Overview for Avaya™ Communication Manager Release 1.
Copyright 2003, Avaya Inc. All Rights Reserved Notice Every effort was made to ensure that the information in this document was complete and accurate at the time of printing. However, information is subject to change. Warranty Avaya Inc. provides a limited warranty on this product. Refer to your sales agreement to establish the terms of the limited warranty.
Safety Requirements for Customer Equipment, ACA Technical Standard (TS) 001 - 1997 One or more of the following Mexican national standards, as applicable: NOM 001 SCFI 1993, NOM SCFI 016 1993, NOM 019 SCFI 1998 The equipment described in this document may contain Class 1 LASER Device(s). These devices comply with the following standards: • EN 60825-1, Edition 1.1, 1998-01 • 21 CFR 1040.10 and CFR 1040.11.
Means of Connection Connection of this equipment to the telephone network is shown in the following tables. For MCC1, SCC1, G600, and CMC1 Media Gateways: Manufacturer’s Port Identifier FIC Code SOC/REN/ Network A.S. Code Jacks Off/On premises station OL13C 9.0F RJ2GX, RJ21X, RJ11C DID trunk 02RV2-T 0.0B RJ2GX, RJ21X CO trunk 02GS2 0.3A RJ21X 02LS2 0.3A RJ21X Tie trunk TL31M 9.0F RJ2GX Basic Rate Interface 02IS5 6.0F, 6.0Y RJ49C 1.544 digital interface 04DU9-BN 6.
European Union Declarations of Conformity Avaya Inc. declares that the equipment specified in this document bearing the “CE” (Conformité Europeénne) mark conforms to the European Union Radio and Telecommunications Terminal Equipment Directive (1999/5/EC), including the Electromagnetic Compatibility Directive (89/336/EEC) and Low Voltage Directive (73/23/EEC).
Contents About this book ■ ■ ■ ■ ■ ■ ■ ■ ■ 1 What is the purpose of this book? Who should read this book? What is in this book? Conventions used in this book Admonishments Trademarks How to obtain Avaya books on the Web How to order documentation How to comment on this book How to get help 25 25 25 26 27 27 28 28 28 29 Overview of Avaya Communication Manager ■ ■ ■ ■ ■ ■ 2 25 31 Communication Manager basic offering and advanced offering Optional software Capacities Avaya Installation Wizard (AIW)
Contents 3 Attendant features ■ ■ ■ ■ ■ ■ ■ ■ ■ 8 43 Accessing the attendant Dial access to attendant Individual attendant access Recall Attendant backup Attendant room status Attendant functions using Distributed Communications System (DCS) protocol Control of trunk group access Direct trunk group selection Inter-PBX attendant calls Call handling Attendant lockout — privacy Attendant split swap Attendant vectoring Automated attendant Backup alerting Call waiting Calling of inward restricted statio
Contents 4 Call center 53 Computer Telephony Integration (CTI) ■ 53 Adjunct Switch Application Interface (ASAI) Co-resident DEFINITY LAN Gateway Direct Agent Announcement (DAA) Flexible billing Pending work mode change Trunk group identification User-to-User Information (UUI) propagation during manual transfer/conference operations VDN override for ASAI messages 54 54 55 55 55 55 56 56 Automatic Call Distribution (ACD) 56 Abandoned call search Adjunct routing Auto-Available Split (AAS) Queue statu
Contents ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Expected Wait Time (EWT) routing Call center messaging Holiday vectoring Vector Directory Number (VDN) Class of Restriction (COR) for VDN Display VDN for route-to DAC VDN in a coverage path VDN of origin announcement VDN return destination Call Work Codes (CWC) Circular station hunt group CMS measurement of ATM Dialed Number Identification Service (DNIS) Direct agent calling Dual links to CMS Duplicate agent login ID administration Agent-loginID skill pair increa
Contents ■ ■ 5 User-to-user information (UUI) over the public network Voice Response Integration (VRI) Collaboration Conferencing ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Abort conference on hang-up Conference — three party Conference — six party Conference/transfer display prompts Conference/transfer toggle/swap Group listen Hold/unhold conference Meet-me conference No dial tone conferencing No hold conference Select line appearance conferencing Selective conference party display, drop, and mute Selective conference m
Contents 6 Communication device support ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ 7 Hospitality ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ 12 2420 DCP telephones Personalized labels Voice mail retrieval button 3410 wireless telephone 3606 wireless VoIP telephone 4600-series IP telephones Katakana character set Voice mail retrieval button 6200-series analog telephones 6400-series DCP telephones 6400 tip/ring interface module 8400-series telephones Attendant console Avaya IP Agent Avaya IP Softphone Avaya IP Softphone for
Contents ■ ■ ■ ■ 8 Localization ■ ■ ■ ■ ■ ■ ■ ■ ■ 9 Suite check-in VIP wakeup Wake-up activation using confirmation tones Xiox call accounting Administrable language displays Katakana character set Administrable loss plan Bellcore calling name ID Block collect call Busy tone disconnect Country-specific localization Italy Distributed Communications Systems (DCS) protocol Japan National private networking support Russia Central Office (CO) support on G700 Media Gateway ISDN/DATS network support Multi-
Contents ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Leave Word Calling (LWC) Leave Word Calling (LWC) — QSIG/DCS Manual message waiting Message demand print Message retrieval Display retrieval Speak-to-me Mode code interface Octel integration QSIG/DCS voice mail interworking Voice mail retrieval button Voice message retrieval Voice messaging and call coverage 10 Mobility ■ ■ ■ ■ ■ ■ ■ ■ Administration Without Hardware (AWOH) Automatic Customer Telephone Rearrangement (ACTR) Avaya Wireless Telephone Solutions (AWTS) 3410 w
Contents Internet Protocol (IP) 4600-series IP telephones Katakana character set Voice mail retrieval button Avaya IP agent Avaya IP Softphone IP endpoint — road-warrior mode IP endpoint — telecommuter mode Wireless 3410 wireless telephone 3606 wireless VoIP telephone 120 120 120 120 120 120 120 121 121 121 121 Port network and gateway connectivity 121 ■ ■ ■ ■ ■ ■ Asynchronous Transfer Mode (ATM) ATM WAN Spare Processor (WSP) Manager Port Network Connectivity (ATM-PNC) Port Network Connectivity (
Contents Trunk types and signaling ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Auxiliary trunks Advanced Private Line Termination (APLT) Central Office (CO) Central Office (CO) support on G700 Media Gateway — Russia Digital multiplexed interface Bit-oriented signalling Message-oriented signalling Direct Inward Dialing (DID) Direct Inward/Outward Dialing (DIOD) E&M signaling — continuous and pulsed E911 CAMA trunk group Foreign Exchange (FX) ISDN trunks Automatic Termination Endpoint Identifier (TEI) Call-by-call
Contents ■ Local exchange trunks 800-service trunks Central Office (CO) trunks Digital Service 1 (DS1) trunks Direct Inward Dialing (DID) trunks Direct Inward/Outward Dialing (DIOD) trunks Foreign Exchange (FX) trunks Wide Area Telecommunications Service (WATS) Intelligent networking ■ ■ ■ ■ ■ Avaya VoIP Monitoring Manager (VMON) Distributed Communications System (DCS) protocol Attendant with DCS Direct trunk group selection Display DCS automatic circuit assurance DCS over ISDN-PRI D-channel (DCS+) DC
Contents ■ ■ Variable length ping Variable Length Subnet Mask (VLSM) QSIG Basic Call completion Call forwarding (diversion) Call Independent Signaling Connections (CISC) Call offer Call transfer Called name ID Centralized Attendant Service (CAS) Attendant display of Class of Restriction (COR) Attendant return call Priority queue RLT emulation through a PRI Communication Manager/Octel QSIG integration Leave Word Calling (LWC) Manufacturer-Specific Information (MSI) Message Waiting Indication (MWI) Name an
Contents ■ ■ ■ ■ Modem pooling Multimedia application server interface Multimedia calling Multimedia call early answer on vectors and stations Multimedia Call Handling (MMCH) Multimedia call redirection to multimedia endpoint Multimedia data conferencing (T.
Contents 13 Security, privacy, and safety System administrator ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Access security gateway Alternate facility restriction levels Alternate operations support system alarm number Privacy — attendant lockout Authorization codes — 13 digits Call restrictions Class of Restriction (COR) Block collect call Customer-provided equipment alarm Data privacy Data restriction Facility restriction levels and traveling class marks Malicious call trace Media encryption Restriction — controll
Contents 14 Special applications 179 15 System management 183 ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Administration Without Hardware (AWOH) Alternate facility restriction levels Announcements Announcement sources for the G700 Media Gateway Avaya Voice Announcement over LAN (VAL) Avaya Voice Announcement over LAN (VAL) Manager Local announcements on the G700 Media Gateway Authorization codes — 13 digits Automatic circuit assurance Automatic transmission measurement system Barrier c
Contents ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Music-on-hold Local music-on-hold Multiple music sources Restriction — controlled Scheduling Security Violation Notification (SVN) Station security codes Tenant partitioning Terminal Translation Initialization (TTI) Time of day clock synchronization through a LAN source Linux platforms UNIX platforms Trunk group circuits Variable length ping Variable Length Subnet Mask (VLSM) Avaya VisAbility management suite ■ ■ ■ ■ ■ ■ ATM WAN Spare Processor (WSP) Manager Avaya Commun
Contents 17 Telephony ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Abbreviated Dialing (AD) Abbreviated dialing labeling Abbreviated dialing on-hook programming Active dialing Administrable timeout on call timer Alphanumeric dialing Automatic Call Back (ACB) Automatic Call Back (ACB) for analog telephones Automatic hold Bellcore calling name ID Bridged call appearance — multi-appearance telephone Bridged call appearance — single-line telephone Call coverage Alphanumeric field designation C
Contents ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ ■ Last number dialed Local call timer automatic start/stop Long hold recall Manual originating line service Misoperation handling Multiappearance preselection and preference Night service Enhanced night service Personalized ringing Posted messages Priority calling Pull transfer Recall signaling Recorded telephone dictation access Reset shift call Ringback queuing Ringer cutoff Ringing — abbreviated and delayed Ringing options Send al
About this book What is the purpose of this book? This book provides general information about the features and capabilities of Avaya™ Communication Manager. It also discusses practical and creative applications for the various platforms that run Communication Manager. For details about changes for the most current release, see the Highlights of Avaya™ Communication Manager, 555-233-783.
Conventions used in this book Become familiar with the following terms and conventions. They help you use this book with Communication Manager. ■ Commands are printed in bold face as follows: command. We show complete commands in this book, but you can usually type an abbreviated version of the command. For example, list configuration station can be typed as list config sta. ■ Screen displays and names of fields are printed in constant width as follows: screen display.
Trademarks Admonishments Admonishments in this book have the following meanings: Tip: Draws attention to information that you may find helpful. NOTE: Draws attention to information that you must heed. ! CAUTION: Denotes possible harm to software, possible loss of data, or possible service interruptions. ! WARNING: Denotes possible harm to hardware or equipment. ! DANGER: Denotes possible harm or injury to your body.
How to obtain Avaya books on the Web If you have internet access, you can view and download the latest version of Avaya documentation products. To view any book, you must have a copy of Adobe Acrobat Reader. NOTE: If you don’t have Acrobat Reader, you can get a free copy at http://www.adobe.com. For example, to access an electronic version of this book: 1. Access the Avaya Web site at http://www.avaya.com/support/. 2. Click Product Documentation. 3.
How to get help How to get help If you suspect that you are being victimized by toll fraud and you need technical assistance or support in the United States and Canada, call the Technical Service Center’s Toll Fraud Intervention Hotline at 1-800-643-2353. If you need additional help, the following resources are available. You may need to purchase an extended service agreement to use some of these resources. See your Avaya representative for more information.
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Overview of Avaya Communication Manager 1 Avaya™ Communication Manager organizes and routes voice, data, image and video transmissions. It can connect to private and public telephone networks, ethernet LANs, ATM networks, and the Internet. Communication Manager seeks to solve business challenges by powering voice communications and integrating with value-added applications. Communication Manager is an open, scalable, highly reliable and secure telephony application.
Overview of Avaya Communication Manager Communication Manager basic offering and advanced offering Communication Manager is available as a basic offering (called “Category B”) and as an advanced offering (called “Category A”). This book describes all of the advanced Communication Manager features. Some of these features are not available with the basic offering — which includes DEFINITY BCS and GuestWorks. NOTE: For a list of features not supported in the basic offering, see your Avaya representative.
Avaya Installation Wizard (AIW) To view the system capacity limits: 1. Go to http://www.avaya.com/support. 2. Type security capacity table in the Search text box, and then click Go. 3. Locate the latest version of the system capacities table document, and then click the title of the document to download the information. Avaya Installation Wizard (AIW) The Avaya™ Installation Wizard (AIW) is a tool to be used in installations (not upgrades) of Communication Manager in S8300/G700 system configurations.
Overview of Avaya Communication Manager The Installation Wizard for Communication Manager has these features: ■ The Installation Wizard supports a stack of up to 10 G700 Media Gateways. ■ Technicians are able to load updated media module firmware versions from their laptop as part of the Installation Wizard process. ■ Installation of the BRI Media Module is supported. ■ The Installation Wizard supports installation of a G700 Media Gateway with a Local Survivable Processor (LSP).
Avaya Media Servers and Media Gateways Avaya Media Servers and Media Gateways The following hardware products are components of Communication Manager: ■ Avaya S8300 Media Server and Avaya G700 Media Gateway ■ Avaya S8700 Media Server for IP Connect configurations ■ Avaya S8700 Media Server for Multi-Connect configurations ■ Avaya S8700 Media Server controlling a remote G700 Media Gateway (with or without an Avaya™ S8300 Media Server configured as an LSP).
Overview of Avaya Communication Manager Avaya S8700 Media Server configurations The following hardware products are components of Communication Manager: ■ Avaya S8700 Media Server for IP Connect Configurations comprises an Avaya S8700 Media Server with an Avaya™ G600 Media Gateway. ■ Avaya S8700 Media Server for Multi-Connect Configurations comprises an Avaya S8700 Media Server with an MCC1 or SCC1 Media Gateway.
Avaya Media Servers and Media Gateways The IP Connect control network is comprised of the customer LAN, and the IP Server interface connectivity via an IP Switch Interface (IPSI) board. The IPSI (TN2312) provides control network connectivity and Tone Clock/Global Call Classifier functionality. For more information about the high-level capabilities of S8700 IP Connect, refer to the Avaya MultiVantage™ Solutions Hardware Guide. Also please see the capacities table for the entire list of updated capacities.
Overview of Avaya Communication Manager S8700 Media Server supports 250 G700 Media Gateways With Communication Manager, a total of 250 G700 Media Gateway can be supported by the S8700 Media Servers in an ECC configuration. NOTE: This feature allows for an average of 40 users (or stations) for each G700 Media Gateway. Thus a system with 250 G700 Media Gateways can be engineered to support a total of 6000 users across all the G700 Media Gateways in an ECC configuration.
MultiTech gateway support MultiTech gateway support Communication Manager supports a voice over IP (VoIP) gateway from MultiTech, a third-party vendor. Any system running Communication Manager can connect and run a MultiTech gateway. With a 2-port, 4-port, or 8-port MultiTech gateway, Communication Manager offers a cost-effective and survivable VoIP gateway solution within a Communication Manager environment for a client’s branch or remote office with fewer then ten stations.
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Application programming interface (API) 2 An application programming inteface (API) allows numerous software applications to work with Avaya™ Communication Manager. Adjunct Switch Application Interface (ASAI) See ‘‘Adjunct Switch Application Interface (ASAI)’’ on page 54. DAPI DEFINITY application programming inteface (DAPI) is an object-oriented application programming interface (API) for accessing control and data paths within Communication Manager.
Application programming interface (API) JTAPI Java telephony application programming interface (JTAPI) is an open API supported by Avaya computer telephony that enables integration to Communication Manager ASAI. It is an object-oriented programming interfaces favored for the development of multimedia solutions. JTAPI applications are supported on any clients that supports a JAVA virtual machine (this includes Windows, UnixWare, and Solaris platforms), or a Java-compatible Web browser.
Attendant features 3 Avaya™ Communication Manager contains many exciting features that provide easy ways to communicate through your telephone system’s attendant (operator). In addition, attendants can connect to their console (switchboard) from other telephones in your system, thereby expanding the attendant capabilities. Accessing the attendant Dial access to attendant The dial access to attendant feature allows you to reach an attendant by dialing an access code.
Attendant features Attendant backup The attendant backup feature allows you to access most attendant console features from one or more specially-administered backup telephones. This allows you to answer calls more promptly, thus providing better service to your guests and prospective clients. When the attendant console is busy, you can answer overflow calls from the backup telephones by pressing a button or dialing a feature access code.
Call handling Inter-PBX attendant calls Inter-PBX attendant calls allows attendants for multiple branches to be concentrated at a main location. Incoming trunk calls to the branch, as well as attendant-seeking voice-terminal calls, route over tie trunks to the main location. Call handling Attendant lockout — privacy This feature prevents an attendant from re-entering a multiple-party connection held on the console unless recalled by a telephone user. This feature is administered on a system-wide basis.
Attendant features Backup alerting The backup alerting feature notifies backup attendants that the primary attendant cannot pick up a call. It provides both audible and visual alerting to backup stations when the attendant queue reaches its queue warning level. When the queue drops below the queue warning level, alerting stops. Audible alerting also occurs when the attendant console is in night mode, regardless of the attendant queue size.
Call handling Override of diversion features The override of diversion feature allows an attendant to bypass diversion features such as send all calls and call coverage by putting a call through to an extension even when these diversion features are on. This feature, together with attendant intrusion, can be used to get an emergency or urgent call through to a telephone user.
Attendant features Timed reminder and attendant timers Attendant timers automatically alert the attendant after an administered time interval for the following types of calls: ■ Extended calls to be answered or waiting to be connected to a busy single-line telephone ■ One-party calls placed on hold on the console ■ Transferred calls that have not been answered after transfer The timed reminder feature informs the attendant that a call requires additional attention.
Monitoring calls Auto start and don’t split The auto start feature allows the attendant to make a telephone call without pushing the start button first. If the attendant is on an active call and presses digits on the keypad, the system automatically splits the call and begins dialing the second call. The don’t split feature deactivates the auto start feature and allows the sending of touch tones over the line for the purposes of such things as picking up messages.
Attendant features Direct extension selection with busy lamp field This feature allows the attendant to keep track of extension status — whether the extension is idle or busy — and to place or extend calls to extension numbers without having to dial the extension number.
Monitoring calls Visually Impaired Attendant Service (VIAS) Visually Impaired Attendant Service (VIAS) provides voice feedback to a visually impaired attendant. Each voice phrase is a sequence of one or more single-voiced messages. This feature defines six attendant buttons to aid visually impaired attendants: ■ Visually impaired service activation/deactivation button: activates or deactivates the feature. All ringers previously disabled (for example, recall and incoming calls) become reenabled.
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Call center 4 The Avaya call center provides a fully integrated telecommunications platform that supports a powerful assortment of features, capabilities, and applications designed to meet all of your customers’ call center needs. Computer Telephony Integration (CTI) Computer Telephony Integration (CTI) enables Avaya™ Communication Manager features to be controlled by external applications, and allows integration of customer databases of information with call control features.
Call center Adjunct Switch Application Interface (ASAI) Adjunct Switch Application Interface (ASAI) allows adjunct applications to access a collection of Communication Manager features and services. Integration with adjuncts occurs through APIs. ASAI is part of Avaya computer telephony. ASAI links Communication Manager and adjunct applications. The interface allows adjunct applications to access Communication Manager features and supply routing information to the system.
Adjunct Switch Application Interface (ASAI) For more information on co-resident DLG and the G700 Media Gateway, see chapters “DEFINITY LAN Gateway and ASAI-Ethernet,” and “Installation and Test for CallVisor ASAI,” in the Avaya MultiVantage™ Software CallVisor ASAI Technical Reference, 555-230-220.
Call center User-to-User Information (UUI) propagation during manual transfer/conference operations This feature enables UUI, specifically used by ASAI, to be propagated to the new call during a manual transfer or conference operation. Previously, ASAI UUI could not be sent in a setup message when the call was transferred to another system, so the ASAI UUI was never passed to an application monitoring calls on the system receiving the transfer.
Adjunct Switch Application Interface (ASAI) A hunt group is especially useful when you expect a high number of calls to a particular phone number. A hunt group might consist of people trained to handle calls on specific topics. For example, the group might be: ■ A benefits department within your company ■ A service department for products you sell ■ A travel reservations service ■ A pool of attendants In addition, a hunt group might consist of a group of shared telecommunications facilities.
Call center 2 1 3 4 5 7 6 6 8 cydfauto KLC 030102 1 2 3 4 System running Avaya™ Communication Manager Incoming lines Hunt group A: business travel Hunt group B: personal travel 5 Hunt group C: general information 6 Queues Call coverage to hunt group C Voice mail 7 8 Figure 2. A basic example of automatic call distribution Abandoned call search Abandoned call search allows a central office that does not provide timely disconnect supervision to identify abandoned calls.
Auto-Available Split (AAS) Auto-Available Split (AAS) Auto-Available Split (AAS) allows members of an Automatic Call Distribution (ACD) split to be continuously in auto-in work mode. An agent in auto-in work mode becomes available for another ACD call immediately after disconnecting from an ACD call. You can use AAS to bring ACD-split members back into auto-in work mode after a system restart.
Call center ■ Historical reports, such as: — Agent — Agent summary — Split — Split summary — Trunk group — Vector directory number Avaya Business Advocate Avaya Business Advocate is the collection of features that provide flexibility in the way a call is selected for an agent in a call surplus situation, and in the way an agent is selected for a call.
Avaya Business Advocate Dynamic threshold adjustment Dynamic threshold adjustment allows the system to compare actual levels of service with service targets, and to adjust overload thresholds. This feature makes the use of overload agents more efficient.
Call center Avaya virtual routing Avaya™ virtual routing (formerly known as Look-Ahead Interflow or LAI) balances the load of ACD calls across multiple locations. Virtual routing helps customers balance call loads among their locations by analyzing demand and directing each call to the location best able to handle it — for example, based on call volume, waiting time in queue, or the time of day.
Call vectoring Data collection Data collection allows the calling party to enter data that can then be used by a host computer application to assist in call handling. For example, this data may be the calling party’s account number, which could then be used to support an inquiry/response application. Data In/Voice Answer (DIVA) Data In/Voice Answer (DIVA) allows the calling party to hear selected announcements based on the digits that he or she enters.
Call center Best service routing (BSR) Best service routing (BSR) distributes the call to the best local or remote split/skill among the resources to be considered, based on expected wait time (EWT) or available agent characteristics. Best service routing (BSR) polling over IP without B-channel Best service routing (BSR) polling over IP without B-channel provides the ability to do BSR polling between multiple sites over H.323 IP trunks without requiring an ISDN PRI B-channel.
Call vectoring A VDN can be accessed in almost any way that an extension can be accessed. When answering a call, the answering agent sees the information (such as the name) associated with the VDN on their display, and can respond to the call with knowledge of the dialed number. This operation provides dialed number identification service (DNIS), allowing the agent to identify the purpose of the incoming call. Class of Restriction (COR) for VDN Class of Restriction (COR) is checked for transfer to the VDN.
Call center VDN return destination VDN return destination is an optional feature that re-routes a call that has been processed through a vector, to the administered return destination. This step occurs once all parties, except the originator, have dropped. The return destination must be a VDN extension. Call Work Codes (CWC) Call Work Codes (CWC) allows ACD agents to enter digits for an ACD call to record the occurrence of a customer-defined event, such as a social security numbers or phone numbers.
Direct agent calling Direct agent calling Direct agent calling lets the customer’s callers automatically go directly to the same agent whenever they call for prompt, personalized service. These direct-to-the-agent calls are also included in their call center measurement statistics. Dual links to CMS The dual links to CMS feature provides an additional TCP/IP link to a separate CMS for full, duplicated CMS data collection functionality and high availability CMS configuration.
Call center Expert Agent Selection (EAS) Expert Agent Selection (EAS) enables certain skill types to be assigned to a call type or a Vector Directory Number (VDN). Routing calls through vectoring then allows the system administration to direct calls to agents who have the particular agent skills required to complete the customers’ inquiries. Add/remove skills Allows an agent using expert agent selection (EAS) to add or remove skills.
Multiple call handling (forced) Multiple call handling (forced) This feature allows agents to receive an ACD call while other types of calls are alerting, active, or on hold. Multiple split queuing Multiple split queuing lets customers direct a call to several splits at the same time, so that the first available agent can take the call. It can help customers handle the busiest periods with greater ease and provide faster service to their callers.
Call center Site statistics for remote port networks The site statistics for remote port networks feature forwards location IDs to CMS to provide call center site-specific reports. VuStats VuStats presents BCMS statistics on telephone displays. Agents, supervisors, call center managers, and other users can press a button and view statistics for agents, splits or skills, VDNs, and trunk groups. These statistics can help agents monitor their own performance, or respond appropriately to the caller’s request.
Caller Information Forwarding (CINFO) The call center capabilities found in either optional software package (Deluxe or Elite) allow Communication Manager Call Center customers to enhance their customer service, help desk, travel, and other operations by providing powerful, integrated call routing via “call vectoring” and resources selection.
Call center Network call redirection 2B-channel transfer This enhancement adds support for the 2-B Channel Transfer PSTN network transfer protocols to the Network Call Redirection (NCR) feature. The protocols that are supported are: ■ Telcordia TBCT (offered by local and inter-exchange PSTNs with Lucent 5Ess or Nortel DMS100 switches in US or Canada) ■ 1998 ANSI Explicit Call Transfer (ECT) for future use.
User-to-user information (UUI) over the public network Service observing by COR Service observing by class of restriction (COR) restricts certain users from using the service observing feature. Service observing of VDNs Service observing of VDNs (also known as VDN observing on agent answer) allows a supervisor to start observing a call to the VDN when the call is delivered to the agent station. The observer will not hear the call during vector processing (announcements, music, and so on).
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Collaboration 5 Avaya™ Communication Manager contains a variety of features aimed at providing easy ways to collaborate with groups of peers, customers, and partners such as executives, sales people, and professional specialists. These key work groups require a high level of effective interaction. Conferencing Abort conference on hang-up When you press the conference button and for any reason you hang up before you complete the conference, you will cancel the conference.
Collaboration Conference/transfer display prompts Conference/transfer display prompts are based on the user’s class of restriction (COR). The display prompts are based on the user’s COR, independent of the select line appearance conferencing and no dial tone conferencing feature. The display messages vary depending on the activation of the two features, but the choice of displaying the additional information or not is dependent on the station user’s COR.
Meet-me conference Meet-me conference The meet-me conference feature allows a person to set up a dial-in conference of up to six parties. The meet-me conference feature uses call vectoring to process the setup of the conference call. Meet-me conference can be optionally set up to require an access code. If an access code is assigned, and if the vector is programmed to expect an access code, each user dialing in to the conference call must enter the correct access code to be added to the call.
Collaboration Selective conference party display, drop, and mute The selective conference party display, drop, and mute feature allows any user on a digital station with display or on an attendant console to use the display to identify all of the other parties on a two-party or conference call. The user would press a feature button while on the call that puts the station or console into conference display mode.
Multimedia Application Server Interface (ASI) Selective conference mute only applies to trunk lines on the conference call, and not to stations. Only one trunk line on the conference call can be selectively muted at a time. This enhanced conferencing feature can be activated from any display station with a “conf-dsp” button and an “fe-mute” button.
Collaboration ■ Voice mail integration — You can access your embedded AUDIX or INTUITY AUDIX voice messaging system from a MultiMedia Communication Exchange (MMCX). Multimedia call early answer on vectors and stations Early answer is a feature applied to multimedia calls in conjunction with conversion to voice.
Multimedia data conferencing (T.120) through an ESM Multimedia data conferencing (T.120) through an ESM The data conference is controlled by an adjunct device called an Expansion Services Module (ESM). The ESM is used to terminate T.120 protocols [including Generalized Conference Call (GCC), a protocol standard for data conference control] and provide data conference control and data distribution. The MultiMedia Interface circuit pack, TN787, is used to rate adapt T.120 data to/from the ESM.
Collaboration Group paging Group paging allows a user to make an announcement to a group of people using speakerphones. The speakerphones are automatically turned on when the user begins the announcement. The recipients can listen to the message over the handset if they wish, but they cannot speak to the user in return. A group page member will not receive the page if the member is active on a call appearance, has a call ringing, is off-hook, has “send-all calls” active, or has “do not disturb” active.
Loudspeaker paging access Loudspeaker paging access Loudspeaker paging access provides attendants and telephone users dial access to voice paging equipment. As many as nine paging zones can be provided by the system, and one zone can be provided that activates all zones at the same time. NOTE: A zone is the location of the loudspeakers — for example, conference rooms, warehouses, or storerooms.
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Communication device support 6 2420 DCP telephones The 2420 Digital Communications Protocol (DCP) telephone is a digital telephone with an optional feature expansion module and downloadable call appearance/feature buttons information. The 2420 DCP phone does not need paper labels. The button information appears on a screen on the phone. The firmware for the 2420 phone can be changed remotely. The 2420 telephone uses icons to indicate the status of call appearances, bridged call appearances, and features.
Communication device support Voice mail retrieval button Avaya™ Communication Manager supports the voice mail retrieval feature as a fixed feature button on the 2420 DCP and the 4602 telephone. A field, “voice-mail Number: _______” appears on the station form for stations of type 2420 and 4602. The allowed values for this field are identical to the values allowed for an autodial feature button number.
6200-series analog telephones This feature requires 4620 firmware version 1.72 or later to work. You can obtain the latest version of 4620 firmware at no charge by going to the Avaya Web site at http://www.avaya.com/support/. Voice mail retrieval button See ‘‘Voice mail retrieval button’’ on page 86. 6200-series analog telephones The 6210, 6211, 6218, 6219, 6220, and 6221 two-wire, analog telephones are designed to take advantage of the many features offered by Communication Manager.
Communication device support 6400 tip/ring interface module This module provides a two-wire analog interface for the 6400 DCP telephones. This allows the operation of an analog adjunct to be independent of the digital telephone’s extension for the use of fax machines or modems without compromising the user’s voice extensions. 8400-series telephones The 8400 digital telephones are versatile two-wire/four-wire DCP telephones.
Avaya IP Softphone for pocket PC Avaya IP Softphone for pocket PC Avaya™ IP Softphone for pocket PC extends the level of Communication Manager services. This feature turns a hand-held personal digital assistant (PDA) into an advanced telephone. Users can place calls, take calls, and handle multiple calls on their PDAs. Avaya Communication Manager PC console The Communication Manager PC console allows your attendants to efficiently handle incoming calls by personal computer.
Communication device support DEFINITY AnyWhere DEFINITY® AnyWhere gives you remote access to the powerful voice and data capabilities of your system running Communication Manager. Communication Manager provides powerful voice features and data collaboration capabilities in your office. With DEFINITY AnyWhere, you can have the same functionality when you are working at your virtual office, traveling, or in your hotel room.
Hospitality 7 Alphanumeric dialing Alphanumeric dialing allows you to place data calls by entering an alphanumeric name rather than a long string of numbers. Attendant room status See ‘‘Attendant room status’’ on page 44. Automatic selection of Direct Inward Dialing (DID) numbers This feature allows the system to automatically choose a number from a list of available Direct Inward Dialing (DID) numbers that will be assigned to a guest’s room extension when checking in.
Hospitality Automatic wakeup The automatic wakeup feature allows attendants, front desk users, and guests to request that one or two wake-up calls be automatically placed to a certain extension number at a later time. When a wakeup call is placed and answered, the system can provide a recorded announcement (which can be a speech synthesis announcement), music, or simply silence.
Dial-by-name Dial-by-name The dial-by-name feature allows callers to the system to access guest rooms simply by dialing the name of the guest they are trying to contact. This feature uses recorded announcements and the call vectoring feature to set up an automatic attendant procedure. This automatic attendant procedure gives callers the ability to enter a guest’s name. When a single or unique match is found, the call is redirected to the guest’s telephone.
Hospitality Names registration The names registration feature automatically sends a guest’s name and room extension from the property management system (PMS) to the switch at check-in, and automatically removes this information at check-out. The information may be displayed on any attendant console or display-equipped telephone at various hotel locations (for example, room service or security).
Single-digit dialing and mixed station numbering Single-digit dialing and mixed station numbering This feature provides hotel staff and guests easy access to internal hotel/motel services, and provides the capability to associate room numbers with guest room telephones. The feature provides the following dial plan types: single-digit dialing, prefixed extensions, and mixed numbering. Suite check-in Suite check-in allows more than one station to be checked in at one time.
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Localization 8 Administrable language displays This feature allows messages that appear on telephone display units to be shown in the language spoken by the user. These messages are available in English (the default), French, Italian, Spanish, or one other user-defined language. The language for display messages is selected by each user. The feature requires 40-character display telephones. Katakana character set See ‘‘Katakana character set’’ on page 86.
Localization Bellcore calling name ID This feature allows the system to accept calling name information from a Local Exchange Carrier (LEC) network that supports the Bellcore calling name specification. The system can send calling name information in the format if Bellcore calling name ID is administered.
Country-specific localization Country-specific localization Italy Distributed Communications Systems (DCS) protocol Enhanced DCS adds features to the existing DCS capabilities and requires the use of Italian TGU/TGE tie trunks.
Localization Multi-Frequency Packet (MFP) signaling Multi-Frequency Packet (MFP) address signaling is provided in Russia on outgoing CO trunks. Calling party number and dialed number information is sent on outgoing links between local and toll switches. Russian MFP is set on each trunk group on the ‘type’ field on the trunk screen. NOTE: Russian MFP does not apply to PCOL trunks.
Message integration 9 Audible message waiting Audible message waiting places a stutter at the beginning of the dial tone when a telephone user picks up the telephone. The stutter dial tone indicates that the user has a message waiting. This feature is particularly useful for visually impaired people who may not be able to see a message light. It is often used with telephones that have no message waiting lights.
Message integration Embedded AUDIX While many voice messaging systems require separate equipment and connections, the embedded AUDIX system easily installs directly into your cabinet to support advanced voice messaging capabilities without the need for an adjunct processor. Each embedded AUDIX system supports up to 2000 mailboxes and stores up to 100 hours of recorded messages. Whenever you call the embedded AUDIX system, you interact with it by entering commands through your telephone’s touch-tone keypad.
INTUITY AUDIX ■ Name record by subscriber lets you record your own name on the system. ■ Automatic message scan can play all new messages in part or in their entirety without requiring you to press additional buttons, which is particularly useful when you are getting messages from your mobile phone. ■ Sending restrictions by community enables you to limit the communities of callers who can communicate using AUDIX voice messaging.
Message integration ■ Turn off AUDIX call answering allows you to turn off call answering in order to conserve system resources. You can create a message that tells callers they cannot leave a message, giving them another number to call, for example. ■ Pre-addressing allows you to address a message before recording it. ■ Integrated messaging allows you access and manage incoming voice, fax, and e-mail messages and file attachments from your personal computer or your telephone.
INTUITY AUDIX The IA770 application consists of license file-activated software residing on the S8300 Media Server, and a small card that can be installed and upgraded in the field. The IA770 application is available in two configurations: ■ 4 ports, 100 users ■ 8 ports, 300 users The IA770 application includes INTUITY™ Message Manager. While the system provides text-to-speech capability in U.S.
Message integration The INTUITY AUDIX mezzanine card also provides the necessary DSP resources for messaging. This hardware eliminates the need for the INTUITY Map 5P adjunct, usually required for this functionality. Also see ‘‘Embedded AUDIX’’ on page 102. AUDIX one-step recording Users can record conversations by pressing a single button. This feature uses AUDIX as the recording device. This feature is not available with Intuity AUDIX through Mode Codes or remote AUDIX.
INTUITY call accounting system The Interval For Applying Periodic Alerting Tone field is used to allow the switch administrator to choose an interval to play an alerting tone to all the parties on the call during recording. Values are 0–60, and the default is 15. This means, if the default value is used, that all parties on the call hear an alerting tone every 15 seconds that indicates the conversation is being recorded. If the value for the field is 0, then no periodic tone is played during recording.
Message integration Hotel guests can leave messages for each other without going through the attendant. For incoming calls, an attendant transfers the call to the appropriate room. If the guest does not answer the call or if the line is busy, the call is automatically transferred to the guest’s voice mailbox, where the caller can leave a voice message. A message-waiting indicator on the guest’s phone notifies the guest that the voice mailbox contains messages.
Leave Word Calling (LWC) Leave Word Calling (LWC) Leave Word Calling (LWC) allows internal system users to leave a short preprogammed message (usually “call” with the calling user’s name, extension number, and the time of the call) for other internal users. When the message is stored, the message lamp on the called telephone automatically lights. LWC messages can be retrieved using a telephone display, voice message retrieval, or AUDIX.
Message integration Display retrieval Users having digital telephones with displays or a personal computer integrated with a telephone can display messages. Speak-to-me Using any touch-tone telephone, employees can dial speak-to-me and hear a synthesized voice read their messages over the telephone. Mode code interface Communication Manager supports an analog mode code interface for communications with INTUITY AUDIX and other voice mail systems using the same interface.
Voice messaging and call coverage a different user’s messages can be retrieved only by a user at a telephone or attendant console in the coverage path, by an administered system-wide message retriever, or by a remote-access user when the extension and associated security code are known. The system restricts unauthorized users from retrieving messages. Voice messaging and call coverage Often an AUDIX system is set up as the last point on a call-coverage path, as shown in Figure 3 on page 111.
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Mobility 10 IP telephones or IP softphones allow you to access the features of Avaya™ Communication Manager without having to be tied to one location. One of the major benefits of IP telephones is that you can move the telephones around on a LAN just by unplugging them and plugging them in somewhere else. One of the main benefits of IP softphones is that you can load them on a laptop PC, and then use the PC's modem to connect them to the switch from almost anywhere.
Mobility ACTR works with the 2420 DCP telephone and the 6400 serialized telephones. The 6400 serialized phone is stamped with the word “serialized” on the faceplate for easy identification. The 6400 serialized phone memory electronically stores its own part ID (comcode) and serial number. ACTR uses the stored information and associates the phone with new port when the phone is moved. ACTR makes it easy to identify and move phones.
Avaya Extension to Cellular 3606 wireless VoIP telephone The 3606 wireless VoIP telephone solution is an IEEE 802.11b standards-based, 2.4 GHz wireless LAN telephone system. Using voice over IP (VoIP) technology, the 3606 wireless VoIP telephone solution provides high quality mobile voice communications throughout the workplace. The 3606 wireless VoIP telephone solution requires one DCP port per handset and emulates an 8410D desk telephone.
Mobility Extension to Cellular offers the following capabilities: 116 ■ XMOBILE station administration enhancements to include the dial prefix, cell phone number, mapping mode, XMOBILE type and configuration set fields ■ User control of Extension to Cellular through “enable” and “disable” feature access codes, or through a configured Extension to Cellular feature button on the office telephone ■ Office caller ID for call originations from the cell phone ■ Extension to Cellular enabled/disabled sta
Personal Station Access (PSA) ■ An optional timer can be included with an administered Extension to Cellular feature button, allowing the user to temporarily disable Extension to Cellular service for one hour. ■ Call classification — Extension to Cellular call filtering uses the same criteria for classifying a call as external or internal as the call coverage feature.
Mobility Terminal Translation Initialization (TTI) Communication Manager provides Terminal Translation Initialization (TTI), a feature that works with Administration Without Hardware (AWOH). TTI associates the terminal translation data with a specific port location through the entry of a special feature-access code, a TTI security code, and an extension number from a terminal that is connected to a wired (but untranslated) jack.
Networking and connectivity 11 Private networking and connectivity Communication device support Circuit switched Analog 6200-series See ‘‘6200-series analog telephones’’ on page 87. Digital telephones 2420 DCP telephones See ‘‘2420 DCP telephones’’ on page 85. Personalized labels See ‘‘Personalized labels’’ on page 85. Voice mail retrieval button See ‘‘Voice mail retrieval button’’ on page 86.
Networking and connectivity 6400-series telephones See ‘‘6400-series DCP telephones’’ on page 87. 6400 tip/ring interface module See ‘‘6400 tip/ring interface module’’ on page 88. 8400-series telephones See ‘‘8400-series telephones’’ on page 88. Internet Protocol (IP) 4600-series IP telephones See ‘‘4600-series IP telephones’’ on page 86. Katakana character set See ‘‘Katakana character set’’ on page 86. Voice mail retrieval button See ‘‘Voice mail retrieval button’’ on page 86.
Wireless The single network connection between the PC and Communication Manager carries two channels, one for the signaling path and one for the voice path. On Communication Manager, the road-warrior application requires the C-LAN circuit pack for signaling and the IP media processor for voice processing. IP endpoint — telecommuter mode IP endpoint — telecommuter mode enables telecommuters to use the full Communication Manager feature set from home.
Networking and connectivity Port Network Connectivity (ATM-PNC) ATM Port Network Connectivity (ATM-PNC) provides an alternative to the Center Stage Switch (CSS) configurations for connecting the Processor Port Network (PPN) to one or more Expansion Port Networks (EPN). ATM-PNC replaces the CSS in a DEFINITY R8r and later network with an ATM switch or network. ATM-PNC is available with all three Communication Manager reliability options — standard, high, and critical.
Circuit switched WAN Spare Processor (WSP) An ATM WAN Spare Processor (WSP) provides a disaster recovery option for a media gateway G3r expansion port networks deployed over an ATM WAN. An ATM WSP acts as a PPN in the event of a catastrophic failure in the network. The ATM WSP continually monitors a path to the PPN to determine if it is on-line and reachable. The WSP functions as a PPN if the main PPN is not functional or is not communicating to one or more of the other EPNs.
Networking and connectivity IP Port Network Connectivity (PNC) Communication Manager allows Control Channel Message Set (CCMS) messages to be packetized over IP LAN and WAN connections to control multiple port networks. Link recovery IP calls must have an H.248 link between the Avaya G700 Media Gateway and the call controller. The H.
Separation of Bearer and Signaling Separation of Bearer and Signaling The Separation of Bearer and Signaling (SBS) feature provides a low cost virtual private network with high voice quality for customers who cannot afford private leased lines. SBS provides a DCS+ VPN replacement for those customers needing Dial Plan Expansion (DPE) functionality. NOTE: QSIG works with six-digit and seven-digit dial plans, but DCS does not. In addition, QSIG does not work over VPNs, but DCS does.
Networking and connectivity ■ The ISDN PUBLIC-UNKNOWN NUMBERING form must be correctly administered to map every SBS extension to the corresponding national public network complete number (that is, the DID/DDI number). This public form is used to develop the complete number even if the incoming SBS trunk group numbering format is administered for private numbering. Trunk connectivity Asynchronous Transfer Mode (ATM) See ‘‘Asynchronous Transfer Mode (ATM)’’ on page 121.
Internet Protocol (IP) T1 See ‘‘T1’’ on page 136. Internet Protocol (IP) IP trunks IP trunk groups may be defined as a virtual private network’s tie lines between systems or ITS-E servers running Communication Manager. Each IP trunk circuit pack provides a basic 12-port package that can be expanded up to a total of 30 ports. The number of ports that are defined will correspond to the total number of simultaneous calls transmitted over the IP trunk interface.
Networking and connectivity Trunk types and signaling Auxiliary trunks Auxiliary trunks connect devices in auxiliary cabinets with Communication Manager. Some of the features that are supported with this type of trunk are recorded announcements, telephone dictation service, malicious call trace, and loudspeaker paging.
Direct Inward Dialing (DID) Communication Manager supports two versions of digital multiplexed interface, each differing in the way information is carried over the 24th channel: ■ Bit-oriented signaling ■ Message-oriented signaling Bit-oriented signalling Digital multiplexed interface bit-oriented signalling carries framing and alarm data and signalling information for connections to host computers and other vendor equipment.
Networking and connectivity E911 CAMA trunk group This form administers the Centralized Automatic Message Accounting (CAMA) trunks and provides Caller Emergency Service Identification (CESID) information to the local community’s enhanced 911 system through the local central office. Foreign Exchange (FX) Foreign Exchange (FX) trunks connect Communication Manager to a Central Office other than the local one. ISDN trunks Gives you access to a variety of public and private network services and facilities.
ISDN trunks ETSI functionality The full set of ETSI public-network and private-network ISDN features is officially supported. This includes Look-Ahead Interflow (LAI), look-ahead routing, and usage allocation.
Networking and connectivity The ISDN-BRI Trunk circuit pack allows Communication Manager to support the T interface and the S/T interface as defined by ISDN standards (ITU-T recommendation I.411). The circuit pack provides eight ports to the network and supports two B channels and one D channel.
ISDN trunks 7 6 1 10 2 8 ? 3 9 8 5 4 4 4 1 System running Communication Manager 6 2 System running Communication Manager 7 3 System running Communication Manager Basic rate interface telephone Passive bus 8 4 5 9 10 Private ISDN (can be carried over ATM-CES) Public ISDN (can be carried over ATM-CES) Public and private networks Central office switch Tandem switch Figure 4.
Networking and connectivity NT interface on TN556C Communication Manager supports the NT (network) side of the T interface using the TN556C circuit pack. This gives the switch full tie trunk capability using BRI trunks. Communication Manager supports leased BRI connections through the public network, with a TN2185 on each end of the leased connection. Communication Manager will not, however, allow customers to administer both endpoints and trunks on the same TN556C circuit pack.
Release Link Trunks (RLT) Release Link Trunks (RLT) Release Link Trunks (RLT) are used between switch locations to provide centralized attendant service or automatic call distribution group availability. Remote access trunks Tie trunks Tie trunks carry communications between Communication Manager and other switches in a private network. Several types of trunks can be used, depending on the type of private network you establish.
Networking and connectivity DS1 trunk service Bit-oriented signaling that multiplexes 24 channels into a single 1.544-Mbps stream. DS1 can be used for voice or voice-grade data and for data-transmission protocols. E1 trunk service is bit-oriented signaling that multiplexes 32 channels into a single 2.048-Mbps stream. Both DS1 and E1 provide a digital interface for trunk groups. Digital Service 1 (DS1) trunks can be used to provide T1 or ISDN Primary Rate Interface (PRI) service.
Local exchange trunks Local exchange trunks Local exchange trunks connect Communication Manager to a central office. The following local exchange trunks are some of the types available. 800-service trunks 800-service trunks let your business pay the charges for inbound long-distance calls so that callers can reach you toll-free. Central Office (CO) trunks See ‘‘Central Office (CO)’’ on page 128. Digital Service 1 (DS1) trunks See ‘‘DS1 trunk service’’ on page 136.
Networking and connectivity Distributed Communications System (DCS) protocol The Distributed Communications System (DCS) protocol allows you to configure two or more switches as if they were a single, large system. DCS provides attendant and voice-terminal features between these switch locations. DCS simplifies dialing procedures and allows transparent use of some of the Communication Manager features.
Electronic Tandem Network (ETN) QSIG/DCS voice mail interworking See ‘‘QSIG/DCS voice mail interworking’’ on page 110. Electronic Tandem Network (ETN) In an Electronic Tandem Network (ETN) — also known as Private Network Access (PNA) — Communication Manager provides a variety of features on a network-wide basis. It allows calls to other systems in a private network. These calls do not use the public network. Instead, they are routed over your dedicated facilities.
Networking and connectivity Internet Protocol (IP) The capabilities and applications of Communication Manager are extended using IP. Communication Manager IP supports audio/voice over a LAN or WAN, and it ensures that remote workers have access to communication system features from their PCs. Communication Manager also provides standards based control between media server and media gateways allowing communications infrastructure to be distributed to the edge of the network.
Internet Protocol (IP) Alternate gatekeeper and registration addresses When an IP endpoint (including softphones, IP phones, and Avaya R300) registers with the switch, the switch sends back an IP registration address. The switch sends a different IP address for each registration according to a cyclic algorithm.
Networking and connectivity Multiple location support for network regions Multiple location support for network regions allows remote Avaya media gateways connected to a central Avaya media server to retain: ■ Local user time ■ Local ARS public analysis tables for local trunking ■ automatic daylight savings time ■ Local touch tone receivers for IP communications devices, such as Avaya IP telephones. Communication Manager allows administrators to map locations to IP network regions.
Internet Protocol (IP) 802.1p/Q IEEE standard 802.1Q and 802.1p provide the means to specify both a Virtual LAN (VLAN) and a frame priority at layer 2 for use by LAN hubs, or bridges, that can do routing based on MAC addresses. 802.1p/Q provides for 8 levels of priority (3 bits) and a large number (12 bits) of VLAN identifiers. The VLAN identifier at layer 2 permits segregation of traffic to reduce traffic on individual links. Because 802.
Networking and connectivity QoS for call control Communication Manager allows QoS for the signaling packets coming from gatekeepers such as the C-LAN by employing the same standards based DiffServ and 802.1p/Q schemes as with audio channels. This QoS services further improve the users VoIP audio experience. QoS for VoIP Communication Manager implements QoS through the selection of audio codec such as G.711, G.723 and G.
Internet Protocol (IP) Shuffling and hairpinning Shuffling and hairpinning can improve traffic handling performance and improve voice quality by more efficiently using both Communication Manager switching fabric by allocating, when possible, available IP network resources. “Shuffling” means rerouting the audio channel connecting two IP endpoints.
Networking and connectivity QSIG Basic QSIG provides compliance to the International Standardization Organization (ISO) ISDN-PRI private-networking specifications. QSIG is defined by ISO as the worldwide standard for private networks. QSIG features are supported on BRI trunks. QSIG is the generic name for a family of signaling protocols. The Q-reference point or interface is the logical point where signaling is passed between 2 peer entities in a private network.
QSIG Call Independent Signaling Connections (CISC) Call Independent Signaling Connections (CISC) are used to pass QSIG supplementary service information that is independent of an active call between two QSIG compliant nodes. Implementation is based on the ISO standard for CISC. It is possible to determine the status of a QSIG TSC by using the “status signaling group” command on the SAT.
Networking and connectivity Attendant return call If a call that is transferred by the attendant goes unanswered for a specified period of time, the call is returned to the attendant. Preferably the call will transfer back to the attendant who transferred the call. Priority queue QSIG MSI will pass more information to the main PBX. This information enables calls coming in from a QSIG CAS branch to be placed in the appropriate place in the queue, as if the call originated on the main PBX.
QSIG Manufacturer-specific supplementary services can be created using specific operations encoded with the manufacturer’s identifier. Communication Manager supports non-QSIG applications that transport information across QSIG networks in MSI. Applications have the same functionality over QSIG networks that they have over non-QSIG networks. Applications that use MSI include Centralized Attendant Service, Transfer to Audix, Best Service Routing, and QSIG VALU.
Networking and connectivity QSIG/DCS voice mail interworking QSIG/DCS Voice Mail Interworking is an enhancement to the current QSIG feature. It integrates DCS and QSIG Centralized Voicemail via the DCS+/QSIG gateway. Switches labeled DCS+/QSIG integrate multi-vendor PBXs into a single voice messaging system. QSIG/DCS Voice Mail Interworking works on G3r, G3si, and G3csi. It provides network flexibility, DCS functionality without a dedicated T1.
Uniform Dial Plan (UDP) VALU Call coverage This feature provides similar call coverage as DCS call coverage and Call Coverage Remote Off Net (C-CRON). The call will come back if covered over QSIG. The functionality will only be complete when all the switches are running under Communication Manager and using QSIG VALU. The covered-to party can still receive distinct alerting.
Networking and connectivity Customers upgrading to Communication Manager can choose to migrate to the 6-digit or 7-digit dial plan or not. Customers who choose not to migrate may convert their dial plans at a later date. Distributed Communications System (DCS) protocol is limited to a dial plan of 3-5 digits, so if your dial plan requires 6 or 7 digits, QSIG — which is the generic name for a family of signaling protocols— is required.
Administered connections Data interfaces Administered connections Automatically establishes an end-to-end connection between two access or data endpoints based on administered attributes.
Networking and connectivity Data modules Data modules connect systems running Communication Manager with other communications equipment, changing protocol, connections, and timing as necessary.
IP asynchronous links IP asynchronous links IP asynchronous links enable Communication Manager to transfer existing asynchronous adjunct connectivity to an Ethernet (TCP/IP) environment. IP asynchronous links support switch server applications, as well as client applications. Systems running Communication Manager can connect to System Management applications such as the Avaya Visibility Suite over the LAN.
Networking and connectivity 1 2 3 4 5 6 System running Communication Manager Asynchronous terminal Digital port Analog trunk Modem Remote application 7 8 9 10 11 12 13 Integrated pooled modem Data line port Analog port 7400A Digital communications protocol Analog EIA standard Figure 5.
Multimedia application server interface Multimedia application server interface The Multimedia Application Server Interface provides a link between Communication Manager and one or more Multimedia Communications eXchange nodes. A Multimedia Communications eXchange is a stand-alone multimedia call processor produced by Avaya. This link to Communication Manager enhances the capabilities of each Multimedia Communications eXchange system by enabling it to share some of the Communication Manager features.
Networking and connectivity For an incoming call, Early Answer answers the dynamic service-link calls when the destination endpoint answers, unless Early Answer is specified during routing or termination processing. NOTE: The “destination voice endpoint” might be an outgoing voice trunk if the destination voice station is forwarded or covered off-premises. Multimedia Call Handling (MMCH) Multimedia Call Handling (MMCH) enables you to control voice, video, and data transmissions using your telephone set.
Pass advice of charge information to world class BRI endpoints Multimedia call redirection to multimedia endpoint A dual port multimedia station may be a destination of call redirection features such as call coverage, forwarding, and station hunting. The station can receive and accept full multimedia calls or data calls converted to multimedia. Multimedia data conferencing (T.120) through ESM The data conference is controlled by an adjunct device called an Expansion Services Module (ESM).
Networking and connectivity Call routing Alternate facility restriction levels Allows Communication Manager to adjust facility restriction levels or authorization codes for lines or trunks. Each line or trunk is normally assigned a facility restriction level. With this feature, Alternate Facility Restriction Levels are also assigned. Attendants can change to the alternates, thus changing access to lines and trunks.
Generalized route selection Automatic Route Selection (ARS) Automatic Route Selection (ARS) selects carriers automatically and routes calls inexpensively over the public network. When there are one or more long-distance carriers and Wide Area Telecommunications Service (WATS) provided, Communication Manager selects the most preferred route for the call. Long-distance carrier-code dialing is not required on routes selected by the system.
Networking and connectivity Look-ahead routing Provides an efficient way to use trunking facilities. It allows you to continue to try to reroute an outgoing ISDN-PRI call that is not completing. When Communication Manager receives a cause value that indicates congestion, Look-Ahead Routing tells the system what to do next. For each routing preference, you can indicate if the next routing-preference should be attempted or if the current routing-preference should be attempted again.
Traveling class marks Traveling class marks Traveling Class Marks are a mechanism for passing a caller’s facility restriction level from one Electronic Tandem Network switch to another. Traveling Class Marks allow privilege checking to be passed across switches through the Electronic Tandem Network. Miscellaneous Answer detection For purposes of Call-Detail Recording (CDR), it is important to know when the called party answers a call.
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Reliability and survivability 12 Alternate gatekeeper The alternate gatekeeper enhancement can provide survivability between Avaya™ Communication Manager and IP communications devices such as IP Telephones and IP softphones. This is accomplished by providing alternate gatekeepers (C-LAN) in the event of network or gatekeeper failure and by load balancing endpoint traffic among multiple gatekeepers.
Reliability and survivability An LSP is a configuration used to provide redundancy of the Avaya call processing application. Usually, a media module serves as the ICC for the system, but it can also serve as a redundant processor for call processing. In the LSP configuration, the processor serves as an alternate controller/gatekeeper for IP entities, such as IP telephones and media gateways. These IP entities use the LSP when they lose connectivity to their primary controller.
Multiple network regions per C-LAN Multiple network regions per C-LAN Multiple network regions per C-LAN enables a single C-LAN to provide registration and call control to IP endpoints in multiple network regions. Communication Manager implements this approach by allowing IP address to be mapped to network regions in a mapping form, instead of just to a C-LAN. When an IP phone registers, the switch will determine the phone’s network region number based on the phone’s IP address.
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Security, privacy, and safety 13 System administrator Access security gateway Access security gateway is an authentication interface used to secure the system administration and maintenance ports and/or logins on the system. Access security gateway employs a challenge/response protocol to confirm the validity of a user and reduce the opportunity for unauthorized access. Successful authentication is accomplished when the feature communicates with a compatible key.
Security, privacy, and safety You might want to use this feature to disable most long-distance calling at night, for example, to prevent unauthorized staff from making long-distance calls. ! CAUTION: This feature may change the AAR and ARS routing preferences. Using it on tandem and tie-trunk applications affects entire networks. Calls that are part of a cross-country private network may be blocked.
Call restrictions Call restrictions By dialing an access code, administrators and attendants have the ability to restrict users from making or receiving certain types of calls. There are five restrictions: ■ Outward — User cannot place external calls. ■ Station-to-station — User cannot place or receive internal calls. ■ Termination — User cannot receive any calls (except priority calls). ■ Toll — User cannot place toll calls but can place local calls.
Security, privacy, and safety Data restriction Data restriction protects analog data calls from being disturbed by any of the system’s overriding or ringing features. It is administered at the system level to selected analog and multi-appearance telephones and trunk groups. Facility restriction levels and traveling class marks Allows certain calls to specific users, while denying the same calls to other users.
Restriction — controlled Restriction — controlled Allows an attendant or telephone user, with console permission, to activate and deactivate for an individual telephone or a group of telephones, the following restrictions: ■ outward ■ total ■ station-to-station ■ termination restrictions Security Violation Notification (SVN) Security Violation Notification (SVN) allows you to set security-related parameters and to receive notification when the limits that you have established are violated.
Security, privacy, and safety End user Backup alerting Notifies backup attendants that the primary attendant cannot pick up a call. It provides both audible and visual alerting to backup stations when the attendant queue reaches its queue warning level. When the queue drops below the queue warning level, alerting stops. Audible alerting also occurs when the attendant console is in night mode, regardless of the attendant queue size.
Crisis alerts to a digital numeric pager Per line CPN restriction Users may block the calling party number when originating calls. For ISDN calls, the CPN presentation indicator is encoded accordingly. For non-ISDN calls, going to a public network that supports the CPN restriction feature, the network specific feature activation code gets passed to the network for interpretation and activation.
Security, privacy, and safety Crisis alerts to an attendant console Crisis alert uses both audible and visual alerting to notify attendant consoles when an emergency call is made. Audible alerting sounds like an ambulance siren. Visual alerting flashes the CRSS-ALRT button lamp and the display of the caller’s name and extension (or room). Crisis alert’s display of the origin of the emergency call enables the attendant or other user to direct emergency-service response to the caller.
Privacy — manual exclusion Privacy — manual exclusion Allows multi-appearance telephone users to keep other users with appearances of the same extension number from bridging onto an existing call. Exclusion is activated by pressing the exclusion button on a per-call basis. Restriction — controlled See ‘‘Restriction — controlled’’ on page 173. Station lock Station lock allows users to lock their phones to prevent unauthorized outgoing calls. Users can block outgoing calls and still receive incoming calls.
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Special applications 14 Special applications are those custom features developed by the Avaya™ global rapid response team to meet a particular customer's need. Each feature is ordered through the rapid response team as an al la carte item. Special ordering and provisioning procedures apply. Contact your Avaya Sales representative or authorized Avaya business partner for more information.
Special applications 180 ■ Display incoming digits for ISDN trunk groups ■ Night service on DID trunk group ■ Display UUI information ■ Enhanced DID routing ■ Vector collect # and * literally option ■ Service observe physical set ■ Busy tones on send all calls with no available coverage points ■ 80000 UDP extension records (DEFINITY Server R only) ■ Dial by name ■ Variable length account codes ■ 25,000 facility busy indicators (DEFINITY Server R only) ■ ISDN redirecting number ■ E
555-233-767 ■ Enhancement to QSIG rerouting for call forwarding - don’t strip ARS/AAR access code (9) when forwarding digits from DEFINITY ECS to IPC turret ■ Expand the number of coverage paths (DEFINITY Server R only) to 2000 and remote cover points Issue 4 May 2003 181
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System management 15 Avaya™ Communication Manager system management provides the administrator powerful tools to maintain their communication solutions and to drive down the total cost of ownership. Administration Without Hardware (AWOH) See ‘‘Administration Without Hardware (AWOH)’’ on page 113. Alternate facility restriction levels This feature allows Communication Manager to adjust facility restriction levels or authorization codes for lines or trunks.
System management Announcements The Announcements feature provides a recorded announcement to a variety of types of calls: calls that cannot be completed as dialed, calls that have been in queue for an assigned interval, any calls whose destination is an announcement, or incoming calls to a user. Announcement sources for the G700 Media Gateway This feature provides an announcement source for each G700 Media Gateway registered to either an S8300 or S8700 server.
Authorization codes — 13 digits Avaya Voice Announcement over LAN (VAL) Manager Avaya™ Voice Announcement over LAN (VAL) Manager is part of the Avaya VisAbility™ Suite of products. It enables you to the use of a LAN to transfer recorded announcements to Avaya media servers. Announcements can be stored in .wav files, which can be sent to a voice announcement over LAN board without conversion.
System management Automatic transmission measurement system Measures voice and data trunk facilities for satisfactory transmission performance. The measurement report contains data on trunk signal loss, noise, signaling return loss, and echo return loss. Acceptable performance, the scheduling of tests, and report contents are administrable. Barrier codes A security code used with remote access to prevent unauthorized access to your system.
Call charge information Call charge information Provides two ways to know the approximate charge for calls made on outgoing trunks: ■ Advice of Charge — for ISDN trunks Advice of Charge (AOC) collects charge information from the public network for each outgoing call. Charge advice is a number representing the cost of a call; it is recorded as either a charging or currency unit.
System management Call restrictions By dialing an access code, administrators and attendants have the ability to restrict users from making or receiving certain types of calls. There are five restrictions: ■ Outward — user cannot place external calls. ■ Station-to-station — user cannot place or receive internal calls. ■ Termination — user cannot receive any calls (except priority calls). ■ Toll — user cannot place toll calls but can place local calls.
Classless Interdomain Routing (CIDR) ■ Off-hook alert ■ Console permission ■ Client room Classless Interdomain Routing (CIDR) See ‘‘Classless Interdomain Routing (CIDR)’’ on page 141. Concurrent user sessions In order to increase the efficiency of administration and maintenance functions, the Communication Manager accommodates multiple concurrent administration and maintenance user sessions.
System management DCS automatic circuit assurance Allows a user or attendant at one node to activate or deactivate automatic circuit assurance referral calls for the entire DCS network. This transparency allows the referral calls to originate at a node other than the node that detects the problem. External device alarming Allows you to assign analog ports to alarm interfaces for external devices.
Firmware download Firmware download The firmware download feature makes it possible to download an image from a remote or local source into the system running Communication Manager, and use that image to reprogram the application code of a port circuit pack. This feature makes updating firmware more cost effective. It also reduces the expense of servicing the system’s port circuit packs because it eliminates the need for a technician to be involved when a board is updated.
System management There is only one maintenance board, which is placed in carrier A. This is the only maintenance board in the cabinet. NOTE: Only two PNs are physically supported in S8700 Media Server IPSI-enabled systems when high/critical reliability options are desired. Only two PNs are physically supported in DEFINITY Server R systems when critical/ATM Network Duplication reliability is desired. The following table shows the number of port networks allowed in an MCC1 Media Gateway.
Information and reports ■ Call coverage reports The call coverage report displays measurements of the distribution of traffic offered to call-coverage groups. Separate reports for all calls and external calls are supplied. ■ Coverage points report The coverage points report differs based on whether all calls or external calls is selected. For each coverage point in the group, the number of calls offered, abandoned while at that coverage point, and overflowing to the next coverage point are listed.
System management ■ Traffic reports Traffic reports show measurements in the form of switch-based reports for local or remote access, and can be collected for subsequent analysis and reporting by adjuncts and operation support systems using the operation support system interface protocol. ■ Trunk group detailed measurements IP asynchronous links See ‘‘IP asynchronous links’’ on page 155. Malicious call trace Allows you to trace malicious calls.
Restriction — controlled For more information, see the Installation for Adjuncts and Peripherals for Avaya™ Communication Manager, 555-233-116. Also see the Administrator’s Guide for Avaya™ Communication Manager, 555-233-506. Multiple music sources On an MCC1, SCC1, CMC1, or G600 Media Gateway, this feature allows the customer to provide multiple distinct music sources for use with the call vectoring features, calls placed on hole, calls awaiting pickup, and so on.
System management Tenant partitioning Allows partitioning of the system in order to lease the system’s services and features to tenants. This provides attractive services and revenue for “virtual” landlords. It provides the robust features of a large system at affordable rates to small business tenants. Communication Manager supports up to 100 partitions and 27 attendant groups. Multiple attendant groups can be assigned to each partition.
Trunk group circuits UNIX platforms Communication Manager running on DEFINITY servers which use an Oryx/Pecos operating system (proprietary UNIX-based OS) receives a command from Avaya site administration to adjust the time. Avaya site administration is synchronized to the LAN PC’s clock. Trunk group circuits Trunks provide the communications links between Communication Manager and other switches, including central office switches and other premises switches.
System management ATM WAN Spare Processor (WSP) Manager ATM WAN Spare Processor (WSP) Manager can be a key part of your emergency restoration and business continuity planning. This application enables users to download translations from a main server running Communication Manager, and simultaneously upload those translations to multiple (up to 15) ATM WAN Spare Processors (WSPs) over a LAN connectivity. This can be done according to a schedule specified by the administrator.
Avaya voice announcement over LAN (VAL) manager Avaya voice announcement over LAN (VAL) manager See ‘‘Avaya Voice Announcement over LAN (VAL) Manager’’ on page 185. Avaya VoIP Monitoring Manager (VMON) Avaya VoIP monitoring manager (VMON) provides the ability to monitor voice over IP (VoIP) network quality.
System management Avaya Directory Enabled Management Avaya Directory Enabled Management (DEM) is part of the Avaya VisAbility management suite, and provides real-time, integrated, directory-based read/write access to Avaya media servers and INTUITY AUDIX messaging servers. It streamlines workflow and information management in an electronic environment using converged networks. DEM creates a meta-directory for converged voice and data networks.
Telecommuting and remote office 16 Avaya R300 remote office communicator (R300) The R300 feature offers a cost-effective method for providing full functionality at a remote site. The R300 provides remote telephony that has all the capabilities of telephony that is connected directly. Through the R300, voice and data can share the same WAN link between Avaya™ Communication Manager and the remote site, thus providing voice and data convergence.
Telecommuting and remote office Coverage of calls redirected off-net (CCRON) Coverage of calls redirected off-net (CCRON) allows calls that have been redirected to locations outside of the switch to return to the switch for further processing. For example, an employee that telecommutes can have two coverage paths. One coverage path is used when the employee is in the office and the other coverage path is used when the employee is working from home.
Off-premises station Off-premises station A trunk-data module connects off-premises private-line trunk facilities and Communication Manager. The trunk-data module converts between the RS-232C and the DCP, and can connect to DDD modems as the DCP member of a modem pool. See ‘‘Call redirection’’ on page 209 and ‘‘Call vectoring’’ on page 63. Remote access Permits authorized callers from remote locations to access the system via the public network and then use its features and services.
Telecommuting and remote office 204 Issue 4 May 2003 555-233-767
Telephony 17 Abbreviated Dialing (AD) Abbreviated Dialing (AD) provides lists of stored numbers you can use to: ■ Place local, long-distance, and international calls ■ Activate features ■ Access remote computer equipment You simply dial the list number and the one-digit, two-digit, or three-digit number associated with the telephone number you want. The number is then automatically dialed by the system.
Telephony Abbreviated dialing on-hook programming On-hook programming allows users of the 2420 DCP telephone, as well as the 4600-series, 6400-series, and 8400-series telephone sets with enabled speakers, to access the programming mode without going off-hook during available call appearances. Signaling changes from DTMF to the S-channel, allowing the use of a longer (60 seconds) time-out period.
Automatic hold Automatic Call Back (ACB) for analog telephones When a person, using an analog telephone, places a call and the line is busy, an announcement prompts the caller to enter the digit 1 to activate ACB, or to enter the digit 2 to route the call to a hunt group extension. Automatic hold Allows attendants and multi-function telephone users to alternate easily between two or more calls.
Telephony Bridged call appearance — single-line telephone Allows single-line telephones users to have a bridged appearance on a multi-appearance telephone. Call coverage Call coverage provides automatic redirection of calls that meet specified criteria to alternate answering positions in a call coverage path.
Call redirection Call redirection Call forward busy/don’t answer Allows calls to be forwarded when the called extension is busy or when the call is not answered after an administrable interval. If the extension is busy, the call forwards immediately. If the extension is not busy, the incoming call rings the called extension, then forwards only if it remains unanswered longer than the administered interval.
Telephony Call pickup Along with directed call pickup, allows you to answer calls for other telephones within your specified call pickup group. Directed call pickup allows you to pick up any call on the system. With this feature, you do not have to leave your telephone to answer a call for a nearby telephone. You simply dial an access code or press a call pickup button. Group call pickup Allows you to dial a feature access code (FAC) and a pickup group number to answer a call from a different group.
Coverage callback Coverage callback Allows a covering user to leave a message for the called party to call back the person who called. Coverage incoming call identification Allows multi-appearance telephones users without a display in a coverage answer group to identify an incoming call to that group. Disconnecting unanswered calls Disconnects unanswered outgoing calls after a predetermined amount of time.
Telephony Enhanced abbreviated dialing Supplements abbreviated dialing by providing one enhanced number per system. Enhanced number lists can contain any number or dial access code. System administrators designate privileges for group number lists, system number lists and enhanced number lists. With privileged lists, users can access otherwise restricted numbers (for example, stations without long-distance access can be programmed to access specified long-distance numbers).
Internal automatic answer Internal automatic answer Allows specific telephones to answer incoming internal calls automatically. This feature is intended for use with telephones that have speakerphones or headsets. You simply press an internal automatic answer feature button, and calls are automatically answered when the telephone is idle.
Telephony Misoperation handling Defines how calls are handled when a misoperation occurs. A misoperation is when calls are left on hold when the controlling station goes on hook. For example, a misoperation can occur under either of the following conditions: ■ If you hang up prior to completing a feature operation (in some cases, hanging up completes the operation, as in call transfer).
Night service Night service There are five night service features: ■ Hunt group night service allows an attendant or a split supervisor to assign a hunt group or split to night service mode. All calls for the hunt group then are redirected to the hunt group’s designated night service extension. When a user activates hunt group night service, the associated button lamp lights. ■ Night console service directs all calls for primary and daytime attendant consoles to a night console.
Telephony Personalized ringing Allows users of certain telephones to uniquely identify their own calls. Each user can choose one of a number of possible ringing patterns. The eight ringing patterns are tone sequences consisting of different combinations of three tones. With this feature, users working closely in the same area can each specify a different ringing pattern in order to better identify their own calls.
Pull transfer Pull transfer Allows either the party who was originally called, or the party to whom the held call will be transferred, to complete the transfer. This is a convenient way to connect a party with someone better qualified to handle the call. Attendant assistance is not required and the call does not have to be redialed. It interfaces with satellite workstations through TGU/TGE trunks and is always available for calls that use TGU/TGE trunks.
Telephony Ringer cutoff Allows the user of a multi-appearance telephone to turn audible ringing signals on and off. Visual alerting is not affected by this feature. When this feature is enabled, only priority (three-burst) ring, redirect notification, intercom ring, and manual signaling ring at the telephone. Internal and external calls do not ring. Ringing — abbreviated and delayed Allows you to manually or automatically assign one of four ring types to each call appearance on a telephone.
Station hunting Station hunting Routes calls made to a busy extension to another extension. To use station hunting, you create a station hunting chain that governs the order in which a call routes from one extension to the next when the called extension is busy. Each extension in the chain links to only one subsequent extension. However, an extension may be linked from any number of extensions.
Telephony Users may select any of the following as the display message language: English (default), French, Italian, or Spanish. In addition, messages can be administered on the system in a fifth language. The language for display messages is selected by each user. Telephone self administration The telephone self administration capability allows you to program feature buttons on the telephone yourself.
Timed call disconnection for outgoing trunk calls Timed call disconnection for outgoing trunk calls This feature provides the capability to automatically disconnect an outgoing trunk call after an administrable amount of time. Warning tones are applied to all parties on the call prior to the disconnection. The amount of time that can elapse before the trunk is dropped can be specified, and can vary between 2-999 minutes.
Telephony Transfer recall Returns the unanswered transfer calls back to the person who transferred the call. Transfer recall uses a priority alerting signal, and the display on the telephone shows “rt”, which indicates a returned call from a failed transfer operation. Transfer upon hang-up Provides you with the ability to transfer a call by hanging up instead of having to press the transfer button a second time.
Index Numerics 2420 DCP digital telephone, 85, 119 personalized labels, 85, 119 voice mail retrieval button, 86, 110, 119 2B-channel transfer, 72 3410 wireless telephone, 86, 114, 121 3606 wireless VoIP telephone, 86, 115, 121 4600-series Internet Protocol (IP) telephones, 86, 120 4620 IP telephone katakana characters, 86, 97, 120 voice mail retrieval button, 87, 120 6200-series analog telephones, 87, 119 6400-series DCP digital telephones, 87, 120 tip/ring interface module, 88, 120 800-service trunks, 137
Index API. See Application Programming Interface (API) APLT. See Advanced Private Line Termination (APLT) Application Programming Interface (API), 41 DAPI, 41 JTAPI, 42 TAPI, 42 TSAPI, 42 Application Server Interface (ASI), 79, 157 approximate charge for calls, 187 ARS. See Automatic Route Selection (ARS) ASA. See Average Speed of Answer (ASA) routing ASAI. See Adjunct Switch Application Interface (ASAI) ASCII character set, 94 ASI.
Index Avaya business advocate, 60 enhancements, 60 auto reserve agents, 60 call selection override per skill, 60 dynamic percentage adjustment, 60 dynamic queue position, 60 dynamic threshold adjustment, 61 Least Occupied Agent (LOA), 68 logged-in advocate agent counting, 61 percent allocation distribution, 61 reserve agent time in queue activation, 61 VuStats, 70 Avaya call center basic, 70 deluxe, 70 elite, 70 features supported on the Avaya G700 Media Gateway, 70 Avaya Call Management System (CMS), 61,
Index call distribution based on skill, 68 call forwarding all calls, 209 busy/don’t answer, 209 diversion, 146 of 18-digits, 209 override, 209 call handling, 45 Call Independent Signaling Connections (CISC), 147 Call Management System (CMS) measurement of ATM, 66, 126 call offer, 147 call park, 209 call pickup, 210 group call pickup, 210 call prompting, 62 call center messaging, 64 data collection, 63 Data In/Voice Answer (DIVA), 63 call redirection intervals, 209 call redirection to multimedia endpoint,
Index conferencing, (continued) conference/transfer display prompts, 76 dial intercom, 82 group listen, 76 group paging, 82 hold/unhold, 76 loudspeaker paging access, 83 manual signaling, 83 meet-me, 77 multimedia, 81, 159 no dial tone, 77 no hold conference, 77 select line appearance, 77 selective party display and drop, 78 six party, 75 three party, 75 transfer toggle/swap, 76 whisper page, 83 with attendant, 46 Connected Party Number (CPN), 55, 174 restriction per call, 174 restriction per line, 175 con
Index dial access to attendant, 43 Dial Plan Expansion (DPE), 125, 151 dial-by-name, 93 Dialed Number Identification Service (DNIS), 66 DID.
Index Expansion Port Network (EPN), 122 Expansion Services Module (ESM), 79, 81, 157, 159 Expected Wait Time (EWT), 64 Expert Agent Selection (EAS), 55, 56, 68 add/remove skills, 68 call distribution based on skill, 68 queue to best ISDN support, 68 extended trunk access, 152 extension number portability, 139 Extension to Cellular. See Avaya Extension to Cellular external device alarming, 190 F FAC.
Index inter-PBX attendant calls, 45 intrusion (call offer), 46 INTUITY AUDIX, 103 call accounting system, 107 Conversant®, 107 lodging, 107 call accounting system, 108 IP Connect, definition of, 36 IP. See Internet Protocol (IP) ISDN. See Integrated Services Digital Network (ISDN) ISDN-BRI. See Integrated Services Digital Network (ISDN), Basic Rate Interface (ISDN-BRI) ISO.
Index misoperation handling, 214 MMCH. See Multimedia Call Handling (MMCH) MMCX. See Multimedia Communications Exchange (MMCX) mobility, 113 mode code centralized voice mail integration, 101 interface, 110 modem pooling, 155 monitoring calls, 49 MSI. See Manufacturer-Specific Information (MSI) MSN. See Multiple Subscriber Number (MSN) multiappearance preselection and preference, 214 Multi-Connect, definition of, 37 Multi-Frequency Packet (MFP) signaling — Russia, 100, 134 multimedia.
Index Personal Station Access (PSA), 117 name/number permanent display, 117 personalized labels, 85, 119 personalized ringing, 216 placing calls, 48 PMS. See Property Management System (PMS) PNA. See Private Network Access (PNA) port network and gateway connectivity, 121 port network and link usage report, 193 Port Network Connectivity (PNC) Asynchronous Transfer Mode (ATM), 122 over WAN, 122 Internet Protocol (IP), 124 posted messages, 216 power failure transfer, 167 PPM.
Index redirection of calls, 209 call forward busy/don’t answer, 209 call forwarding all calls, 209 call forwarding override, 209 call redirection intervals, 209 on no answer, 69 refresh route report, 193 Release Link Trunks (RLT), 135, 147 emulation through a PRI, 148 release loop operation, 47 reliability and survivability, 165 remote access trunks, 135, 203 remote logout of agent, 72 remote office communicator (R300), 201 reports, 192 attendant position, 192 blockage study, 192 call coverage, 193 coverag
Index select line appearance conferencing, 77 selective conference mute, 47, 78 selective conference party display and drop, 78 self-administered telephones, 87, 220 send all calls, 218 Separation of Bearer and Signaling (SBS), 125 serial calling, 47 service observing, 72 by COR, 73 of VDNs, 73 remote, 73 vector-initiated, 73 shuffling, 145 and NAT devices, 145 single-digit dialing and mixed station numbering, 95 site statistics for remote port networks, 70 six party conferencing, 75 skill, 68 speak-to-me
Index transfer, 221 abort, 221 outgoing trunk to outgoing trunk, 221 recall, 222 trunk-to-trunk, 222 upon hang-up, 222 TransTalk 9000 digital wireless system, 118 traveling class marks, 163 tripwire security, 173 trunk call disconnection, 221 trunk connectivity, 126 trunk flash, 222 trunk group access, 50 busy/warning indicators to attendant, 50 circuits, 197 detailed measurement report, 194 identification, 55 trunk identification by attendant, 50 trunk signaling and error recovery, 139 trunk types and sig
Index VLSM. See Variable Length Subnet Mask (VLSM) VMON. See VoIP Monitoring Manager (VMON) Voice Announcement over LAN (VAL), 184 Voice Announcement over LAN (VAL) Manager, 185, 199 voice mail integration, 80 voice mail interworking, 150 QSIG/DCS, 110, 139 voice mail retrieval button, 86, 87, 110, 119, 120 voice mail system (VMS), 215 voice message retrieval, 110 voice messaging and call coverage, 111 Voice Response Integration (VRI), 73 VoIP Monitoring Manager (VMON), 137, 199 VRI.
