TFTP & HTTP Server IP to upgrade IP phone firmware
44 IP Extensions To add User automatically when IP extension is setup System LAN1 VoIP To auto create User for IP Extn.To add User automatically when IP extension is setup
45 IP Extensions By default DHCP Server is ON. System LAN1 DHCP Pools If DHCP to be used only for PhonesDHCP PoolBy default DHCP Server is ON.Optionally, DHCP can lease IPs to Avaya Phones only
46 IP Extensions Change TFTP/HTTP Server IP as IPO Summary:Change TFTP/HTTP Server IP as IPOSelect “H323 Auto-Create User”Set DHCP Option (Optional)Reboot PhoneSpecify Extension on the phoneTo view 16xx phone Config: <MUTE> VIEW#To reset 16xx phone Config: <MUTE> CLEAR#To view/reset 96xx phone Config: <MUTE> #To edit configuration: Press * upon reboot
47 Users Users can be used for extensions, but also for RAS By default, a user is automatically created for each extensionsUsers do not have to have a physical extension (hot desking)
48 Users - Profile User User User Profile User productivityAllows to select type of user whether Receptionist, Mobile Worker etcUser productivity require license
49 Users – appearance buttons Call Appearance Buttons - used to alert the phone user of calls to their extension or hunt group. Multiple call appearance buttons allow the user to handle multiple calls simultaneouslyCall Coverage Buttons - can alert the user when a selected colleague has an unanswered call.Bridged Appearance Buttons - shows the user the status of one of a colleague's call appearance buttons. They can then answer and make calls on the colleague's behalf.Line Appearance Buttons - show the user when a particular IP Office line is in use. The button can also be used to answer and make calls on that line.
50 Users - Phone Button Programming User Select User Button Programming Double click to assign functionDestination ExtensionCall will ring first on “201” and then after an interval will ring on “202”Select Action as “Coverage Appearance” or “Bridged Appearance” to check the difference
51 Users – Hot Desking User Select User Telephony Supervisor Settings Add Hot-Desk passwordLogin Code: *35*<Extn>*<Password># Logout Code: *36Hot-Desking allows users to operate from any phone with their original extension
52 User Rights (templates) Set & lock user settings in a template
53 User Rights – for button prog. User Rights New Add new User GroupAdd new group to assign specific feature or restrictions to set of users
54 User Rights – for button prog. User Rights User Rights Button Programming Select “Apply User Rights Value”Add required button featuresAdd required feature buttonsSelect “Apply User Rights Value” from drop down to lock values
55 User Rights – for button prog. User Rights User Rights Membership Members of this User Rights Select user membersSelect users who are going to receive the button programming
57 Phone Manager Lite PC Call Control (Phone Manager Lite) Dial In/Out TransferConferenceBLF IndicationSpeed DialCall LogVoic ……...NOTE: PM Pro/ Softphone are EndOfSale in March 2012
58 Phone Manager Select Phone Manager Option: Lite Install User Software. (USER4_2_43.exe)Reboot PC/LaptopLogin with User and Password to start managing calls
59 CTI Lite configuration Install TAPI from User CDConfigure TAPI driver:Control panel -> Phone and Modem options > AdvancedSelect „Avaya IP Office TAPI2 Service Provider“Configure usingIP Office IP addressUser name of the IP Office user (NOT the extension number!)User password, if configured on the User / User tabReboot PC or restart Windows Telephony Services after making changes
60 CTI Lite: click-to-dial from Outlook Contacts Autodialer buttonIP Office extensionSelect „Auto-Dialer from contactFrom „Dialling Options“: Select IP Office extensionClick „Start Call“
61 Softconsole Requires Receptionist license User User User Profile Requires Receptionist licenseEasy call transfer using drag & drop
62 Softconsole Start Programs IP Office SoftConsole User name/ extension to be controlled
63 Video Softphone Requires Teleworker or Power User license Required by Soft Video phone User User User Profile System System “Enable Softphone HTTP Provisioning”Requires Teleworker or Power User licenseUses existing user‘s extension, no need to create additional VoIP extension
64 Video Softphone Start Programs IP Office IP Office Softphone IP Office User name & password
65 Example 3rd party SIP Softphone: X-Lite Requires 3rd-part IP Endpoint licenseNeed to create User + VoIP extensionSIP account details:User name = IP Office Extension NumberPassword = Login code (User -> Telephony -> Supervisor Settings)Authorization user name = IP Office user nameDomain = IP Office IP addressConfiguration details for tested devices can be found in the „SIP Extensions“ manual
66 One-X Portal User User User Profile Requires Windows/Linux Server as the server, a browser as the clientRequires Office Worker, Teleworker or Power User licenseTelecommuter mode not available with Office Worker
68 Hunt GroupsA hunt group is a collection of users accessible through a single directory numberCall Presentation: Call ringing fashionAvailability: Additional user assignmentQueuing: Keep calls in queue if users are busyAnnouncements: MoH adjustmentOverflow: Redirect calls to another Hunt GroupVoic Redirect calls to Voic s
69 Hunt Groups - Add Specify hunt group name HuntGroup HuntGroup Hunt Group Create a new hunt groupName the Hunt GroupAssign unique Extension“Collective” to ring simultaneouslySpecify hunt group nameAssign unique extension number to the hunt groupChange ring mode as appropriate
70 Hunt Groups - Members Edit Hunt Group members HuntGroup HuntGroup Hunt Group Edit Edit Hunt Group membersSelect Users and click AppendEdit Hunt Group members
71 Hunt Group - Overflow HuntGroup HuntGroup Hunt Group Overflow Group List Set overflow delayAdd available overflow groupAdd OverflowBeyond Overflow Time call will be presented to Overflow group usersUpon exceeding Voic Answer Time, call will redirected to voic
73 Lines (trunks) Analog ISDN BRI / PRI IP (SIP & H.323) IP DECT Routing on trunk channels is done according to their:Incoming Group IDOutgoing Group ID
74 Analog Lines Lines Analog Line Line Settings Id to which Line belongs.Analog line showed as overhead cable iconLine group to dial outPrefix to incoming number for callbackGroup Id suggested to keep separate for different set of trunks. Exa: 0 for PSTN and 1 for PRIPrefix automatically placed in front of incoming calls for direct callback.
75 Analog Lines Lines Analog Line Analog Options Signal delay from Telco to clear the callFor lines with CLIDFor external call transferTo fix echo problemsEnable modem for remote managementTrunk Type set as “Loop Start ICLID” for lines with CLID. Otherwise set as Loop Start. If not connected; then set as “Out of Service”Disconnect Clear delay to receive clear signal from telco
76 Analog Lines Busy Tone Detection parameters depends upon Locale set System Telephony Tone & Music Busy Tone Detection Single FrequencyLine frequencyOn cadence as per line specOff cadence as per line specBusy Tone Detection parameters depends upon Locale setTo avoid line disconnect issue, Single freq, On/Off width must match with line parameters
77 Analog Lines System System Locale CustomizeTone Plan 2 for UKEnable Busy Tone DetectionImportant: Locale sets telephony plus external line parameters.
78 PRI Lines Lines Line PRI Line PRI line showed as Cat5 cable in blue colorLine group to dial outId to which Line belongsPrefix to incoming number for callbackGroup Id suggested to keep separate for different set of trunks. Exa: 0 for PSTN and 1 for PRIDefined prefix automatically placed in front of incoming calls for direct callback.
79 PRI Lines Lines Line PRI Line PRI line showed as Cat5 cable in blue colorAdjust channels in case Fraction PRIAdjust number of PRI channels; if client has 10-channel fraction PRIIPO require license above 8-channels on a PRI card
80 SIP Trunking – Technical View IP Office Managed ViewSIPProviderSIP phoneDECT R4B2BUAControlCallUASIPProviderH323IP phoneB2BUADCPDigital phoneB2BUAAnalogAnalog phoneSIPProviderIP Office acts a SIP gateway i.e. no need for a SIP phone to make a SIP call with IP OfficeBenefits: SIP trunking 20% to 30% than traditional carriersSIP trunking requires VCM cards + SIP Trunking RFARTP relay is also supported (avoids using VCM resource for IP phones once call is set up)UA (User Agent) is an entity in SIP that generates and terminates sessions.B2BUA (Back to Back User Agent) is an entity in SIP that generates and terminates sessions.IP Office acts a SIP gateway i.e. no need for a SIP phone to make a SIP call with IP OfficeIncoming calls are routed based on IP Office incoming call routesCalls can be matched against ‘Incoming’ (from) and ‘Called’ (to) addressesSIP Trunks are licensed in increments of 1, 5, 10 and 20Additional sizes are available by combining licences
81 Why NATs break VoIP (part of the story) Public InternetNATIP OfficeITSPSIP MessageThe SIP Message, at a higher layer, untouched by the NAT, tells the ISP to set up the RTP toSIP MessageSIP message from 192…1 to 217…180NAT replaces the source address 192…1 with 217…186RTP/RTCPRTP/RTCP✘The ITSP Is expecting RTP fromRTCP & RTP from 192…1 to 217…180NAT replaces the source address 192…1 with 217…186The ITSP Is sends RTP to which miss-routes in the internet✘SIP message from 192…1 to 217…180
82 Request NAT information Retrieve NAT information SIP network topology using STUNPSTNSTUN serverRequest NAT informationSIP Service ProviderRetrieve NAT informationNAT Firewall / RouterSIPIP OfficeLANInternetConnection to the ITSP over NAT using 3rd-party STUN servers in the networkSTUN (Simple Traversal of UDP through NAT) allows to discover the NAT mechanism being used
83 SIP trunks - templatesTemplates are XML files stored in Manager \Templates subfolderTo activate Templates in Manager: File -> Preferences -> Visual Prefernces -> Enable Template OptionsTo enable Template Creation, go to HKEY_CURRENT_USER\Software\Avaya\IP400\Manager and add a new DWORD Value “TemplateProvisioning” if it does not already exist and set its value to 1..
84 SIP trunks Note: license is required for SIP trunks . Lines SIP Line SIP Line SIP server domain name or IP addressSIP line showed as Cat5 cable in red colorNote: license is required for SIP trunks.
85 SIP trunks Lines SIP Line Transport STUN activation (none = not active)
86 SIP trunks Lines SIP Line SIP URI this form links SIP URIs to Incoming / Outgoing Group IDs, so incoming calls to this URI can be routed to a certain destination, or outgoing calls can be placed using a certain URI.
87 SIP trunks On this form the SIP account credentials are entered Lines SIP Line SIP Credentials Provider requires registration?On this form the SIP account credentials are enteredMax. 30 registration accounts per SIP line allowed.
88 SIP trunks: STUN configuration System LAN1 Network Topologyif STUN is used, the STUN server IP address needs to be entered here. “Run STUN” will do a STUN request and will display the results in Manager.
89 SIP trunks: IP routing IP Route IP RouteIP routing needs to be defined to route the outbound SIP traffic through the appropriate gateway. The static route as shown here (with for address & mask) is used to define the default gateway.
92 Incoming Call Route - Form (1) Incoming Call Route 0 Standard Default route “0”Incoming call trunk Line GroupMatches the digits provided by telcoCallers ICLIDSelect Music source
93 Incoming Call Route - Form (2) Incoming Call Route 0 Destinations Routing scheduleEnter manually or select option to redirect calls
94 Incoming Call Route – DID setup Incoming Call Route Incoming Call Route Standard New entryDID number with x wild cards.Exa: 404xxxx for DID range toWild card “x” is used to represent single digit
95 Incoming Call Route – DID setup Incoming Call Route Incoming Call Route Destinations Enter “#” to match all wild cards in incoming numbers# to match all wild cards in incoming DID numbers and route to respective extension
96 Incoming Call Route – DID Setup Tools MSN Configuration Enter first DID numberDigits presented by SPMSN ConfigurationExtension MappingNumber of DIDsPRI/BRI line IdUsed to populate Incoming Call Route with range of DID numbersApplicable on PRI & BRI trunks
98 Short Codes USED FOR: Speed Dials Feature activation/deactivation Call Routing and RestrictionSHORT CODE LEVELS:UserUser Restriction (group of users)SystemARSLineUsed to match the number dialed with an action.
99 Short Codes – Outgoing Call flow used internal extension?YExtension number will be dialledNUser Short Code ?YDial feature routes to ARS form„Dial“featureNUser Rights Short Code?YARS Short Code?„Dial“featureNSystem Short Code ?YY„Dial“featureDial feature routes to trunk directlydial number on specified trunk
100 Short Codes Short Code Short Code Short Code Default entryLine group id to dial out numbersPredefined short codesDefault entry allows to dial out any unmatched short codes through hunt group “Main”In case or Prefix, delete and add a new entry to dial out through default ARS rule “50:Main”
101 Short Codes – with prefix 9 Short Code Short Code Short Code Default entryCreate new entryNumber output of shortcodeLine group id to dial out numbersEntry allows 9 as a prefix followed by numbers to dial out through default ARS rule
102 Short Codes – trunk access (through ARS) Default entry(no trunk access code)0 for trunk access0 as prefixPass numbers followed by 0
103 Short Codes – call restriction Block all international callsNumbers starting with 00Return busy toneRestrict calls with passwordCalls can be restricted using password. Require Force account code checked and passcode via Account Code menu.
104 Short Codes – feature access Speed DialDo Not Disturb
105 Short Codes – feature access Call Pickup AnyForward Number
106 Short Codes - hot desking feature Use own profile from any phoneExample: extension 201, login code 1234:Log In: *35*201*1234#Log Off: *36
107 Short Codes – dial paging To Page on Extensions or Hunt GroupsAdd new short code. Exa: *55*N# / Dial Paging / NExample: To page on hunt group 200Dial: *55*200#
108 Automated Route Selection (default entry) ARS Main ARS Offer secondary dial tone after prefixUnknown digitsPass digits as it isDial out through line id 0ARS executes results received from short code.Default entry allows to dial out unmatched numbers through lines with group id 0.
109 ARS entry for SIP callsN; is used to wait for complete dialled number before sending out on the trunkThis format is required to route the dialled number correctly to the providerSIP calls require special formatting in the ShortCode entry.
110 Short Codes Processing System Telephony Telephony Delay between digitsCount to start short code executionDial Delay Time = 1000 msDial Delay Count = 4Short Code 1 = 01Short Code 2 = 0123
112 Music On Hold Up to 4 sources (max. 1 external) possible System Telephony Tones & Music Incoming Call Route 0 Standard MoH file nameWAV:<MoH file name>Select Music source for incoming callsUp to 4 sources (max. 1 external) possiblePCM, 8kHz 16-bit, mono, maximum length 90 seconds Initial download from TFTP serverAutomatically stored on SD card
123 System Status Application (SSA) The System Status Application (SSA) is a diagnostic tool for system managers and administrators to monitorand check the status of IP Office systems locally or remotely. SSA shows both the current state of an IP Officesystem and details of any problems that have occurred. The information reported is a combination of realtimeevents, historical events, status and configuration data to assist fault finding and diagnosis. SSA providesreal-time status, historic utilization and alarm information for ports, modules and expansion cards on thesystem. SSA connects to all variants of IP Office running release 4.0, using an IP connection that can beremote or local. Modem connections at 14.4kbps or above are supported for remote diagnostics.SSA provides information on the following:· AlarmsSSA displays all alarms which are recorded within IP Office for each device in error. The number, dateand time of the occurrence is recorded. The last 50 alarms are stored within IP Office to avoid need forlocal PC.· Call DetailsInformation on incoming and outgoing calls, including call length, call ID and routing information.· ExtensionsSSA details all extensions (including device type and port location) on the IP Office system. Informationon the current status of a device is also displayed.· TrunksIP Office trunks and connections (VoIP, analog and digital) and their current status are displayed. ForVoIP trunks, QoS information is also displayed (e.g. round trip delay, jitter and packet loss)· System ResourcesIP Office includes central resources that are utilized to perform various functions. Diagnosing theseresources is often critical to the successful operation of the system. This includes details on resourcesfor VCM, Voic and conferencing.SSA can be launched independently or from IP Office Manager and there can be up to two (2) SSA clientsconnected to an IP Office unit at one time.Note: SSA is not a configuration tool for IP Office systems. For information on configuration, refer to IP OfficeManager.Login Id: AdministratorPassword: AdministratorDiagnostic tool to check the status of IPO system including real-time state and problems occurred on Calls, Extensions, Trunks, System Resources
124 Monitor Diagnostic tool to troubleshoot trunk related issues. Used to capture traces and then sent to Avaya Backbone for analysis
126 SMDR – Station Message Detail Recording System SMDR Output SMDR OnlyServer to pull CDR recordsBuffer in case of failureCALL DETAIL RECORDING (CDR)For IP Office customers that want to capture simple call details, the system can output Call Detail Records(CDR) to a designated IP address and port. The records that can be included by IP Office CDR output are listedbelow:· Date RecordsA date record is sent each time a CDR connection is started and then once a day (at midnight). The datecan be in month/day or day/month format, as selected on the System | CDR/SMDR tab.· Call Detail RecordsCall detail records are sent at the termination of a call (in 5 second increments). For some formats,additional fields can be selected using the Normal, Enhanced, or ISDN options on the System | CDR/SMDR tab.Depending upon the selected report format and options, there are a number of different fields available withinthe CDR. The fields are described in the IP Office Manager documentation.STATION MESSAGE DETAIL RECORDING (SMDR)For more formal call logging and reporting, the IP Office Station Message Detail Report (SMDR)output is usedby third party applications for many call accounting applications. IP Office SMDR provides much greater detailsof the call, including duration, ring time, hold time, and transfer information.From Release 4.2 onwards, IP Office can output SMDR events directly as well as through a separate Windowsservice included in the IP Office Delta Server application. To generate SMDR events directly, choose SMDRfrom the System | CDR/SMDR tab in Manager.The IP Office Delta Server (SMDR) application is provided on the Admin portion of the IP Office CD/DVD set. Itallows the detail of all calls to be sent to a file on the PC. Both methods allow the details of calls to be sentover an IP network to a TCP/IP port.Third party applications use this data to allocate costs to departments, analyze trunk capacity, report usageagainst account codes etc. One IP Office SMDR (Delta Server) is required for each site requiring the use of callaccounting software. Please refer to the Technical Specifications section for the Delta Server requirements.Sample IP Office SMDR Information Output
127 SMDR – Station Message Detail Recording HyperTerminal IPO addressSame port as SMDR in IPOHyperTerminal can be used as a CDR server to pull records and display real-time
129 IP Networking Small Community Networking H.323 and Q.SIG High integration between sitesProprietaryH.323 and Q.SIGBasic functionalityOpen standards
130 IP Networking – H323 Trunks PSTNPSTN6+2xxH323 TrunkABSys ASys B2xx2xxHunt group 1Huntgroup 2VoIP over H.323 / QSIG over E1T1 / Standards Based TDMDesk-to-Desk dialingCalling / Connected Party NumberCalling / Connected Name PresentationCall HoldCall Transfer130
131 IP Networking – H323 Trunks on Sys-A Lines Create a New Record VoIP Line Create new entryH323 Line as Cat5 cable in red colorId to which Line belongsLine group to dial outPrefix to add in ICLIDSystem-A:Group Id = 2 to call in/outPrefix = 6 to reach Sys-B
132 IP Networking – H323 Trunks on Sys-A Lines Create a New Record VoIP Settings Remote IPO ipH450 as H323 TrunkSystem-A: parameters of Sys BG/W IP = as remote IPOSupplementary Services = H450
133 IP Networking – H323 Trunks on Sys-A Short Code Create a New Record Short Code Create new entry6 as prefixDial number as it isH323 line id to dial outSystem-A: parameters of Sys BCode = 6NFeature = DialTelephone Number = NLine Group ID = 2 as H323 trunk idAdd short code to reach Sys B via prefix 6
134 IP Networking – H323 Trunks on Sys-B Lines Create a New Record VoIP Line Create new entryH323 Line as Cat5 cable in red colorId to which Line belongsLine group to dial outPrefix to add in ICLIDSystem-B:Group Id = 2 to call in/outPrefix = 6 to reach Sys-B
135 IP Networking – H323 Trunks on Sys-B Lines Create a New Record VoIP Settings Remote IPO ipH450 as H323 TrunkSystem-B: parameters of Sys AG/W IP = as remote IPOSupplementary Services = H450
136 IP Networking – H323 Trunks on Sys-B Short Code Create a New Record Short Code Create new entry6 as prefixDial number as it isH323 line id to dial outSystem-B: parameters of Sys ACode = 6NFeature = DialTelephone Number = NLine Group ID = 2 as H323 trunk idAdd short code to reach Sys A via prefix 6
137 IP Networking – H323 Trunks PSTNPSTN6+2xxH323 TrunkABSys ASys B201202201202Hunt group 1Hunt group 2Test Connectivity:From Sys-A dial to reach 201 on Sys-BFrom Sys-B dial to reach 201 on Sys-AUniform dialing plan is possible through prefix in case of H323 trunk137
138 IP Networking - SCN IP Up to 1000 users across 32 IP Office sites Voice Networking between IP Office sites over IP links (using Linked numbering plan)Call PagingCall Pick-upCall Back when FreeCamp-onConferenceCentralized Voice Mail (using Voic Pro)Dynamic Internal Users DirectoryDistributed Groups across networkDesk-to-desk dialingAbsent Text MessageAnti-TromboningBusy Lamp FieldCall HoldCall TransferCall ForwardHot desking across networkUp to 1000 users across 32 IP Office sitesSCN requires Multisite option on IP500
139 IP Networking – Extension Renumber Tools Extension Renumber Extension RenumberNumber rangeTo changeAllows to renumber extensions in a certain rangeSCN support unique dialing plan
140 IP Networking – SCN on Sys-A Lines Create a New Record VoIP Settings Remote IPO ipIPO Office SCNSystem-A: parameters of Sys BG/W IP = as remote IPOSupplementary Services = IP Office SCN
141 IP Networking – SCN on Sys-B Lines Create a New Record VoIP Settings Remote IPO ipIPO Office SCNSystem-B: parameters of Sys AG/W IP = as remote IPOSupplementary Services = IP Office SCN
142 IP Networking – SCN Fallback Lines Create a New Record VoIP Settings Remote IPO ipIPO Office SCN – FallbackBackup serviceSCN Fallback allows to fail over Ip phones, hunt groups, voic s (Pref. Edition) services3-minutes delay in fallback
144 Embedded Voicemail - Visual Voice System Voic Voic Type Control voic s through phone displayEmbedded VoicSelect voic type as Embedded VoicMessage Button to Visual Voice or User Button (Emulation -> Visual Voice)Max. 6 voic channels and up to 15 hours of Voic s are stored on the SD Card
145 Embedded Voicemail – User setting User User Voic Default is ONVoic passwordVoic to idPersonal Auto AttendantPersonal Auto Attendant: press *0, *2 or *3 to transferSMTP Integration to send Voic as
146 Embedded Voicemail – Access Press *17 from phone to access voic menuLogin using passwordRecord greetingRetrieve voic sTip:To prompt for Mailbox ID and Password, modify in *17 short code “Telephone Number” as “?”Voic s are deleted once user checks themTo retain the voic Save option must be selected
147 Announcements Record with *91N; and *92N; ShortCodes User User Announcement Enable AnnouncementRecord with *91N; and *92N; ShortCodesExa: *91201# for 1st annoucement on extn 201Exa: *92201# for 2nd annoucement on extn 201Announcement played on calls waiting to be answered & before sending to voic
149 Auto Attendant – add Time Profile Time Profile Create New Time Profile 1. Specify name3. Select the period2. Add recurring periodAdd a new time profile to play Auto Attendant during specified period
150 Auto Attendant Up to 40 Auto Attendant entries Auto Attendant Auto Attendant Auto Attendant Name to AADirect Dial-by-NumberSchedule to play AAShort code to record greetingsUp to 40 Auto Attendant entriesNew feature Direct Dial-by-Number
151 Auto Attendant Add actions to keys Auto Attendant Auto Attendant Actions Add actions to keysTransfer call to another AAAvailable actionsAdd actions to keysSystem detects Fax tones to route fax calls
152 Auto Attendant Short Codes Short Code Short Code Auto AttendantAA:<AAName>Internally to test: create Shortcode with Feature: Auto Attendant and Telephone Number:AA:<name of your Auto Attendant>
153 Auto Attendant Incoming Call Route 0 Destinations AA:<AAName> to direct incoming calls to AADirect calls to auto attendant using Incoming call routeDestination: AA:<name of your Auto Attendant>
154 LVM Greeting Utility Used for AutoAttendant Greetings only Files Advanced LVM Greeting Utility .wav file to be converted Auto Attendant Auto Attendant Auto Attendant Case sensitive nameSelect greetingUsed for AutoAttendant Greetings onlyConvert WAV (PCM/Uncompressed,8000 KHz, 16bit, Mono) to special embedded VM formatTransfer .c11 file in to “\dynamic \ lvmail \AAG“ folder on SDCard
156 VM Pro Installation Use VM Pro CD Select required languages from Wizard„Web Voic “ (UMS) option only appears if IIS installed„Allow VM Pro service to interact with desktop“ can give helpful troubleshooting info for lab purposesTTS on separate CDs
157 Voicemail Type Change Voicemail Type as “Voicemail Lite/Pro“ System Voic Voic Type Voic Lite/ProVM Pro Server IPChange Voic Type as “Voic Lite/Pro“Specify Preferred Editition (VMPro) server IP as Voic IP Address
158 Basic Auto AttendantNeeds ShortCode in Manager: Telephone Number = CallFlow name, Feature = Voic nodeOr Incoming Route Destination: VM:CallFlowName??? for 3-digit number$ for any number
159 Basic Auto Attendant Login to VMPro Server to create call flow Start Programs IP Office Voic Pro Client Log InVM Pro LoginVM Pro Server IPLogin to VMPro Server to create call flow
160 Basic Auto Attendant Open VM Pro Client and add Call Flow module Start Programs IP Office Voic Pro Client 2. Call flow name. Exa: AA1. Add call flow moduleOpen VM Pro Client and add Call Flow module
161 Basic Auto Attendant Add Menu action with Touch Tones Start Programs IP Office Voic Pro Client 4. Select Menu from Basic Actions3. Select call flow5. Double click and select Touch Tones6. Select 1,2,3Add Menu action with Touch Tones
162 Basic Auto Attendant Add Transfer action for each menu touch tone Start Programs IP Office Voic Pro Client 7. Add Transfer action two timesTransfer action for 1,2 touch tone menuAdd Transfer action for each menu touch tone
163 Basic Auto Attendant Add name and destination for each Transfer action Start Programs IP Office Voic Pro Client 8. Double click9. Name each action10. Specify destinationAdd name and destination for each Transfer action
164 Basic Auto Attendant Start Programs IP Office Voic Pro Client 13. Add Transfer action and double click11. Add a new Menu and double click12. “???” as wildcard14. “$KEY” as DestinationAdd a new Menu action with “???“ Toch Tone for Dial-by-NumberAdd Transfer action with “$KEY“ as destination for Dial-by-Number
165 Basic Auto Attendant Join all actions using Connector Start Programs IP Office Voic Pro Client 15. Select Connector icon17. “Save & Make Live”16. Connect all menu actions with respective transfer actionsJoin all actions using ConnectorSave changes and make call flow <AA> live
166 How to test Auto Attendant - Internal Short Codes Short Code Short Code Voic CollectCall flow name. Exa: AAInternally to test: create Shortcode with Feature: Voic Collect and Telephone Number: <name of your Call Flow>
167 How to test Auto Attendant - External Incoming Call Route 0 Destinations VM:<CF Name>to direct incoming calls to AADirect calls to auto attendant using Incoming call routeDestination: VM:<Call Flow Name>
173 Server Edition Demo Kit option Place a regular order for the pre-staged Server Edition server, 2 options:270395 DL120G7 SRVR IPO R8.1+ ME EXP $3.500,00270393 DL360G7 SRVR IPO R8.1+ ME PRIMARY $8.100,00The shipment will have the System ID of the server printed on a label on the box.Request the required demo licenses for the Server Edition System ID using Appendix B of the standard 8.1 Power Demo Kit document.https://avaya.my.salesforce.com/sfc/servlet.shepherd/version/download/ iGKNAA2A maximum of 4 Server Edition licenses can be requested
174 Demo Tool: Demo Anywhere Server running on LinuxPreconfigured image for VMWare playerAllows demonstration of UC features on a laptop:VM ProOne-X PortalSoftphoneFlare CommunicatorOne-X MobileReceptionistIP phones / SIP trunksFully configurablehttps://avaya.my.salesforce.com/apex/sp_ViewDetailPage?Id=a3j LEBuAAO
178 Avaya University Training Training available from Avaya University:Avaya Connect Competency Model has 3 SME credentials:Avaya Professional Sales Specialists (APSS)Avaya Certified Implementation Specialist (ACIS)Avaya Certified Solutions Specialist (ACSS)
180 Transition Skills to IP Office 8 Transition Skills to IP Office 8.1 IP Office Avayalive™ Engage Technical CenterCourse NumberCourse NameDurationAvailabilityDescription1S00010EIP Office Technical Center18 hrsNowAllows students to find content on IP Office ACIS & ACSS certification - Access to 9 hrs of free pre-requisite knowledge on IP Office basics of implementation, maintenance and troubleshooting is provided with ACIS or ACSS registration - Find additional self paced content: IP Office Technical Delta Release IP Office Technical Delta Release IP Office Technical Delta Release IP Office Essential Edition / Partner Version IP Office Video Installation IP Office CCR and other optional topicsIP Office Technical Center Delta Course Collection 1S00010E: Contains 7.0, and 8.1 deltas with a 6 or 12 month subscriptions to AvayaLive™ Engage; $459 US List; targeted at PA Implement transition3 month Free subscription to AvayaLive™ Engage with ACIS/ACSS registration8.0 Delta Bundle 4S00003G: Includes the 8.0 delta and all refreshed ACIS and ACSS courseware on DVD for self paced learning; $433 US List8.1 Slipsheets with additional content now in Learner Kits, ACIS/ACSS courseware8.1 Delta WBT 4S00006: Refresh your knowledge on the new release; $125 US ListAvaya Inc. – Proprietary. Use pursuant to the terms of your signed agreement or Avaya policy.
181 IP Office Server Edition Technical Training 4-hour web-based course: 4S00006W IP Office Release 8.1, Mid-Market and Linux - Technical Overview
182 ACIS – Avaya Certified Implementation Specialist Virtual and instructor led Boot CampsIP Office Technical CenterSelf-paced DVD’s
183 ACIS Test – Pearson Vue www.pearsonvue.com/avaya Tests are taken at a Pearson Vue testing Center – you must complete this step to achieve the ACIS Certification