When all agents in an ACD Group are unavailable, an incoming call will queue and cause the Queue Status Display to occur on the ACD Group Supervisor and/or agent’s display. The display helps the supervisor keep track of the traffic load within their group. In addition, any display keyset can have a Queue Status Display Check programmable function key. The keyset user can press this key any time while idle, and using the VOL and VOL , scroll through the Queue Status Displays of all the ACD Groups. The Queue Status Displays shows (see the Queue Status Display illustration below): The number of calls queued for an available agent in the group. The trunk that has been waiting the longest, and how long it has been waiting. The number of calls in queue. 2 LINE-001 01:30 Name of trunk that has been queued the longest. How long the longest queued call has been waiting. For each ACD Group, you can set the following conditions: The number of trunks that can wait in queue before the Queue Status Display occurs. How often the time in queue portion of the display reoccurs (see the Queue Status display Timing illustration below). Queue Status Display holding time. Queue Status Alarm enable/disable. Queue Status Alarm sending time.
U-NT and U-LT loops
U-NT and U-LT loops can be used in combination to provide D-packet service for a point-of-sale terminal adapter (POSTA) or other D-packet device. D-packet service is a 16 kbps data transmission service that uses the D-channel of an ISDN line. To deliver D-packet service, a network connection (U-NT) is programmed to work with a terminal connection (U-LT). The loops must be on the same physical card. For example, if the network connection is a loop found on the BRI Card in Slot 1, the terminal connection must be a loop found on the same card. S reference point The S reference point connection provides either a point-topoint or point-to-multipoint digital connection between Norstar and ISDN terminal equipment (TE) that uses an S interface. S loops support up to seven ISDN DNs, which identify TE to the ICS.
Networking Norstar
Networking with Norstar There are a number of ways you can network Norstar systems together, or network Norstar systems with other Nortel systems into private networks. What types of lines you use to perform the networking will determine the type of services that can be shared between systems. Keep in mind that each node (Norstar system) is considered an external system by every other node within the network, even though, to the users, it appears to be all one system. This affects how you configure call transfer and call out features on each system. On the home node, all features are configured as local numbers. On all other nodes, all features directed to the home node are configured with external numbers. As well, each node must have a unique identifying code. What this code will be, and how it is configured for the user, depends on what type of trunks and dialing rules you choose to use. If the network has a Meridian as part of the network, the Meridian administrator will determine identification codes for the systems. This section describes various configurations of private networks. The general settings that are required to set up the home node for each system are provided to give you a sense of what is required for each type of network. The common goal is to provide the user with the sense that the network is one large system that provides common access to colleagues in other buildings, cities, or countries. In some systems they may need to enter a destination code before the local number to route the call to the correct system. In other systems, using a common dialing plan allows users to dial colleagues at any location simply by entering the same number of digits they would use to dial a colleague at the next desk.
SIE keys for ACD Groups
Any Multiline Terminal can have SIE keys for ACD Groups. When a call comes into a covered ACD Group, the SIE key will ring immediately, ring after a delay or just flash (depending on system programming and user-set options). The Multiline Terminal user can answer the call by just lifting the handset and pressing the SIE key. ACD SIE keys help maximize ACD service during high traffic periods or when agents are unavailable. The covering extension does not have to be a member of the ACD Group. In addition, an extension can have SIE keys for as many ACD Groups as it has available programmable keys. An ACD Group SIE key also allows for one-button Transfer to an ACD Group. Conditions Ringing for SIE keys may need to be programmed through the telephone
DISA
Place call to DISA facility of the system. 2. At receipt of Intercom dial tone/AA announcement, dial as desired. If DND Warning tone is received, enter an Authorization Code, to receive Intercom dial tone. Conditions 1. Each CO/IP path is separately assigned for DISA operation during Day, Night and/or Timed system operation mode. DISA operation is active only when the system is in the assigned operating mode(s). 2. DISA callers can be routed to a VMIM/VSF Auto Attendant announcement in place of Intercom dial tone. The announcement can be associated with a CCR Table or assigned to disconnect after playback (‘#’). 3. A DISA caller can be required to enter an Authorization Code to access the system’s external outgoing resources, facilities or features. If required, the caller is permitted to retry entry of a valid Authorization Code based on the DISA Retry count. Continued failure results in disconnect. 4. DISA callers are subject to COS dialing restrictions. If Authorization Codes are required and the code entered matches a Station Authorization Code, the station’s COS will apply. Otherwise, the assigned DISA COS will apply. In both cases, the CO/IP COS for the outgoing path will be active. 5. The system will disconnect an outgoing DISA call if the Unsupervised Conference timer expires or disconnect supervision is received. A disconnect warning tone is provided 15 seconds prior to disconnect. 6. If a DISA caller encounters a system All Lines Busy, busy tone is received for 5 seconds before ICM Dial tone is again presented and the DISA caller may try another call. 7. LEDs associated with the DISA CO Line appearance will provide normal status indications at all stations except the Attendants. The LED for the line at an Attendant will flutter at 240 ipm when busy. 8. An iPECS Phone user can only receive a DISA call with an available CO/IP appearance button.
DISA
Place call to DISA facility of the system. 2. At receipt of Intercom dial tone/AA announcement, dial as desired. If DND Warning tone is received, enter an Authorization Code, to receive Intercom dial tone. Conditions 1. Each CO/IP path is separately assigned for DISA operation during Day, Night and/or Timed system operation mode. DISA operation is active only when the system is in the assigned operating mode(s). 2. DISA callers can be routed to a VMIM/VSF Auto Attendant announcement in place of Intercom dial tone. The announcement can be associated with a CCR Table or assigned to disconnect after playback (‘#’). 3. A DISA caller can be required to enter an Authorization Code to access the system’s external outgoing resources, facilities or features. If required, the caller is permitted to retry entry of a valid Authorization Code based on the DISA Retry count. Continued failure results in disconnect. 4. DISA callers are subject to COS dialing restrictions. If Authorization Codes are required and the code entered matches a Station Authorization Code, the station’s COS will apply. Otherwise, the assigned DISA COS will apply. In both cases, the CO/IP COS for the outgoing path will be active. 5. The system will disconnect an outgoing DISA call if the Unsupervised Conference timer expires or disconnect supervision is received. A disconnect warning tone is provided 15 seconds prior to disconnect. 6. If a DISA caller encounters a system All Lines Busy, busy tone is received for 5 seconds before ICM Dial tone is again presented and the DISA caller may try another call. 7. LEDs associated with the DISA CO Line appearance will provide normal status indications at all stations except the Attendants. The LED for the line at an Attendant will flutter at 240 ipm when busy. 8. An iPECS Phone user can only receive a DISA call with an available CO/IP appearance button.
CIX License Control
CIX License Control The system size and feature capability is controlled using a software License Key Code. This key code is obtained from the Toshiba Internet FYI site during the ordering process and is installed onto the system processor via CIX eManager. Processor license codes activate system hardware capacities. Additional sets of four CO line/digital station ports beyond the Basic bundled number of ports requires one LIC-4 BASIC license. See table below. CIX System Processor Basic Bundled Port Licenses Maximum Ports CIX670 CIX200 CIX100 CIX100-S CIX40 BCTU2A LCTU1A ACTU3A ACTU3A-S GCTU2A 64 32 32 16 192 or 6721 192 112 1122 LIC-4 BASIC license not required 1. The BEXU2A sub-assembly can be added to expand capacity from 192 to 672 ports. 2. The upgrade from 16 to 24 ports and from 24 to 32 ports requires the eight port upgrade LIC100S-8 PORTS license. Each additional set of 4 line/station ports requires the four port upgrade LIC-4 BASIC license (maximum of 112 ports). DTMF tone receiver circuits are required for standard telephones, Voice Mail DTMF integration, Tie, DID and DNIS line service. 16 DTMF built-in receiver hardware circuits and 16 ABR circuits – The first four DTMF circuits and all ABR circuits do not require a license. Each additional set of four DTMF receiver circuits require one LIC-4 DTMF license (maximum of 16 DTMF circuits). IP End Point licenses are required for IPT, SIP and SoftIPT phones. The optional RS-232 serial port interface (BSIS) provides two circuits to interface with SMDI or Toshiba Proprietary Voice Mail integration, Call Accounting SMDR, and two for future applications. The first circuit does not require a license, but circuits two through four each require one LIC-SER PORT license. Refer to the “Strata CIX Software License Requirements” on page 177 for license part numbers and hardware configurations.
PSTN switching
To assure that the PSTN switching equipment has sufficient time to restore to the idle condition, the system will hold analog CO lines in a busy state to users after release of a CO line by a station. The time between the station disconnect and when the system changes the CO line status from busy to idle is the CO Line Release Guard time. Operation System Operation of this feature is automatic. Conditions Programming SYSTEM Related Features
System Administration
Launching the System Administration Application 1. 2. 3. 4. To start the System Administration application, perform one of the following steps: Double-click on the MERLIN Messaging Administration desktop short cut or the PARTNER Messaging Administration desktop short cut. Select the MERLIN Messaging Administration short cut or the PARTNER Messaging Administration short cut from the Start menu. From the Start menu, select Programs > MERLIN Messaging Release 4.0 or PARTNER Messaging Release 7.0>System Administration. (This is the default location.) The Messaging System Administration window appears, displaying the Messaging Login dialog box. In the IP address or host name box, enter the Fixed IP address of the messaging system module that was loaded into the message system. (See Section 2.) In the Login box, enter sysadmin. In the Password box, enter the system administration password. (If this is your first time logging into the system, click the OK button. You are prompted to enter the password.) 5. Click the OK button. Once you log in successfully, you can start administering the messaging system. The System Administration application windows display the current settings for the messaging system. For more information regarding installation and use of the System Administration application, see the MERLIN Messaging System Administration Getting Started Guide or the PARTNER Messaging System Administration Getting Started Guide under Documentation, System Administration contained on the Library CD
SIE Key for ACD Groups
SIE Key for ACD Groups Description Any Multiline Terminal can have SIE keys for ACD Groups. When a call comes into a covered ACD Group, the SIE key will ring immediately, ring after a delay or just flash (depending on system programming and user-set options). The Multiline Terminal user can answer the call by just lifting the handset and pressing the SIE key. ACD SIE keys help maximize ACD service during high traffic periods or when agents are unavailable. The covering extension does not have to be a member of the ACD Group. In addition, an extension can have SIE keys for as many ACD Groups as it has available programmable keys. An ACD Group SIE key also allows for one-button Transfer to an ACD Group.
Linked stations
Linked-Pair Station Description A station can be logically linked to a primary station so that the two stations function as a single station. When linked, the two stations effectively act as a single station with the station attributes of the primary station. The status of one station is reflected in the status of the other and features activated at one are active at the other. All internal or external calls to a linked pair station will ring both stations. All features available to the primary station are available and controllable by the secondary station, one station may activate Call Forward and the other may cancel the forward. The displays of the linked stations will display the status of the linked pair. If one of the linked stations is busy, the LCD of the other station will display “IN USE AT LINK STA”. When a linked station is busy, the other idle linked station will not receive ring for CO lines, transferred ring or intercom calls. Consideration A station can be linked with only one station. Any combination of DKTs and SLTs may be assigned as Linked pairs. A DSS Console, Door Phone Box or Port Blocked Station may not be assigned as a linked pair station. Linked pair stations are treated as having a single station number for all features including LCD displays, station programming, ADMIN access, ACD statistics, SMDR, etc. Intercom calls to the Linked stations always signal in the Tone ring mode and cannot be changed using the Forced Hands-free feature. The station attributes of the secondary station will follow attributes of the primary station, i.e. Day/Night COS, CO Warning Tone, CO Auto Hold, CO Call Drop, DID Call Waiting, Speed Access, Alarm, VMIB Access, DND, FWD, Paging, CO Access, CO Ring, etc. An Attendant station can be linked with another station but, the linked station cannot use attendant features (refer to Ref. D). Calls can be placed or transferred between the stations of a Linked pair using the primary station number.
ARS time
NetworkOutgoingInter-DigitARSTimeWithNetworking,thistimereplaces20-03-04whendeterminingifallnet-workprotocoldigitshavebeenre-ceived.IfARSisenabledatSiteB,thistimecanbeprogrammedfor5(500ms)atSiteA.IfARSisdisabledandSiteBisusingF-Routeforout-bounddialing,thistimeshouldbeprogrammedfor30(threeseconds)atSiteA.
Transferring Calls
Transferring Calls to an Attendant Normally calls are directed to an auto attendant by an IP Office incoming call route. However it can also be useful to transfer calls received at an internal extension to an auto attendant. You can transfer calls to an Auto Attendant by: · Using Programmed Buttons · Using Phone Manager · Using SoftConsole · Using Short Codes 40. 40 39.. 39. Using Programmed Buttons On Avaya phones with programmable buttons, those buttons can be programmed to access auto attendant services. To create an auto attendant button: 1.From the IP Office system configuration, set the action of one of the users programmable buttons to Dial. 2.Set the associated telephone number to AA:Name where Name matches the name of the auto attendant. 3.Save this configuration change back to the IP Office. When the user receives a call they want to transfer to the auto-attendant, they can use a programmed button. To transfer a call using the programmed button: 1.Place the call on hold. 2.Press the button programmed for the auto-attendant. 3.Hang-up the call at their extension. This will cause a blind transfer of the held call to the auto-attendant. Using Phone Manager To create an auto attendant speed dial: 1.Within the user's Phone Manager, click the Speed Dials tab. 2.Right-click the speed dial panel and select New > Speed Dial Group Member. The Speed Dial window opens. 3.In the Name field, enter a name for the Auto Attendant. 4.In the Number field, enter AA:Name where Name matches the name of the auto attendant. 5.Click OK. To transfer a call using the Speed Dial: 1.During a call that you want to transfer to the auto attendant click Hold to place the call on hold. 2.Click the Speed Dials tab. 3.Click the speed dial created for the auto attendant. 4.Click Complete Transfer to transfer the held caller.
Programmable Buttons
Line/Programmable Buttons. Used for individual outside lines or (if no line is assigned on a button) for programming telephone or extension numbers, or system features (such as Last Number Redial). When a line is assigned, press the line button to make a call on that specific line (lights show status of line). When a number feature is programmed, press the button to dial the number or use the feature. The PARTNER-34D has 36 programmable buttons (32 with lights and 4 without lights); the PARTNER-18D has 20 programmable buttons (16 with lights and 4 without lights); the PARTNER-18 has 16 programmable buttons (all with lights); the PARTNER-6 has 4 programmable buttons (all with lights). ■ Fixed Buttons. In addition to the line buttons, the telephones have some or all of the following fixed buttons, which are already imprinted: — Intercom Buttons. Press to make (or answer) a call to (or from) another extension in the system. If you receive a call on a T1 line with Direct Inward Dialing (DID), and you cannot access that line from the line or pool buttons on your telephone, the call will appear on your Intercom button. Press the button to answer this outside call. — Feature. Press to change programmed settings or use system features. — Conf (Conference). Press to add other parties to your call. — Transfr (Transfer). Press to pass a call to another extension. — Hold. Press to put a call on hold. — Spkr (Speaker). Press to turn on and off the speaker and microphone (if available), so you can dial and have a conversation without lifting the handset. The light next to this button shows when the speaker is turned on. — Mic/HFAI. Press to turn the microphone on and off. The light next to this button shows when the microphone is turned on. Leave on to use Hands-Free Answer on Intercom (HFAI) feature. — Volume Control Buttons. Press – to decrease or + to increase the volume as follows: ■ To adjust ringer volume, press while the telephone is idle and the handset is in the cradle. ■ To adjust speaker volume, press while listening to a call through the speaker. ■ To adjust handset volume, press while listening through the handset. ■ To adjust background music volume, press while listening to music through the telephone’s speaker.
eMG80
The iPECS eMG80 is a richly-featured hybrid IP/TDM communications platform for voice and mobility services, optimized for small and growing businesses. With its modular and flexible design, businesses can easily and affordably expand into premium UC and more sophisticated enterprise applications.
Feature name
SL1100 Feature NameV1.0V1.5V2.0V3.0Loop KeysSSSSMaintenanceSSSEMeet Me ConferenceSSSSMeet Me PagingSSSSMeet Me Paging TransferSSSSMemo DialSSSSMessage WaitingSSSSMicrophone CutoffSSSSMobile ExtensionSSSSMobile Extension - Callback to Mobile PhoneSSSSMultiple Trunk TypesSSSSMusic on HoldSSSSName StoringSSSSNavigation KeySSESNight ServiceSSSSOff-Hook SignalingSSSSOne-Touch CallingSSSSOperatorSSSSPaging, ExternalSSSSPaging, External (VRS)SSSSPaging, InternalSSSSParkSSSSPBX Compatibility/Behind PBXSSSSPC ProgrammingSSSEPC Programming - Intuition SetupSSSEPower Failure TransferSSSSPrime Line SelectionSSSSPrivate LineSSSSProgrammable Function KeysSSSSProgramming from a Multiline TerminalSSSSPulse to Tone ConversionSSSSRedial FunctionSSSSRemote (System) UpgradeSSSSRepeat RedialSSSSResident System ProgramSSSSReverse Voice OverSSSSRing GroupsSSSSRingdown Extension (Hotline), Internal/ExternalSSSSRoom MonitorSSSSSave Number DialedSSSSSecondary Incoming ExtensionSSSSSecretary Call (Buzzer)SSSSSecretary Call Pickup
full power of iPECS
Unleash the full power of iPECS, and experience the next generation in business communications technology. With a full range of powerful software applications, the true potential of your iPECS voice platforms can be realized. As your business grows and your technology needs become more sophisticated, Ericsson applications leverage your current iPECS investment. Applications deliver features that empower your employees to be more productive, more mobile and more collaborative. They can also enhance your business’ ability to deliver a more responsive and superior customer service experience beyond traditional voice-only customer contact. Whether your business is faced with the continued adoption and resulting challenges of “bring-your-own-device” (BYOD), increased need for Unified Communications (UC) or the complexity of managing a geographically dispersed and disparate voice and data network, iPECS applications
Data channel
channel (Data channel): An ISDN standard transmission channel which is packet-switched, and is used for call setup, signalling and data transmission. Data channel: See D channel. Data Communications Interface (DCI): A Norstar device that allows you to attach an RS-232 data device to the Norstar system. data terminal: A device, such as a modem, that can be used to transfer data instead of sound over a telephone network. You cannot use Norstar programming to set up such devices. See the documentation that accompanies the device. date: See Show Time or Time and Date. defaults: The settings for all Norstar features when the system is first installed. Settings are changed from their defaults in programming. In this manual, default settings are shown in bold text. Delayed Ring Transfer (DRT) to prime: After a specified number of rings, this feature transfers an unanswered call on an external line, to the prime telephone associated with that line. This feature is activated under Feature settings in Sys prgrmmng. destination code: A two- to 12-digit number that the system interprets and then translates into the digits that you want dialed out. Both the code and its associated dialed digits are assigned under Routing service in Services programming. DID trunk: See Direct Inward Dial trunks. DID Trunk Cartridge: The Trunk Cartridge that allows you to connect DID trunks to the Norstar system. dialing restriction: See Restriction filter. dialing modes: ≤•°¤ This feature allows you to set the dialing mode of your telephone. Norstar supports three dialing modes: Automatic Dial, Pre-Dial, and Standard Dial. All three modes support on-hook dialing, meaning you can dial a call without picking up the receiver. The special features of the Automatic and Pre-Dial modes are available only when you dial on-hook. Digital Mobility phones 74XX: These telephones connect to the system through station modules connected to a Nortel Networks Digital Mobility controller. Digital Trunk Interface: The Trunk Cartridge connects digital T1 AND ISDN trunks to the Norstar system. Direct-dial: A feature that allows you to dial a designated telephone in your Norstar system with a single digit, such as the main receptionist. As many as five direct dial sets can be established. Each telephone in the system is assigned to one direct-dial telephone. There is a single, system wide digit for calling the assigned direct-dial telephone of any telephone. Direct-dial telephones are established in System programming. Telephones are assigned to a direct-dial telephone under Capabilities in Terminals&Sets programming. Direct-dial #: A digit used system- wide to call the Direct-dial telephone. The digit is assigned under Access codes in Sys prgrmmng. Direct-dial number: The digit used
System response to an incoming DID call
System response to an incoming DID call (analogue CO line): 1. set-up a connection based on the defined Start signal, 2. collect incoming digits based on the programmed Receive Digit Count, 3. handle digits based on the Conversion type (0-2), 4. route the call to assigned destination. System response to an incoming DID call (ISDN line): 1. set-up a connection based on the received call set-up messages, 2. collect incoming digits and delete digits from left based on the programmed ISDN Remove Digit Count, 3. handle digits based on the Conversion type (0-2), 4. route the call to assigned destination. Conditions 1. If ICLID routing is assigned for the CO/IP Line, the received Caller Id is compared to the ICLID Table for routing first. If Caller Id does not match an entry in the ICLID Table, the normal DID call processes are used. 2. DID calls that encounter a busy signal, are not answered in the DID/DISA No Answer Timer, or are received at a vacant or invalid number can be routed to the Attendant, a tone, Station group, or VMIM/VSF announcement. When the Attendant receives such calls, the call is appropriately identified by the Attendant iPECS Phone display. 3. For a station that is part of a non-pilot Station Hunt group, DID calls will follow the group hunt process if the Station is busy or does not answer the call. 4. DID calls are subject to Group Call Pick-up and Directed Call Pick-up. 5. If a VMIM/VSF announcement is defined as the destination in the Flexible DID Destination Table, a Caller Controlled Routing Table for the announcement can be defined. iPECS can be configured to drop (disconnect) the call after playing the recorded announcement.
eMG80
The iPECS eMG80 is a richly-featured hybrid IP/TDM communications platform for voice and mobility services, optimized for small and growing businesses. With its modular and flexible design, businesses can easily and affordably expand into premium UC and more sophisticated enterprise applications.
PRI
Features and Specifications The Total Access 900e Series products have the following features: • Support for 4 DS1 (or 3 DS1 plus 1 PRI/CAS, or 2 DS1 plus 2 PRI/CAS) interfaces • Support for a single built-in FXO interface • Support for up to 24 FXS ports with octal FXS daughter board • Support for up to 16 FXS ports and 8 FXO ports with octal FXO daughter board (Total Access 924e only) • Supports Primary Rate ISDN (PRI) or Robbed Bit Signaling (RBS) on the PRI/CAS interfaces • Support for a two auto MDI/MDX 10/100BaseT Ethernet ports (RJ-48C) • Full-featured AOS IP router/firewall • QoS/NAT/DHCP client, server, and relay • Support for SIP trunks • Support for up to 6 Mbps of multi-link Frame Relay, multi-link PPP • Support for optional VPN - 500 IPSec tunnels using DES/3DES/AES encryption • Support for 3-way conferencing • Support for caller ID, message waiting, and stutter dial tone • Fax and analog modem compatible (V.90) • Support for local station to station calls • Up to 48 channels of G.711 (µ-law) • Up to 48 channels on G.726 (32K ADPCM) • Up to 48 channels on G.729 • Up to 48 channels of DTMF detection/generation
MAKE CALL Description
MAKE CALL Description There are three types of call setup – Station Call, CO Call, System Call Feature Implementation. Operation Making Station Call Setup 1. Dial Station Number. 2. If ‘Dial Digit Map’ is programmed, SIP Phone will send call setup immediately. 3. If ‘Dial Digit Map’ is not programmed, press the [SEND] button or “#” key for send out call setup. Making CO Call Setup 1. CO Access Code + Dial Number + [SEND] ex) CO Access Code “9”, Dial Number ‘450-4500’, dial ‘94504500’ and press [SEND] or “#” button NOTE If you program ‘Second Dial Tone Digit Map’ on SIP Phone, you will hear self dial tone from SIP Phone. CO Access Code + [SEND], after hearing CO dial tone, press Dial Number ex) CO Access Code “9”, Dial Number ‘450-4500’, dial “9” and press [SEND] or “#” hear CO dial tone from system dial ‘4504500’ Making System Call Feature Setup 1. System Call Feature by Numbering Code : System Numbering Plan (PGM106-109) 2. Enblock Dialing : System Call Feature numbering code + data + [SEND] button. 3. Supported Call Features by Numbering are, Internal Page Zones Internal All Call Page Meet Me Page Internal All Call Page Meet Me Page Internal All Call Page External Page Zone External All Call Page All Call Page SMDR Account Code Enter SLT Last Number Redial Do-Not-Disturb(DND) Call Forward Speed Dial Program SLT Speed Dial Access
Make call
MAKE CALL Description There are three types of call setup – Station Call, CO Call, System Call Feature Implementation. Operation Making Station Call Setup 1. Dial Station Number. 2. If ‘Dial Digit Map’ is programmed, SIP Phone will send call setup immediately. 3. If ‘Dial Digit Map’ is not programmed, press the [SEND] button or “#” key for send out call setup. Making CO Call Setup 1. CO Access Code + Dial Number + [SEND] ex) CO Access Code “9”, Dial Number ‘450-4500’, dial ‘94504500’ and press [SEND] or “#” button NOTE If you program ‘Second Dial Tone Digit Map’ on SIP Phone, you will hear self dial tone from SIP Phone. CO Access Code + [SEND], after hearing CO dial tone, press Dial Number ex) CO Access Code “9”, Dial Number ‘450-4500’, dial “9” and press [SEND] or “#” hear CO dial tone from system dial ‘4504500’ Making System Call Feature Setup 1. System Call Feature by Numbering Code : System Numbering Plan (PGM106-109) 2. Enblock Dialing : System Call Feature numbering code + data + [SEND] button. 3. Supported Call Features by Numbering are, Internal Page Zones Internal All Call Page Meet Me Page Internal All Call Page Meet Me Page Internal All Call Page External Page Zone External All Call Page All Call Page SMDR Account Code Enter SLT Last Number Redial Do-Not-Disturb(DND) Call Forward Speed Dial Program SLT Speed Dial Access
Dial the VM Pilot Number
TO FORWARD ALL INCOMING CALLS TO YOUR MAILBOX ❍ Press the Speaker key ● Dial 741 or press the Call Forward Immediate Function Key (if one is programmed on the phone)● Dial 1 to Set ● Dial the VM Pilot Number ● Hang up TO FORWARD INCOMING CALLS TO YOUR MAILBOX WHEN YOUR PHONE IS BUSY ❍ Press the Speaker key ● Dial 742 or press the Call Forward Busy Function Key (if one is programmed on the phone)● Dial 1 to Set ● Dial the VM Pilot Number ● Hang up TO FORWARD INCOMING CALLS TO YOUR MAILBOX WHEN YOU DO NOT ANSWER ❍ Press the Speaker key ● Dial 743 or press the Call Forward No Answer Function Key (if one is programmed on the phone)● Dial 1 to Set ● Dial the VM Pilot Number ● Hang up TO FORWARD INCOMING CALLS TO YOUR MAILBOX WHEN YOUR PHONE IS BUSY OR YOU DO NOT ANSWER ❍ Press the Speaker key ● Dial 744 or press the Call Forward Busy/No Answer Function Key (if one is programmed on the phone)● Dial 1 to Set ● Dial the VM Pilot Number ● Hang up
Logging
Initially Logging in to System Administration............................................... 22Selecting the System Administration Prompt Language............................. 23Programming System Parameters.............................................................. 24Programming the System Language Mode and System Language......... 24Setting the System Language Mode ......................................................25Setting the System Language ................................................................25Programming the Call Answer Service Operator Extension .....................28Programming the General Mailbox Owners............................................. 29Programming the Maximum Extension Length
AA delivers recorded announcement
Auto Attendant / Voice Mail Application –AA delivers recorded announcement to direct callers to the proper destination –Voice Mail includes message broadcast, email and mobile notification –Offers all the common VM functionality –Both provide multi-language support › Built-in Automatic Call Distribution (ACD) –Flexible incoming call routing –Real-time agent monitoring and call record statistics –Event messages for management reporting › Mobile Extension –Allows the mobile to place and receive calls through the system –Calls sent to a user’s iPECS phone and mobile simultaneously › Centralized Control T-NET (Transparent Network) Central UCP controls all modules and terminals located in remote offices providing all the features and functions of the central UCP Local survivability is provided with a second call server located at a remote site Power redundancy available when UCP100/600/2400 installed in main cabinet
UcP
iPECS UCP is designed to deliver the flexibility you need as your organization grows Simple Unified Communications Built-in Integrated Applications Tailored to your Needs › Users can access voice, video, instant messaging, conference calling and visual voicemail, all on a simple and easy-to-use platform Leverage the Latest Standards-Based Technologies › iPECS helps you make the most of the latest network technologies such as SIP, optimize call costs using Wi-Fi or use built-in voice conferencing –Provides capacity for up to 2,200 devices, allowing it to handle most any need iPECS-UCP Anywhere, Anytime Connectivity › Access the power of your iPECS call servers your way regardless of your device or location using smartphone, tablet or PC applications › iPECS offers a range of enhanced applications from Ericsson-LG and other specialist application providers, including Microsoft Outlook or Lync as well as others Reliable and Resilient › Total reliability is the only option for your communications. With inherent modular architecture, iPECS UCP provides geographic redundancy, hot standby power redundancy and Central Control T-Net
distributed amplified paging
Installation/Connection Cabling Category 3 or 5 twisted pair cable is recommended for all Valcom distributed amplified paging installations. Screw terminals are provided for the basic connections. RJ45 jacks are provided for chaining multiple V-9964 units together. Removing the narrow right side panel of the V-9964 provides access to controls, connections and option switches. To remove the panel, loosen the two screws holding the panel in place and lift the panel. Mounting The V-9964 may be wall mounted or rack mounted in a standard 19 inch equipment rack using the brackets included. Connections See Figure 1 for a connection diagram. Tip 1, Ring 1 INPUT 1 is the normal Primary or Call Stacker system input, and connects to a Loop Start Trunk Port, 600 Ohm Page Port or some Valcom Page Controls. Note: Do not connect to a C. O. Line. Control Input 1 Provides contact closure activation when using a Page Port. Tip 2, Ring 2 INPUT 2 is the Override page or Call Stacker line two input. If desired, connect this to a second Loop Start Trunk Port or Page Port. Note: Do not connect to a C. O. Line. Provides contact closure activation when using a Page Port. Background Music Input Connection for external line level music source (Example: V-2952, FM Tuner). NOTE: If using multiple V-9964 units in a chained configuration, all speakers must connect to the output of the last unit in the chain. Line Out Output connections to Valcom amplified speakers or 70 Volt amplifier Aux input. Loop Out Connects to Tip and Ring input on a Valcom multi-zone page control unit. Expansion In RJ45 connection from the previous V-9964 in a chained configuration. Expansion Out RJ45 connection to the next V-9964 in a chained configuration. Closing a switch connected to pin 7 and pin 3 of either Expansion In or Expansion Out will hold all queued recorded messages, on all linked V-9964s. During this time, additional messages may still be recorded into the V-9964(s). Once the switch is opened, all queued messages, including DTMF, will play in their entirety. Abort To abort a message during play, connect an external relay contact across the two abort terminals. NOTE: To abort a message during the record sequence, press any DTMF button on the dial pad of the access telephone. Relay Closure Outputs (PLAY) PLYSW and PLYMK Normally open relay contact that closes while a message is being broadcast. (RECORD) RECSW and RECMK Normally open relay contact that closes while a message is being recorded
Do not disturb
Do Not Disturb Function The N.E.C key sets have a function so that anyone trying to ring the key set receives a BUSY tone or routes to voicemail even if it is not being used. This function is useful if a room is needed for an important meeting and the keyset needs to be available for outgoing calls but needs to be silent so as not to disrupt the meeting. Pressing the DND key then choosing one of the following options does this: 1 - External calls only 2 - Intercom calls only 3 – All calls 0 - Cancel
on-premises single line telephones (SLTs)
The system is compatible with 500 type (dial pulse) and 2500 type (DTMF) analog telephone devices. This includes on-premises single line telephones (SLTs), fax machines, and modems. In DSX-40, SLTs connect to analog ports in the main equipment cabinet. In DSX-80/160, SLTs connect to SLIU PCBs. Each analog port provides power and ring voltage for the connected SLT. The analog ports use DTMF receivers. Each system provides 10 DTMF receivers that are shared by all connected analog ports. Message Waiting Both DSX-40 and DSX-80/160 support FSK Message Waiting lamps. DSX-80/160 also provides support for high voltage Message Waiting lamps – while DSX-40 does not. Ringing For Incoming Calls Single line extensions ring according to the settings in 2132-[01-64]: Line Ringing Stations: Config: Ring Line Ringing (2132): Ring Assignment] Assign: . It is not necessary to assign single line sets to Ring Groups to make them ring for incoming calls; they follow Key Ring instead. • In DSX-80/160 by default, the first 16 extensions (300-315) ring (option 2) for lines 1-12 and flash (option 1) for lines 13-64. All other extensions have lamp only (no ringing) for all lines. • In DSX-40 by default, all extensions (including single line sets) have immediate ring for all lines. Ringer Equivalence Number (REN) Considerations DSX-40 Single line telephones assigned to Key Ring or the same Ring Group will ring simultaneously. This is also true for single line telephones connected to the same port. Since the Ringer Equivalence Numbers of connected single line telephones are cumulative, you must do the following: • Add up the RENs of all connected single line telephones. • Be sure the total REN does not exceed 4 on any single port or system-wide. Note that a REN of 1 is normal for an industry standard 2500 set with electromechanical ringer. Many phones with electronic ringers have significantly lower RENs. Check the label on the bottom of each single line telephone for the REN value. DSX-80/160 Single line telephones assigned to Key Ring or the same Ring Group ring in pairs according to their SLIU PCB port assignment. For example, ports 1 and 2 ring together, followed by 3 and 4, 5 and 6, and finally 7 and 8. If the system has more than one SLIU PCB installed, the respective port pairs ring simultaneously on each card (e.g., ports 1 and 2 ring simultaneously on each PCB). The SLIU provides the capability to support this rin
Description the system provides flexible routing
Description The system provides flexible routing of incoming CO (trunks) calls to meet the exact site requirements. This lets trunk calls ring and be answered at any combination of system extensions. A maximum of 84 trunks are available. For additional information on making trunks ring, refer to Ring Groups on page 1-763. Delayed Ringing Extensions in a Ring Group can have delayed ringing for trunks. If the trunk is not answered at its original destination, it rings the DIL No Answer Ring Group (this ring group applies to DIL or non-DIL trunks). This could help a secretary that covers calls for their boss. If the boss does not answer the call, it rings the secretary’s telephone after a programmable interval. Universal Answer Universal Answer allows an employee to answer a call by going to any Multiline Terminal and dialing a unique Universal Answer code. The employee does not have to know the trunk number or dial any other codes to pick up the ringing trunk. You normally set up Universal Answer along with Universal Night Answer (refer to Night Service on page 1-663). When a Universal Night Answer call rings the External Paging, an employee can answer the call from the first available telephone. You might also want to use Universal Answer in a noisy warehouse or machine shop where the volume of normal telephone ringing is not adequate. After hearing the ringing over the Paging, an employee can then easily pick up the call from a shop telephone. The Automatic Off-Hook Answer of Universal Answer Call options (PRG 20-10-07) determines whether or not the extension has the Auto Answer feature for ringing calls. This option allows a user to simply lift the handset to answer a ringing call; dialing the service code is not necessary. Additional Trunk Ring Tones Various ring tone patterns and melodies for incoming calls are available (PRG 22-03-01); Ring Tone Patterns 1~8 and Melodies 1~5 (V3.0 or higher) can be configurable, refer below chart.
please reset the number and reboot the PC
Line Information: Shows the terminal that is going to be used. You can set the following items: ❒ ❒ ❒ Extension Number: Shows the extension number you installed. If you change the terminal or number please reset the number and reboot the PC in order for the change to take affect. Extension Name: Shows the name of the Extension. Line Configuration: Shows the Line Configuration screen. You can't press the button unless you are connected to PBX. ❍ Network Information: ❒ ❒ ❒ PBX IP Address: Shows the IP Address which you entered during installation. If you change the PBX IP Address within the UX5000 programming, change this setting as well and reboot your PC. PBX TCP Port: Shows the TCP Port Address you entered during installation. If you change the PBX TCP Port Address within the UX5000 programming, change this setting as well and reboot your PC. My Host Name / IP Address: Shows the My Host Name/IP Address you entered during installation. We recommend using a "Localhost" (Default) or the PC name. Otherwise, enter the IP address of the network card you are using. If you change the IP address of the card you’re using, change this setting as well and reboot your PC. ❍ Operation Mode: Shows the Operation Mode you selected during the installation. ❒ Multi Line: Send event to CTI that occurs on Speaker Key/Answer Key/Trunk Key/Virtual Extension Key/Yellow Key. Use this setting if the application shows multiple call information.
LINE INFORMATION
Line Information: Shows the terminal that is going to be used. You can set the following items: ❒ ❒ ❒ Extension Number: Shows the extension number you installed. If you change the terminal or number please reset the number and reboot the PC in order for the change to take affect. Extension Name: Shows the name of the Extension. Line Configuration: Shows the Line Configuration screen. You can't press the button unless you are connected to PBX. ❍ Network Information: ❒ ❒ ❒ PBX IP Address: Shows the IP Address which you entered during installation. If you change the PBX IP Address within the UX5000 programming, change this setting as well and reboot your PC. PBX TCP Port: Shows the TCP Port Address you entered during installation. If you change the PBX TCP Port Address within the UX5000 programming, change this setting as well and reboot your PC. My Host Name / IP Address: Shows the My Host Name/IP Address you entered during installation. We recommend using a "Localhost" (Default) or the PC name. Otherwise, enter the IP address of the network card you are using. If you change the IP address of the card you’re using, change this setting as well and reboot your PC. ❍ Operation Mode: Shows the Operation Mode you selected during the installation. ❒ Multi Line: Send event to CTI that occurs on Speaker Key/Answer Key/Trunk Key/Virtual Extension Key/Yellow Key. Use this setting if the application shows multiple call information.
Comparing ISDN to Analog
Comparing ISDN to Analog 54 Type of ISDN service 54 B channels 55 D channels 55 ISDN layers 55 ISDN bearer capability 56 Services and features for ISDN PRI and BRI 57 PRI services and features 57 BRI services and features 58 Feature descriptions 59 Network name display 59 Message Waiting Indicator (MWI) 60 Name and number blocking 60 External call forwarding 61 MCDN trunk features 61 Call by Call service selection for PRI 61 Emergency 911 dialing 62 MCID (Profile 2) 63 Network Call Diversion (Profile 2) 63 DTI card configured as a PRI card 64 ISDN hardware 64 DTI card configured as PRI 64 BRI Card 65 BRI-U2 and BRI-U4 card 65 BRI-ST card 65 U-LT reference point 66 U-NT reference points 66 S reference point 67 T reference points 68 Clock source for ISDN cards 69 Other ISDN BRI equipment: NT1 70 ISDN standards compatibility 71
Slt massege wait
SLT MESSAGE WAIT INDICATION Description All SLT devices will receive a “Stutter” dial tone as an audible Message Wait Indication. In addition, industry standard Message Waiting telephones may be connected to the system. Software will cause the lamp to flash when a messaging is waiting. Operation System The system switches the 90 VDC lamp On and Off for assigned SLTs indicating a Message Wait. Conditions 1. The system switches a 90 VDC supply On and Off to flash the SLT’s neon lamp. 2. Although the SLT Battery Feed is removed during the 90 VDC On cycle, the system will recognize an SLT Off-hook event. 3. The SLT must incorporate a 90 VDC neon lamp that is connected directly across the tip and ring of the voice network.
The calling phone sends out an invite
The calling phone sends out an invite
The called phone sends an information response 100 � Trying � back.
When the called phone starts ringing a response 180 � Ringing � is sent back
When the caller picks up the phone, the called phone sends a response 200 � OK
The calling phone responds with ACK � acknowledgement
Now the actual conversation is transmitted as data via RTP
When the person calling hangs up, a BYE request is sent to the calling phone
The calling phone responds with a 200 � OK.
It�s as simple as that! The SIP protocol is easy to understand and logical.
menu structure
following table shows the menu structure of “Menu” Soft Key. You can reach the desired feature using the following operation. 2-8 TUE 3:03PM 200 Menu Dir VM:00 CL:00 Prev Next Select Exit Prev Next Select Back It is possible to search the desired feature by pressing Cursor the Keys (Up / Down / Right / Left) several times instead of “Prev” or “Next” Soft Keys, or it’s possible to access the desired feature directly by dialing the 2 digit Menu Code after pressing the “Menu” Soft Key. Item Menu Code Next Operation after pressing the “Select” 10 : Volume Preference 11 : Ring 12 : Off-Hook Ring Press “Down” or “Up” to adjust the selected option. 20 : Display Preference 21 : Contrast 22 : Min Brightness 23 : Max Brightness Press “Down” or “Up” to adjust the selected option. 30 : Feature Preference 31 : Voice Announce 32 : Handsfree Reply 33 : Auto Call Timer 34 : Preview Dial 35 : Illuminated Dialpad 36 : Auto Call Screening 37 : Incoming Page 38 : Ringing Line Preference For the selected option, press “On” (enable) or “Off” (disable). 39 : Auto Backlit 40 : Ring Preference 41 : Intercom 42 : Line Keys 50 : Key Assignment 51 : Feature Keys 52 : Primeline Key Press “<<” or “>>” to select and save option. Press “<<” or “>>” to select and save option. 60 : Call Forwarding 61 : Immediate 62 : Ring No Ans 63 : Busy No Ans 64 : Call Forward AME 65 : Display Message 66 : Follow Me 67 : Both Ring Press “Set” or “Cancel”, enter the destination and select option to save. 70 : Speed Dial 71 : Personal Speed Dial 72 : Company Speed Dial Enter Bin number and Phone number, Name and save. 80 : Name and Language 81 : Extension Name 82 : Display Language For Name, enter the name using Alphanumeric Characters, For Language, press “<<” or “>>” to select and save. 90 : Option Preference 91 : Headset Mode 92 : Headset Voice Announce 93 : System Information 94 : VoIPDB Information 95 : Auto Backlit (Threshold) 96 : IP Address Information For Headset option, press “On” (enable) or “Off” (disable). For System / VoIPDB information (IP Address, MAC Address), press “Select”. For Auto Backlit, select threshold option to save. 00 : Admin 01 : Time 02 : Date 03 : Extension Name 04 : Clear All Call Fwd 05 : System Night Key Mode For Time, Date and Extension Name, enter the Time, Date and Extension Number and Name to save. For Clear All Call Fwd, press “Yes”.
ASUS WL-330N provides
Operation modes ASUS WL-330N provides five operation modes: Wireless Router, Access Point (AP), Hotspot (WiFi Account Sharing), Repeater, and Wireless Network Adapter. NOTES: • Use the Device Discovery utility in the support CD to get the WL-330N's WL-330N's's dynamic IP address. • Before setting up the WL-330N to Hotspot, Repeater or Network Adapter WL-330NtoHotspot,RepeaterorNetworkAdapter to Hotspot, Repeater or Network Adapter mode, ensure that you connect the computer and the WL-330N using a network cable. • If you cannot switch modes successfully, reset the system to its factory default settings through pressing the Restore button while the ASUS WL330N is ON. Wireless Router mode In the Router mode, connect the WL-330N to an ADSL or a cable modem, and the network clients share an IP address for Internet connection. In this mode, the WL-330N provides wireless signals, NAT, firewall, and IP sharing functions to the wireless clients.
Administration Software on a LAN
Apendix 2 – Installing the System Administration Software on a LAN
The PARTNER or MERLIN Messaging System Administration (SA) application software can be used instead of the Touch-tone user interface to configure and program the messaging system. You must use the SA application to configure the Unified Messaging application.
Installing the SA Application
The SA application is installed on a PC that is connected to the LAN. Only a user on that PC can install the SA application. The SA application will communicate with the Messaging system through its LAN port. Installation of the MERLIN Messaging or PARTNER Messaging SA application can be performed from the MERLIN Messaging System Release 3.0 or PARTNER Messaging Release 6.0 Library CD. You select “PARTNER Messaging or MERLIN Messaging System Administration” under “Install Software” from the main Library screen. This will automatically launch a windows software install wizard. Follow the instructions displayed by the install wizard to complete the installation.
Launching the SA Application
To 1. start the SA application, perform one of the following steps:
Dou▪ ble-click on the MERLIN or PARTNER Messaging Administration desktop short cut.
Sel▪ ect the MERLIN or PARTNER Messaging Administration short cut from the Start menu.
Fro▪ m the Start menu, select Programs > MERLIN Messaging Release 3.0 or PARTNER Messaging Release 6.0. (This is the default location.)
The2. Messaging Login window appears, displaying the Messaging Login dialog box.
In 3. the IP address or host name box, enter the Fixed IP address of the Messaging System module that was loaded into the message system (See Section 2)
In 4. the Login box, enter sysadmin.
In 5. the Password box, enter the system administration password. (If this is your first time logging in to the system, click the OK button. You are prompted to enter the password.)
Cli6. ck the OK button.
Once you log in successfully, you can start administering the Messaging system. The SA application windows display the current Messaging settings. For more information regarding installation and use of the SA application, see the MERLIN or PARTNER Messaging System Administration Getting Started Guide under Documentation, System Administration contained on the Library CD.
Technical Specifications
Technical Specifications
AC / DC Power Adapter
› 100~ 240 V AC @ 50/60Hz › DC48V, 1A
Physical Dimensions & Weight
› Width: 278mm
› Depth: 233mm
› Height: 34mm
› Weight: 0.86kg
Firewall
› General security policy, access control › Web site restriction, port forwarding, port triggering › NAT / NAPT, DMZ host, rule-based packet filtering › Connection information, security log
Security
› Virtual Private Network (IPSec, PPTP, L2TP) › Remote administration access control › Digital certificate management
Quality of Service (QoS)
› General QoS profile, bandwidth restriction › Rule-based traffic priority and traffic shaping › DSCP / 802.1p / priority queue configuration › Connection utilization and statistics
Routing
› Static routing (routing table management) › Dynamic routing (RIP v1/v2) › NAT / NAPT, IGMP / Multi-cast
L2 Switching
› 8-port 10 / 100 BASE-TX with 4 built-in PoE (total PoE budget: 20 Watts)
› STP / RSTP, VLAN, LAN bridge › Broadcast and multi-cast storm control, loop detect
Wireless LAN
› 802.11 b/g/n (2.4 GHz)
› WEP, WPA, WPA2 or WPA / WPA2 and web authentication
› 802.1x for enterprise
› Multiple SSIDs (virtual APs), MAC filtering › WPS (WiFi protected set up) support › Wireless multi-media (WMM) › Channel width and frequency selection
IP-PBX / SIP
› Ericsson-LG advanced IP-PBX features (call transfer, call forward, call park, call pick up, call waiting camp on, CO queuing, speed dial, station groups, mobile extension, 3-party voice conference, IP fax relay [T.38])
› SIP trunk – 4 trunks with DECT, 6 trunks w/o DECT › CO trunk – only one option can be mounted on the SBG-1000 either in the factory or locally (1CO, 2CO, 4CO, 1 BRI or 2BRI)
› Extension – 23 IP extensions – 6 Ericsson-LG proprietary DECT terminals › Built-in SIP proxy, registrar, user agent, failover to PSTN
Administration
› Web-based administration (HTTP / HTTPS) › CLI (telnet and telnet over SSH) › User management (role and permissions) › Date and time (NTP / TOD with daylight savings option)
› Smart installation wizard
Services
› File server (disk management, back up and restore) › Printer server (LPD, IPP, Microsoft shared printing support)
› DHCP / DNS server, dynamic DNS, UPnP
Management
› Device information and map view, SNMP, TR-069 › Network connection management, monitoring and diagnostics
› Email notification and syslog support, system log
Service code setup
Use Program 11-15 : Service Code Setup, Administrative (for Special Access) to customize the special access Service Codes which are used by the administrator in the Hotel/Motel feature. You can customize additional Service Codes in Programs 11-10 ~ 11-14 and 11-16. The following chart shows: • The number of each code (01 ~ 14). • The function of the Service Code. • What type of telephones can use the Service Code. • The default entry. • Programs that may be affected when changing the code.
Press the programmed button to deactivate
PARTNER ® Advanced Communications System Installation, Programming, and Use 3. Press the programmed button to deactivate the feature.
The feature is deactivated automatically if you hang up the handset or press any button other than a line, pool, or i button. The green light is off when the feature is deactivated.
Caller ID Call Logging and Dialing (F23) 8 Once the system administrator assigns the Caller ID Call Log Line Association, Caller ID Log Answered Calls, and/or the Caller ID Log All Calls features to log Caller ID calls, you use Caller ID Logging and Dialing (F23) to view the log. Caller ID Call Logging and Dialing is available on system display telephones for all lines for which you subscribe to a Caller ID service. Use this feature to view Caller ID information for central office calls.
You must program Caller ID Logging and Dialing onto a line button with LEDs to use
the feature.
Up to 400 call records can be stored for the system. Each line associated with an extension to log
Caller ID calls is guaranteed a minimum of 20 call records.
You also can automatically dial the number stored in the log.
The call records stored in each user’s call log and available for viewing depend on the following:
■
■
■
■
■
Unanswered transferred calls are logged automatically, whether or not the line and extension are associated with the Call Logging features.
If Caller ID Log Answered Calls is used alone, all Caller ID calls that are answered by that extension are logged.
If Caller ID Log Line Association is used alone, all unanswered Caller ID calls that ring on a line associated with the extension are logged.
If both Caller ID Log Answered Calls and Caller ID Log Line Association are used, all
Caller ID calls that are answered by that extension and all unanswered Caller ID calls that ring on a line associated with the extension are logged.
If both Caller ID Log Line Association and Caller ID Log All Calls are used, all answered
Caller ID calls and all unanswered Caller ID calls received at any extension on specific lines are logged. This combination can be assigned to a maximum of one extension per system. The Caller ID information appears on three screens:
■ The first screen shows the caller’s number (or the reason that the number is not available).
■ The second screen shows the caller’s name (or the reason that the name is not available).
■ The third screen shows the date and time of the call. In addition, the system logs the line the call came in on, whether the log entry was viewed, whether the call was answered or not answered, and whether an atte
Battery Replacement
Battery Replacement 1 1 The PARTNER ACS processor module uses two user-replaceable AAA alkaline batteries. These batteries provide enough power to retain the system programming settings during a power failure for 45 days to six months, depending on the freshness of the batteries. When battery power is getting low, the system displays a ChgBat W/PowerOn or ReplaceSysBat W/Power On message on the top line of display telephones at extensions 10 and 11 in place of the default day/ date/time message. Users at these extensions should be instructed to notify you when they see this message. You should replace the batteries within 45 days of seeing the message.
Message Waiting Light
Programming the General Mailbox Owners
Touch-Tone Input
777
0#
[nnnnnn] #
9
1
3
[nn] #
PARTNER Messaging provides four General Mailboxes–one for each Automated Attendant. The General Mailbox extensions are 9991, 9992, 9993, and 9994. You cannot delete these extensions or change their phone status. Automated Attendant Service calls are directed to the Automated Attendant’s General Mailbox when the Automated Attendant’s Dial 0/Timeout Action is set to record a message in the General Mailbox and: • Caller does not make a selection from an Automated Attendant Service Menu.
• Caller presses 0 while in Automated Attendant Service.
• Caller presses 0 while using the Directory to transfer.
The General Mailbox Owner is the extension whose Message Waiting Light is turned on whenever a message is placed in the General Mailbox. The factory setting for the General Mailbox Owner is extension 10. You can program a different General Mailbox Owner for each General Mailbox, or you can have the same owner for all the General Mailboxes.
In Appendix B, refer to PARTNER Messaging Planning Form 1 — System Parameters for the General Mailbox Owners.
NEC SL1100 features
• Conference, Remote
• Conference, Voice Call/Privacy
Release
• Continued Dialing
• Data Line Security
• Delayed Ringing
• Department Calling
• Department Step Calling
• Dial Pad Confirmation Tone
• Dial Tone Detection
• Dialing Number Preview
• Digital Trunk Clocking
• Direct Inward Dialing (DID)
• Direct Inward Line (DIL)
• Direct Inward System Access (DISA)
• Direct Station Selection (DSS)
Console
• Directed Call Pickup
• Directory Dialing
• Distinctive Ringing, Tones and
Flash Patterns
• Do Not Disturb (DND)
• Door Box
• Drop Key
• Ecologically Sound Power Saving
ModeE911 Compatibility
• Flash
• Flexible System Numbering
• Flexible Timeouts
• Forced Trunk Disconnect
• Group Call Pickup
• Group Listen
• Handset Mute/Handset Cutoff
• Hands-free and Monitor
• Hands-free Answerback/Forced
Intercom Ringing
• Headset Operation
• Hold
• Hotel/Motel
• Hotel/Motel - Do Not Disturb
• Hotel/Motel - DSS Console
Monitoring
• Hotel/Motel - Message Waiting
• Hotel/Motel - Room Status
• Hotel/Motel - Room Status Printout
• Hotel/Motel - Room-to-Room Call
Restriction
• Hotel/Motel - Single Digit Dialing
• Hotel/Motel - Toll Restriction (When
Checked In)
• Hotel/Motel - Wake Up Call
• Hot Key-Pad
• Hotline
• Howler Tone Service
• Illuminated Dial Pad
• InMail
• InMail-Automatic Access to VM
by Caller ID
• InMail-Cascade Message
Notification
• InMail-Email Notification
• InMail-Find-Me Follow-Me
• InMail - Language Setting
• InMail Park and Page
• InMail Upload Download Audio
• Intercom
• IP Multiline Station (SIP)
• IP Single Line Telephone (SIP)
• IP Trunk - (SIP) Session Initiation
Protocol
• ISDN Compatibility
• Last Number Redial
• Licensing
• Line Preference
• Long Conversation Cutoff
• Loop Keys
• Maintenance
• Meet Me Conference
• Meet Me Paging
• Meet Me Paging Transfer
• Memo Dial
• Message Waiting
• Microphone Cutoff
• Mobile Extension
• Mobile Extension - Callback to
Mobile Phone
• Multiple Trunk Types
• Music on Hold
• Name Storing
• Navigation Key
• Night Service
Handsfree, using the headset
Call Forwarding with Follow Me
Call Forwarding with Follow Me
Description
While at a co-worker’s desk, a user can have Call Forwarding with Follow Me redirect their calls to the co-worker’s extension. This helps an employee who gets detained at a co-worker’s desk longer than expected. To prevent losing important calls, the employee can activate Call Forwarding with Follow Me from the co-worker’s telephone.
Call Forwarding with Follow Me reroutes calls from the destination extension. To reroute calls from the initiating (forwarding) extension, use Call Forwarding.
Conditions
•Call Forwarding an extension in a Department Group prevents that extension from receiving Department Pilot Calls.
•Multiple Stations can set Call Forward Follow Me to one station.
•Calls to extensions with DND active do not follow Call Forwarding programming. DIL calls ring an idle Department Group member, and then follow PRG 22-08 programming then PRG 22-05
Press the [MUTE] button
An iPECS Phone can turn off audio transmission from the handset, speakerphone or headset microphone, “Mic Mute”.
Operation iPECS Phone To Mute the Microphone;
1. Press the [MUTE] button, the [MUTE] button LED is on and the microphone (Handset, Speakerphone, Headset) is muted; the connected party receives silence.
To activate the microphone;
1. Press the illuminated [MUTE] button, the [MUTE] button LED is off, and the microphone is activated, transmitting audio to the connected party.
Conditions
1. Changing from speakerphone to handset or vice versa during a mute condition will eliminate the mute status.
2. Returning to idle or placing another CO/IP or intercom call will change the mute status to its normal (active microphone) condition.
Back up the system voice prompts
Determine the version of system voice prompts currently installed on the InMail CompactFlash card. Refer to Selecting the CompactFlash Card on page
• Back up the system voice prompts and recorded names, messages and greetings stored on the In- Mail CompactFlash card to your PC hard disk. Refer to Backing Up the InMail CompactFlash Card on page
• Restore the system voice prompts and recorded names, messages, and greetings stored on the your PC hard disk to the InMail CompactFlash card. Refer to Restoring a Backed-up Database to the In- Mail CompactFlash Card on page
• Update the system voice prompts stored on the InMail CompactFlash Card (required for feature up- grades). Refer to Initializing the InMail CompactFlash Card on page
• Optionally, initialize (completely erase) the InMail CompactFlash card and load the latest system voice prompts. Refer to Initializing the InMail CompactFlash Card on page
• Optionally add or remove any of the supported language prompt sets. Refer to Managing Language Prompt Sets on SL1100 InMail
DOOR OPEN
DOOR OPEN
Description
The iPECS hardware is equipped with relays that activate External Control Contacts. The contacts can be assigned to one of several functions including a Door Open Contact. When used as a Door Open Contact, the contact is connected to a door-lock release mechanism. When assigned stations receive the Door Bell signal, the user may dial the Door Open code to activate the contact.
Expansion Module
The IX-EXPML2 Expansion Module is used exclusively when adding a fourth shelf to the system. The IX-EXPML1 Expansion module must be used when adding a second, third, or fifth shelf to the system.
The IX-EXPML2 expansion module adds six universal card slots and 96 universal ports to the ADIX APS system. In IX-EXPML2, the first card slot supports a maximum of 24 ports. The second card slot supports a maximum of 8 ports. The third through sixth card slots each support a maximum of 16 ports. This unit has dedicated space for the IX-PWSL main power supply as well as the IX-DCDCM, IX-RNGUM power supplies. An internally-mounted IX-PWSL power supply is required in the IX-EXPML2 expansion module.
touch-tone or rotary
Dial Mode (#201) 4
Use this feature to identify individual lines as touch-tone or rotary. Check with your local telephone company if you are not sure which type of line is being provided to you.
Considerations 4
■ If you are having difficulty using touch-tone telephones on rotary lines, you may need to adjust the Rotary Dialing Timeout (#108).
■ If the system has rotary lines, you can use Touch-Tone Enable (F08) to send touch-tone signals over a rotary line (for example, to access bank-by-telephone services).
■ If Dial Mode is set to Rotary, star codes are entered by dialing 11 instead of *. If you have users at extensions with Outgoing Call Restriction (#401) set to Local Only who are calling out on lines with the Dial Mode set to Rotary, you should add “11” to an Allowed Phone Numbers List (#407) and assign the list to these extensions. Otherwise, when the users at the restricted extensions dial 1 to begin a star code, the system interprets this as an attempt to place a long-distance call, the call is blocked, and the user hears the reorder tone.
■ The 1600 DSL module supports only touch-tone signaling.
Programming
Department Step Calling
If you and your co-workers handle
each other's calls, you might want to be in a Department Calling group Someone calling your group's number goes through to any- one who's available. You can even have Department Step Calling () send your personal calls to your group when you're not avail- able. To answer a call already ring- ing a co-worker's phone, use Group Call Pickup ().
When you're on a call and you want the others in your area to listen in on the conversation, activate Group Listen . Your co-work- ers hear the call through your telephone's speaker.
If you frequently call the same co-worker, you can have Ringdown automatically call them for you. All you have to do is lift youndset.
call a door phone
Door Phone(s)
Door phones can be used to call phones selected in system programming. When a door phone calls, you hear a distinctive ringing tone, one or five times (set in system programming). You can also call a door phone and monitor the surrounding area.
LCD telephones display the door phone name ID when calls are made to or from door phones.
The number of possible door phones varies by Strata CIX system, 01~24 maximum for larger systems. Check with your System Administrator to find out the names and locations of your system’s door phones and record them below.
Unforced account Codes
Optional (Unforced) Account Codes
Optional Account Codes allow a keyset extension user to enter an Account Code while placing a trunk call or any time while on a call. This type of Account Code is optional: the system does not require the user to enter it. If the keyset user is already talking on a trunk call, their conversation continues uninterrupted while they enter an Account Code.
Single line telephone users can only enter an Account Code while placing their trunk call. Forced Account Codes
Forced Account Codes require an extension user to enter an Account Code every time they place a trunk call. If the user doesn’t enter the code, the system prevents the call. The system can require Forced Account Codes for all trunk calls, or just for toll calls (as determined by Toll Restriction programming). Note that Forced Account Codes do not pertain to incoming calls.
Verified Account Codes
With Verified Account Codes, the system compares the Account Code the user dials with a list of codes programmed into the Verified Account Code Table. If the Account Code is in the table, the call goes through (provided it is not prevented by an extension’s Toll Restriction programming). If the code is not in the table, the system prevents the call. Verified Account Codes, if enabled, apply only to Forced Account Codes.
Using Account Codes and Speed Dial
To simplify Account Code operation, Personal and System Speed Dial bins can contain Account Codes. Keep the following in mind when using Speed Dial and Account Codes:
● The Account Code can be either the first or last entry in the bin, and must be preceded and fol- lowed by the # character. For example, the Account Code 1234 must be entered as #1234#.
● The Program 0201 - # Key to Enter Account Codes (page 629) option must be enabled in sys- tem programming. In addition, the Program 0201 - Enable Account Codes in Speed Dial (page 629) option must also be enabled.
● The Speed Dial bin can contain an Account Code followed by an outside number, or just the Account Code. The Account Code must be preceded and followed by a # entry. If the bin contains just the Account Code, the user must be sure to press the bin key before dialing the outside number.
● If the system has Verified Account Codes enabled, the Account Code entered in the Speed Dial bin must match an entry in the Verified Account Code Table.
IP-telephony platform, SIP and DECT
iPECS SBG-1000 is a truly converged business communications services platform in a single appliance
IP-telephony platform, SIP and DECT, a broad range of terminals, and advanced telecom applications
Dynamic embedded routing protocols, integrated 802.11b/g/n WiFi, QoS and built-in NAT
Built-in VPN, enhanced security protocols, firewall, DMZ and rules-based packet filtering
Embedded high-speed USB 2.0 port for IT services, connection to local servers for file, fax and printer sharing
Soft client makes smartphones an extension of the communications platform, 3G / 4G mobile broadband plug-in option
Advanced Qsig and IP networking, local and remote management and interoperability with TR-069 and SNMP
UCS Client
iPECS UCS Client: Basic voice calling plus advanced functionality such as video, IM, file sharing, audio conferencing, visual voice mail
• Multi-Tier Mobility: DECT and Wi-Fi phones for in-office cordless, mobile extensions to extend voice features to any phone and iPECS Communicator for full smartphone integration
• Integrated Auto Attendant / Voice Mail (AA/VM): Multi-level, multi-language AA, built- in Voice Store and Forward (VSF) Gateway
• Voice Mail Notifications: Email with or without .wav file attachment or as an alert to mobile phone
• IP Attendant: Windows-based application with powerful monitoring, Busy Lamp Field (BLF) status and attendant console features
• Automatic Call Distribution (ACD): Flexible incoming call routing, real-time agent monitoring and supervision and call record statistics
• Centrally-Controlled T-Net: Greatly enhances survivability, server and geographic redundancy and management of local and remote sites
• System Geographic Redundancy: Hot standby call server for seamless hand-over in case of main server failure
• Multi-Language: Simultaneous support for up to 16 languages in all features
• Power Backup: Power redundancy features wit
Conference call and Collaboration
iPECS UCP – Premium UC Features
› Conference call and Collaboration ̶ Max six members in video conference ̶ File transfer among colleagues
̶ Application sharing for simultaneous access and editing of any type document in real time
̶ Share desktop with other UCS users
̶ Share web page address with other UCS
users
̶ Whiteboard to share drawings and free-form
text in real-time
› Microsoft Exchange Server Integration
– More precise schedule synchronization with Exchange Server
Changing Settings
Changing Settings to Support
PBX or Centrex Services 3
Your system may work behind a PBX or Centrex system:
■ PBX services are provided by a private telephone switch.
■ Centrex services are provided by your local telephone company from a Central Office (CO) outside your premises. These services include the Centrex lines connected to your control unit modules and some set of features—such as hold, conference, or transfer—available on those lines. Centrex services may be offered in your area under a different name. For specific Centrex features to be available to you, your company must subscribe to those features. For specific information about using Centrex features, see the Centrex documentation provided by your local telephone company.
Pool Access Restriction
Pool Access Restriction (#315) 4
Use this feature to restrict a pooled extension from receiving and/or making outside calls on all lines belonging to a specific pool. For example, you may want customer service representatives to make calls using the WATS lines that belong to auxiliary pool 881, but not receive incoming calls on the lines in that pool; in this case, you assign pool 881 to the customer service representative’s telephones and restrict the pool to Out Only.
This procedure is the most extreme way to restrict dialing. For example, an extension set to In Only or No Access for a particular pool cannot select that pool to dial out—even for numbers on the Emergency Phone Number
Data Call Button
Data Call Button
A flexible button on a digital telephone can be assigned as a 'DWD&DOO button, which can be used to dial internal data calls. The telephone must be equipped with an RPCI-DI for 'DWD&DOO button applications.
Direct Station Selection (DSS) Buttons
Digital telephone users can ring selected stations by pressing a flexible feature button assigned for a DSS function. The LED associated with the button provides the busy status of the station and the station’s [PDN]. Each flexible button can be assigned as a DSS button to a different station [PDN]. DSS buttons can also be assigned on DADMs.
Direct Station Selection (DSS) Console Features
On DK40i and DK424, digital and electronic telephones can operate with DSS consoles, which offer the following features:
o Automatic CO line Hold
display different languages
change the language in which display messages appear if the extension has a system display telephone. The language is set for each extension, so telephones in the same system can display different languages.
Considerations 4
If SMDR is used, the call report header is printed in the language specified for extension
T1 module
T1 module is required for Direct Inward Dialing (DID).
■ You must use the PARTNER ACS R7.0 PC Administration software to program Direct Inward Dialing (DID) on T1 lines.
■ You can have a call on a T1 line with Direct Inward Dialing (DID) ring at an extension that cannot access a T1 line from a line or pool button. If the line or pool button is not programmed at an extension, or the line or pool button is busy, the call rings at one of the extension’s Intercom buttons.
■ Outgoing calls can be made on T1 lines administered for Direct Inward Dialing (DID).
■ Automatic System Answer features are disabled on T1 lines with Direct Inward Dialing (DID).
■ Direct Extension Dialing features are disabled on T1 lines with Direct Inward Dialing (DID).
■ Caller ID information is unavailable on T1 lines. The message “Direct In Dial” or “DID” is displayed for incoming calls on T1 lines with Direct Inward Dialing (DID).
■ You should not assign T1 lines with Direct Inward Dialing (DID) to Hunt Groups. If a Hunt Group contains T1 lines with Direct Inward Dialing (DID), callers will receive busy signal when all extensions in the Hunt Group are busy (for example, off-hook), have Do Not Disturb activated, or are in programming mode.
Call Pick-up
Stations can be grouped for incoming call routing and Call Pick-up purposes. Ten types of groups can be defined:
Circular Terminal ACD Ring Pick-Up External Voice Mail VMIM/VSF-Voice Mail Feature Server UMS Group Net VM (Centralized External VM) Unified Communication Solution Server Circular Station Group
In Circular Hunt, calls to a station in the group will go to the station, if unavailable or unanswered in the hunt no answer time; the call will be directed to the next station defined in the group. The call will continue to hunt until each station in the group has been tried. The call remains at the last station or passes to a designated overflow station or group.
A Circular Station Group can be assigned with a pilot number (the Station Group Number) so that calls to the pilot number will hunt. In this case, the call will be directed to the first station in the group and, if needed, hunt through each station in the group until reaching the last station. The call may remain at the last station, passed to an overflow destination or sent to a voice mailbox.
Terminal Station Group
Calls to a station in a Terminal Station Group that encounter an unavailable or unanswered status will be routed through the hunt process. The call will proceed to the next listed station in the group until reaching the last listed station in the group. The call may remain at the last station or be routed to an Overflow destination.
A Terminal Hunt Group can be assigned with a pilot number (the Station Group number) so that calls to the pilot number will hunt. In this case, the call will route as described for Circular Pilot Number hunting.
ACD Station Group
Calls can be sent to an ACD group by dialing the Station Group Number or assigning CO/IP lines to ring directly to the Station Group. Calls are directed to the station in the group that has been idle for the longest continuous time, Uniform Call Distribution. If all stations are busy or unavailable when the call is received, the call may be routed to an alternate location or may continue to wait (queue) for an available station in the group. After queuing to the group, the caller may be routed to an overflow destination, which can be a Station, Station Group or Voice Mailbox.
An ACD supervisor can be assigned to monitor the group and act to oversee operations of the group. The ACD Supervisor can print group statistics and activate alternate routing as well as assist agents.
External hotline
External Hotline (#311) 4
Use this feature to identify an external hotline extension. When a user lifts the handset of an external hotline, a predetermined outside number is dialed automatically. The external number might be, for example, a frequently called service bureau. The external hotline must be a single- line telephone, not a system telephone, and should not have a dialpad.
After you identify an external hotline extension, you must store the external telephone number for the hotline extension as Personal Speed Dial code 80.
Considerations 4
■ Under certain conditions of heavy telephone usage, the external hotline may be unable to dial the programmed number immediately (for example, if many of the tip/ring devices connected to your system dial out at the same time).
■ You can identify several extensions as external hotlines.
■ Do not assign an external hotline to extension 10, 11, or to the first two extensions of any 206 or 308EC module, which are reserved as power-failure extensions.
■ Use Hotline (#603) to identify an internal hotline extension.
■ Use Line Assignment (#301) to assign outside lines to the external hotline extension, Pool Extension Assignment (#314) to assign pools to the external hotline extension, and Automatic Line Selection to set the extension to select outside lines or pools first.
If your use of the external hotline requires immediate dialing of the programmed number, assign a line for use only by this extension.
■ Set Line Ringing to No Ring for all lines or pools assigned to the external hotline to prevent incoming calls from ringing at the extension.
■ Make sure there are no call, line, or pool restrictions assigned to an external hotline.
■ Do not use Station Lock at an external hotline because it will prevent the outside number from being dialed.
■ Remove external hotlines from Night Service Group Extensions (#504), Calling Group Extensions (#502), and Hunt Group Extensions (#505).
■ Do not assign Forced Account Code Entry (#307) to the external hotline.
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